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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 "internal.h"
44 #include "mpegaudio.h"
45 #include "mpegaudiodsp.h"
46 #include "rdft.h"
47 
48 #include "qdm2_tablegen.h"
49 
50 #define QDM2_LIST_ADD(list, size, packet) \
51 do { \
52  if (size > 0) { \
53  list[size - 1].next = &list[size]; \
54  } \
55  list[size].packet = packet; \
56  list[size].next = NULL; \
57  size++; \
58 } while(0)
59 
60 // Result is 8, 16 or 30
61 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
62 
63 #define FIX_NOISE_IDX(noise_idx) \
64  if ((noise_idx) >= 3840) \
65  (noise_idx) -= 3840; \
66 
67 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
68 
69 #define SAMPLES_NEEDED \
70  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
71 
72 #define SAMPLES_NEEDED_2(why) \
73  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
74 
75 #define QDM2_MAX_FRAME_SIZE 512
76 
77 typedef int8_t sb_int8_array[2][30][64];
78 
79 /**
80  * Subpacket
81  */
82 typedef struct QDM2SubPacket {
83  int type; ///< subpacket type
84  unsigned int size; ///< subpacket size
85  const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
87 
88 /**
89  * A node in the subpacket list
90  */
91 typedef struct QDM2SubPNode {
92  QDM2SubPacket *packet; ///< packet
93  struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
94 } QDM2SubPNode;
95 
96 typedef struct QDM2Complex {
97  float re;
98  float im;
99 } QDM2Complex;
100 
101 typedef struct FFTTone {
102  float level;
104  const float *table;
105  int phase;
107  int duration;
108  short time_index;
109  short cutoff;
110 } FFTTone;
111 
112 typedef struct FFTCoefficient {
113  int16_t sub_packet;
115  int16_t offset;
116  int16_t exp;
119 
120 typedef struct QDM2FFT {
122 } QDM2FFT;
123 
124 /**
125  * QDM2 decoder context
126  */
127 typedef struct QDM2Context {
128  /// Parameters from codec header, do not change during playback
129  int nb_channels; ///< number of channels
130  int channels; ///< number of channels
131  int group_size; ///< size of frame group (16 frames per group)
132  int fft_size; ///< size of FFT, in complex numbers
133  int checksum_size; ///< size of data block, used also for checksum
134 
135  /// Parameters built from header parameters, do not change during playback
136  int group_order; ///< order of frame group
137  int fft_order; ///< order of FFT (actually fftorder+1)
138  int frame_size; ///< size of data frame
140  int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
141  int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
142  int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
143 
144  /// Packets and packet lists
145  QDM2SubPacket sub_packets[16]; ///< the packets themselves
146  QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
147  QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
148  int sub_packets_B; ///< number of packets on 'B' list
149  QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
150  QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
151 
152  /// FFT and tones
163 
164  /// I/O data
168 
169  /// Synthesis filter
175 
176  /// Mixed temporary data used in decoding
177  float tone_level[MPA_MAX_CHANNELS][30][64];
186 
187  // Flags
188  int has_errors; ///< packet has errors
189  int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
190  int do_synth_filter; ///< used to perform or skip synthesis filter
191 
193  int noise_idx; ///< index for dithering noise table
194 } QDM2Context;
195 
196 static const int switchtable[23] = {
197  0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
198 };
199 
200 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
201 {
202  int value;
203 
204  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
205 
206  /* stage-2, 3 bits exponent escape sequence */
207  if (value-- == 0)
208  value = get_bits(gb, get_bits(gb, 3) + 1);
209 
210  /* stage-3, optional */
211  if (flag) {
212  int tmp;
213 
214  if (value >= 60) {
215  av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
216  return 0;
217  }
218 
219  tmp= vlc_stage3_values[value];
220 
221  if ((value & ~3) > 0)
222  tmp += get_bits(gb, (value >> 2));
223  value = tmp;
224  }
225 
226  return value;
227 }
228 
229 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
230 {
231  int value = qdm2_get_vlc(gb, vlc, 0, depth);
232 
233  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
234 }
235 
236 /**
237  * QDM2 checksum
238  *
239  * @param data pointer to data to be checksummed
240  * @param length data length
241  * @param value checksum value
242  *
243  * @return 0 if checksum is OK
244  */
245 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
246 {
247  int i;
248 
249  for (i = 0; i < length; i++)
250  value -= data[i];
251 
252  return (uint16_t)(value & 0xffff);
253 }
254 
255 /**
256  * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
257  *
258  * @param gb bitreader context
259  * @param sub_packet packet under analysis
260  */
262  QDM2SubPacket *sub_packet)
263 {
264  sub_packet->type = get_bits(gb, 8);
265 
266  if (sub_packet->type == 0) {
267  sub_packet->size = 0;
268  sub_packet->data = NULL;
269  } else {
270  sub_packet->size = get_bits(gb, 8);
271 
272  if (sub_packet->type & 0x80) {
273  sub_packet->size <<= 8;
274  sub_packet->size |= get_bits(gb, 8);
275  sub_packet->type &= 0x7f;
276  }
277 
278  if (sub_packet->type == 0x7f)
279  sub_packet->type |= (get_bits(gb, 8) << 8);
280 
281  // FIXME: this depends on bitreader-internal data
282  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
283  }
284 
285  av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
286  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
287 }
288 
289 /**
290  * Return node pointer to first packet of requested type in list.
291  *
292  * @param list list of subpackets to be scanned
293  * @param type type of searched subpacket
294  * @return node pointer for subpacket if found, else NULL
295  */
297  int type)
298 {
299  while (list && list->packet) {
300  if (list->packet->type == type)
301  return list;
302  list = list->next;
303  }
304  return NULL;
305 }
306 
307 /**
308  * Replace 8 elements with their average value.
309  * Called by qdm2_decode_superblock before starting subblock decoding.
310  *
311  * @param q context
312  */
314 {
315  int i, j, n, ch, sum;
316 
318 
319  for (ch = 0; ch < q->nb_channels; ch++)
320  for (i = 0; i < n; i++) {
321  sum = 0;
322 
323  for (j = 0; j < 8; j++)
324  sum += q->quantized_coeffs[ch][i][j];
325 
326  sum /= 8;
327  if (sum > 0)
328  sum--;
329 
330  for (j = 0; j < 8; j++)
331  q->quantized_coeffs[ch][i][j] = sum;
332  }
333 }
334 
335 /**
336  * Build subband samples with noise weighted by q->tone_level.
337  * Called by synthfilt_build_sb_samples.
338  *
339  * @param q context
340  * @param sb subband index
341  */
343 {
344  int ch, j;
345 
347 
348  if (!q->nb_channels)
349  return;
350 
351  for (ch = 0; ch < q->nb_channels; ch++) {
352  for (j = 0; j < 64; j++) {
353  q->sb_samples[ch][j * 2][sb] =
354  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
355  q->sb_samples[ch][j * 2 + 1][sb] =
356  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
357  }
358  }
359 }
360 
361 /**
362  * Called while processing data from subpackets 11 and 12.
363  * Used after making changes to coding_method array.
