FFmpeg  4.4.4
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 
25 /**
26  * @file
27  * QDM2 decoder
28  * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29  *
30  * The decoder is not perfect yet, there are still some distortions
31  * especially on files encoded with 16 or 8 subbands.
32  */
33 
34 #include <math.h>
35 #include <stddef.h>
36 #include <stdio.h>
37 
39 #include "libavutil/mem_internal.h"
40 #include "libavutil/thread.h"
41 
42 #define BITSTREAM_READER_LE
43 #include "avcodec.h"
44 #include "get_bits.h"
45 #include "bytestream.h"
46 #include "internal.h"
47 #include "mpegaudio.h"
48 #include "mpegaudiodsp.h"
49 #include "rdft.h"
50 
51 #include "qdm2_tablegen.h"
52 
53 #define QDM2_LIST_ADD(list, size, packet) \
54 do { \
55  if (size > 0) { \
56  list[size - 1].next = &list[size]; \
57  } \
58  list[size].packet = packet; \
59  list[size].next = NULL; \
60  size++; \
61 } while(0)
62 
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
65 
66 #define FIX_NOISE_IDX(noise_idx) \
67  if ((noise_idx) >= 3840) \
68  (noise_idx) -= 3840; \
69 
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
71 
72 #define SAMPLES_NEEDED \
73  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
74 
75 #define SAMPLES_NEEDED_2(why) \
76  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
77 
78 #define QDM2_MAX_FRAME_SIZE 512
79 
80 typedef int8_t sb_int8_array[2][30][64];
81 
82 /**
83  * Subpacket
84  */
85 typedef struct QDM2SubPacket {
86  int type; ///< subpacket type
87  unsigned int size; ///< subpacket size
88  const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
90 
91 /**
92  * A node in the subpacket list
93  */
94 typedef struct QDM2SubPNode {
95  QDM2SubPacket *packet; ///< packet
96  struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
97 } QDM2SubPNode;
98 
99 typedef struct QDM2Complex {
100  float re;
101  float im;
102 } QDM2Complex;
103 
104 typedef struct FFTTone {
105  float level;
107  const float *table;
108  int phase;
110  int duration;
111  short time_index;
112  short cutoff;
113 } FFTTone;
114 
115 typedef struct FFTCoefficient {
116  int16_t sub_packet;
118  int16_t offset;
119  int16_t exp;
122 
123 typedef struct QDM2FFT {
125 } QDM2FFT;
126 
127 /**
128  * QDM2 decoder context
129  */
130 typedef struct QDM2Context {
131  /// Parameters from codec header, do not change during playback
132  int nb_channels; ///< number of channels
133  int channels; ///< number of channels
134  int group_size; ///< size of frame group (16 frames per group)
135  int fft_size; ///< size of FFT, in complex numbers
136  int checksum_size; ///< size of data block, used also for checksum
137 
138  /// Parameters built from header parameters, do not change during playback
139  int group_order; ///< order of frame group
140  int fft_order; ///< order of FFT (actually fftorder+1)
141  int frame_size; ///< size of data frame
143  int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
144  int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
145  int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
146 
147  /// Packets and packet lists
148  QDM2SubPacket sub_packets[16]; ///< the packets themselves
149  QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
150  QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
151  int sub_packets_B; ///< number of packets on 'B' list
152  QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
153  QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
154 
155  /// FFT and tones
166 
167  /// I/O data
171 
172  /// Synthesis filter
178 
179  /// Mixed temporary data used in decoding
180  float tone_level[MPA_MAX_CHANNELS][30][64];
189 
190  // Flags
191  int has_errors; ///< packet has errors
192  int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
193  int do_synth_filter; ///< used to perform or skip synthesis filter
194 
196  int noise_idx; ///< index for dithering noise table
197 } QDM2Context;
198 
199 static const int switchtable[23] = {
200  0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
201 };
202 
203 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
204 {
205  int value;
206 
207  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
208 
209  /* stage-2, 3 bits exponent escape sequence */
210  if (value < 0)
211  value = get_bits(gb, get_bits(gb, 3) + 1);
212 
213  /* stage-3, optional */
214  if (flag) {
215  int tmp;
216 
217  if (value >= 60) {
218  av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
219  return 0;
220  }
221 
223 
224  if ((value & ~3) > 0)
225  tmp += get_bits(gb, (value >> 2));
226  value = tmp;
227  }
228 
229  return value;
230 }
231 
232 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
233 {
234  int value = qdm2_get_vlc(gb, vlc, 0, depth);
235 
236  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
237 }
238 
239 /**
240  * QDM2 checksum
241  *
242  * @param data pointer to data to be checksummed
243  * @param length data length
244  * @param value checksum value
245  *
246  * @return 0 if checksum is OK
247  */
248 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
249 {
250  int i;
251 
252  for (i = 0; i < length; i++)
253  value -= data[i];
254 
255  return (uint16_t)(value & 0xffff);
256 }
257 
258 /**
259  * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
260  *
261  * @param gb bitreader context
262  * @param sub_packet packet under analysis
263  */
265  QDM2SubPacket *sub_packet)
266 {
267  sub_packet->type = get_bits(gb, 8);
268 
269  if (sub_packet->type == 0) {
270  sub_packet->size = 0;
271  sub_packet->data = NULL;
272  } else {
273  sub_packet->size = get_bits(gb, 8);
274 
275  if (sub_packet->type & 0x80) {
276  sub_packet->size <<= 8;
277  sub_packet->size |= get_bits(gb, 8);
278  sub_packet->type &= 0x7f;
279  }
280 
281  if (sub_packet->type == 0x7f)
282  sub_packet->type |= (get_bits(gb, 8) << 8);
283 
284  // FIXME: this depends on bitreader-internal data
285  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
286  }
287 
288  av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
289  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
290 }
291 
292 /**
293  * Return node pointer to first packet of requested type in list.
294  *
295  * @param list list of subpackets to be scanned
296  * @param type type of searched subpacket
297  * @return node pointer for subpacket if found, else NULL
298  */
300  int type)
301 {
302  while (list && list->packet) {
303  if (list->packet->type == type)
304  return list;
305  list = list->next;
306  }
307  return NULL;
308 }
309 
310 /**
311  * Replace 8 elements with their average value.
312  * Called by qdm2_decode_superblock before starting subblock decoding.
313  *
314  * @param q context
315  */
317 {
318  int i, j, n, ch, sum;
319 
321 
322  for (ch = 0; ch < q->nb_channels; ch++)
323  for (i = 0; i < n; i++) {
324  sum = 0;
325 
326  for (j = 0; j < 8; j++)
327  sum += q->quantized_coeffs[ch][i][j];
328 
329  sum /= 8;
330  if (sum > 0)
331  sum--;
332 
333  for (j = 0; j < 8; j++)
334  q->quantized_coeffs[ch][i][j] = sum;
335  }
336 }
337 
338 /**
339  * Build subband samples with noise weighted by q->tone_level.
340  * Called by synthfilt_build_sb_samples.
341  *
342  * @param q context
343  * @param sb subband index
344  */
346 {
347  int ch, j;
348 
350 
351  if (!q->nb_channels)
352  return;
353 
354  for (ch = 0; ch < q->nb_channels; ch++) {
355  for (j = 0; j < 64; j++) {
356  q->sb_samples[ch][j * 2][sb] =
357  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
358  q->sb_samples[ch][j * 2 + 1][sb] =
359  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
360  }
361  }
362 }
363 
364 /**
365  * Called while processing data from subpackets 11 and 12.
366  * Used after making changes to coding_method array.
