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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 
40 #define BITSTREAM_READER_LE
41 #include "avcodec.h"
42 #include "get_bits.h"
43 #include "bytestream.h"
44 #include "internal.h"
45 #include "mpegaudio.h"
46 #include "mpegaudiodsp.h"
47 #include "rdft.h"
48 
49 #include "qdm2_tablegen.h"
50 
51 #define QDM2_LIST_ADD(list, size, packet) \
52 do { \
53  if (size > 0) { \
54  list[size - 1].next = &list[size]; \
55  } \
56  list[size].packet = packet; \
57  list[size].next = NULL; \
58  size++; \
59 } while(0)
60 
61 // Result is 8, 16 or 30
62 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
63 
64 #define FIX_NOISE_IDX(noise_idx) \
65  if ((noise_idx) >= 3840) \
66  (noise_idx) -= 3840; \
67 
68 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
69 
70 #define SAMPLES_NEEDED \
71  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
72 
73 #define SAMPLES_NEEDED_2(why) \
74  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
75 
76 #define QDM2_MAX_FRAME_SIZE 512
77 
78 typedef int8_t sb_int8_array[2][30][64];
79 
80 /**
81  * Subpacket
82  */
83 typedef struct QDM2SubPacket {
84  int type; ///< subpacket type
85  unsigned int size; ///< subpacket size
86  const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
88 
89 /**
90  * A node in the subpacket list
91  */
92 typedef struct QDM2SubPNode {
93  QDM2SubPacket *packet; ///< packet
94  struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
95 } QDM2SubPNode;
96 
97 typedef struct QDM2Complex {
98  float re;
99  float im;
100 } QDM2Complex;
101 
102 typedef struct FFTTone {
103  float level;
105  const float *table;
106  int phase;
108  int duration;
109  short time_index;
110  short cutoff;
111 } FFTTone;
112 
113 typedef struct FFTCoefficient {
114  int16_t sub_packet;
116  int16_t offset;
117  int16_t exp;
120 
121 typedef struct QDM2FFT {
123 } QDM2FFT;
124 
125 /**
126  * QDM2 decoder context
127  */
128 typedef struct QDM2Context {
129  /// Parameters from codec header, do not change during playback
130  int nb_channels; ///< number of channels
131  int channels; ///< number of channels
132  int group_size; ///< size of frame group (16 frames per group)
133  int fft_size; ///< size of FFT, in complex numbers
134  int checksum_size; ///< size of data block, used also for checksum
135 
136  /// Parameters built from header parameters, do not change during playback
137  int group_order; ///< order of frame group
138  int fft_order; ///< order of FFT (actually fftorder+1)
139  int frame_size; ///< size of data frame
141  int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
142  int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
143  int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
144 
145  /// Packets and packet lists
146  QDM2SubPacket sub_packets[16]; ///< the packets themselves
147  QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
148  QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
149  int sub_packets_B; ///< number of packets on 'B' list
150  QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
151  QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
152 
153  /// FFT and tones
164 
165  /// I/O data
169 
170  /// Synthesis filter
176 
177  /// Mixed temporary data used in decoding
178  float tone_level[MPA_MAX_CHANNELS][30][64];
187 
188  // Flags
189  int has_errors; ///< packet has errors
190  int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
191  int do_synth_filter; ///< used to perform or skip synthesis filter
192 
194  int noise_idx; ///< index for dithering noise table
195 } QDM2Context;
196 
197 static const int switchtable[23] = {
198  0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
199 };
200 
201 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
202 {
203  int value;
204 
205  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
206 
207  /* stage-2, 3 bits exponent escape sequence */
208  if (value-- == 0)
209  value = get_bits(gb, get_bits(gb, 3) + 1);
210 
211  /* stage-3, optional */
212  if (flag) {
213  int tmp;
214 
215  if (value >= 60) {
216  av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
217  return 0;
218  }
219 
220  tmp= vlc_stage3_values[value];
221 
222  if ((value & ~3) > 0)
223  tmp += get_bits(gb, (value >> 2));
224  value = tmp;
225  }
226 
227  return value;
228 }
229 
230 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
231 {
232  int value = qdm2_get_vlc(gb, vlc, 0, depth);
233 
234  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
235 }
236 
237 /**
238  * QDM2 checksum
239  *
240  * @param data pointer to data to be checksummed
241  * @param length data length
242  * @param value checksum value
243  *
244  * @return 0 if checksum is OK
245  */
246 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
247 {
248  int i;
249 
250  for (i = 0; i < length; i++)
251  value -= data[i];
252 
253  return (uint16_t)(value & 0xffff);
254 }
255 
256 /**
257  * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
258  *
259  * @param gb bitreader context
260  * @param sub_packet packet under analysis
261  */
263  QDM2SubPacket *sub_packet)
264 {
265  sub_packet->type = get_bits(gb, 8);
266 
267  if (sub_packet->type == 0) {
268  sub_packet->size = 0;
269  sub_packet->data = NULL;
270  } else {
271  sub_packet->size = get_bits(gb, 8);
272 
273  if (sub_packet->type & 0x80) {
274  sub_packet->size <<= 8;
275  sub_packet->size |= get_bits(gb, 8);
276  sub_packet->type &= 0x7f;
277  }
278 
279  if (sub_packet->type == 0x7f)
280  sub_packet->type |= (get_bits(gb, 8) << 8);
281 
282  // FIXME: this depends on bitreader-internal data
283  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
284  }
285 
286  av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
287  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
288 }
289 
290 /**
291  * Return node pointer to first packet of requested type in list.
292  *
293  * @param list list of subpackets to be scanned
294  * @param type type of searched subpacket
295  * @return node pointer for subpacket if found, else NULL
296  */
298  int type)
299 {
300  while (list && list->packet) {
301  if (list->packet->type == type)
302  return list;
303  list = list->next;
304  }
305  return NULL;
306 }
307 
308 /**
309  * Replace 8 elements with their average value.
310  * Called by qdm2_decode_superblock before starting subblock decoding.
311  *
312  * @param q context
313  */
315 {
316  int i, j, n, ch, sum;
317 
319 
320  for (ch = 0; ch < q->nb_channels; ch++)
321  for (i = 0; i < n; i++) {
322  sum = 0;
323 
324  for (j = 0; j < 8; j++)
325  sum += q->quantized_coeffs[ch][i][j];
326 
327  sum /= 8;
328  if (sum > 0)
329  sum--;
330 
331  for (j = 0; j < 8; j++)
332  q->quantized_coeffs[ch][i][j] = sum;
333  }
334 }
335 
336 /**
337  * Build subband samples with noise weighted by q->tone_level.
338  * Called by synthfilt_build_sb_samples.
339  *
340  * @param q context
341  * @param sb subband index
342  */
344 {
345  int ch, j;
346 
348 
349  if (!q->nb_channels)
350  return;
351 
352  for (ch = 0; ch < q->nb_channels; ch++) {
353  for (j = 0; j < 64; j++) {
354  q->sb_samples[ch][j * 2][sb] =
355  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
356  q->sb_samples[ch][j * 2 + 1][sb] =
357  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
358  }
359  }
360 }
361 
362 /**
363  * Called while processing data from subpackets 11 and 12.
364  * Used after making changes to coding_method array.