364  *
365  * @param sb subband index
366  * @param channels number of channels
367  * @param coding_method q->coding_method[0][0][0]
368  */
369 static int fix_coding_method_array(int sb, int channels,
370  sb_int8_array coding_method)
371 {
372  int j, k;
373  int ch;
374  int run, case_val;
375 
376  for (ch = 0; ch < channels; ch++) {
377  for (j = 0; j < 64; ) {
378  if (coding_method[ch][sb][j] < 8)
379  return -1;
380  if ((coding_method[ch][sb][j] - 8) > 22) {
381  run = 1;
382  case_val = 8;
383  } else {
384  switch (switchtable[coding_method[ch][sb][j] - 8]) {
385  case 0: run = 10;
386  case_val = 10;
387  break;
388  case 1: run = 1;
389  case_val = 16;
390  break;
391  case 2: run = 5;
392  case_val = 24;
393  break;
394  case 3: run = 3;
395  case_val = 30;
396  break;
397  case 4: run = 1;
398  case_val = 30;
399  break;
400  case 5: run = 1;
401  case_val = 8;
402  break;
403  default: run = 1;
404  case_val = 8;
405  break;
406  }
407  }
408  for (k = 0; k < run; k++) {
409  if (j + k < 128) {
410  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
411  if (k > 0) {
413  //not debugged, almost never used
414  memset(&coding_method[ch][sb][j + k], case_val,
415  k *sizeof(int8_t));
416  memset(&coding_method[ch][sb][j + k], case_val,
417  3 * sizeof(int8_t));
418  }
419  }
420  }
421  }
422  j += run;
423  }
424  }
425  return 0;
426 }
427 
428 /**
429  * Related to synthesis filter
430  * Called by process_subpacket_10
431  *
432  * @param q context
433  * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
434  */
436 {
437  int i, sb, ch, sb_used;
438  int tmp, tab;
439 
440  for (ch = 0; ch < q->nb_channels; ch++)
441  for (sb = 0; sb < 30; sb++)
442  for (i = 0; i < 8; i++) {
444  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
446  else
447  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
448  if(tmp < 0)
449  tmp += 0xff;
450  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
451  }
452 
453  sb_used = QDM2_SB_USED(q->sub_sampling);
454 
455  if ((q->superblocktype_2_3 != 0) && !flag) {
456  for (sb = 0; sb < sb_used; sb++)
457  for (ch = 0; ch < q->nb_channels; ch++)
458  for (i = 0; i < 64; i++) {
459  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
460  if (q->tone_level_idx[ch][sb][i] < 0)
461  q->tone_level[ch][sb][i] = 0;
462  else
463  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
464  }
465  } else {
466  tab = q->superblocktype_2_3 ? 0 : 1;
467  for (sb = 0; sb < sb_used; sb++) {
468  if ((sb >= 4) && (sb <= 23)) {
469  for (ch = 0; ch < q->nb_channels; ch++)
470  for (i = 0; i < 64; i++) {
471  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
472  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
473  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
474  q->tone_level_idx_hi2[ch][sb - 4];
475  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
476  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
477  q->tone_level[ch][sb][i] = 0;
478  else
479  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
480  }
481  } else {
482  if (sb > 4) {
483  for (ch = 0; ch < q->nb_channels; ch++)
484  for (i = 0; i < 64; i++) {
485  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
486  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
487  q->tone_level_idx_hi2[ch][sb - 4];
488  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
489  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
490  q->tone_level[ch][sb][i] = 0;
491  else
492  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
493  }
494  } else {
495  for (ch = 0; ch < q->nb_channels; ch++)
496  for (i = 0; i < 64; i++) {
497  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
498  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
499  q->tone_level[ch][sb][i] = 0;
500  else
501  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
502  }
503  }
504  }
505  }
506  }
507 }
508 
509 /**
510  * Related to synthesis filter
511  * Called by process_subpacket_11
512  * c is built with data from subpacket 11
513  * Most of this function is used only if superblock_type_2_3 == 0,
514  * never seen it in samples.
515  *
516  * @param tone_level_idx
517  * @param tone_level_idx_temp
518  * @param coding_method q->coding_method[0][0][0]
519  * @param nb_channels number of channels
520  * @param c coming from subpacket 11, passed as 8*c
521  * @param superblocktype_2_3 flag based on superblock packet type
522  * @param cm_table_select q->cm_table_select
523  */
524 static void fill_coding_method_array(sb_int8_array tone_level_idx,
525  sb_int8_array tone_level_idx_temp,
526  sb_int8_array coding_method,
527  int nb_channels,
528  int c, int superblocktype_2_3,
529  int cm_table_select)
530 {
531  int ch, sb, j;
532  int tmp, acc, esp_40, comp;
533  int add1, add2, add3, add4;
534  int64_t multres;
535 
536  if (!superblocktype_2_3) {
537  /* This case is untested, no samples available */
538  avpriv_request_sample(NULL, "!superblocktype_2_3");
539  return;
540  for (ch = 0; ch < nb_channels; ch++)
541  for (sb = 0; sb < 30; sb++) {
542  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
543  add1 = tone_level_idx[ch][sb][j] - 10;
544  if (add1 < 0)
545  add1 = 0;
546  add2 = add3 = add4 = 0;
547  if (sb > 1) {
548  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
549  if (add2 < 0)
550  add2 = 0;
551  }
552  if (sb > 0) {
553  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
554  if (add3 < 0)
555  add3 = 0;
556  }
557  if (sb < 29) {
558  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
559  if (add4 < 0)
560  add4 = 0;
561  }
562  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
563  if (tmp < 0)
564  tmp = 0;
565  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
566  }
567  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
568  }
569  acc = 0;
570  for (ch = 0; ch < nb_channels; ch++)
571  for (sb = 0; sb < 30; sb++)
572  for (j = 0; j < 64; j++)
573  acc += tone_level_idx_temp[ch][sb][j];
574 
575  multres = 0x66666667LL * (acc * 10);
576  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
577  for (ch = 0; ch < nb_channels; ch++)
578  for (sb = 0; sb < 30; sb++)
579  for (j = 0; j < 64; j++) {
580  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
581  if (comp < 0)
582  comp += 0xff;
583  comp /= 256; // signed shift
584  switch(sb) {
585  case 0:
586  if (comp < 30)
587  comp = 30;
588  comp += 15;
589  break;
590  case 1:
591  if (comp < 24)
592  comp = 24;
593  comp += 10;
594  break;
595  case 2:
596  case 3:
597  case 4:
598  if (comp < 16)
599  comp = 16;
600  }
601  if (comp <= 5)
602  tmp = 0;
603  else if (comp <= 10)
604  tmp = 10;
605  else if (comp <= 16)
606  tmp = 16;
607  else if (comp <= 24)
608  tmp = -1;
609  else
610  tmp = 0;
611  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
612  }
613  for (sb = 0; sb < 30; sb++)
614  fix_coding_method_array(sb, nb_channels, coding_method);
615  for (ch = 0; ch < nb_channels; ch++)
616  for (sb = 0; sb < 30; sb++)
617  for (j = 0; j < 64; j++)
618  if (sb >= 10) {
619  if (coding_method[ch][sb][j] < 10)
620  coding_method[ch][sb][j] = 10;
621  } else {
622  if (sb >= 2) {
623  if (coding_method[ch][sb][j] < 16)
624  coding_method[ch][sb][j] = 16;
625  } else {
626  if (coding_method[ch][sb][j] < 30)
627  coding_method[ch][sb][j] = 30;
628  }
629  }
630  } else { // superblocktype_2_3 != 0
631  for (ch = 0; ch < nb_channels; ch++)
632  for (sb = 0; sb < 30; sb++)
633  for (j = 0; j < 64; j++)
634  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
635  }
636 }
637 
638 /**
639  * Called by process_subpacket_11 to process more data from subpacket 11
640  * with sb 0-8.