367  *
368  * @param sb subband index
369  * @param channels number of channels
370  * @param coding_method q->coding_method[0][0][0]
371  */
372 static int fix_coding_method_array(int sb, int channels,
373  sb_int8_array coding_method)
374 {
375  int j, k;
376  int ch;
377  int run, case_val;
378 
379  for (ch = 0; ch < channels; ch++) {
380  for (j = 0; j < 64; ) {
381  if (coding_method[ch][sb][j] < 8)
382  return -1;
383  if ((coding_method[ch][sb][j] - 8) > 22) {
384  run = 1;
385  case_val = 8;
386  } else {
387  switch (switchtable[coding_method[ch][sb][j] - 8]) {
388  case 0: run = 10;
389  case_val = 10;
390  break;
391  case 1: run = 1;
392  case_val = 16;
393  break;
394  case 2: run = 5;
395  case_val = 24;
396  break;
397  case 3: run = 3;
398  case_val = 30;
399  break;
400  case 4: run = 1;
401  case_val = 30;
402  break;
403  case 5: run = 1;
404  case_val = 8;
405  break;
406  default: run = 1;
407  case_val = 8;
408  break;
409  }
410  }
411  for (k = 0; k < run; k++) {
412  if (j + k < 128) {
413  int sbjk = sb + (j + k) / 64;
414  if (sbjk > 29) {
416  continue;
417  }
418  if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
419  if (k > 0) {
421  //not debugged, almost never used
422  memset(&coding_method[ch][sb][j + k], case_val,
423  k *sizeof(int8_t));
424  memset(&coding_method[ch][sb][j + k], case_val,
425  3 * sizeof(int8_t));
426  }
427  }
428  }
429  }
430  j += run;
431  }
432  }
433  return 0;
434 }
435 
436 /**
437  * Related to synthesis filter
438  * Called by process_subpacket_10
439  *
440  * @param q context
441  * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
442  */
444 {
445  int i, sb, ch, sb_used;
446  int tmp, tab;
447 
448  for (ch = 0; ch < q->nb_channels; ch++)
449  for (sb = 0; sb < 30; sb++)
450  for (i = 0; i < 8; i++) {
452  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
454  else
456  if(tmp < 0)
457  tmp += 0xff;
458  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
459  }
460 
461  sb_used = QDM2_SB_USED(q->sub_sampling);
462 
463  if ((q->superblocktype_2_3 != 0) && !flag) {
464  for (sb = 0; sb < sb_used; sb++)
465  for (ch = 0; ch < q->nb_channels; ch++)
466  for (i = 0; i < 64; i++) {
467  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
468  if (q->tone_level_idx[ch][sb][i] < 0)
469  q->tone_level[ch][sb][i] = 0;
470  else
471  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
472  }
473  } else {
474  tab = q->superblocktype_2_3 ? 0 : 1;
475  for (sb = 0; sb < sb_used; sb++) {
476  if ((sb >= 4) && (sb <= 23)) {
477  for (ch = 0; ch < q->nb_channels; ch++)
478  for (i = 0; i < 64; i++) {
479  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
480  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
481  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
482  q->tone_level_idx_hi2[ch][sb - 4];
483  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
484  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
485  q->tone_level[ch][sb][i] = 0;
486  else
487  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
488  }
489  } else {
490  if (sb > 4) {
491  for (ch = 0; ch < q->nb_channels; ch++)
492  for (i = 0; i < 64; i++) {
493  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
494  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
495  q->tone_level_idx_hi2[ch][sb - 4];
496  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
497  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
498  q->tone_level[ch][sb][i] = 0;
499  else
500  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
501  }
502  } else {
503  for (ch = 0; ch < q->nb_channels; ch++)
504  for (i = 0; i < 64; i++) {
505  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
506  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
507  q->tone_level[ch][sb][i] = 0;
508  else
509  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
510  }
511  }
512  }
513  }
514  }
515 }
516 
517 /**
518  * Related to synthesis filter
519  * Called by process_subpacket_11
520  * c is built with data from subpacket 11
521  * Most of this function is used only if superblock_type_2_3 == 0,
522  * never seen it in samples.
523  *
524  * @param tone_level_idx
525  * @param tone_level_idx_temp
526  * @param coding_method q->coding_method[0][0][0]
527  * @param nb_channels number of channels
528  * @param c coming from subpacket 11, passed as 8*c
529  * @param superblocktype_2_3 flag based on superblock packet type
530  * @param cm_table_select q->cm_table_select
531  */
532 static void fill_coding_method_array(sb_int8_array tone_level_idx,
533  sb_int8_array tone_level_idx_temp,
534  sb_int8_array coding_method,
535  int nb_channels,
536  int c, int superblocktype_2_3,
537  int cm_table_select)
538 {
539  int ch, sb, j;
540  int tmp, acc, esp_40, comp;
541  int add1, add2, add3, add4;
542  int64_t multres;
543 
544  if (!superblocktype_2_3) {
545  /* This case is untested, no samples available */
546  avpriv_request_sample(NULL, "!superblocktype_2_3");
547  return;
548  for (ch = 0; ch < nb_channels; ch++) {
549  for (sb = 0; sb < 30; sb++) {
550  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
551  add1 = tone_level_idx[ch][sb][j] - 10;
552  if (add1 < 0)
553  add1 = 0;
554  add2 = add3 = add4 = 0;
555  if (sb > 1) {
556  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
557  if (add2 < 0)
558  add2 = 0;
559  }
560  if (sb > 0) {
561  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
562  if (add3 < 0)
563  add3 = 0;
564  }
565  if (sb < 29) {
566  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
567  if (add4 < 0)
568  add4 = 0;
569  }
570  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
571  if (tmp < 0)
572  tmp = 0;
573  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
574  }
575  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
576  }
577  }
578  acc = 0;
579  for (ch = 0; ch < nb_channels; ch++)
580  for (sb = 0; sb < 30; sb++)
581  for (j = 0; j < 64; j++)
582  acc += tone_level_idx_temp[ch][sb][j];
583 
584  multres = 0x66666667LL * (acc * 10);
585  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
586  for (ch = 0; ch < nb_channels; ch++)
587  for (sb = 0; sb < 30; sb++)
588  for (j = 0; j < 64; j++) {
589  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
590  if (comp < 0)
591  comp += 0xff;
592  comp /= 256; // signed shift
593  switch(sb) {
594  case 0:
595  if (comp < 30)
596  comp = 30;
597  comp += 15;
598  break;
599  case 1:
600  if (comp < 24)
601  comp = 24;
602  comp += 10;
603  break;
604  case 2:
605  case 3:
606  case 4:
607  if (comp < 16)
608  comp = 16;
609  }
610  if (comp <= 5)
611  tmp = 0;
612  else if (comp <= 10)
613  tmp = 10;
614  else if (comp <= 16)
615  tmp = 16;
616  else if (comp <= 24)
617  tmp = -1;
618  else
619  tmp = 0;
620  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
621  }
622  for (sb = 0; sb < 30; sb++)
623  fix_coding_method_array(sb, nb_channels, coding_method);
624  for (ch = 0; ch < nb_channels; ch++)
625  for (sb = 0; sb < 30; sb++)
626  for (j = 0; j < 64; j++)
627  if (sb >= 10) {
628  if (coding_method[ch][sb][j] < 10)
629  coding_method[ch][sb][j] = 10;
630  } else {
631  if (sb >= 2) {
632  if (coding_method[ch][sb][j] < 16)
633  coding_method[ch][sb][j] = 16;
634  } else {
635  if (coding_method[ch][sb][j] < 30)
636  coding_method[ch][sb][j] = 30;
637  }
638  }
639  } else { // superblocktype_2_3 != 0
640  for (ch = 0; ch < nb_channels; ch++)
641  for (sb = 0; sb < 30; sb++)
642  for (j = 0; j < 64; j++)
643  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
644  }
645 }
646 
647 /**
648  * Called by process_subpacket_11 to process more data from subpacket 11
649  * with sb 0-8.
650  * Called by process_subpacket_12 to process data from subpacket 12 with
651  * sb 8-sb_used.