365  *
366  * @param sb subband index
367  * @param channels number of channels
368  * @param coding_method q->coding_method[0][0][0]
369  */
370 static int fix_coding_method_array(int sb, int channels,
371  sb_int8_array coding_method)
372 {
373  int j, k;
374  int ch;
375  int run, case_val;
376 
377  for (ch = 0; ch < channels; ch++) {
378  for (j = 0; j < 64; ) {
379  if (coding_method[ch][sb][j] < 8)
380  return -1;
381  if ((coding_method[ch][sb][j] - 8) > 22) {
382  run = 1;
383  case_val = 8;
384  } else {
385  switch (switchtable[coding_method[ch][sb][j] - 8]) {
386  case 0: run = 10;
387  case_val = 10;
388  break;
389  case 1: run = 1;
390  case_val = 16;
391  break;
392  case 2: run = 5;
393  case_val = 24;
394  break;
395  case 3: run = 3;
396  case_val = 30;
397  break;
398  case 4: run = 1;
399  case_val = 30;
400  break;
401  case 5: run = 1;
402  case_val = 8;
403  break;
404  default: run = 1;
405  case_val = 8;
406  break;
407  }
408  }
409  for (k = 0; k < run; k++) {
410  if (j + k < 128) {
411  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
412  if (k > 0) {
414  //not debugged, almost never used
415  memset(&coding_method[ch][sb][j + k], case_val,
416  k *sizeof(int8_t));
417  memset(&coding_method[ch][sb][j + k], case_val,
418  3 * sizeof(int8_t));
419  }
420  }
421  }
422  }
423  j += run;
424  }
425  }
426  return 0;
427 }
428 
429 /**
430  * Related to synthesis filter
431  * Called by process_subpacket_10
432  *
433  * @param q context
434  * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
435  */
437 {
438  int i, sb, ch, sb_used;
439  int tmp, tab;
440 
441  for (ch = 0; ch < q->nb_channels; ch++)
442  for (sb = 0; sb < 30; sb++)
443  for (i = 0; i < 8; i++) {
445  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
447  else
448  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
449  if(tmp < 0)
450  tmp += 0xff;
451  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
452  }
453 
454  sb_used = QDM2_SB_USED(q->sub_sampling);
455 
456  if ((q->superblocktype_2_3 != 0) && !flag) {
457  for (sb = 0; sb < sb_used; sb++)
458  for (ch = 0; ch < q->nb_channels; ch++)
459  for (i = 0; i < 64; i++) {
460  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
461  if (q->tone_level_idx[ch][sb][i] < 0)
462  q->tone_level[ch][sb][i] = 0;
463  else
464  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
465  }
466  } else {
467  tab = q->superblocktype_2_3 ? 0 : 1;
468  for (sb = 0; sb < sb_used; sb++) {
469  if ((sb >= 4) && (sb <= 23)) {
470  for (ch = 0; ch < q->nb_channels; ch++)
471  for (i = 0; i < 64; i++) {
472  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
473  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
474  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
475  q->tone_level_idx_hi2[ch][sb - 4];
476  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
477  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
478  q->tone_level[ch][sb][i] = 0;
479  else
480  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
481  }
482  } else {
483  if (sb > 4) {
484  for (ch = 0; ch < q->nb_channels; ch++)
485  for (i = 0; i < 64; i++) {
486  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
487  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
488  q->tone_level_idx_hi2[ch][sb - 4];
489  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
490  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
491  q->tone_level[ch][sb][i] = 0;
492  else
493  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
494  }
495  } else {
496  for (ch = 0; ch < q->nb_channels; ch++)
497  for (i = 0; i < 64; i++) {
498  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
499  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
500  q->tone_level[ch][sb][i] = 0;
501  else
502  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
503  }
504  }
505  }
506  }
507  }
508 }
509 
510 /**
511  * Related to synthesis filter
512  * Called by process_subpacket_11
513  * c is built with data from subpacket 11
514  * Most of this function is used only if superblock_type_2_3 == 0,
515  * never seen it in samples.
516  *
517  * @param tone_level_idx
518  * @param tone_level_idx_temp
519  * @param coding_method q->coding_method[0][0][0]
520  * @param nb_channels number of channels
521  * @param c coming from subpacket 11, passed as 8*c
522  * @param superblocktype_2_3 flag based on superblock packet type
523  * @param cm_table_select q->cm_table_select
524  */
525 static void fill_coding_method_array(sb_int8_array tone_level_idx,
526  sb_int8_array tone_level_idx_temp,
527  sb_int8_array coding_method,
528  int nb_channels,
529  int c, int superblocktype_2_3,
530  int cm_table_select)
531 {
532  int ch, sb, j;
533  int tmp, acc, esp_40, comp;
534  int add1, add2, add3, add4;
535  int64_t multres;
536 
537  if (!superblocktype_2_3) {
538  /* This case is untested, no samples available */
539  avpriv_request_sample(NULL, "!superblocktype_2_3");
540  return;
541  for (ch = 0; ch < nb_channels; ch++) {
542  for (sb = 0; sb < 30; sb++) {
543  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
544  add1 = tone_level_idx[ch][sb][j] - 10;
545  if (add1 < 0)
546  add1 = 0;
547  add2 = add3 = add4 = 0;
548  if (sb > 1) {
549  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
550  if (add2 < 0)
551  add2 = 0;
552  }
553  if (sb > 0) {
554  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
555  if (add3 < 0)
556  add3 = 0;
557  }
558  if (sb < 29) {
559  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
560  if (add4 < 0)
561  add4 = 0;
562  }
563  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
564  if (tmp < 0)
565  tmp = 0;
566  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
567  }
568  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
569  }
570  }
571  acc = 0;
572  for (ch = 0; ch < nb_channels; ch++)
573  for (sb = 0; sb < 30; sb++)
574  for (j = 0; j < 64; j++)
575  acc += tone_level_idx_temp[ch][sb][j];
576 
577  multres = 0x66666667LL * (acc * 10);
578  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
579  for (ch = 0; ch < nb_channels; ch++)
580  for (sb = 0; sb < 30; sb++)
581  for (j = 0; j < 64; j++) {
582  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
583  if (comp < 0)
584  comp += 0xff;
585  comp /= 256; // signed shift
586  switch(sb) {
587  case 0:
588  if (comp < 30)
589  comp = 30;
590  comp += 15;
591  break;
592  case 1:
593  if (comp < 24)
594  comp = 24;
595  comp += 10;
596  break;
597  case 2:
598  case 3:
599  case 4:
600  if (comp < 16)
601  comp = 16;
602  }
603  if (comp <= 5)
604  tmp = 0;
605  else if (comp <= 10)
606  tmp = 10;
607  else if (comp <= 16)
608  tmp = 16;
609  else if (comp <= 24)
610  tmp = -1;
611  else
612  tmp = 0;
613  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
614  }
615  for (sb = 0; sb < 30; sb++)
616  fix_coding_method_array(sb, nb_channels, coding_method);
617  for (ch = 0; ch < nb_channels; ch++)
618  for (sb = 0; sb < 30; sb++)
619  for (j = 0; j < 64; j++)
620  if (sb >= 10) {
621  if (coding_method[ch][sb][j] < 10)
622  coding_method[ch][sb][j] = 10;
623  } else {
624  if (sb >= 2) {
625  if (coding_method[ch][sb][j] < 16)
626  coding_method[ch][sb][j] = 16;
627  } else {
628  if (coding_method[ch][sb][j] < 30)
629  coding_method[ch][sb][j] = 30;
630  }
631  }
632  } else { // superblocktype_2_3 != 0
633  for (ch = 0; ch < nb_channels; ch++)
634  for (sb = 0; sb < 30; sb++)
635  for (j = 0; j < 64; j++)
636  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
637  }
638 }
639 
640 /**
641  * Called by process_subpacket_11 to process more data from subpacket 11
642  * with sb 0-8.
643  * Called by process_subpacket_12 to process data from subpacket 12 with
644  * sb 8-sb_used.