641  * Called by process_subpacket_12 to process data from subpacket 12 with
642  * sb 8-sb_used.
643  *
644  * @param q context
645  * @param gb bitreader context
646  * @param length packet length in bits
647  * @param sb_min lower subband processed (sb_min included)
648  * @param sb_max higher subband processed (sb_max excluded)
649  */
651  int length, int sb_min, int sb_max)
652 {
653  int sb, j, k, n, ch, run, channels;
654  int joined_stereo, zero_encoding;
655  int type34_first;
656  float type34_div = 0;
657  float type34_predictor;
658  float samples[10];
659  int sign_bits[16] = {0};
660 
661  if (length == 0) {
662  // If no data use noise
663  for (sb=sb_min; sb < sb_max; sb++)
665 
666  return 0;
667  }
668 
669  for (sb = sb_min; sb < sb_max; sb++) {
670  channels = q->nb_channels;
671 
672  if (q->nb_channels <= 1 || sb < 12)
673  joined_stereo = 0;
674  else if (sb >= 24)
675  joined_stereo = 1;
676  else
677  joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
678 
679  if (joined_stereo) {
680  if (get_bits_left(gb) >= 16)
681  for (j = 0; j < 16; j++)
682  sign_bits[j] = get_bits1(gb);
683 
684  for (j = 0; j < 64; j++)
685  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
686  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
687 
689  q->coding_method)) {
690  av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
692  continue;
693  }
694  channels = 1;
695  }
696 
697  for (ch = 0; ch < channels; ch++) {
699  zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
700  type34_predictor = 0.0;
701  type34_first = 1;
702 
703  for (j = 0; j < 128; ) {
704  switch (q->coding_method[ch][sb][j / 2]) {
705  case 8:
706  if (get_bits_left(gb) >= 10) {
707  if (zero_encoding) {
708  for (k = 0; k < 5; k++) {
709  if ((j + 2 * k) >= 128)
710  break;
711  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
712  }
713  } else {
714  n = get_bits(gb, 8);
715  if (n >= 243) {
716  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
717  return AVERROR_INVALIDDATA;
718  }
719 
720  for (k = 0; k < 5; k++)
721  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
722  }
723  for (k = 0; k < 5; k++)
724  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
725  } else {
726  for (k = 0; k < 10; k++)
727  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
728  }
729  run = 10;
730  break;
731 
732  case 10:
733  if (get_bits_left(gb) >= 1) {
734  float f = 0.81;
735 
736  if (get_bits1(gb))
737  f = -f;
738  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
739  samples[0] = f;
740  } else {
741  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
742  }
743  run = 1;
744  break;
745 
746  case 16:
747  if (get_bits_left(gb) >= 10) {
748  if (zero_encoding) {
749  for (k = 0; k < 5; k++) {
750  if ((j + k) >= 128)
751  break;
752  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
753  }
754  } else {
755  n = get_bits (gb, 8);
756  if (n >= 243) {
757  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
758  return AVERROR_INVALIDDATA;
759  }
760 
761  for (k = 0; k < 5; k++)
762  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
763  }
764  } else {
765  for (k = 0; k < 5; k++)
766  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
767  }
768  run = 5;
769  break;
770 
771  case 24:
772  if (get_bits_left(gb) >= 7) {
773  n = get_bits(gb, 7);
774  if (n >= 125) {
775  av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
776  return AVERROR_INVALIDDATA;
777  }
778 
779  for (k = 0; k < 3; k++)
780  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
781  } else {
782  for (k = 0; k < 3; k++)
783  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
784  }
785  run = 3;
786  break;
787 
788  case 30:
789  if (get_bits_left(gb) >= 4) {
790  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
791  if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
792  av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
793  return AVERROR_INVALIDDATA;
794  }
795  samples[0] = type30_dequant[index];
796  } else
797  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
798 
799  run = 1;
800  break;
801 
802  case 34:
803  if (get_bits_left(gb) >= 7) {
804  if (type34_first) {
805  type34_div = (float)(1 << get_bits(gb, 2));
806  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
807  type34_predictor = samples[0];
808  type34_first = 0;
809  } else {
810  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
811  if (index >= FF_ARRAY_ELEMS(type34_delta)) {
812  av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
813  return AVERROR_INVALIDDATA;
814  }
815  samples[0] = type34_delta[index] / type34_div + type34_predictor;
816  type34_predictor = samples[0];
817  }
818  } else {
819  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
820  }
821  run = 1;
822  break;
823 
824  default:
825  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
826  run = 1;
827  break;
828  }
829 
830  if (joined_stereo) {
831  for (k = 0; k < run && j + k < 128; k++) {
832  q->sb_samples[0][j + k][sb] =
833  q->tone_level[0][sb][(j + k) / 2] * samples[k];
834  if (q->nb_channels == 2) {
835  if (sign_bits[(j + k) / 8])
836  q->sb_samples[1][j + k][sb] =
837  q->tone_level[1][sb][(j + k) / 2] * -samples[k];
838  else
839  q->sb_samples[1][j + k][sb] =
840  q->tone_level[1][sb][(j + k) / 2] * samples[k];
841  }
842  }
843  } else {
844  for (k = 0; k < run; k++)
845  if ((j + k) < 128)
846  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
847  }
848 
849  j += run;
850  } // j loop
851  } // channel loop
852  } // subband loop
853  return 0;
854 }
855 
856 /**
857  * Init the first element of a channel in quantized_coeffs with data
858  * from packet 10 (quantized_coeffs[ch][0]).
859  * This is similar to process_subpacket_9, but for a single channel
860  * and for element [0]
861  * same VLC tables as process_subpacket_9 are used.