652  *
653  * @param q context
654  * @param gb bitreader context
655  * @param length packet length in bits
656  * @param sb_min lower subband processed (sb_min included)
657  * @param sb_max higher subband processed (sb_max excluded)
658  */
660  int length, int sb_min, int sb_max)
661 {
662  int sb, j, k, n, ch, run, channels;
663  int joined_stereo, zero_encoding;
664  int type34_first;
665  float type34_div = 0;
666  float type34_predictor;
667  float samples[10];
668  int sign_bits[16] = {0};
669 
670  if (length == 0) {
671  // If no data use noise
672  for (sb=sb_min; sb < sb_max; sb++)
674 
675  return 0;
676  }
677 
678  for (sb = sb_min; sb < sb_max; sb++) {
679  channels = q->nb_channels;
680 
681  if (q->nb_channels <= 1 || sb < 12)
682  joined_stereo = 0;
683  else if (sb >= 24)
684  joined_stereo = 1;
685  else
686  joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
687 
688  if (joined_stereo) {
689  if (get_bits_left(gb) >= 16)
690  for (j = 0; j < 16; j++)
691  sign_bits[j] = get_bits1(gb);
692 
693  for (j = 0; j < 64; j++)
694  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
695  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
696 
698  q->coding_method)) {
699  av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
701  continue;
702  }
703  channels = 1;
704  }
705 
706  for (ch = 0; ch < channels; ch++) {
708  zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
709  type34_predictor = 0.0;
710  type34_first = 1;
711 
712  for (j = 0; j < 128; ) {
713  switch (q->coding_method[ch][sb][j / 2]) {
714  case 8:
715  if (get_bits_left(gb) >= 10) {
716  if (zero_encoding) {
717  for (k = 0; k < 5; k++) {
718  if ((j + 2 * k) >= 128)
719  break;
720  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
721  }
722  } else {
723  n = get_bits(gb, 8);
724  if (n >= 243) {
725  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
726  return AVERROR_INVALIDDATA;
727  }
728 
729  for (k = 0; k < 5; k++)
730  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
731  }
732  for (k = 0; k < 5; k++)
733  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
734  } else {
735  for (k = 0; k < 10; k++)
736  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
737  }
738  run = 10;
739  break;
740 
741  case 10:
742  if (get_bits_left(gb) >= 1) {
743  float f = 0.81;
744 
745  if (get_bits1(gb))
746  f = -f;
747  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
748  samples[0] = f;
749  } else {
750  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
751  }
752  run = 1;
753  break;
754 
755  case 16:
756  if (get_bits_left(gb) >= 10) {
757  if (zero_encoding) {
758  for (k = 0; k < 5; k++) {
759  if ((j + k) >= 128)
760  break;
761  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
762  }
763  } else {
764  n = get_bits (gb, 8);
765  if (n >= 243) {
766  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
767  return AVERROR_INVALIDDATA;
768  }
769 
770  for (k = 0; k < 5; k++)
771  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
772  }
773  } else {
774  for (k = 0; k < 5; k++)
775  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
776  }
777  run = 5;
778  break;
779 
780  case 24:
781  if (get_bits_left(gb) >= 7) {
782  n = get_bits(gb, 7);
783  if (n >= 125) {
784  av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
785  return AVERROR_INVALIDDATA;
786  }
787 
788  for (k = 0; k < 3; k++)
789  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
790  } else {
791  for (k = 0; k < 3; k++)
792  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
793  }
794  run = 3;
795  break;
796 
797  case 30:
798  if (get_bits_left(gb) >= 4) {
799  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
801  av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
802  return AVERROR_INVALIDDATA;
803  }
804  samples[0] = type30_dequant[index];
805  } else
806  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
807 
808  run = 1;
809  break;
810 
811  case 34:
812  if (get_bits_left(gb) >= 7) {
813  if (type34_first) {
814  type34_div = (float)(1 << get_bits(gb, 2));
815  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
816  type34_predictor = samples[0];
817  type34_first = 0;
818  } else {
819  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
821  av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
822  return AVERROR_INVALIDDATA;
823  }
824  samples[0] = type34_delta[index] / type34_div + type34_predictor;
825  type34_predictor = samples[0];
826  }
827  } else {
828  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
829  }
830  run = 1;
831  break;
832 
833  default:
834  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
835  run = 1;
836  break;
837  }
838 
839  if (joined_stereo) {
840  for (k = 0; k < run && j + k < 128; k++) {
841  q->sb_samples[0][j + k][sb] =
842  q->tone_level[0][sb][(j + k) / 2] * samples[k];
843  if (q->nb_channels == 2) {
844  if (sign_bits[(j + k) / 8])
845  q->sb_samples[1][j + k][sb] =
846  q->tone_level[1][sb][(j + k) / 2] * -samples[k];
847  else
848  q->sb_samples[1][j + k][sb] =
849  q->tone_level[1][sb][(j + k) / 2] * samples[k];
850  }
851  }
852  } else {
853  for (k = 0; k < run; k++)
854  if ((j + k) < 128)
855  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
856  }
857 
858  j += run;
859  } // j loop
860  } // channel loop
861  } // subband loop
862  return 0;
863 }
864 
865 /**
866  * Init the first element of a channel in quantized_coeffs with data
867  * from packet 10 (quantized_coeffs[ch][0]).
868  * This is similar to process_subpacket_9, but for a single channel
869  * and for element [0]
870  * same VLC tables as process_subpacket_9 are used.
871  *
872  * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
873  * @param gb bitreader context
874  */
875 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
876  GetBitContext *gb)
877 {
878  int i, k, run, level, diff;
879 
880  if (get_bits_left(gb) < 16)
881  return -1;
882  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
883 
884  quantized_coeffs[0] = level;
885 
886  for (i = 0; i < 7; ) {
887  if (get_bits_left(gb) < 16)
888  return -1;
889  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
890 
891  if (i + run >= 8)
892  return -1;
893 
894  if (get_bits_left(gb) < 16)
895  return -1;
896  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
897 
898  for (k = 1; k <= run; k++)
899  quantized_coeffs[i + k] = (level + ((k * diff) / run));
900 
901  level += diff;
902  i += run;
903  }
904  return 0;
905 }
906 
907 /**
908  * Related to synthesis filter, process data from packet 10
909  * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
910  * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
911  * data from packet 10
912  *
913  * @param q context
914  * @param gb bitreader context
915  */
917 {
918  int sb, j, k, n, ch;
919 
920  for (ch = 0; ch < q->nb_channels; ch++) {
922 
923  if (get_bits_left(gb) < 16) {
924  memset(q->quantized_coeffs[ch][0], 0, 8);
925  break;
926  }
927  }
928 
929  n = q->sub_sampling + 1;
930 
931  for (sb = 0; sb < n; sb++)
932  for (ch = 0; ch < q->nb_channels; ch++)
933  for (j = 0; j < 8; j++) {
934  if (get_bits_left(gb) < 1)
935  break;
936  if (get_bits1(gb)) {
937  for (k=0; k < 8; k++) {
938  if (get_bits_left(gb) < 16)
939  break;
940  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
941  }
942  } else {
943  for (k=0; k < 8; k++)
944  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
945  }
946  }
947 
948  n = QDM2_SB_USED(q->sub_sampling) - 4;
949 
950  for (sb = 0; sb < n; sb++)
951  for (ch = 0; ch < q->nb_channels; ch++) {
952  if (get_bits_left(gb) < 16)
953  break;
955  if (sb > 19)
956  q->tone_level_idx_hi2[ch][sb] -= 16;
957  else
958  for (j = 0; j < 8; j++)
959  q->tone_level_idx_mid[ch][sb][j] = -16;
960  }
961 
962  n = QDM2_SB_USED(q->sub_sampling) - 5;
963 
964  for (sb = 0; sb < n; sb++)
965  for (ch = 0; ch < q->nb_channels; ch++)
966  for (j = 0; j < 8; j++) {
967  if (get_bits_left(gb) < 16)
968  break;
969  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
970  }
971 }
972 
973 /**
974  * Process subpacket 9, init quantized_coeffs with data from it
975  *
976  * @param q context
977  * @param node pointer to node with packet
978  */
980 {
981  GetBitContext gb;
982  int i, j, k, n, ch, run, level, diff;
983 
984  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
985 
987 
988  for (i = 1; i < n; i++)
989  for (ch = 0; ch < q->nb_channels; ch++) {
990  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
991  q->quantized_coeffs[ch][i][0] = level;
992 
993  for (j = 0; j < (8 - 1); ) {
994  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