645  *
646  * @param q context
647  * @param gb bitreader context
648  * @param length packet length in bits
649  * @param sb_min lower subband processed (sb_min included)
650  * @param sb_max higher subband processed (sb_max excluded)
651  */
653  int length, int sb_min, int sb_max)
654 {
655  int sb, j, k, n, ch, run, channels;
656  int joined_stereo, zero_encoding;
657  int type34_first;
658  float type34_div = 0;
659  float type34_predictor;
660  float samples[10];
661  int sign_bits[16] = {0};
662 
663  if (length == 0) {
664  // If no data use noise
665  for (sb=sb_min; sb < sb_max; sb++)
667 
668  return 0;
669  }
670 
671  for (sb = sb_min; sb < sb_max; sb++) {
672  channels = q->nb_channels;
673 
674  if (q->nb_channels <= 1 || sb < 12)
675  joined_stereo = 0;
676  else if (sb >= 24)
677  joined_stereo = 1;
678  else
679  joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
680 
681  if (joined_stereo) {
682  if (get_bits_left(gb) >= 16)
683  for (j = 0; j < 16; j++)
684  sign_bits[j] = get_bits1(gb);
685 
686  for (j = 0; j < 64; j++)
687  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
688  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
689 
691  q->coding_method)) {
692  av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
694  continue;
695  }
696  channels = 1;
697  }
698 
699  for (ch = 0; ch < channels; ch++) {
701  zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
702  type34_predictor = 0.0;
703  type34_first = 1;
704 
705  for (j = 0; j < 128; ) {
706  switch (q->coding_method[ch][sb][j / 2]) {
707  case 8:
708  if (get_bits_left(gb) >= 10) {
709  if (zero_encoding) {
710  for (k = 0; k < 5; k++) {
711  if ((j + 2 * k) >= 128)
712  break;
713  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
714  }
715  } else {
716  n = get_bits(gb, 8);
717  if (n >= 243) {
718  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
719  return AVERROR_INVALIDDATA;
720  }
721 
722  for (k = 0; k < 5; k++)
723  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
724  }
725  for (k = 0; k < 5; k++)
726  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
727  } else {
728  for (k = 0; k < 10; k++)
729  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
730  }
731  run = 10;
732  break;
733 
734  case 10:
735  if (get_bits_left(gb) >= 1) {
736  float f = 0.81;
737 
738  if (get_bits1(gb))
739  f = -f;
740  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
741  samples[0] = f;
742  } else {
743  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
744  }
745  run = 1;
746  break;
747 
748  case 16:
749  if (get_bits_left(gb) >= 10) {
750  if (zero_encoding) {
751  for (k = 0; k < 5; k++) {
752  if ((j + k) >= 128)
753  break;
754  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
755  }
756  } else {
757  n = get_bits (gb, 8);
758  if (n >= 243) {
759  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
760  return AVERROR_INVALIDDATA;
761  }
762 
763  for (k = 0; k < 5; k++)
764  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
765  }
766  } else {
767  for (k = 0; k < 5; k++)
768  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
769  }
770  run = 5;
771  break;
772 
773  case 24:
774  if (get_bits_left(gb) >= 7) {
775  n = get_bits(gb, 7);
776  if (n >= 125) {
777  av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
778  return AVERROR_INVALIDDATA;
779  }
780 
781  for (k = 0; k < 3; k++)
782  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
783  } else {
784  for (k = 0; k < 3; k++)
785  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
786  }
787  run = 3;
788  break;
789 
790  case 30:
791  if (get_bits_left(gb) >= 4) {
792  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
793  if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
794  av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
795  return AVERROR_INVALIDDATA;
796  }
797  samples[0] = type30_dequant[index];
798  } else
799  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
800 
801  run = 1;
802  break;
803 
804  case 34:
805  if (get_bits_left(gb) >= 7) {
806  if (type34_first) {
807  type34_div = (float)(1 << get_bits(gb, 2));
808  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
809  type34_predictor = samples[0];
810  type34_first = 0;
811  } else {
812  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
813  if (index >= FF_ARRAY_ELEMS(type34_delta)) {
814  av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
815  return AVERROR_INVALIDDATA;
816  }
817  samples[0] = type34_delta[index] / type34_div + type34_predictor;
818  type34_predictor = samples[0];
819  }
820  } else {
821  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
822  }
823  run = 1;
824  break;
825 
826  default:
827  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
828  run = 1;
829  break;
830  }
831 
832  if (joined_stereo) {
833  for (k = 0; k < run && j + k < 128; k++) {
834  q->sb_samples[0][j + k][sb] =
835  q->tone_level[0][sb][(j + k) / 2] * samples[k];
836  if (q->nb_channels == 2) {
837  if (sign_bits[(j + k) / 8])
838  q->sb_samples[1][j + k][sb] =
839  q->tone_level[1][sb][(j + k) / 2] * -samples[k];
840  else
841  q->sb_samples[1][j + k][sb] =
842  q->tone_level[1][sb][(j + k) / 2] * samples[k];
843  }
844  }
845  } else {
846  for (k = 0; k < run; k++)
847  if ((j + k) < 128)
848  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
849  }
850 
851  j += run;
852  } // j loop
853  } // channel loop
854  } // subband loop
855  return 0;
856 }
857 
858 /**
859  * Init the first element of a channel in quantized_coeffs with data
860  * from packet 10 (quantized_coeffs[ch][0]).
861  * This is similar to process_subpacket_9, but for a single channel
862  * and for element [0]
863  * same VLC tables as process_subpacket_9 are used.
864  *
865  * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
866  * @param gb bitreader context
867  */
868 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
869  GetBitContext *gb)
870 {
871  int i, k, run, level, diff;
872 
873  if (get_bits_left(gb) < 16)
874  return -1;
875  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
876 
877  quantized_coeffs[0] = level;
878 
879  for (i = 0; i < 7; ) {
880  if (get_bits_left(gb) < 16)
881  return -1;
882  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
883 
884  if (i + run >= 8)
885  return -1;
886 
887  if (get_bits_left(gb) < 16)
888  return -1;
889  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
890 
891  for (k = 1; k <= run; k++)
892  quantized_coeffs[i + k] = (level + ((k * diff) / run));
893 
894  level += diff;
895  i += run;
896  }
897  return 0;
898 }
899 
900 /**
901  * Related to synthesis filter, process data from packet 10
902  * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
903  * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
904  * data from packet 10
905  *
906  * @param q context
907  * @param gb bitreader context
908  */
910 {
911  int sb, j, k, n, ch;
912 
913  for (ch = 0; ch < q->nb_channels; ch++) {
915 
916  if (get_bits_left(gb) < 16) {
917  memset(q->quantized_coeffs[ch][0], 0, 8);
918  break;
919  }
920  }
921 
922  n = q->sub_sampling + 1;
923 
924  for (sb = 0; sb < n; sb++)
925  for (ch = 0; ch < q->nb_channels; ch++)
926  for (j = 0; j < 8; j++) {
927  if (get_bits_left(gb) < 1)
928  break;
929  if (get_bits1(gb)) {
930  for (k=0; k < 8; k++) {
931  if (get_bits_left(gb) < 16)
932  break;
933  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
934  }
935  } else {
936  for (k=0; k < 8; k++)
937  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
938  }
939  }
940 
941  n = QDM2_SB_USED(q->sub_sampling) - 4;
942 
943  for (sb = 0; sb < n; sb++)
944  for (ch = 0; ch < q->nb_channels; ch++) {
945  if (get_bits_left(gb) < 16)
946  break;
948  if (sb > 19)
949  q->tone_level_idx_hi2[ch][sb] -= 16;
950  else
951  for (j = 0; j < 8; j++)
952  q->tone_level_idx_mid[ch][sb][j] = -16;
953  }
954 
955  n = QDM2_SB_USED(q->sub_sampling) - 5;
956 
957  for (sb = 0; sb < n; sb++)
958  for (ch = 0; ch < q->nb_channels; ch++)
959  for (j = 0; j < 8; j++) {
960  if (get_bits_left(gb) < 16)
961  break;
962  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
963  }
964 }
965 
966 /**
967  * Process subpacket 9, init quantized_coeffs with data from it
968  *
969  * @param q context
970  * @param node pointer to node with packet
971  */
973 {
974  GetBitContext gb;
975  int i, j, k, n, ch, run, level, diff;
976 
977  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
978 
980 
981  for (i = 1; i < n; i++)
982  for (ch = 0; ch < q->nb_channels; ch++) {
983  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
984  q->quantized_coeffs[ch][i][0] = level;
985 
986  for (j = 0; j < (8 - 1); ) {
987  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
988  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
989 
990  if (j + run >= 8)
991  return -1;
992 
993  for (k = 1; k <= run; k++)
994  q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
995 
996  level += diff;
997  j += run;
998  }
999  }
1000 
1001  for (ch = 0; ch < q->nb_channels; ch++)
1002  for (i = 0; i < 8; i++)
1003  q->quantized_coeffs[ch][0][i] = 0;
1004 
1005  return 0;
1006 }
1007 
1008 /**
1009  * Process subpacket 10 if not null, else
1010  *
1011  * @param q context
1012  * @param node pointer to node with packet
1013  */
1015 {
1016  GetBitContext gb;
1017 
1018  if (node) {
1019  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1021  fill_tone_level_array(q, 1);
1022  } else {
1023  fill_tone_level_array(q, 0);
1024  }
1025 }
1026 
1027 /**
1028  * Process subpacket 11
1029  *
1030  * @param q context
1031  * @param node pointer to node with packet
1032  */
1034 {
1035  GetBitContext gb;
1036  int length = 0;
1037 
1038  if (node) {
1039  length = node->packet->size * 8;
1040  init_get_bits(&gb, node->packet->data, length);
1041  }
1042 
1043  if (length >= 32) {
1044  int c = get_bits(&gb, 13);
1045 
1046  if (c > 3)
1049  q->nb_channels, 8 * c,
1051  }
1052 
1053  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1054 }
1055 
1056 /**
1057  * Process subpacket 12
1058  *
1059  * @param q context
1060  * @param node pointer to node with packet
1061  */
1063 {
1064  GetBitContext gb;
1065  int length = 0;
1066 
1067  if (node) {
1068  length = node->packet->size * 8;
1069  init_get_bits(&gb, node->packet->data, length);
1070  }
1071 
1072  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1073 }
1074 
1075 /**
1076  * Process new subpackets for synthesis filter
1077  *
1078  * @param q context
1079  * @param list list with synthesis filter packets (list D)
1080  */
1082 {
1083  QDM2SubPNode *nodes[4];
1084 
1085  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1086  if (nodes[0])
1087  process_subpacket_9(q, nodes[0]);
1088 
1089  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1090  if (nodes[1])
1091  process_subpacket_10(q, nodes[1]);
1092  else
1094 
1095  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1096  if (nodes[0] && nodes[1] && nodes[2])
1097  process_subpacket_11(q, nodes[2]);
1098  else
1100 
1101  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1102  if (nodes[0] && nodes[1] && nodes[3])
1103  process_subpacket_12(q, nodes[3]);
1104  else
1106 }
1107 
1108 /**
1109  * Decode superblock, fill packet lists.