862  *
863  * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
864  * @param gb bitreader context
865  */
866 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
867  GetBitContext *gb)
868 {
869  int i, k, run, level, diff;
870 
871  if (get_bits_left(gb) < 16)
872  return -1;
873  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
874 
875  quantized_coeffs[0] = level;
876 
877  for (i = 0; i < 7; ) {
878  if (get_bits_left(gb) < 16)
879  return -1;
880  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
881 
882  if (i + run >= 8)
883  return -1;
884 
885  if (get_bits_left(gb) < 16)
886  return -1;
887  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
888 
889  for (k = 1; k <= run; k++)
890  quantized_coeffs[i + k] = (level + ((k * diff) / run));
891 
892  level += diff;
893  i += run;
894  }
895  return 0;
896 }
897 
898 /**
899  * Related to synthesis filter, process data from packet 10
900  * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
901  * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
902  * data from packet 10
903  *
904  * @param q context
905  * @param gb bitreader context
906  */
908 {
909  int sb, j, k, n, ch;
910 
911  for (ch = 0; ch < q->nb_channels; ch++) {
913 
914  if (get_bits_left(gb) < 16) {
915  memset(q->quantized_coeffs[ch][0], 0, 8);
916  break;
917  }
918  }
919 
920  n = q->sub_sampling + 1;
921 
922  for (sb = 0; sb < n; sb++)
923  for (ch = 0; ch < q->nb_channels; ch++)
924  for (j = 0; j < 8; j++) {
925  if (get_bits_left(gb) < 1)
926  break;
927  if (get_bits1(gb)) {
928  for (k=0; k < 8; k++) {
929  if (get_bits_left(gb) < 16)
930  break;
931  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
932  }
933  } else {
934  for (k=0; k < 8; k++)
935  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
936  }
937  }
938 
939  n = QDM2_SB_USED(q->sub_sampling) - 4;
940 
941  for (sb = 0; sb < n; sb++)
942  for (ch = 0; ch < q->nb_channels; ch++) {
943  if (get_bits_left(gb) < 16)
944  break;
946  if (sb > 19)
947  q->tone_level_idx_hi2[ch][sb] -= 16;
948  else
949  for (j = 0; j < 8; j++)
950  q->tone_level_idx_mid[ch][sb][j] = -16;
951  }
952 
953  n = QDM2_SB_USED(q->sub_sampling) - 5;
954 
955  for (sb = 0; sb < n; sb++)
956  for (ch = 0; ch < q->nb_channels; ch++)
957  for (j = 0; j < 8; j++) {
958  if (get_bits_left(gb) < 16)
959  break;
960  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
961  }
962 }
963 
964 /**
965  * Process subpacket 9, init quantized_coeffs with data from it
966  *
967  * @param q context
968  * @param node pointer to node with packet
969  */
971 {
972  GetBitContext gb;
973  int i, j, k, n, ch, run, level, diff;
974 
975  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
976 
978 
979  for (i = 1; i < n; i++)
980  for (ch = 0; ch < q->nb_channels; ch++) {
981  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
982  q->quantized_coeffs[ch][i][0] = level;
983 
984  for (j = 0; j < (8 - 1); ) {
985  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
986  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
987 
988  if (j + run >= 8)
989  return -1;
990 
991  for (k = 1; k <= run; k++)
992  q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
993 
994  level += diff;
995  j += run;
996  }
997  }
998 
999  for (ch = 0; ch < q->nb_channels; ch++)
1000  for (i = 0; i < 8; i++)
1001  q->quantized_coeffs[ch][0][i] = 0;
1002 
1003  return 0;
1004 }
1005 
1006 /**
1007  * Process subpacket 10 if not null, else
1008  *
1009  * @param q context
1010  * @param node pointer to node with packet
1011  */
1013 {
1014  GetBitContext gb;
1015 
1016  if (node) {
1017  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1019  fill_tone_level_array(q, 1);
1020  } else {
1021  fill_tone_level_array(q, 0);
1022  }
1023 }
1024 
1025 /**
1026  * Process subpacket 11
1027  *
1028  * @param q context
1029  * @param node pointer to node with packet
1030  */
1032 {
1033  GetBitContext gb;
1034  int length = 0;
1035 
1036  if (node) {
1037  length = node->packet->size * 8;
1038  init_get_bits(&gb, node->packet->data, length);
1039  }
1040 
1041  if (length >= 32) {
1042  int c = get_bits(&gb, 13);
1043 
1044  if (c > 3)
1047  q->nb_channels, 8 * c,
1049  }
1050 
1051  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1052 }
1053 
1054 /**
1055  * Process subpacket 12
1056  *
1057  * @param q context
1058  * @param node pointer to node with packet
1059  */
1061 {
1062  GetBitContext gb;
1063  int length = 0;
1064 
1065  if (node) {
1066  length = node->packet->size * 8;
1067  init_get_bits(&gb, node->packet->data, length);
1068  }
1069 
1070  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1071 }
1072 
1073 /**
1074  * Process new subpackets for synthesis filter
1075  *
1076  * @param q context
1077  * @param list list with synthesis filter packets (list D)
1078  */
1080 {
1081  QDM2SubPNode *nodes[4];
1082 
1083  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1084  if (nodes[0])
1085  process_subpacket_9(q, nodes[0]);
1086 
1087  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1088  if (nodes[1])
1089  process_subpacket_10(q, nodes[1]);
1090  else
1092 
1093  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1094  if (nodes[0] && nodes[1] && nodes[2])
1095  process_subpacket_11(q, nodes[2]);
1096  else
1098 
1099  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1100  if (nodes[0] && nodes[1] && nodes[3])
1101  process_subpacket_12(q, nodes[3]);
1102  else
1104 }
1105 
1106 /**
1107  * Decode superblock, fill packet lists.
1108  *
1109  * @param q context
1110  */
1112 {
1113  GetBitContext gb;
1114  QDM2SubPacket header, *packet;
1115  int i, packet_bytes, sub_packet_size, sub_packets_D;
1116  unsigned int next_index = 0;
1117 
1118  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1119  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1120  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1121 
1122  q->sub_packets_B = 0;
1123  sub_packets_D = 0;
1124 
1125  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1126 
1128  qdm2_decode_sub_packet_header(&gb, &header);
1129 
1130  if (header.type < 2 || header.type >= 8) {
1131  q->has_errors = 1;
1132  av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1133  return;
1134  }
1135 
1136  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1137  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1138 
1139  init_get_bits(&gb, header.data, header.size * 8);
1140 
1141  if (header.type == 2 || header.type == 4 || header.type == 5) {
1142  int csum = 257 * get_bits(&gb, 8);
1143  csum += 2 * get_bits(&gb, 8);
1144 
1145  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1146 
1147  if (csum != 0) {
1148  q->has_errors = 1;
1149  av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1150  return;
1151  }
1152  }
1153 
1154  q->sub_packet_list_B[0].packet = NULL;
1155  q->sub_packet_list_D[0].packet = NULL;
1156 
1157  for (i = 0; i < 6; i++)
1158  if (--q->fft_level_exp[i] < 0)
1159  q->fft_level_exp[i] = 0;
1160 
1161  for (i = 0; packet_bytes > 0; i++) {
1162  int j;
1163 
1164  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1165  SAMPLES_NEEDED_2("too many packet bytes");
1166  return;
1167  }
1168 
1169  q->sub_packet_list_A[i].next = NULL;
1170 
1171  if (i > 0) {
1172  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1173 
1174  /* seek to next block */
1175  init_get_bits(&gb, header.data, header.size * 8);
1176  skip_bits(&gb, next_index * 8);
1177 
1178  if (next_index >= header.size)
1179  break;
1180  }
1181 
1182  /* decode subpacket */
1183  packet = &q->sub_packets[i];
1184  qdm2_decode_sub_packet_header(&gb, packet);
1185  next_index = packet->size + get_bits_count(&gb) / 8;
1186  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1187 
1188  if (packet->type == 0)
1189  break;
1190 
1191  if (sub_packet_size > packet_bytes) {
1192  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1193  break;
1194  packet->size += packet_bytes - sub_packet_size;
1195  }
1196 
1197  packet_bytes -= sub_packet_size;
1198 
1199  /* add subpacket to 'all subpackets' list */