995  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
996 
997  if (j + run >= 8)
998  return -1;
999 
1000  for (k = 1; k <= run; k++)
1001  q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1002 
1003  level += diff;
1004  j += run;
1005  }
1006  }
1007 
1008  for (ch = 0; ch < q->nb_channels; ch++)
1009  for (i = 0; i < 8; i++)
1010  q->quantized_coeffs[ch][0][i] = 0;
1011 
1012  return 0;
1013 }
1014 
1015 /**
1016  * Process subpacket 10 if not null, else
1017  *
1018  * @param q context
1019  * @param node pointer to node with packet
1020  */
1022 {
1023  GetBitContext gb;
1024 
1025  if (node) {
1026  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1028  fill_tone_level_array(q, 1);
1029  } else {
1030  fill_tone_level_array(q, 0);
1031  }
1032 }
1033 
1034 /**
1035  * Process subpacket 11
1036  *
1037  * @param q context
1038  * @param node pointer to node with packet
1039  */
1041 {
1042  GetBitContext gb;
1043  int length = 0;
1044 
1045  if (node) {
1046  length = node->packet->size * 8;
1047  init_get_bits(&gb, node->packet->data, length);
1048  }
1049 
1050  if (length >= 32) {
1051  int c = get_bits(&gb, 13);
1052 
1053  if (c > 3)
1056  q->nb_channels, 8 * c,
1058  }
1059 
1060  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1061 }
1062 
1063 /**
1064  * Process subpacket 12
1065  *
1066  * @param q context
1067  * @param node pointer to node with packet
1068  */
1070 {
1071  GetBitContext gb;
1072  int length = 0;
1073 
1074  if (node) {
1075  length = node->packet->size * 8;
1076  init_get_bits(&gb, node->packet->data, length);
1077  }
1078 
1079  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1080 }
1081 
1082 /**
1083  * Process new subpackets for synthesis filter
1084  *
1085  * @param q context
1086  * @param list list with synthesis filter packets (list D)
1087  */
1089 {
1090  QDM2SubPNode *nodes[4];
1091 
1092  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1093  if (nodes[0])
1094  process_subpacket_9(q, nodes[0]);
1095 
1096  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1097  if (nodes[1])
1098  process_subpacket_10(q, nodes[1]);
1099  else
1101 
1102  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1103  if (nodes[0] && nodes[1] && nodes[2])
1104  process_subpacket_11(q, nodes[2]);
1105  else
1107 
1108  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1109  if (nodes[0] && nodes[1] && nodes[3])
1110  process_subpacket_12(q, nodes[3]);
1111  else
1113 }
1114 
1115 /**
1116  * Decode superblock, fill packet lists.
1117  *
1118  * @param q context
1119  */
1121 {
1122  GetBitContext gb;
1123  QDM2SubPacket header, *packet;
1124  int i, packet_bytes, sub_packet_size, sub_packets_D;
1125  unsigned int next_index = 0;
1126 
1127  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1128  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1129  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1130 
1131  q->sub_packets_B = 0;
1132  sub_packets_D = 0;
1133 
1134  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1135 
1138 
1139  if (header.type < 2 || header.type >= 8) {
1140  q->has_errors = 1;
1141  av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1142  return;
1143  }
1144 
1145  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1146  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1147 
1148  init_get_bits(&gb, header.data, header.size * 8);
1149 
1150  if (header.type == 2 || header.type == 4 || header.type == 5) {
1151  int csum = 257 * get_bits(&gb, 8);
1152  csum += 2 * get_bits(&gb, 8);
1153 
1154  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1155 
1156  if (csum != 0) {
1157  q->has_errors = 1;
1158  av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1159  return;
1160  }
1161  }
1162 
1163  q->sub_packet_list_B[0].packet = NULL;
1164  q->sub_packet_list_D[0].packet = NULL;
1165 
1166  for (i = 0; i < 6; i++)
1167  if (--q->fft_level_exp[i] < 0)
1168  q->fft_level_exp[i] = 0;
1169 
1170  for (i = 0; packet_bytes > 0; i++) {
1171  int j;
1172 
1173  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1174  SAMPLES_NEEDED_2("too many packet bytes");
1175  return;
1176  }
1177 
1178  q->sub_packet_list_A[i].next = NULL;
1179 
1180  if (i > 0) {
1181  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1182 
1183  /* seek to next block */
1184  init_get_bits(&gb, header.data, header.size * 8);
1185  skip_bits(&gb, next_index * 8);
1186 
1187  if (next_index >= header.size)
1188  break;
1189  }
1190 
1191  /* decode subpacket */
1192  packet = &q->sub_packets[i];
1193  qdm2_decode_sub_packet_header(&gb, packet);
1194  next_index = packet->size + get_bits_count(&gb) / 8;
1195  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1196 
1197  if (packet->type == 0)
1198  break;
1199 
1200  if (sub_packet_size > packet_bytes) {
1201  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1202  break;
1203  packet->size += packet_bytes - sub_packet_size;
1204  }
1205 
1206  packet_bytes -= sub_packet_size;
1207 
1208  /* add subpacket to 'all subpackets' list */
1209  q->sub_packet_list_A[i].packet = packet;
1210 
1211  /* add subpacket to related list */
1212  if (packet->type == 8) {
1213  SAMPLES_NEEDED_2("packet type 8");
1214  return;
1215  } else if (packet->type >= 9 && packet->type <= 12) {
1216  /* packets for MPEG Audio like Synthesis Filter */
1217  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1218  } else if (packet->type == 13) {
1219  for (j = 0; j < 6; j++)
1220  q->fft_level_exp[j] = get_bits(&gb, 6);
1221  } else if (packet->type == 14) {
1222  for (j = 0; j < 6; j++)
1223  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1224  } else if (packet->type == 15) {
1225  SAMPLES_NEEDED_2("packet type 15")
1226  return;
1227  } else if (packet->type >= 16 && packet->type < 48 &&
1228  !fft_subpackets[packet->type - 16]) {
1229  /* packets for FFT */
1231  }
1232  } // Packet bytes loop
1233 
1234  if (q->sub_packet_list_D[0].packet) {
1236  q->do_synth_filter = 1;
1237  } else if (q->do_synth_filter) {
1241  }
1242 }
1243 
1244 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1245  int offset, int duration, int channel,
1246  int exp, int phase)
1247 {
1248  if (q->fft_coefs_min_index[duration] < 0)
1250 
1252  ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1255  q->fft_coefs[q->fft_coefs_index].exp = exp;
1256  q->fft_coefs[q->fft_coefs_index].phase = phase;
1257  q->fft_coefs_index++;
1258 }
1259 
1261  GetBitContext *gb, int b)
1262 {
1263  int channel, stereo, phase, exp;
1264  int local_int_4, local_int_8, stereo_phase, local_int_10;
1265  int local_int_14, stereo_exp, local_int_20, local_int_28;
1266  int n, offset;
1267 
1268  local_int_4 = 0;
1269  local_int_28 = 0;
1270  local_int_20 = 2;
1271  local_int_8 = (4 - duration);
1272  local_int_10 = 1 << (q->group_order - duration - 1);
1273  offset = 1;
1274 
1275  while (get_bits_left(gb)>0) {
1276  if (q->superblocktype_2_3) {
1277  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1278  if (get_bits_left(gb)<0) {
1279  if(local_int_4 < q->group_size)
1280  av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1281  return;
1282  }
1283  offset = 1;
1284  if (n == 0) {
1285  local_int_4 += local_int_10;
1286  local_int_28 += (1 << local_int_8);
1287  } else {
1288  local_int_4 += 8 * local_int_10;
1289  local_int_28 += (8 << local_int_8);
1290  }
1291  }
1292  offset += (n - 2);
1293  } else {
1294  if (local_int_10 <= 2) {
1295  av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
1296  return;
1297  }
1298  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1299  while (offset >= (local_int_10 - 1)) {
1300  offset += (1 - (local_int_10 - 1));
1301  local_int_4 += local_int_10;
1302  local_int_28 += (1 << local_int_8);
1303  }
1304  }
1305 
1306  if (local_int_4 >= q->group_size)
1307  return;
1308 
1309  local_int_14 = (offset >> local_int_8);
1310  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1311  return;
1312 
1313  if (q->nb_channels > 1) {
1314  channel = get_bits1(gb);
1315  stereo = get_bits1(gb);
1316  } else {
1317  channel = 0;
1318  stereo = 0;
1319  }
1320 
1322  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1323  exp = (exp < 0) ? 0 : exp;
1324 
1325  phase = get_bits(gb, 3);
1326  stereo_exp = 0;
1327  stereo_phase = 0;
1328 
1329  if (stereo) {
1330  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1331  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1332  if (stereo_phase < 0)
1333  stereo_phase += 8;
1334  }
1335 
1336  if (q->frequency_range > (local_int_14 + 1)) {
1337  int sub_packet = (local_int_20 + local_int_28);
1338 
1339  if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
1340  return;
1341 