1110  *
1111  * @param q context
1112  */
1114 {
1115  GetBitContext gb;
1116  QDM2SubPacket header, *packet;
1117  int i, packet_bytes, sub_packet_size, sub_packets_D;
1118  unsigned int next_index = 0;
1119 
1120  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1121  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1122  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1123 
1124  q->sub_packets_B = 0;
1125  sub_packets_D = 0;
1126 
1127  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1128 
1130  qdm2_decode_sub_packet_header(&gb, &header);
1131 
1132  if (header.type < 2 || header.type >= 8) {
1133  q->has_errors = 1;
1134  av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1135  return;
1136  }
1137 
1138  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1139  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1140 
1141  init_get_bits(&gb, header.data, header.size * 8);
1142 
1143  if (header.type == 2 || header.type == 4 || header.type == 5) {
1144  int csum = 257 * get_bits(&gb, 8);
1145  csum += 2 * get_bits(&gb, 8);
1146 
1147  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1148 
1149  if (csum != 0) {
1150  q->has_errors = 1;
1151  av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1152  return;
1153  }
1154  }
1155 
1156  q->sub_packet_list_B[0].packet = NULL;
1157  q->sub_packet_list_D[0].packet = NULL;
1158 
1159  for (i = 0; i < 6; i++)
1160  if (--q->fft_level_exp[i] < 0)
1161  q->fft_level_exp[i] = 0;
1162 
1163  for (i = 0; packet_bytes > 0; i++) {
1164  int j;
1165 
1166  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1167  SAMPLES_NEEDED_2("too many packet bytes");
1168  return;
1169  }
1170 
1171  q->sub_packet_list_A[i].next = NULL;
1172 
1173  if (i > 0) {
1174  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1175 
1176  /* seek to next block */
1177  init_get_bits(&gb, header.data, header.size * 8);
1178  skip_bits(&gb, next_index * 8);
1179 
1180  if (next_index >= header.size)
1181  break;
1182  }
1183 
1184  /* decode subpacket */
1185  packet = &q->sub_packets[i];
1186  qdm2_decode_sub_packet_header(&gb, packet);
1187  next_index = packet->size + get_bits_count(&gb) / 8;
1188  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1189 
1190  if (packet->type == 0)
1191  break;
1192 
1193  if (sub_packet_size > packet_bytes) {
1194  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1195  break;
1196  packet->size += packet_bytes - sub_packet_size;
1197  }
1198 
1199  packet_bytes -= sub_packet_size;
1200 
1201  /* add subpacket to 'all subpackets' list */
1202  q->sub_packet_list_A[i].packet = packet;
1203 
1204  /* add subpacket to related list */
1205  if (packet->type == 8) {
1206  SAMPLES_NEEDED_2("packet type 8");
1207  return;
1208  } else if (packet->type >= 9 && packet->type <= 12) {
1209  /* packets for MPEG Audio like Synthesis Filter */
1210  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1211  } else if (packet->type == 13) {
1212  for (j = 0; j < 6; j++)
1213  q->fft_level_exp[j] = get_bits(&gb, 6);
1214  } else if (packet->type == 14) {
1215  for (j = 0; j < 6; j++)
1216  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1217  } else if (packet->type == 15) {
1218  SAMPLES_NEEDED_2("packet type 15")
1219  return;
1220  } else if (packet->type >= 16 && packet->type < 48 &&
1221  !fft_subpackets[packet->type - 16]) {
1222  /* packets for FFT */
1224  }
1225  } // Packet bytes loop
1226 
1227  if (q->sub_packet_list_D[0].packet) {
1229  q->do_synth_filter = 1;
1230  } else if (q->do_synth_filter) {
1234  }
1235 }
1236 
1237 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1238  int offset, int duration, int channel,
1239  int exp, int phase)
1240 {
1241  if (q->fft_coefs_min_index[duration] < 0)
1243 
1245  ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1248  q->fft_coefs[q->fft_coefs_index].exp = exp;
1249  q->fft_coefs[q->fft_coefs_index].phase = phase;
1250  q->fft_coefs_index++;
1251 }
1252 
1254  GetBitContext *gb, int b)
1255 {
1256  int channel, stereo, phase, exp;
1257  int local_int_4, local_int_8, stereo_phase, local_int_10;
1258  int local_int_14, stereo_exp, local_int_20, local_int_28;
1259  int n, offset;
1260 
1261  local_int_4 = 0;
1262  local_int_28 = 0;
1263  local_int_20 = 2;
1264  local_int_8 = (4 - duration);
1265  local_int_10 = 1 << (q->group_order - duration - 1);
1266  offset = 1;
1267 
1268  while (get_bits_left(gb)>0) {
1269  if (q->superblocktype_2_3) {
1270  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1271  if (get_bits_left(gb)<0) {
1272  if(local_int_4 < q->group_size)
1273  av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1274  return;
1275  }
1276  offset = 1;
1277  if (n == 0) {
1278  local_int_4 += local_int_10;
1279  local_int_28 += (1 << local_int_8);
1280  } else {
1281  local_int_4 += 8 * local_int_10;
1282  local_int_28 += (8 << local_int_8);
1283  }
1284  }
1285  offset += (n - 2);
1286  } else {
1287  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1288  while (offset >= (local_int_10 - 1)) {
1289  offset += (1 - (local_int_10 - 1));
1290  local_int_4 += local_int_10;
1291  local_int_28 += (1 << local_int_8);
1292  }
1293  }
1294 
1295  if (local_int_4 >= q->group_size)
1296  return;
1297 
1298  local_int_14 = (offset >> local_int_8);
1299  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1300  return;
1301 
1302  if (q->nb_channels > 1) {
1303  channel = get_bits1(gb);
1304  stereo = get_bits1(gb);
1305  } else {
1306  channel = 0;
1307  stereo = 0;
1308  }
1309 
1310  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1311  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1312  exp = (exp < 0) ? 0 : exp;
1313 
1314  phase = get_bits(gb, 3);
1315  stereo_exp = 0;
1316  stereo_phase = 0;
1317 
1318  if (stereo) {
1319  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1320  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1321  if (stereo_phase < 0)
1322  stereo_phase += 8;
1323  }
1324 
1325  if (q->frequency_range > (local_int_14 + 1)) {
1326  int sub_packet = (local_int_20 + local_int_28);
1327 
1328  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1329  channel, exp, phase);
1330  if (stereo)
1331  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1332  1 - channel,
1333  stereo_exp, stereo_phase);
1334  }
1335  offset++;
1336  }
1337 }
1338 
1340 {
1341  int i, j, min, max, value, type, unknown_flag;
1342  GetBitContext gb;
1343 
1344  if (!q->sub_packet_list_B[0].packet)
1345  return;
1346 
1347  /* reset minimum indexes for FFT coefficients */
1348  q->fft_coefs_index = 0;
1349  for (i = 0; i < 5; i++)
1350  q->fft_coefs_min_index[i] = -1;
1351 
1352  /* process subpackets ordered by type, largest type first */
1353  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1354  QDM2SubPacket *packet = NULL;
1355 
1356  /* find subpacket with largest type less than max */
1357  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1358  value = q->sub_packet_list_B[j].packet->type;
1359  if (value > min && value < max) {
1360  min = value;
1361  packet = q->sub_packet_list_B[j].packet;
1362  }
1363  }
1364 
1365  max = min;
1366 
1367  /* check for errors (?) */
1368  if (!packet)
1369  return;
1370 
1371  if (i == 0 &&
1372  (packet->type < 16 || packet->type >= 48 ||
1373  fft_subpackets[packet->type - 16]))
1374  return;
1375 
1376  /* decode FFT tones */
1377  init_get_bits(&gb, packet->data, packet->size * 8);
1378 