1200  q->sub_packet_list_A[i].packet = packet;
1201 
1202  /* add subpacket to related list */
1203  if (packet->type == 8) {
1204  SAMPLES_NEEDED_2("packet type 8");
1205  return;
1206  } else if (packet->type >= 9 && packet->type <= 12) {
1207  /* packets for MPEG Audio like Synthesis Filter */
1208  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1209  } else if (packet->type == 13) {
1210  for (j = 0; j < 6; j++)
1211  q->fft_level_exp[j] = get_bits(&gb, 6);
1212  } else if (packet->type == 14) {
1213  for (j = 0; j < 6; j++)
1214  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1215  } else if (packet->type == 15) {
1216  SAMPLES_NEEDED_2("packet type 15")
1217  return;
1218  } else if (packet->type >= 16 && packet->type < 48 &&
1219  !fft_subpackets[packet->type - 16]) {
1220  /* packets for FFT */
1222  }
1223  } // Packet bytes loop
1224 
1225  if (q->sub_packet_list_D[0].packet) {
1227  q->do_synth_filter = 1;
1228  } else if (q->do_synth_filter) {
1232  }
1233 }
1234 
1235 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1236  int offset, int duration, int channel,
1237  int exp, int phase)
1238 {
1239  if (q->fft_coefs_min_index[duration] < 0)
1241 
1243  ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1244  q->fft_coefs[q->fft_coefs_index].channel = channel;
1246  q->fft_coefs[q->fft_coefs_index].exp = exp;
1247  q->fft_coefs[q->fft_coefs_index].phase = phase;
1248  q->fft_coefs_index++;
1249 }
1250 
1252  GetBitContext *gb, int b)
1253 {
1254  int channel, stereo, phase, exp;
1255  int local_int_4, local_int_8, stereo_phase, local_int_10;
1256  int local_int_14, stereo_exp, local_int_20, local_int_28;
1257  int n, offset;
1258 
1259  local_int_4 = 0;
1260  local_int_28 = 0;
1261  local_int_20 = 2;
1262  local_int_8 = (4 - duration);
1263  local_int_10 = 1 << (q->group_order - duration - 1);
1264  offset = 1;
1265 
1266  while (get_bits_left(gb)>0) {
1267  if (q->superblocktype_2_3) {
1268  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1269  if (get_bits_left(gb)<0) {
1270  if(local_int_4 < q->group_size)
1271  av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1272  return;
1273  }
1274  offset = 1;
1275  if (n == 0) {
1276  local_int_4 += local_int_10;
1277  local_int_28 += (1 << local_int_8);
1278  } else {
1279  local_int_4 += 8 * local_int_10;
1280  local_int_28 += (8 << local_int_8);
1281  }
1282  }
1283  offset += (n - 2);
1284  } else {
1285  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1286  while (offset >= (local_int_10 - 1)) {
1287  offset += (1 - (local_int_10 - 1));
1288  local_int_4 += local_int_10;
1289  local_int_28 += (1 << local_int_8);
1290  }
1291  }
1292 
1293  if (local_int_4 >= q->group_size)
1294  return;
1295 
1296  local_int_14 = (offset >> local_int_8);
1297  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1298  return;
1299 
1300  if (q->nb_channels > 1) {
1301  channel = get_bits1(gb);
1302  stereo = get_bits1(gb);
1303  } else {
1304  channel = 0;
1305  stereo = 0;
1306  }
1307 
1308  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1309  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1310  exp = (exp < 0) ? 0 : exp;
1311 
1312  phase = get_bits(gb, 3);
1313  stereo_exp = 0;
1314  stereo_phase = 0;
1315 
1316  if (stereo) {
1317  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1318  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1319  if (stereo_phase < 0)
1320  stereo_phase += 8;
1321  }
1322 
1323  if (q->frequency_range > (local_int_14 + 1)) {
1324  int sub_packet = (local_int_20 + local_int_28);
1325 
1326  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1327  channel, exp, phase);
1328  if (stereo)
1329  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1330  1 - channel,
1331  stereo_exp, stereo_phase);
1332  }
1333  offset++;
1334  }
1335 }
1336 
1338 {
1339  int i, j, min, max, value, type, unknown_flag;
1340  GetBitContext gb;
1341 
1342  if (!q->sub_packet_list_B[0].packet)
1343  return;
1344 
1345  /* reset minimum indexes for FFT coefficients */
1346  q->fft_coefs_index = 0;
1347  for (i = 0; i < 5; i++)
1348  q->fft_coefs_min_index[i] = -1;
1349 
1350  /* process subpackets ordered by type, largest type first */
1351  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1352  QDM2SubPacket *packet = NULL;
1353 
1354  /* find subpacket with largest type less than max */
1355  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1356  value = q->sub_packet_list_B[j].packet->type;
1357  if (value > min && value < max) {
1358  min = value;
1359  packet = q->sub_packet_list_B[j].packet;
1360  }
1361  }
1362 
1363  max = min;
1364 
1365  /* check for errors (?) */
1366  if (!packet)
1367  return;
1368 
1369  if (i == 0 &&
1370  (packet->type < 16 || packet->type >= 48 ||
1371  fft_subpackets[packet->type - 16]))
1372  return;
1373 
1374  /* decode FFT tones */
1375  init_get_bits(&gb, packet->data, packet->size * 8);
1376 
1377  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1378  unknown_flag = 1;
1379  else
1380  unknown_flag = 0;
1381 
1382  type = packet->type;
1383 
1384  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1385  int duration = q->sub_sampling + 5 - (type & 15);
1386 
1387  if (duration >= 0 && duration < 4)
1388  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1389  } else if (type == 31) {
1390  for (j = 0; j < 4; j++)
1391  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1392  } else if (type == 46) {
1393  for (j = 0; j < 6; j++)
1394  q->fft_level_exp[j] = get_bits(&gb, 6);
1395  for (j = 0; j < 4; j++)
1396  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1397  }
1398  } // Loop on B packets
1399 
1400  /* calculate maximum indexes for FFT coefficients */
1401  for (i = 0, j = -1; i < 5; i++)
1402  if (q->fft_coefs_min_index[i] >= 0) {
1403  if (j >= 0)
1405  j = i;
1406  }
1407  if (j >= 0)
1409 }
1410 
1412 {
1413  float level, f[6];
1414  int i;
1415  QDM2Complex c;
1416  const double iscale = 2.0 * M_PI / 512.0;
1417 
1418  tone->phase += tone->phase_shift;
1419 
1420  /* calculate current level (maximum amplitude) of tone */
1421  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1422  c.im = level * sin(tone->phase * iscale);
1423  c.re = level * cos(tone->phase * iscale);
1424 
1425  /* generate FFT coefficients for tone */
1426  if (tone->duration >= 3 || tone->cutoff >= 3) {
1427  tone->complex[0].im += c.im;
1428  tone->complex[0].re += c.re;
1429  tone->complex[1].im -= c.im;
1430  tone->complex[1].re -= c.re;
1431  } else {
1432  f[1] = -tone->table[4];
1433  f[0] = tone->table[3] - tone->table[0];
1434  f[2] = 1.0 - tone->table[2] - tone->table[3];
1435  f[3] = tone->table[1] + tone->table[4] - 1.0;
1436  f[4] = tone->table[0] - tone->table[1];
1437  f[5] = tone->table[2];
1438  for (i = 0; i < 2; i++) {
1439  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1440  c.re * f[i];
1441  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1442  c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1443  }
1444  for (i = 0; i < 4; i++) {
1445  tone->complex[i].re += c.re * f[i + 2];
1446  tone->complex[i].im += c.im * f[i + 2];
1447  }
1448  }
1449 
1450  /* copy the tone if it has not yet died out */
1451  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1452  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1453  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1454  }
1455 }
1456 
1457 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1458 {
1459  int i, j, ch;
1460  const double iscale = 0.25 * M_PI;
1461 
1462  for (ch = 0; ch < q->channels; ch++) {
1463  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1464  }
1465 
1466 
1467  /* apply FFT tones with duration 4 (1 FFT period) */
1468  if (q->fft_coefs_min_index[4] >= 0)
1469  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1470  float level;
1471  QDM2Complex c;
1472 
1473  if (q->fft_coefs[i].sub_packet != sub_packet)
1474  break;
1475 
1476  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1477  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];
1478 