1342  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1343  channel, exp, phase);
1344  if (stereo)
1345  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1346  1 - channel,
1347  stereo_exp, stereo_phase);
1348  }
1349  offset++;
1350  }
1351 }
1352 
1354 {
1355  int i, j, min, max, value, type, unknown_flag;
1356  GetBitContext gb;
1357 
1358  if (!q->sub_packet_list_B[0].packet)
1359  return;
1360 
1361  /* reset minimum indexes for FFT coefficients */
1362  q->fft_coefs_index = 0;
1363  for (i = 0; i < 5; i++)
1364  q->fft_coefs_min_index[i] = -1;
1365 
1366  /* process subpackets ordered by type, largest type first */
1367  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1368  QDM2SubPacket *packet = NULL;
1369 
1370  /* find subpacket with largest type less than max */
1371  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1373  if (value > min && value < max) {
1374  min = value;
1375  packet = q->sub_packet_list_B[j].packet;
1376  }
1377  }
1378 
1379  max = min;
1380 
1381  /* check for errors (?) */
1382  if (!packet)
1383  return;
1384 
1385  if (i == 0 &&
1386  (packet->type < 16 || packet->type >= 48 ||
1387  fft_subpackets[packet->type - 16]))
1388  return;
1389 
1390  /* decode FFT tones */
1391  init_get_bits(&gb, packet->data, packet->size * 8);
1392 
1393  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1394  unknown_flag = 1;
1395  else
1396  unknown_flag = 0;
1397 
1398  type = packet->type;
1399 
1400  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1401  int duration = q->sub_sampling + 5 - (type & 15);
1402 
1403  if (duration >= 0 && duration < 4)
1404  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1405  } else if (type == 31) {
1406  for (j = 0; j < 4; j++)
1407  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1408  } else if (type == 46) {
1409  for (j = 0; j < 6; j++)
1410  q->fft_level_exp[j] = get_bits(&gb, 6);
1411  for (j = 0; j < 4; j++)
1412  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1413  }
1414  } // Loop on B packets
1415 
1416  /* calculate maximum indexes for FFT coefficients */
1417  for (i = 0, j = -1; i < 5; i++)
1418  if (q->fft_coefs_min_index[i] >= 0) {
1419  if (j >= 0)
1421  j = i;
1422  }
1423  if (j >= 0)
1425 }
1426 
1428 {
1429  float level, f[6];
1430  int i;
1431  QDM2Complex c;
1432  const double iscale = 2.0 * M_PI / 512.0;
1433 
1434  tone->phase += tone->phase_shift;
1435 
1436  /* calculate current level (maximum amplitude) of tone */
1437  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1438  c.im = level * sin(tone->phase * iscale);
1439  c.re = level * cos(tone->phase * iscale);
1440 
1441  /* generate FFT coefficients for tone */
1442  if (tone->duration >= 3 || tone->cutoff >= 3) {
1443  tone->complex[0].im += c.im;
1444  tone->complex[0].re += c.re;
1445  tone->complex[1].im -= c.im;
1446  tone->complex[1].re -= c.re;
1447  } else {
1448  f[1] = -tone->table[4];
1449  f[0] = tone->table[3] - tone->table[0];
1450  f[2] = 1.0 - tone->table[2] - tone->table[3];
1451  f[3] = tone->table[1] + tone->table[4] - 1.0;
1452  f[4] = tone->table[0] - tone->table[1];
1453  f[5] = tone->table[2];
1454  for (i = 0; i < 2; i++) {
1455  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1456  c.re * f[i];
1457  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1458  c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1459  }
1460  for (i = 0; i < 4; i++) {
1461  tone->complex[i].re += c.re * f[i + 2];
1462  tone->complex[i].im += c.im * f[i + 2];
1463  }
1464  }
1465 
1466  /* copy the tone if it has not yet died out */
1467  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1468  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1469  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1470  }
1471 }
1472 
1473 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1474 {
1475  int i, j, ch;
1476  const double iscale = 0.25 * M_PI;
1477 
1478  for (ch = 0; ch < q->channels; ch++) {
1479  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1480  }
1481 
1482 
1483  /* apply FFT tones with duration 4 (1 FFT period) */
1484  if (q->fft_coefs_min_index[4] >= 0)
1485  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1486  float level;
1487  QDM2Complex c;
1488 
1489  if (q->fft_coefs[i].sub_packet != sub_packet)
1490  break;
1491 
1492  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1493  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];
1494 
1495  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1496  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1497  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1498  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1499  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1500  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1501  }
1502 
1503  /* generate existing FFT tones */
1504  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1506  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1507  }
1508 
1509  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1510  for (i = 0; i < 4; i++)
1511  if (q->fft_coefs_min_index[i] >= 0) {
1512  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1513  int offset, four_i;
1514  FFTTone tone;
1515 
1516  if (q->fft_coefs[j].sub_packet != sub_packet)
1517  break;
1518 
1519  four_i = (4 - i);
1520  offset = q->fft_coefs[j].offset >> four_i;
1521  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1522 
1523  if (offset < q->frequency_range) {
1524  if (offset < 2)
1525  tone.cutoff = offset;
1526  else
1527  tone.cutoff = (offset >= 60) ? 3 : 2;
1528 
1529  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];
1530  tone.complex = &q->fft.complex[ch][offset];
1531  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1532  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1533  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1534  tone.duration = i;
1535  tone.time_index = 0;
1536 
1537  qdm2_fft_generate_tone(q, &tone);
1538  }
1539  }
1540  q->fft_coefs_min_index[i] = j;
1541  }
1542 }
1543 
1544 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1545 {
1546  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1547  float *out = q->output_buffer + channel;
1548  int i;
1549  q->fft.complex[channel][0].re *= 2.0f;
1550  q->fft.complex[channel][0].im = 0.0f;
1552  /* add samples to output buffer */
1553  for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1554  out[0] += q->fft.complex[channel][i].re * gain;
1555  out[q->channels] += q->fft.complex[channel][i].im * gain;
1556  out += 2 * q->channels;
1557  }
1558 }
1559 
1560 /**
1561  * @param q context
1562  * @param index subpacket number
1563  */
1565 {
1566  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1567 
1568  /* copy sb_samples */
1569  sb_used = QDM2_SB_USED(q->sub_sampling);
1570 
1571  for (ch = 0; ch < q->channels; ch++)
1572  for (i = 0; i < 8; i++)
1573  for (k = sb_used; k < SBLIMIT; k++)
1574  q->sb_samples[ch][(8 * index) + i][k] = 0;
1575 
1576  for (ch = 0; ch < q->nb_channels; ch++) {
1577  float *samples_ptr = q->samples + ch;
1578 
1579  for (i = 0; i < 8; i++) {
1581  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1582  ff_mpa_synth_window_float, &dither_state,
1583  samples_ptr, q->nb_channels,
1584  q->sb_samples[ch][(8 * index) + i]);
1585  samples_ptr += 32 * q->nb_channels;
1586  }
1587  }
1588 
1589  /* add samples to output buffer */
1590  sub_sampling = (4 >> q->sub_sampling);
1591 
1592  for (ch = 0; ch < q->channels; ch++)
1593  for (i = 0; i < q->frame_size; i++)
1594  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1595 }
1596 
1597 /**
1598  * Init static data (does not depend on specific file)
1599  */
1600 static av_cold void qdm2_init_static_data(void) {
1601  qdm2_init_vlc();
1603  rnd_table_init();
1605 
1607 }
1608 
1609 /**
1610  * Init parameters from codec extradata
1611  */
1613 {
1614  static AVOnce init_static_once = AV_ONCE_INIT;
1615  QDM2Context *s = avctx->priv_data;
1616  int tmp_val, tmp, size;
1617  GetByteContext gb;
1618 
1619  /* extradata parsing
1620 
1621  Structure:
1622  wave {
1623  frma (QDM2)
1624  QDCA
1625  QDCP
1626  }
1627 
1628  32 size (including this field)
1629  32 tag (=frma)
1630  32 type (=QDM2 or QDMC)
1631 
1632  32 size (including this field, in bytes)
1633  32 tag (=QDCA) // maybe mandatory parameters
1634  32 unknown (=1)
1635  32 channels (=2)
1636  32 samplerate (=44100)
1637  32 bitrate (=96000)
1638  32 block size (=4096)
1639  32 frame size (=256) (for one channel)
1640  32 packet size (=1300)
1641 
1642  32 size (including this field, in bytes)
1643  32 tag (=QDCP) // maybe some tuneable parameters
1644  32 float1 (=1.0)
1645  32 zero ?