1379  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1380  unknown_flag = 1;
1381  else
1382  unknown_flag = 0;
1383 
1384  type = packet->type;
1385 
1386  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1387  int duration = q->sub_sampling + 5 - (type & 15);
1388 
1389  if (duration >= 0 && duration < 4)
1390  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1391  } else if (type == 31) {
1392  for (j = 0; j < 4; j++)
1393  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1394  } else if (type == 46) {
1395  for (j = 0; j < 6; j++)
1396  q->fft_level_exp[j] = get_bits(&gb, 6);
1397  for (j = 0; j < 4; j++)
1398  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1399  }
1400  } // Loop on B packets
1401 
1402  /* calculate maximum indexes for FFT coefficients */
1403  for (i = 0, j = -1; i < 5; i++)
1404  if (q->fft_coefs_min_index[i] >= 0) {
1405  if (j >= 0)
1407  j = i;
1408  }
1409  if (j >= 0)
1411 }
1412 
1414 {
1415  float level, f[6];
1416  int i;
1417  QDM2Complex c;
1418  const double iscale = 2.0 * M_PI / 512.0;
1419 
1420  tone->phase += tone->phase_shift;
1421 
1422  /* calculate current level (maximum amplitude) of tone */
1423  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1424  c.im = level * sin(tone->phase * iscale);
1425  c.re = level * cos(tone->phase * iscale);
1426 
1427  /* generate FFT coefficients for tone */
1428  if (tone->duration >= 3 || tone->cutoff >= 3) {
1429  tone->complex[0].im += c.im;
1430  tone->complex[0].re += c.re;
1431  tone->complex[1].im -= c.im;
1432  tone->complex[1].re -= c.re;
1433  } else {
1434  f[1] = -tone->table[4];
1435  f[0] = tone->table[3] - tone->table[0];
1436  f[2] = 1.0 - tone->table[2] - tone->table[3];
1437  f[3] = tone->table[1] + tone->table[4] - 1.0;
1438  f[4] = tone->table[0] - tone->table[1];
1439  f[5] = tone->table[2];
1440  for (i = 0; i < 2; i++) {
1441  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1442  c.re * f[i];
1443  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1444  c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1445  }
1446  for (i = 0; i < 4; i++) {
1447  tone->complex[i].re += c.re * f[i + 2];
1448  tone->complex[i].im += c.im * f[i + 2];
1449  }
1450  }
1451 
1452  /* copy the tone if it has not yet died out */
1453  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1454  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1455  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1456  }
1457 }
1458 
1459 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1460 {
1461  int i, j, ch;
1462  const double iscale = 0.25 * M_PI;
1463 
1464  for (ch = 0; ch < q->channels; ch++) {
1465  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1466  }
1467 
1468 
1469  /* apply FFT tones with duration 4 (1 FFT period) */
1470  if (q->fft_coefs_min_index[4] >= 0)
1471  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1472  float level;
1473  QDM2Complex c;
1474 
1475  if (q->fft_coefs[i].sub_packet != sub_packet)
1476  break;
1477 
1478  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1479  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];
1480 
1481  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1482  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1483  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1484  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1485  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1486  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1487  }
1488 
1489  /* generate existing FFT tones */
1490  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1492  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1493  }
1494 
1495  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1496  for (i = 0; i < 4; i++)
1497  if (q->fft_coefs_min_index[i] >= 0) {
1498  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1499  int offset, four_i;
1500  FFTTone tone;
1501 
1502  if (q->fft_coefs[j].sub_packet != sub_packet)
1503  break;
1504 
1505  four_i = (4 - i);
1506  offset = q->fft_coefs[j].offset >> four_i;
1507  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1508 
1509  if (offset < q->frequency_range) {
1510  if (offset < 2)
1511  tone.cutoff = offset;
1512  else
1513  tone.cutoff = (offset >= 60) ? 3 : 2;
1514 
1515  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];
1516  tone.complex = &q->fft.complex[ch][offset];
1517  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1518  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1519  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1520  tone.duration = i;
1521  tone.time_index = 0;
1522 
1523  qdm2_fft_generate_tone(q, &tone);
1524  }
1525  }
1526  q->fft_coefs_min_index[i] = j;
1527  }
1528 }
1529 
1530 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1531 {
1532  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1533  float *out = q->output_buffer + channel;
1534  int i;
1535  q->fft.complex[channel][0].re *= 2.0f;
1536  q->fft.complex[channel][0].im = 0.0f;
1537  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1538  /* add samples to output buffer */
1539  for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1540  out[0] += q->fft.complex[channel][i].re * gain;
1541  out[q->channels] += q->fft.complex[channel][i].im * gain;
1542  out += 2 * q->channels;
1543  }
1544 }
1545 
1546 /**
1547  * @param q context
1548  * @param index subpacket number
1549  */
1551 {
1552  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1553 
1554  /* copy sb_samples */
1555  sb_used = QDM2_SB_USED(q->sub_sampling);
1556 
1557  for (ch = 0; ch < q->channels; ch++)
1558  for (i = 0; i < 8; i++)
1559  for (k = sb_used; k < SBLIMIT; k++)
1560  q->sb_samples[ch][(8 * index) + i][k] = 0;
1561 
1562  for (ch = 0; ch < q->nb_channels; ch++) {
1563  float *samples_ptr = q->samples + ch;
1564 
1565  for (i = 0; i < 8; i++) {
1567  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1568  ff_mpa_synth_window_float, &dither_state,
1569  samples_ptr, q->nb_channels,
1570  q->sb_samples[ch][(8 * index) + i]);
1571  samples_ptr += 32 * q->nb_channels;
1572  }
1573  }
1574 
1575  /* add samples to output buffer */
1576  sub_sampling = (4 >> q->sub_sampling);
1577 
1578  for (ch = 0; ch < q->channels; ch++)
1579  for (i = 0; i < q->frame_size; i++)
1580  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1581 }
1582 
1583 /**
1584  * Init static data (does not depend on specific file)
1585  *
1586  * @param q context
1587  */
1588 static av_cold void qdm2_init_static_data(void) {
1589  static int done;
1590 
1591  if(done)
1592  return;
1593 
1594  qdm2_init_vlc();
1597  rnd_table_init();
1599 
1600  done = 1;
1601 }
1602 
1603 /**
1604  * Init parameters from codec extradata
1605  */
1607 {
1608  QDM2Context *s = avctx->priv_data;
1609  int tmp_val, tmp, size;
1610  GetByteContext gb;
1611 
1613 
1614  /* extradata parsing
1615 
1616  Structure:
1617  wave {
1618  frma (QDM2)
1619  QDCA
1620  QDCP
1621  }
1622 
1623  32 size (including this field)
1624  32 tag (=frma)
1625  32 type (=QDM2 or QDMC)
1626 
1627  32 size (including this field, in bytes)
1628  32 tag (=QDCA) // maybe mandatory parameters
1629  32 unknown (=1)
1630  32 channels (=2)
1631  32 samplerate (=44100)
1632  32 bitrate (=96000)
1633  32 block size (=4096)
1634  32 frame size (=256) (for one channel)
1635  32 packet size (=1300)
1636 
1637  32 size (including this field, in bytes)
1638  32 tag (=QDCP) // maybe some tuneable parameters
1639  32 float1 (=1.0)
1640  32 zero ?