1479  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1480  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1481  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1482  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1483  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1484  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1485  }
1486 
1487  /* generate existing FFT tones */
1488  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1490  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1491  }
1492 
1493  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1494  for (i = 0; i < 4; i++)
1495  if (q->fft_coefs_min_index[i] >= 0) {
1496  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1497  int offset, four_i;
1498  FFTTone tone;
1499 
1500  if (q->fft_coefs[j].sub_packet != sub_packet)
1501  break;
1502 
1503  four_i = (4 - i);
1504  offset = q->fft_coefs[j].offset >> four_i;
1505  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1506 
1507  if (offset < q->frequency_range) {
1508  if (offset < 2)
1509  tone.cutoff = offset;
1510  else
1511  tone.cutoff = (offset >= 60) ? 3 : 2;
1512 
1513  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];
1514  tone.complex = &q->fft.complex[ch][offset];
1515  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1516  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1517  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1518  tone.duration = i;
1519  tone.time_index = 0;
1520 
1521  qdm2_fft_generate_tone(q, &tone);
1522  }
1523  }
1524  q->fft_coefs_min_index[i] = j;
1525  }
1526 }
1527 
1528 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1529 {
1530  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1531  float *out = q->output_buffer + channel;
1532  int i;
1533  q->fft.complex[channel][0].re *= 2.0f;
1534  q->fft.complex[channel][0].im = 0.0f;
1535  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1536  /* add samples to output buffer */
1537  for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1538  out[0] += q->fft.complex[channel][i].re * gain;
1539  out[q->channels] += q->fft.complex[channel][i].im * gain;
1540  out += 2 * q->channels;
1541  }
1542 }
1543 
1544 /**
1545  * @param q context
1546  * @param index subpacket number
1547  */
1549 {
1550  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1551 
1552  /* copy sb_samples */
1553  sb_used = QDM2_SB_USED(q->sub_sampling);
1554 
1555  for (ch = 0; ch < q->channels; ch++)
1556  for (i = 0; i < 8; i++)
1557  for (k = sb_used; k < SBLIMIT; k++)
1558  q->sb_samples[ch][(8 * index) + i][k] = 0;
1559 
1560  for (ch = 0; ch < q->nb_channels; ch++) {
1561  float *samples_ptr = q->samples + ch;
1562 
1563  for (i = 0; i < 8; i++) {
1565  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1566  ff_mpa_synth_window_float, &dither_state,
1567  samples_ptr, q->nb_channels,
1568  q->sb_samples[ch][(8 * index) + i]);
1569  samples_ptr += 32 * q->nb_channels;
1570  }
1571  }
1572 
1573  /* add samples to output buffer */
1574  sub_sampling = (4 >> q->sub_sampling);
1575 
1576  for (ch = 0; ch < q->channels; ch++)
1577  for (i = 0; i < q->frame_size; i++)
1578  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1579 }
1580 
1581 /**
1582  * Init static data (does not depend on specific file)
1583  *
1584  * @param q context
1585  */
1586 static av_cold void qdm2_init_static_data(void) {
1587  static int done;
1588 
1589  if(done)
1590  return;
1591 
1592  qdm2_init_vlc();
1595  rnd_table_init();
1597 
1598  done = 1;
1599 }
1600 
1601 /**
1602  * Init parameters from codec extradata
1603  */
1605 {
1606  QDM2Context *s = avctx->priv_data;
1607  uint8_t *extradata;
1608  int extradata_size;
1609  int tmp_val, tmp, size;
1610 
1612 
1613  /* extradata parsing
1614 
1615  Structure:
1616  wave {
1617  frma (QDM2)
1618  QDCA
1619  QDCP
1620  }
1621 
1622  32 size (including this field)
1623  32 tag (=frma)
1624  32 type (=QDM2 or QDMC)
1625 
1626  32 size (including this field, in bytes)
1627  32 tag (=QDCA) // maybe mandatory parameters
1628  32 unknown (=1)
1629  32 channels (=2)
1630  32 samplerate (=44100)
1631  32 bitrate (=96000)
1632  32 block size (=4096)
1633  32 frame size (=256) (for one channel)
1634  32 packet size (=1300)
1635 
1636  32 size (including this field, in bytes)
1637  32 tag (=QDCP) // maybe some tuneable parameters
1638  32 float1 (=1.0)
1639  32 zero ?
1640  32 float2 (=1.0)
1641  32 float3 (=1.0)
1642  32 unknown (27)
1643  32 unknown (8)
1644  32 zero ?
1645  */
1646 
1647  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1648  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1649  return AVERROR_INVALIDDATA;
1650  }
1651 
1652  extradata = avctx->extradata;
1653  extradata_size = avctx->extradata_size;
1654 
1655  while (extradata_size > 7) {
1656  if (!memcmp(extradata, "frmaQDM", 7))
1657  break;
1658  extradata++;
1659  extradata_size--;
1660  }
1661 
1662  if (extradata_size < 12) {
1663  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1664  extradata_size);
1665  return AVERROR_INVALIDDATA;
1666  }
1667 
1668  if (memcmp(extradata, "frmaQDM", 7)) {
1669  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1670  return AVERROR_INVALIDDATA;
1671  }
1672 
1673  if (extradata[7] == 'C') {
1674 // s->is_qdmc = 1;
1675  avpriv_report_missing_feature(avctx, "QDMC version 1");
1676  return AVERROR_PATCHWELCOME;
1677  }
1678 
1679  extradata += 8;
1680  extradata_size -= 8;
1681 
1682  size = AV_RB32(extradata);
1683 
1684  if(size > extradata_size){
1685  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1686  extradata_size, size);
1687  return AVERROR_INVALIDDATA;
1688  }
1689 
1690  extradata += 4;
1691  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1692  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1693  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1694  return AVERROR_INVALIDDATA;
1695  }
1696 
1697  extradata += 8;
1698 
1699  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1700  extradata += 4;
1701  if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1702  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1703  return AVERROR_INVALIDDATA;
1704  }
1705  avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1707 
1708  avctx->sample_rate = AV_RB32(extradata);
1709  extradata += 4;
1710 
1711  avctx->bit_rate = AV_RB32(extradata);
1712  extradata += 4;
1713 
1714  s->group_size = AV_RB32(extradata);
1715  extradata += 4;
1716 
1717  s->fft_size = AV_RB32(extradata);
1718  extradata += 4;
1719 
1720  s->checksum_size = AV_RB32(extradata);
1721  if (s->checksum_size >= 1U << 28) {
1722  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1723  return AVERROR_INVALIDDATA;
1724  }
1725 
1726  s->fft_order = av_log2(s->fft_size) + 1;
1727 
1728  // something like max decodable tones
1729  s->group_order = av_log2(s->group_size) + 1;
1730  s->frame_size = s->group_size / 16; // 16 iterations per super block
1731 
1733  return AVERROR_INVALIDDATA;
1734 
1735  s->sub_sampling = s->fft_order - 7;
1736  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1737 
1738  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1739  case 0: tmp = 40; break;
1740  case 1: tmp = 48; break;
1741  case 2: tmp = 56; break;
1742  case 3: tmp = 72; break;
1743  case 4: tmp = 80; break;
1744  case 5: tmp = 100;break;
1745  default: tmp=s->sub_sampling; break;
1746  }
1747  tmp_val = 0;
1748  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1749  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1750  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1751  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1752  s->cm_table_select = tmp_val;
1753 
1754  if (avctx->bit_rate <= 8000)
1755  s->coeff_per_sb_select = 0;
1756  else if (avctx->bit_rate < 16000)
1757  s->coeff_per_sb_select = 1;
1758  else
1759  s->coeff_per_sb_select = 2;
1760 
1761  // Fail on unknown fft order
1762  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1763  avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1764  return AVERROR_PATCHWELCOME;
1765  }