1646  32 float2 (=1.0)
1647  32 float3 (=1.0)
1648  32 unknown (27)
1649  32 unknown (8)
1650  32 zero ?
1651  */
1652 
1653  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1654  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1655  return AVERROR_INVALIDDATA;
1656  }
1657 
1658  bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1659 
1660  while (bytestream2_get_bytes_left(&gb) > 8) {
1661  if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1662  (uint64_t)MKBETAG('Q','D','M','2')))
1663  break;
1664  bytestream2_skip(&gb, 1);
1665  }
1666 
1667  if (bytestream2_get_bytes_left(&gb) < 12) {
1668  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1670  return AVERROR_INVALIDDATA;
1671  }
1672 
1673  bytestream2_skip(&gb, 8);
1674  size = bytestream2_get_be32(&gb);
1675 
1676  if (size > bytestream2_get_bytes_left(&gb)) {
1677  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1679  return AVERROR_INVALIDDATA;
1680  }
1681 
1682  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1683  if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1684  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1685  return AVERROR_INVALIDDATA;
1686  }
1687 
1688  bytestream2_skip(&gb, 4);
1689 
1690  avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1691  if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1692  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1693  return AVERROR_INVALIDDATA;
1694  }
1695  avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1697 
1698  avctx->sample_rate = bytestream2_get_be32(&gb);
1699  avctx->bit_rate = bytestream2_get_be32(&gb);
1700  s->group_size = bytestream2_get_be32(&gb);
1701  s->fft_size = bytestream2_get_be32(&gb);
1702  s->checksum_size = bytestream2_get_be32(&gb);
1703  if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
1704  av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
1705  return AVERROR_INVALIDDATA;
1706  }
1707 
1708  s->fft_order = av_log2(s->fft_size) + 1;
1709 
1710  // Fail on unknown fft order
1711  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1712  avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1713  return AVERROR_PATCHWELCOME;
1714  }
1715 
1716  // something like max decodable tones
1717  s->group_order = av_log2(s->group_size) + 1;
1718  s->frame_size = s->group_size / 16; // 16 iterations per super block
1719 
1720  if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1721  return AVERROR_INVALIDDATA;
1722 
1723  s->sub_sampling = s->fft_order - 7;
1724  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1725 
1726  if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
1727  avpriv_request_sample(avctx, "large frames");
1728  return AVERROR_PATCHWELCOME;
1729  }
1730 
1731  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1732  case 0: tmp = 40; break;
1733  case 1: tmp = 48; break;
1734  case 2: tmp = 56; break;
1735  case 3: tmp = 72; break;
1736  case 4: tmp = 80; break;
1737  case 5: tmp = 100;break;
1738  default: tmp=s->sub_sampling; break;
1739  }
1740  tmp_val = 0;
1741  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1742  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1743  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1744  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1745  s->cm_table_select = tmp_val;
1746 
1747  if (avctx->bit_rate <= 8000)
1748  s->coeff_per_sb_select = 0;
1749  else if (avctx->bit_rate < 16000)
1750  s->coeff_per_sb_select = 1;
1751  else
1752  s->coeff_per_sb_select = 2;
1753 
1754  if (s->fft_size != (1 << (s->fft_order - 1))) {
1755  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1756  return AVERROR_INVALIDDATA;
1757  }
1758 
1759  ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1760  ff_mpadsp_init(&s->mpadsp);
1761 
1762  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1763 
1764  ff_thread_once(&init_static_once, qdm2_init_static_data);
1765 
1766  return 0;
1767 }
1768 
1770 {
1771  QDM2Context *s = avctx->priv_data;
1772 
1773  ff_rdft_end(&s->rdft_ctx);
1774 
1775  return 0;
1776 }
1777 
1778 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1779 {
1780  int ch, i;
1781  const int frame_size = (q->frame_size * q->channels);
1782 
1783  if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1784  return -1;
1785 
1786  /* select input buffer */
1787  q->compressed_data = in;
1789 
1790  /* copy old block, clear new block of output samples */
1791  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1792  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1793 
1794  /* decode block of QDM2 compressed data */
1795  if (q->sub_packet == 0) {
1796  q->has_errors = 0; // zero it for a new super block
1797  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1799  }
1800 
1801  /* parse subpackets */
1802  if (!q->has_errors) {
1803  if (q->sub_packet == 2)
1805 
1807  }
1808 
1809  /* sound synthesis stage 1 (FFT) */
1810  for (ch = 0; ch < q->channels; ch++) {
1811  qdm2_calculate_fft(q, ch, q->sub_packet);
1812 
1813  if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1814  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1815  return -1;
1816  }
1817  }
1818 
1819  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1820  if (!q->has_errors && q->do_synth_filter)
1822 
1823  q->sub_packet = (q->sub_packet + 1) % 16;
1824 
1825  /* clip and convert output float[] to 16-bit signed samples */
1826  for (i = 0; i < frame_size; i++) {
1827  int value = (int)q->output_buffer[i];
1828 
1831  else if (value < -SOFTCLIP_THRESHOLD)
1833 
1834  out[i] = value;
1835  }
1836 
1837  return 0;
1838 }
1839 
1840 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1841  int *got_frame_ptr, AVPacket *avpkt)
1842 {
1843  AVFrame *frame = data;
1844  const uint8_t *buf = avpkt->data;
1845  int buf_size = avpkt->size;
1846  QDM2Context *s = avctx->priv_data;
1847  int16_t *out;
1848  int i, ret;
1849 
1850  if(!buf)
1851  return 0;
1852  if(buf_size < s->checksum_size)
1853  return -1;
1854 
1855  /* get output buffer */
1856  frame->nb_samples = 16 * s->frame_size;
1857  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1858  return ret;
1859  out = (int16_t *)frame->data[0];
1860 
1861  for (i = 0; i < 16; i++) {
1862  if ((ret = qdm2_decode(s, buf, out)) < 0)
1863  return ret;
1864  out += s->channels * s->frame_size;
1865  }
1866 
1867  *got_frame_ptr = 1;
1868 
1869  return s->checksum_size;
1870 }
1871 
1873  .name = "qdm2",
1874  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1875  .type = AVMEDIA_TYPE_AUDIO,
1876  .id = AV_CODEC_ID_QDM2,
1877  .priv_data_size = sizeof(QDM2Context),
1879  .close = qdm2_decode_close,
1881  .capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1882  .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
1883 };
channels
Definition: aptx.h:33
#define av_cold
Definition: attributes.h:88
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
uint8_t
Libavcodec external API header.