1641  32 float2 (=1.0)
1642  32 float3 (=1.0)
1643  32 unknown (27)
1644  32 unknown (8)
1645  32 zero ?
1646  */
1647 
1648  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1649  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1650  return AVERROR_INVALIDDATA;
1651  }
1652 
1653  bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1654 
1655  while (bytestream2_get_bytes_left(&gb) > 8) {
1656  if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1657  (uint64_t)MKBETAG('Q','D','M','2')))
1658  break;
1659  bytestream2_skip(&gb, 1);
1660  }
1661 
1662  if (bytestream2_get_bytes_left(&gb) < 12) {
1663  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1665  return AVERROR_INVALIDDATA;
1666  }
1667 
1668  bytestream2_skip(&gb, 8);
1669  size = bytestream2_get_be32(&gb);
1670 
1671  if (size > bytestream2_get_bytes_left(&gb)) {
1672  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1673  bytestream2_get_bytes_left(&gb), size);
1674  return AVERROR_INVALIDDATA;
1675  }
1676 
1677  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1678  if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1679  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1680  return AVERROR_INVALIDDATA;
1681  }
1682 
1683  bytestream2_skip(&gb, 4);
1684 
1685  avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1686  if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1687  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1688  return AVERROR_INVALIDDATA;
1689  }
1690  avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1692 
1693  avctx->sample_rate = bytestream2_get_be32(&gb);
1694  avctx->bit_rate = bytestream2_get_be32(&gb);
1695  s->group_size = bytestream2_get_be32(&gb);
1696  s->fft_size = bytestream2_get_be32(&gb);
1697  s->checksum_size = bytestream2_get_be32(&gb);
1698  if (s->checksum_size >= 1U << 28) {
1699  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1700  return AVERROR_INVALIDDATA;
1701  }
1702 
1703  s->fft_order = av_log2(s->fft_size) + 1;
1704 
1705  // something like max decodable tones
1706  s->group_order = av_log2(s->group_size) + 1;
1707  s->frame_size = s->group_size / 16; // 16 iterations per super block
1708 
1710  return AVERROR_INVALIDDATA;
1711 
1712  s->sub_sampling = s->fft_order - 7;
1713  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1714 
1715  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1716  case 0: tmp = 40; break;
1717  case 1: tmp = 48; break;
1718  case 2: tmp = 56; break;
1719  case 3: tmp = 72; break;
1720  case 4: tmp = 80; break;
1721  case 5: tmp = 100;break;
1722  default: tmp=s->sub_sampling; break;
1723  }
1724  tmp_val = 0;
1725  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1726  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1727  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1728  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1729  s->cm_table_select = tmp_val;
1730 
1731  if (avctx->bit_rate <= 8000)
1732  s->coeff_per_sb_select = 0;
1733  else if (avctx->bit_rate < 16000)
1734  s->coeff_per_sb_select = 1;
1735  else
1736  s->coeff_per_sb_select = 2;
1737 
1738  // Fail on unknown fft order
1739  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1740  avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1741  return AVERROR_PATCHWELCOME;
1742  }
1743  if (s->fft_size != (1 << (s->fft_order - 1))) {
1744  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1745  return AVERROR_INVALIDDATA;
1746  }
1747 
1749  ff_mpadsp_init(&s->mpadsp);
1750 
1751  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1752 
1753  return 0;
1754 }
1755 
1757 {
1758  QDM2Context *s = avctx->priv_data;
1759 
1760  ff_rdft_end(&s->rdft_ctx);
1761 
1762  return 0;
1763 }
1764 
1765 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1766 {
1767  int ch, i;
1768  const int frame_size = (q->frame_size * q->channels);
1769 
1770  if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1771  return -1;
1772 
1773  /* select input buffer */
1774  q->compressed_data = in;
1776 
1777  /* copy old block, clear new block of output samples */
1778  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1779  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1780 
1781  /* decode block of QDM2 compressed data */
1782  if (q->sub_packet == 0) {
1783  q->has_errors = 0; // zero it for a new super block
1784  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1786  }
1787 
1788  /* parse subpackets */
1789  if (!q->has_errors) {
1790  if (q->sub_packet == 2)
1792 
1794  }
1795 
1796  /* sound synthesis stage 1 (FFT) */
1797  for (ch = 0; ch < q->channels; ch++) {
1798  qdm2_calculate_fft(q, ch, q->sub_packet);
1799 
1800  if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1801  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1802  return -1;
1803  }
1804  }
1805 
1806  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1807  if (!q->has_errors && q->do_synth_filter)
1809 
1810  q->sub_packet = (q->sub_packet + 1) % 16;
1811 
1812  /* clip and convert output float[] to 16-bit signed samples */
1813  for (i = 0; i < frame_size; i++) {
1814  int value = (int)q->output_buffer[i];
1815 
1816  if (value > SOFTCLIP_THRESHOLD)
1817  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1818  else if (value < -SOFTCLIP_THRESHOLD)
1819  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1820 
1821  out[i] = value;
1822  }
1823 
1824  return 0;
1825 }
1826 
1827 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1828  int *got_frame_ptr, AVPacket *avpkt)
1829 {
1830  AVFrame *frame = data;
1831  const uint8_t *buf = avpkt->data;
1832  int buf_size = avpkt->size;
1833  QDM2Context *s = avctx->priv_data;
1834  int16_t *out;
1835  int i, ret;
1836 
1837  if(!buf)
1838  return 0;
1839  if(buf_size < s->checksum_size)
1840  return -1;
1841 
1842  /* get output buffer */
1843  frame->nb_samples = 16 * s->frame_size;
1844  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1845  return ret;
1846  out = (int16_t *)frame->data[0];
1847 
1848  for (i = 0; i < 16; i++) {
1849  if ((ret = qdm2_decode(s, buf, out)) < 0)
1850  return ret;
1851  out += s->channels * s->frame_size;
1852  }
1853 
1854  *got_frame_ptr = 1;
1855 
1856  return s->checksum_size;
1857 }
1858 
1860  .name = "qdm2",
1861  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1862  .type = AVMEDIA_TYPE_AUDIO,
1863  .id = AV_CODEC_ID_QDM2,
1864  .priv_data_size = sizeof(QDM2Context),
1866  .close = qdm2_decode_close,
1868  .capabilities = AV_CODEC_CAP_DR1,
1869 };
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:112
#define NULL
Definition: coverity.c:32
const char * s
Definition: avisynth_c.h:768
#define AVERROR_INVALIDDATA
Invalid data found when processing input.
Definition: error.h:59
#define SBLIMIT
Definition: mpegaudio.h:44
FFTTone fft_tones[1000]
FFT and tones.
Definition: qdm2.c:154
A node in the subpacket list.
Definition: qdm2.c:92
This structure describes decoded (raw) audio or video data.
Definition: frame.h:194
QDM2FFT fft
Definition: qdm2.c:163
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:370
ptrdiff_t const GLvoid * data
Definition: opengl_enc.c:101
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:868
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:261
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1811
static const float fft_tone_level_table[2][64]
Definition: qdm2data.h:438
static av_cold int init(AVCodecContext *avctx)
Definition: avrndec.c:35
static void average_quantized_coeffs(QDM2Context *q)
Replace 8 elements with their average value.
Definition: qdm2.c:314
Subpacket.