1766  if (s->fft_size != (1 << (s->fft_order - 1))) {
1767  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1768  return AVERROR_INVALIDDATA;
1769  }
1770 
1772  ff_mpadsp_init(&s->mpadsp);
1773 
1774  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1775 
1776  return 0;
1777 }
1778 
1780 {
1781  QDM2Context *s = avctx->priv_data;
1782 
1783  ff_rdft_end(&s->rdft_ctx);
1784 
1785  return 0;
1786 }
1787 
1788 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1789 {
1790  int ch, i;
1791  const int frame_size = (q->frame_size * q->channels);
1792 
1793  if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1794  return -1;
1795 
1796  /* select input buffer */
1797  q->compressed_data = in;
1799 
1800  /* copy old block, clear new block of output samples */
1801  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1802  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1803 
1804  /* decode block of QDM2 compressed data */
1805  if (q->sub_packet == 0) {
1806  q->has_errors = 0; // zero it for a new super block
1807  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1809  }
1810 
1811  /* parse subpackets */
1812  if (!q->has_errors) {
1813  if (q->sub_packet == 2)
1815 
1817  }
1818 
1819  /* sound synthesis stage 1 (FFT) */
1820  for (ch = 0; ch < q->channels; ch++) {
1821  qdm2_calculate_fft(q, ch, q->sub_packet);
1822 
1823  if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1824  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1825  return -1;
1826  }
1827  }
1828 
1829  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1830  if (!q->has_errors && q->do_synth_filter)
1832 
1833  q->sub_packet = (q->sub_packet + 1) % 16;
1834 
1835  /* clip and convert output float[] to 16-bit signed samples */
1836  for (i = 0; i < frame_size; i++) {
1837  int value = (int)q->output_buffer[i];
1838 
1839  if (value > SOFTCLIP_THRESHOLD)
1840  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1841  else if (value < -SOFTCLIP_THRESHOLD)
1842  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1843 
1844  out[i] = value;
1845  }
1846 
1847  return 0;
1848 }
1849 
1850 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1851  int *got_frame_ptr, AVPacket *avpkt)
1852 {
1853  AVFrame *frame = data;
1854  const uint8_t *buf = avpkt->data;
1855  int buf_size = avpkt->size;
1856  QDM2Context *s = avctx->priv_data;
1857  int16_t *out;
1858  int i, ret;
1859 
1860  if(!buf)
1861  return 0;
1862  if(buf_size < s->checksum_size)
1863  return -1;
1864 
1865  /* get output buffer */
1866  frame->nb_samples = 16 * s->frame_size;
1867  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1868  return ret;
1869  out = (int16_t *)frame->data[0];
1870 
1871  for (i = 0; i < 16; i++) {
1872  if ((ret = qdm2_decode(s, buf, out)) < 0)
1873  return ret;
1874  out += s->channels * s->frame_size;
1875  }
1876 
1877  *got_frame_ptr = 1;
1878 
1879  return s->checksum_size;
1880 }
1881 
1883  .name = "qdm2",
1884  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1885  .type = AVMEDIA_TYPE_AUDIO,
1886  .id = AV_CODEC_ID_QDM2,
1887  .priv_data_size = sizeof(QDM2Context),
1889  .close = qdm2_decode_close,
1891  .capabilities = AV_CODEC_CAP_DR1,
1892 };
av_cold void ff_rdft_end(RDFTContext *s)
Definition: rdft.c:132
#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:43
FFTTone fft_tones[1000]
FFT and tones.
Definition: qdm2.c:153
A node in the subpacket list.
Definition: qdm2.c:91
This structure describes decoded (raw) audio or video data.
Definition: frame.h:184
QDM2FFT fft
Definition: qdm2.c:162
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:369
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:866
static unsigned int get_bits(GetBitContext *s, int n)
Read 1-25 bits.
Definition: get_bits.h:247
int64_t bit_rate
the average bitrate
Definition: avcodec.h:1740
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:313
Subpacket.
Definition: qdm2.c:82
int acc
Definition: yuv2rgb.c:546
int fft_coefs_index
Definition: qdm2.c:157
static VLC vlc_tab_tone_level_idx_hi2
#define QDM2_MAX_FRAME_SIZE
Definition: qdm2.c:75
float synth_buf[MPA_MAX_CHANNELS][512 *2]
Definition: qdm2.c:171
int size
Definition: avcodec.h:1601
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:180
const float * table
Definition: qdm2.c:104
int coeff_per_sb_select
selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
Definition: qdm2.c:141
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
Definition: qdm2.c:1779
short cutoff
Definition: qdm2.c:109
unsigned int size
subpacket size
Definition: qdm2.c:84
int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26]
Definition: qdm2.c:183
int sub_packet
Definition: qdm2.c:192
uint8_t run
Definition: svq3.c:206
float sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]
Definition: qdm2.c:173
#define AV_CH_LAYOUT_STEREO
int frequency_range
Definition: qdm2.c:139
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:261
AVCodec.
Definition: avcodec.h:3589
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet, int offset, int duration, int channel, int exp, int phase)
Definition: qdm2.c:1235
QDM2SubPNode sub_packet_list_C[16]
packets with errors?
Definition: qdm2.c:149
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:296
static VLC vlc_tab_type30
float re
Definition: qdm2.c:97
int phase
Definition: qdm2.c:105
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
Init parameters from codec extradata.
Definition: qdm2.c:1604
QDM2 decoder context.
Definition: qdm2.c:127
static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
Definition: qdm2.c:200
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:2445
uint8_t
#define av_cold
Definition: attributes.h:82
int fft_order
order of FFT (actually fftorder+1)
Definition: qdm2.c:137
static void qdm2_decode_fft_packets(QDM2Context *q)
Definition: qdm2.c:1337
int sub_sampling
subsampling: 0=25%, 1=50%, 2=100% */
Definition: qdm2.c:140
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:1790
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
Definition: qdm2.c:1457
uint64_t_TMPL AV_WL64 unsigned int_TMPL AV_WL32 unsigned int_TMPL AV_WL24 unsigned int_TMPL AV_WL16 uint64_t_TMPL AV_WB64 unsigned int_TMPL AV_RB32
Definition: bytestream.h:87
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:101
static const float fft_tone_envelope_table[4][31]
Definition: qdm2data.h:476
uint8_t * data
Definition: avcodec.h:1600
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:133
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:1060
static int qdm2_decode_frame(AVCodecContext *avctx, void *data, int *got_frame_ptr, AVPacket *avpkt)
Definition: qdm2.c:1850
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:1586
int channels
number of channels
Definition: qdm2.c:130
static void fill_tone_level_array(QDM2Context *q, int flag)
Related to synthesis filter Called by process_subpacket_10.
Definition: qdm2.c:435
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:650
#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:568
int synth_buf_offset[MPA_MAX_CHANNELS]
Definition: qdm2.c:172
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:166
const uint8_t * data
pointer to subpacket data (points to input data buffer, it's not a private copy)
Definition: qdm2.c:85
static VLC vlc_tab_tone_level_idx_mid
int16_t offset
Definition: qdm2.c:115
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:176
static const int switchtable[23]
Definition: qdm2.c:196
int group_size
size of frame group (16 frames per group)
Definition: qdm2.c:131
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
void ff_mpa_synth_filter_float(MPADSPContext *s, float *synth_buf_ptr, int *synth_buf_offset, float *window, int *dither_state, float *samples, int incr, float *sb_samples)
int sub_packets_B
number of packets on 'B' list
Definition: qdm2.c:148
QDM2SubPNode sub_packet_list_A[16]
list of all packets
Definition: qdm2.c:146
int noise_idx
index for dithering noise table
Definition: qdm2.c:193
GLsizei GLsizei * length
Definition: opengl_enc.c:115
Definition: avfft.h:73
const char * name
Name of the codec implementation.