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:31
static av_always_inline int bytestream2_get_bytes_left(GetByteContext *g)
Definition: bytestream.h:158
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:137
static av_always_inline void bytestream2_skip(GetByteContext *g, unsigned int size)
Definition: bytestream.h:168
#define flag(name)
Definition: cbs_av1.c:553
#define s(width, name)
Definition: cbs_vp9.c:257
#define f(width, name)
Definition: cbs_vp9.c:255
int nb_channels
audio channel layout utility functions
#define MKBETAG(a, b, c, d)
Definition: common.h:479
#define NULL
Definition: coverity.c:32
#define max(a, b)
Definition: cuda_runtime.h:33
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1900
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:71
static AVFrame * frame
static void comp(unsigned char *dst, ptrdiff_t dst_stride, unsigned char *src, ptrdiff_t src_stride, int add)
Definition: eamad.c:85
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
double value
Definition: eval.c:98
int8_t exp
Definition: eval.c:72
int
bitstream reader API header.
static av_always_inline int get_vlc2(GetBitContext *s, VLC_TYPE(*table)[2], int bits, int max_depth)
Parse a vlc code.
Definition: get_bits.h:797
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:849
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:498
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:467
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:219
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:379
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:659
#define AV_CH_LAYOUT_MONO
#define AV_CH_LAYOUT_STEREO
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() or get_encode_buffer() for allocating buffers and supports custom allocators.
Definition: codec.h:52
#define AV_CODEC_CAP_CHANNEL_CONF
Codec should fill in channel configuration and samplerate instead of container.
Definition: codec.h:104
@ AV_CODEC_ID_QDM2
Definition: codec_id.h:443
float FFTSample
Definition: avfft.h:35
@ IDFT_C2R
Definition: avfft.h:73
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:215
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:194
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:117
@ AVMEDIA_TYPE_AUDIO
Definition: avutil.h:202
@ AV_SAMPLE_FMT_S16
signed 16 bits
Definition: samplefmt.h:61
int index
Definition: gxfenc.c:89
for(j=16;j >0;--j)
cl_device_type type
int i
Definition: input.c:407
#define av_log2
Definition: intmath.h:83
static const uint8_t dequant_table[64]
Definition: 4xm.c:115
#define FF_CODEC_CAP_INIT_THREADSAFE
The codec does not modify any global variables in the init function, allowing to call the init functi...
Definition: internal.h:41
common internal API header
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.
Definition: internal.h:117
#define AVOnce
Definition: thread.h:172
static int ff_thread_once(char *control, void(*routine)(void))
Definition: thread.h:175
#define AV_ONCE_INIT
Definition: thread.h:173
#define FFALIGN(x, a)
Definition: macros.h:48
#define M_PI
Definition: mathematics.h:52
mpeg audio declarations for both encoder and decoder.
#define MPA_FRAME_SIZE
Definition: mpegaudio.h:37
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:42
#define SBLIMIT
Definition: mpegaudio.h:44
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:81
float ff_mpa_synth_window_float[]
void ff_mpa_synth_init_float(void)
void ff_mpa_synth_filter_float(MPADSPContext *s, float *synth_buf_ptr, int *synth_buf_offset, float *window, int *dither_state, float *samples, ptrdiff_t incr, float *sb_samples)
const char data[16]
Definition: mxf.c:142
int frame_size
Definition: mxfenc.c:2206
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 10 if not null, else.
Definition: qdm2.c:1021
static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
QDM2 checksum.
Definition: qdm2.c:248
static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 9, init quantized_coeffs with data from it.
Definition: qdm2.c:979
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 12.
Definition: qdm2.c:1069
#define QDM2_SB_USED(sub_sampling)
Definition: qdm2.c:64
static int fix_coding_method_array(int sb, int channels, sb_int8_array coding_method)
Called while processing data from subpackets 11 and 12.
Definition: qdm2.c:372
#define SB_DITHERING_NOISE(sb, noise_idx)
Definition: qdm2.c:70
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
Definition: qdm2.c:1544
#define SAMPLES_NEEDED_2(why)
Definition: qdm2.c:75
#define QDM2_LIST_ADD(list, size, packet)
Definition: qdm2.c:53
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 11.
Definition: qdm2.c:1040
static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
Related to synthesis filter, process data from packet 10 Init part of quantized_coeffs via function i...
Definition: qdm2.c:916
static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: qdm2.c:1840
static void qdm2_decode_fft_packets(QDM2Context *q)
Definition: qdm2.c:1353
static void average_quantized_coeffs(QDM2Context *q)
Replace 8 elements with their average value.
Definition: qdm2.c:316
#define FIX_NOISE_IDX(noise_idx)
Definition: qdm2.c:66
static av_cold void qdm2_init_static_data(void)
Init static data (does not depend on specific file)
Definition: qdm2.c:1600
static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs, GetBitContext *gb)
Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch...
Definition: qdm2.c:875
static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
Definition: qdm2.c:203
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
Init parameters from codec extradata.
Definition: qdm2.c:1612
AVCodec ff_qdm2_decoder
Definition: qdm2.c:1872
static void qdm2_fft_decode_tones(QDM2Context *q, int duration, GetBitContext *gb, int b)
Definition: qdm2.c:1260
#define QDM2_MAX_FRAME_SIZE
Definition: qdm2.c:78
static const int switchtable[23]
Definition: qdm2.c:199
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, int offset, int duration, int channel, int exp, int phase)
Definition: qdm2.c:1244
int8_t sb_int8_array[2][30][64]
Definition: qdm2.c:80
static void fill_tone_level_array(QDM2Context *q, int flag)
Related to synthesis filter Called by process_subpacket_10.
Definition: qdm2.c:443
static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
Definition: qdm2.c:232
#define SAMPLES_NEEDED
Definition: qdm2.c:72
static void qdm2_synthesis_filter(QDM2Context *q, int index)
Definition: qdm2.c:1564
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
Definition: qdm2.c:1427
static void qdm2_decode_sub_packet_header(GetBitContext *gb, QDM2SubPacket *sub_packet)
Fill a QDM2SubPacket structure with packet type, size, and data pointer.
Definition: qdm2.c:264
static void build_sb_samples_from_noise(QDM2Context *q, int sb)
Build subband samples with noise weighted by q->tone_level.
Definition: qdm2.c:345
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
Process new subpackets for synthesis filter.
Definition: qdm2.c:1088
static void fill_coding_method_array(sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp, sb_int8_array coding_method, int nb_channels, int c, int superblocktype_2_3, int cm_table_select)
Related to synthesis filter Called by process_subpacket_11 c is built with data from subpacket 11 Mos...
Definition: qdm2.c:532
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
Definition: qdm2.c:1473
static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8.
Definition: qdm2.c:659
static void qdm2_decode_super_block(QDM2Context *q)
Decode superblock, fill packet lists.
Definition: qdm2.c:1120
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
Definition: qdm2.c:1778
static QDM2SubPNode * qdm2_search_subpacket_type_in_list(QDM2SubPNode *list, int type)
Return node pointer to first packet of requested type in list.