Definition: qdm2.c:83
int acc
Definition: yuv2rgb.c:546
int fft_coefs_index
Definition: qdm2.c:158
static VLC vlc_tab_tone_level_idx_hi2
#define QDM2_MAX_FRAME_SIZE
Definition: qdm2.c:76
float synth_buf[MPA_MAX_CHANNELS][512 *2]
Definition: qdm2.c:172
int size
Definition: avcodec.h:1672
const char * b
Definition: vf_curves.c:113
int flag
Definition: cpu.c:34
const uint8_t * buffer
Definition: get_bits.h:56
int av_log2(unsigned v)
Definition: intmath.c:26
int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8]
Definition: qdm2.c:181
const float * table
Definition: qdm2.c:105
static av_always_inline void bytestream2_init(GetByteContext *g, const uint8_t *buf, int buf_size)
Definition: bytestream.h:133
int coeff_per_sb_select
selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
Definition: qdm2.c:142
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
Definition: qdm2.c:1756
short cutoff
Definition: qdm2.c:110
unsigned int size
subpacket size
Definition: qdm2.c:85
int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]
Definition: qdm2.c:184
int sub_packet
Definition: qdm2.c:193
uint8_t run
Definition: svq3.c:206
float sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]
Definition: qdm2.c:174
#define AV_CH_LAYOUT_STEREO
int frequency_range
Definition: qdm2.c:140
static VLC fft_stereo_exp_vlc
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:262
AVCodec.
Definition: avcodec.h:3722
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, int offset, int duration, int channel, int exp, int phase)
Definition: qdm2.c:1237
QDM2SubPNode sub_packet_list_C[16]
packets with errors?
Definition: qdm2.c:150
static void decode(AVCodecContext *dec_ctx, AVPacket *pkt, AVFrame *frame, FILE *outfile)
Definition: decode_audio.c:42
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:297
static VLC vlc_tab_type30
float re
Definition: qdm2.c:98
int phase
Definition: qdm2.c:106
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
Init parameters from codec extradata.
Definition: qdm2.c:1606
QDM2 decoder context.
Definition: qdm2.c:128
static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
Definition: qdm2.c:201
void void avpriv_request_sample(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
enum AVSampleFormat sample_fmt
audio sample format
Definition: avcodec.h:2516
uint8_t
#define av_cold
Definition: attributes.h:82
int fft_order
order of FFT (actually fftorder+1)
Definition: qdm2.c:138
static void qdm2_decode_fft_packets(QDM2Context *q)
Definition: qdm2.c:1339
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(constuint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(constint16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(constint32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(constint64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(constint64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(constfloat *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(constdouble *) pi *(INT64_C(1)<< 63)))#defineFMT_PAIR_FUNC(out, in) staticconv_func_type *constfmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};staticvoidcpy1(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, len);}staticvoidcpy2(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 2 *len);}staticvoidcpy4(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 4 *len);}staticvoidcpy8(uint8_t **dst, constuint8_t **src, intlen){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, constint *ch_map, intflags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) returnNULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) returnNULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case1:ctx->simd_f=cpy1;break;case2:ctx->simd_f=cpy2;break;case4:ctx->simd_f=cpy4;break;case8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);returnctx;}voidswri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}intswri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, intlen){intch;intoff=0;constintos=(out->planar?1:out->ch_count)*out->bps;unsignedmisaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){intplanes=in->planar?in->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){intplanes=out->planar?out->ch_count:1;unsignedm=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){intplanes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
int sub_sampling
subsampling: 0=25%, 1=50%, 2=100% */
Definition: qdm2.c:141
void ff_mpa_synth_init_float(float *window)
#define SOFTCLIP_THRESHOLD
Definition: qdm2_tablegen.h:31
uint8_t * extradata
some codecs need / can use extradata like Huffman tables.
Definition: avcodec.h:1861
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
Definition: qdm2.c:1459
int64_t duration
Definition: movenc.c:63
static AVFrame * frame
static const int16_t fft_level_index_table[256]
Definition: qdm2data.h:238
#define DECLARE_ALIGNED(n, t, v)
Declare a variable that is aligned in memory.
Definition: mem.h:104
static const float fft_tone_envelope_table[4][31]
Definition: qdm2data.h:476
uint8_t * data
Definition: avcodec.h:1671
static int get_bits_count(const GetBitContext *s)
Definition: get_bits.h:199
bitstream reader API header.
static const uint8_t coeff_per_sb_for_dequant[3][30]
Definition: qdm2data.h:300
int checksum_size
size of data block, used also for checksum
Definition: qdm2.c:134
ptrdiff_t size
Definition: opengl_enc.c:101
static const uint8_t header[24]
Definition: sdr2.c:67
#define FFALIGN(x, a)
Definition: macros.h:48
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 12.
Definition: qdm2.c:1062
static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: qdm2.c:1827
static const uint8_t fft_subpackets[32]
Definition: qdm2data.h:510
#define av_log(a,...)
static av_cold void qdm2_init_static_data(void)
Init static data (does not depend on specific file)
Definition: qdm2.c:1588
int channels
number of channels
Definition: qdm2.c:131
static void fill_tone_level_array(QDM2Context *q, int flag)
Related to synthesis filter Called by process_subpacket_10.
Definition: qdm2.c:436
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:652
#define U(x)
Definition: vp56_arith.h:37
static av_cold void qdm2_init_vlc(void)
static int get_bits_left(GetBitContext *gb)
Definition: get_bits.h:587
int synth_buf_offset[MPA_MAX_CHANNELS]
Definition: qdm2.c:173
static VLC fft_level_exp_vlc
static av_cold void rnd_table_init(void)
Definition: qdm2_tablegen.h:57
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
static uint8_t random_dequant_type24[128][3]
Definition: qdm2_tablegen.h:44
int compressed_size
Definition: qdm2.c:167
const uint8_t * data
pointer to subpacket data (points to input data buffer, it's not a private copy)
Definition: qdm2.c:86
static VLC vlc_tab_tone_level_idx_mid
int16_t offset
Definition: qdm2.c:116
static av_always_inline void bytestream2_skip(GetByteContext *g, unsigned int size)
Definition: bytestream.h:164
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:179
static const int switchtable[23]
Definition: qdm2.c:197
int group_size
size of frame group (16 frames per group)
Definition: qdm2.c:132
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
static av_always_inline unsigned int bytestream2_get_bytes_left(GetByteContext *g)
Definition: bytestream.h:154
int sub_packets_B
number of packets on 'B' list
Definition: qdm2.c:149
QDM2SubPNode sub_packet_list_A[16]
list of all packets
Definition: qdm2.c:147
int noise_idx
index for dithering noise table
Definition: qdm2.c:194
GLsizei GLsizei * length
Definition: opengl_enc.c:115
Definition: avfft.h:73
const char * name
Name of the codec implementation.
Definition: avcodec.h:3729
uint8_t channel
Definition: qdm2.c:115
int duration
Definition: qdm2.c:108
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
Definition: qdm2.c:121
float tone_level[MPA_MAX_CHANNELS][30][64]
Mixed temporary data used in decoding.
Definition: qdm2.c:178
float FFTSample
Definition: avfft.h:35
int8_t exp
Definition: eval.c:65
int depth
Definition: v4l.c:62
RDFTContext rdft_ctx
Definition: qdm2.c:162
Definition: vlc.h:26
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2559
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:38
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]
Definition: qdm2.c:180
static void qdm2_synthesis_filter(QDM2Context *q, int index)
Definition: qdm2.c:1550
static VLC vlc_tab_tone_level_idx_hi1
#define QDM2_SB_USED(sub_sampling)
Definition: qdm2.c:62
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:42
int group_order
Parameters built from header parameters, do not change during playback.
Definition: qdm2.c:137
static VLC fft_level_exp_alt_vlc
audio channel layout utility functions
static float noise_samples[128]
Definition: qdm2_tablegen.h:45
QDM2SubPNode sub_packet_list_B[16]
FFT packets B are on list.
Definition: qdm2.c:148
struct QDM2SubPNode * next
pointer to next packet in the list, NULL if leaf node
Definition: qdm2.c:94
GLsizei GLboolean const GLfloat * value
Definition: opengl_enc.c:109
static const int8_t tone_level_idx_offset_table[30][4]
Definition: qdm2data.h:307
float ff_mpa_synth_window_float[]
static void qdm2_decode_super_block(QDM2Context *q)
Decode superblock, fill packet lists.
Definition: qdm2.c:1113
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:554
#define SAMPLES_NEEDED_2(why)
Definition: qdm2.c:73
static const int8_t coding_method_table[5][30]
Definition: qdm2data.h:342
static VLC fft_stereo_phase_vlc
int n
Definition: avisynth_c.h:684
static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
QDM2 checksum.