Definition: avcodec.h:3596
uint8_t channel
Definition: qdm2.c:114
int duration
Definition: qdm2.c:107
static const uint8_t offset[127][2]
Definition: vf_spp.c:92
Definition: qdm2.c:120
float tone_level[MPA_MAX_CHANNELS][30][64]
Mixed temporary data used in decoding.
Definition: qdm2.c:177
float FFTSample
Definition: avfft.h:35
int8_t exp
Definition: eval.c:64
int depth
Definition: v4l.c:62
RDFTContext rdft_ctx
Definition: qdm2.c:161
Definition: vlc.h:26
uint64_t channel_layout
Audio channel layout.
Definition: avcodec.h:2488
void(* rdft_calc)(struct RDFTContext *s, FFTSample *z)
Definition: rdft.h:60
int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8]
Definition: qdm2.c:179
static void qdm2_synthesis_filter(QDM2Context *q, int index)
Definition: qdm2.c:1548
static VLC vlc_tab_tone_level_idx_hi1
#define QDM2_SB_USED(sub_sampling)
Definition: qdm2.c:61
#define MPA_MAX_CHANNELS
Definition: mpegaudio.h:41
int group_order
Parameters built from header parameters, do not change during playback.
Definition: qdm2.c:136
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:147
struct QDM2SubPNode * next
pointer to next packet in the list, NULL if leaf node
Definition: qdm2.c:93
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:1111
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:535
#define SAMPLES_NEEDED_2(why)
Definition: qdm2.c:72
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:245
#define QDM2_LIST_ADD(list, size, packet)
Definition: qdm2.c:50
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:155
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:121
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
#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:342
float samples[MPA_MAX_CHANNELS *MPA_FRAME_SIZE]
Definition: qdm2.c:174
static const uint8_t last_coeff[3]
Definition: qdm2data.h:257
int frame_size
Definition: mxfenc.c:1822
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:2437
#define SAMPLES_NEEDED
Definition: qdm2.c:69
static void qdm2_fft_decode_tones(QDM2Context *q, int duration, GetBitContext *gb, int b)
Definition: qdm2.c:1251
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:182
main external API structure.
Definition: avcodec.h:1675
static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
Definition: qdm2.c:229
float output_buffer[QDM2_MAX_FRAME_SIZE *MPA_MAX_CHANNELS *2]
Definition: qdm2.c:167
int ff_get_buffer(AVCodecContext *avctx, AVFrame *frame, int flags)
Get a buffer for a frame.
Definition: utils.c:934
AVCodec ff_qdm2_decoder
Definition: qdm2.c:1882
uint8_t phase
Definition: qdm2.c:117
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:158
void * buf
Definition: avisynth_c.h:690
GLint GLenum type
Definition: opengl_enc.c:105
FFTCoefficient fft_coefs[1000]
Definition: qdm2.c:156
int extradata_size
Definition: avcodec.h:1791
static unsigned int get_bits1(GetBitContext *s)
Definition: get_bits.h:299
static void skip_bits(GetBitContext *s, int n)
Definition: get_bits.h:292
int index
Definition: gxfenc.c:89
int has_errors
packet has errors
Definition: qdm2.c:188
static const uint8_t dequant_table[64]
Definition: 4xm.c:114
int fft_level_exp[6]
Definition: qdm2.c:160
static int init_get_bits(GetBitContext *s, const uint8_t *buffer, int bit_size)
Initialize GetBitContext.
Definition: get_bits.h:406
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:524
int16_t sub_packet
Definition: qdm2.c:113
#define HARDCLIP_THRESHOLD
Definition: qdm2_tablegen.h:32
float im
Definition: qdm2.c:98
int16_t exp
Definition: qdm2.c:116
int8_t coding_method[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:178
void avpriv_report_missing_feature(void *avc, const char *msg,...) av_printf_format(2
Log a generic warning message about a missing feature.
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 10 if not null, else.
Definition: qdm2.c:1012
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:198
uint8_t level
Definition: svq3.c:207
int fft_size
size of FFT, in complex numbers
Definition: qdm2.c:132
int type
subpacket type
Definition: qdm2.c:83
int fft_coefs_max_index[5]
Definition: qdm2.c:159
static int decode(AVCodecContext *avctx, void *data, int *got_sub, AVPacket *avpkt)
Definition: ccaption_dec.c:750
int frame_size
size of data frame
Definition: qdm2.c:138
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
Definition: qdm2.c:1788
#define FIX_NOISE_IDX(noise_idx)
Definition: qdm2.c:63
static const float fft_tone_sample_table[4][16][5]
Definition: qdm2data.h:368
Definition: qdm2.c:101
int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8]
Definition: qdm2.c:181
int nb_channels
Parameters from codec header, do not change during playback.
Definition: qdm2.c:129
int superblocktype_2_3
select fft tables and some algorithm based on superblock type
Definition: qdm2.c:189
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:142
signed 16 bits
Definition: samplefmt.h:61
static double c[64]
QDM2SubPacket * packet
packet
Definition: qdm2.c:92
QDM2SubPacket sub_packets[16]
Packets and packet lists.
Definition: qdm2.c:145
static const int vlc_stage3_values[60]
Definition: qdm2data.h:360
mpeg audio declarations for both encoder and decoder.
QDM2Complex * complex
Definition: qdm2.c:103
int do_synth_filter
used to perform or skip synthesis filter
Definition: qdm2.c:190
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_YASM &&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
const uint8_t * compressed_data
I/O data.
Definition: qdm2.c:165
int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:185
static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
Process subpacket 9, init quantized_coeffs with data from it.
Definition: qdm2.c:970
#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:1031
MPADSPContext mpadsp
Synthesis filter.
Definition: qdm2.c:170
void * priv_data
Definition: avcodec.h:1717
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:907
static av_always_inline int diff(const uint32_t a, const uint32_t b)
int channels
number of audio channels
Definition: avcodec.h:2438
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
Definition: qdm2.c:1411
QDM2SubPNode sub_packet_list_D[16]
DCT packets.
Definition: qdm2.c:150
static uint8_t tmp[8]
Definition: des.c:38
int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64]
Definition: qdm2.c:184
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:108
static void comp(unsigned char *dst, int dst_stride, unsigned char *src, int src_stride, int add)
Definition: eamad.c:83
int8_t sb_int8_array[2][30][64]
Definition: qdm2.c:77
#define M_PI
Definition: mathematics.h:52
#define SB_DITHERING_NOISE(sb, noise_idx)
Definition: qdm2.c:67
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
Definition: qdm2.c:1528
int nb_channels
int phase_shift
Definition: qdm2.c:106
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
Process new subpackets for synthesis filter.
Definition: qdm2.c:1079
#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:99
#define MPA_FRAME_SIZE
Definition: mpegaudio.h:36
float min
This structure stores compressed data.
Definition: avcodec.h:1577
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:241
#define AV_CODEC_CAP_DR1
Codec uses get_buffer() for allocating buffers and supports custom allocators.
Definition: avcodec.h:958
static VLC vlc_tab_type34
for(j=16;j >0;--j)
float level
Definition: qdm2.c:102
int fft_tone_start
Definition: qdm2.c:154