Definition: qdm2.c:299
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
Definition: qdm2.c:1769
#define SOFTCLIP_THRESHOLD
Definition: qdm2_tablegen.h:31
static VLC vlc_tab_diff
Definition: qdm2_tablegen.h:99
static av_cold void init_noise_samples(void)
Definition: qdm2_tablegen.h:88
static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD+1]
Definition: qdm2_tablegen.h:41
static float noise_samples[128]
Definition: qdm2_tablegen.h:45
static uint8_t random_dequant_index[256][5]
Definition: qdm2_tablegen.h:43
static VLC fft_stereo_exp_vlc
static VLC fft_stereo_phase_vlc
static VLC vlc_tab_run
static VLC vlc_tab_fft_tone_offset[5]
static VLC vlc_tab_level
Definition: qdm2_tablegen.h:98
static VLC vlc_tab_type34
static VLC vlc_tab_tone_level_idx_mid
static VLC fft_level_exp_alt_vlc
static VLC vlc_tab_type30
static av_cold void qdm2_init_vlc(void)
static av_cold void rnd_table_init(void)
Definition: qdm2_tablegen.h:57
static uint8_t random_dequant_type24[128][3]
Definition: qdm2_tablegen.h:44
static av_cold void softclip_table_init(void)
Definition: qdm2_tablegen.h:47
static VLC fft_level_exp_vlc
static VLC vlc_tab_tone_level_idx_hi2
#define HARDCLIP_THRESHOLD
Definition: qdm2_tablegen.h:32
static VLC vlc_tab_tone_level_idx_hi1
static const int16_t fft_level_index_table[256]
Definition: qdm2data.h:168
static const uint8_t fft_subpackets[32]
Definition: qdm2data.h:440
static const int vlc_stage3_values[60]
Definition: qdm2data.h:290
static const int fft_cutoff_index_table[4][2]
Definition: qdm2data.h:164
static const float fft_tone_envelope_table[4][31]
Definition: qdm2data.h:406
static const int8_t tone_level_idx_offset_table[30][4]
Definition: qdm2data.h:237
static const uint8_t coeff_per_sb_for_dequant[3][30]
Definition: qdm2data.h:230
static const uint8_t coeff_per_sb_for_avg[3][30]
Definition: qdm2data.h:191
static const float fft_tone_sample_table[4][16][5]
Definition: qdm2data.h:298
static const float type34_delta[10]
Definition: qdm2data.h:456
static const float type30_dequant[8]
Definition: qdm2data.h:451
static const float dequant_1bit[2][3]
Definition: qdm2data.h:446
static const uint8_t last_coeff[3]
Definition: qdm2data.h:187
static const float fft_tone_level_table[2][64]
Definition: qdm2data.h:368
static const int8_t coding_method_table[5][30]
Definition: qdm2data.h:272
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
Definition: rdft.c:88
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:114
static const uint8_t header[24]
Definition: sdr2.c:67
#define FF_ARRAY_ELEMS(a)
main external API structure.
Definition: avcodec.h:536
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:1204
int64_t bit_rate
the average bitrate
Definition: avcodec.h:586
int sample_rate
samples per second
Definition: avcodec.h:1196
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:637
int channels
number of audio channels
Definition: avcodec.h:1197
int extradata_size
Definition: avcodec.h:638
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:1247
void * priv_data
Definition: avcodec.h:563
AVCodec.
Definition: codec.h:197
const char * name
Name of the codec implementation.
Definition: codec.h:204
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:384
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:332
This structure stores compressed data.
Definition: packet.h:346
int size
Definition: packet.h:370
uint8_t * data
Definition: packet.h:369
int16_t offset
Definition: qdm2.c:118
uint8_t channel
Definition: qdm2.c:117
uint8_t phase
Definition: qdm2.c:120
int16_t sub_packet
Definition: qdm2.c:116
int16_t exp
Definition: qdm2.c:119
Definition: qdm2.c:104
int duration
Definition: qdm2.c:110
int phase
Definition: qdm2.c:108
const float * table
Definition: qdm2.c:107
float level
Definition: qdm2.c:105
short cutoff
Definition: qdm2.c:112
int phase_shift
Definition: qdm2.c:109
short time_index
Definition: qdm2.c:111
QDM2Complex * complex
Definition: qdm2.c:106
const uint8_t * buffer
Definition: get_bits.h:62
float im
Definition: qdm2.c:101
float re
Definition: qdm2.c:100
QDM2 decoder context.
Definition: qdm2.c:130
int coeff_per_sb_select
selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
Definition: qdm2.c:144
FFTTone fft_tones[1000]
FFT and tones.
Definition: qdm2.c:156
int do_synth_filter
used to perform or skip synthesis filter
Definition: qdm2.c:193
int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]
Definition: qdm2.c:183
int superblocktype_2_3
select fft tables and some algorithm based on superblock type
Definition: qdm2.c:192
int fft_tone_end
Definition: qdm2.c:158
float output_buffer[QDM2_MAX_FRAME_SIZE *MPA_MAX_CHANNELS *2]
Definition: qdm2.c:170
float tone_level[MPA_MAX_CHANNELS][30][64]
Mixed temporary data used in decoding.
Definition: qdm2.c:180
int synth_buf_offset[MPA_MAX_CHANNELS]
Definition: qdm2.c:175
int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]
Definition: qdm2.c:185
int fft_size
size of FFT, in complex numbers
Definition: qdm2.c:135
float synth_buf[MPA_MAX_CHANNELS][512 *2]
Definition: qdm2.c:174
QDM2SubPNode sub_packet_list_A[16]
list of all packets
Definition: qdm2.c:149
QDM2FFT fft
Definition: qdm2.c:165
int channels
number of channels
Definition: qdm2.c:133
int8_t coding_method[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:181
float sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]
Definition: qdm2.c:176
int noise_idx
index for dithering noise table
Definition: qdm2.c:196
int fft_tone_start
Definition: qdm2.c:157
MPADSPContext mpadsp
Synthesis filter.
Definition: qdm2.c:173
float samples[MPA_MAX_CHANNELS *MPA_FRAME_SIZE]
Definition: qdm2.c:177
const uint8_t * compressed_data
I/O data.
Definition: qdm2.c:168
int frequency_range
Definition: qdm2.c:142
int fft_coefs_min_index[5]
Definition: qdm2.c:161
int sub_packets_B
number of packets on 'B' list
Definition: qdm2.c:151
int fft_order
order of FFT (actually fftorder+1)
Definition: qdm2.c:140
int group_order
Parameters built from header parameters, do not change during playback.
Definition: qdm2.c:139
int fft_level_exp[6]
Definition: qdm2.c:163
FFTCoefficient fft_coefs[1000]
Definition: qdm2.c:159
int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]
Definition: qdm2.c:184
int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]
Definition: qdm2.c:186
int frame_size
size of data frame
Definition: qdm2.c:141
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:188
int compressed_size
Definition: qdm2.c:169
int fft_coefs_index
Definition: qdm2.c:160
RDFTContext rdft_ctx
Definition: qdm2.c:164
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]
Definition: qdm2.c:182
int fft_coefs_max_index[5]
Definition: qdm2.c:162
QDM2SubPNode sub_packet_list_D[16]
DCT packets.
Definition: qdm2.c:153
int sub_sampling
subsampling: 0=25%, 1=50%, 2=100% *‍/
Definition: qdm2.c:143
QDM2SubPNode sub_packet_list_C[16]
packets with errors?
Definition: qdm2.c:152
int nb_channels
Parameters from codec header, do not change during playback.
Definition: qdm2.c:132
QDM2SubPacket sub_packets[16]
Packets and packet lists.
Definition: qdm2.c:148
int cm_table_select
selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
Definition: qdm2.c:145
int checksum_size
size of data block, used also for checksum
Definition: qdm2.c:136
int has_errors
packet has errors
Definition: qdm2.c:191
int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:187
QDM2SubPNode sub_packet_list_B[16]
FFT packets B are on list.
Definition: qdm2.c:150
int group_size
size of frame group (16 frames per group)
Definition: qdm2.c:134
int sub_packet
Definition: qdm2.c:195
Definition: qdm2.c:123
QDM2Complex complex[MPA_MAX_CHANNELS][256]
Definition: qdm2.c:124
A node in the subpacket list.
Definition: qdm2.c:94
QDM2SubPacket * packet
packet
Definition: qdm2.c:95
struct QDM2SubPNode * next
pointer to next packet in the list, NULL if leaf node
Definition: qdm2.c:96
Subpacket.
Definition: qdm2.c:85
int type
subpacket type
Definition: qdm2.c:86
const uint8_t * data
pointer to subpacket data (points to input data buffer, it's not a private copy)
Definition: qdm2.c:88
unsigned int size
subpacket size
Definition: qdm2.c:87
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:38
Definition: vlc.h:26
int bits
Definition: vlc.h:27
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
uint8_t run
Definition: svq3.c:205
uint8_t level
Definition: svq3.c:206
#define avpriv_request_sample(...)
#define av_log(a,...)
static uint8_t tmp[11]
Definition: aes_ctr.c:27
FILE * out
Definition: movenc.c:54
int64_t duration
Definition: movenc.c:64
int size
static const struct twinvq_data tab
const char * b
Definition: vf_curves.c:118
if(ret< 0)
Definition: vf_mcdeint.c:282
static av_always_inline int diff(const uint32_t a, const uint32_t b)
static const uint8_t offset[127][2]
Definition: vf_spp.c:107
float min
static double c[64]
int acc
Definition: yuv2rgb.c:555