Definition: qdm2.c:246
#define QDM2_LIST_ADD(list, size, packet)
Definition: qdm2.c:51
static uint8_t random_dequant_index[256][5]
Definition: qdm2_tablegen.h:43
static const float type30_dequant[8]
Definition: qdm2data.h:521
int fft_tone_end
Definition: qdm2.c:156
static uint16_t softclip_table[HARDCLIP_THRESHOLD-SOFTCLIP_THRESHOLD+1]
Definition: qdm2_tablegen.h:41
#define FF_ARRAY_ELEMS(a)
QDM2Complex complex[MPA_MAX_CHANNELS][256]
Definition: qdm2.c:122
static const float type34_delta[10]
Definition: qdm2data.h:526
static VLC vlc_tab_fft_tone_offset[5]
int bits
Definition: vlc.h:27
static const float dequant_1bit[2][3]
Definition: qdm2data.h:516
static void comp(unsigned char *dst, ptrdiff_t dst_stride, unsigned char *src, ptrdiff_t src_stride, int add)
Definition: eamad.c:83
#define AVERROR_PATCHWELCOME
Not yet implemented in FFmpeg, patches welcome.
Definition: error.h:62
static void build_sb_samples_from_noise(QDM2Context *q, int sb)
Build subband samples with noise weighted by q->tone_level.
Definition: qdm2.c:343
float samples[MPA_MAX_CHANNELS *MPA_FRAME_SIZE]
Definition: qdm2.c:175
static const uint8_t last_coeff[3]
Definition: qdm2data.h:257
int frame_size
Definition: mxfenc.c:1820
Libavcodec external API header.
static const int fft_cutoff_index_table[4][2]
Definition: qdm2data.h:234
int sample_rate
samples per second
Definition: avcodec.h:2508
#define SAMPLES_NEEDED
Definition: qdm2.c:70
static void qdm2_fft_decode_tones(QDM2Context *q, int duration, GetBitContext *gb, int b)
Definition: qdm2.c:1253
static const uint8_t coeff_per_sb_for_avg[3][30]
Definition: qdm2data.h:261
int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8]
Definition: qdm2.c:183
main external API structure.
Definition: avcodec.h:1746
static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
Definition: qdm2.c:230
float output_buffer[QDM2_MAX_FRAME_SIZE *MPA_MAX_CHANNELS *2]
Definition: qdm2.c:168
AVCodec ff_qdm2_decoder
Definition: qdm2.c:1859
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: decode.c:1748
uint8_t phase
Definition: qdm2.c:118
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(constuint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(constint16_t *) pi >>8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t,*(constint16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(constint32_t *) pi >>24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t,*(constint32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(constfloat *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(constfloat *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(constfloat *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(constdouble *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(constdouble *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(constdouble *) pi *(1U<< 31))))#defineSET_CONV_FUNC_GROUP(ofmt, ifmt) staticvoidset_generic_function(AudioConvert *ac){}voidff_audio_convert_free(AudioConvert **ac){if(!*ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);}AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enumAVSampleFormatout_fmt, enumAVSampleFormatin_fmt, intchannels, intsample_rate, intapply_map){AudioConvert *ac;intin_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) returnNULL;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);returnNULL;}returnac;}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;}elseif(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;elseac->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);returnac;}intff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in){intuse_generic=1;intlen=in->nb_samples;intp;if(ac->dc){av_log(ac->avr, AV_LOG_TRACE,"%dsamples-audio_convert:%sto%s(dithered)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));returnff_convert_dither(ac-> in
int fft_coefs_min_index[5]
Definition: qdm2.c:159
void * buf
Definition: avisynth_c.h:690
GLint GLenum type
Definition: opengl_enc.c:105
FFTCoefficient fft_coefs[1000]
Definition: qdm2.c:157
int extradata_size
Definition: avcodec.h:1862
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:313
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:306
int index
Definition: gxfenc.c:89
int has_errors
packet has errors
Definition: qdm2.c:189
static const uint8_t dequant_table[64]
Definition: 4xm.c:114
int fft_level_exp[6]
Definition: qdm2.c:161
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:425
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:525
int16_t sub_packet
Definition: qdm2.c:114
#define HARDCLIP_THRESHOLD
Definition: qdm2_tablegen.h:32
float im
Definition: qdm2.c:99
int16_t exp
Definition: qdm2.c:117
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)
int8_t coding_method[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:179
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 10 if not null, else.
Definition: qdm2.c:1014
static av_cold void softclip_table_init(void)
Definition: qdm2_tablegen.h:47
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:208
uint8_t level
Definition: svq3.c:207
int fft_size
size of FFT, in complex numbers
Definition: qdm2.c:133
int type
subpacket type
Definition: qdm2.c:84
int fft_coefs_max_index[5]
Definition: qdm2.c:160
int frame_size
size of data frame
Definition: qdm2.c:139
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
Definition: qdm2.c:1765
#define FIX_NOISE_IDX(noise_idx)
Definition: qdm2.c:64
static const float fft_tone_sample_table[4][16][5]
Definition: qdm2data.h:368
int
Definition: qdm2.c:102
int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]
Definition: qdm2.c:182
int nb_channels
Parameters from codec header, do not change during playback.
Definition: qdm2.c:130
int superblocktype_2_3
select fft tables and some algorithm based on superblock type
Definition: qdm2.c:190
common internal api header.
if(ret< 0)
Definition: vf_mcdeint.c:282
static VLC vlc_tab_diff
Definition: qdm2_tablegen.h:99
int cm_table_select
selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
Definition: qdm2.c:143
signed 16 bits
Definition: samplefmt.h:61
static double c[64]
channel
Use these values when setting the channel map with ebur128_set_channel().
Definition: ebur128.h:39
QDM2SubPacket * packet
packet
Definition: qdm2.c:93
QDM2SubPacket sub_packets[16]
Packets and packet lists.
Definition: qdm2.c:146
static const int vlc_stage3_values[60]
Definition: qdm2data.h:360
mpeg audio declarations for both encoder and decoder.
QDM2Complex * complex
Definition: qdm2.c:104
int do_synth_filter
used to perform or skip synthesis filter
Definition: qdm2.c:191
const uint8_t * compressed_data
I/O data.
Definition: qdm2.c:166
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:186
static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 9, init quantized_coeffs with data from it.
Definition: qdm2.c:972
#define MKBETAG(a, b, c, d)
Definition: common.h:343
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 11.
Definition: qdm2.c:1033
MPADSPContext mpadsp
Synthesis filter.
Definition: qdm2.c:171
void * priv_data
Definition: avcodec.h:1788
static VLC vlc_tab_level
Definition: qdm2_tablegen.h:98
static VLC vlc_tab_run
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:909
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
Definition: avcodec.h:2509
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
Definition: qdm2.c:1413
QDM2SubPNode sub_packet_list_D[16]
DCT packets.
Definition: qdm2.c:151
int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:185
VLC_TYPE(* table)[2]
code, bits
Definition: vlc.h:28
static const struct twinvq_data tab
FILE * out
Definition: movenc.c:54
short time_index
Definition: qdm2.c:109
int8_t sb_int8_array[2][30][64]
Definition: qdm2.c:78
#define M_PI
Definition: mathematics.h:52
#define SB_DITHERING_NOISE(sb, noise_idx)
Definition: qdm2.c:68
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
Definition: qdm2.c:1530
int nb_channels
int phase_shift
Definition: qdm2.c:107
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
Process new subpackets for synthesis filter.
Definition: qdm2.c:1081
#define AV_CH_LAYOUT_MONO
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
Set up a real FFT.
Definition: rdft.c:86
#define MPA_FRAME_SIZE
Definition: mpegaudio.h:37
float min
This structure stores compressed data.
Definition: avcodec.h:1648
av_cold void ff_mpadsp_init(MPADSPContext *s)
Definition: mpegaudiodsp.c:27
static av_cold void init_noise_samples(void)
Definition: qdm2_tablegen.h:88
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:260
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:1001
static VLC vlc_tab_type34
for(j=16;j >0;--j)
float level
Definition: qdm2.c:103
int fft_tone_start
Definition: qdm2.c:155
static uint8_t tmp[11]
Definition: aes_ctr.c:26