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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 
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
40 #include "avcodec.h"
41 #include "get_bits.h"
42 #include "internal.h"
43 #include "rdft.h"
44 #include "mpegaudiodsp.h"
45 #include "mpegaudio.h"
46 
47 #include "qdm2data.h"
48 #include "qdm2_tablegen.h"
49 
50 #undef NDEBUG
51 #include <assert.h>
52 
53 
54 #define QDM2_LIST_ADD(list, size, packet) \
55 do { \
56  if (size > 0) { \
57  list[size - 1].next = &list[size]; \
58  } \
59  list[size].packet = packet; \
60  list[size].next = NULL; \
61  size++; \
62 } while(0)
63 
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66 
67 #define FIX_NOISE_IDX(noise_idx) \
68  if ((noise_idx) >= 3840) \
69  (noise_idx) -= 3840; \
70 
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72 
73 #define SAMPLES_NEEDED \
74  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
75 
76 #define SAMPLES_NEEDED_2(why) \
77  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
78 
79 #define QDM2_MAX_FRAME_SIZE 512
80 
81 typedef int8_t sb_int8_array[2][30][64];
82 
83 /**
84  * Subpacket
85  */
86 typedef struct {
87  int type; ///< subpacket type
88  unsigned int size; ///< subpacket size
89  const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
90 } QDM2SubPacket;
91 
92 /**
93  * A node in the subpacket list
94  */
95 typedef struct QDM2SubPNode {
96  QDM2SubPacket *packet; ///< packet
97  struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
98 } QDM2SubPNode;
99 
100 typedef struct {
101  float re;
102  float im;
103 } QDM2Complex;
104 
105 typedef struct {
106  float level;
107  QDM2Complex *complex;
108  const float *table;
109  int phase;
110  int phase_shift;
111  int duration;
112  short time_index;
113  short cutoff;
114 } FFTTone;
115 
116 typedef struct {
117  int16_t sub_packet;
118  uint8_t channel;
119  int16_t offset;
120  int16_t exp;
121  uint8_t phase;
122 } FFTCoefficient;
123 
124 typedef struct {
125  DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
126 } QDM2FFT;
127 
128 /**
129  * QDM2 decoder context
130  */
131 typedef struct {
132  /// Parameters from codec header, do not change during playback
133  int nb_channels; ///< number of channels
134  int channels; ///< number of channels
135  int group_size; ///< size of frame group (16 frames per group)
136  int fft_size; ///< size of FFT, in complex numbers
137  int checksum_size; ///< size of data block, used also for checksum
138 
139  /// Parameters built from header parameters, do not change during playback
140  int group_order; ///< order of frame group
141  int fft_order; ///< order of FFT (actually fftorder+1)
142  int frame_size; ///< size of data frame
143  int frequency_range;
144  int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
145  int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
146  int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
147 
148  /// Packets and packet lists
149  QDM2SubPacket sub_packets[16]; ///< the packets themselves
150  QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
151  QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
152  int sub_packets_B; ///< number of packets on 'B' list
153  QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
154  QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
155 
156  /// FFT and tones
157  FFTTone fft_tones[1000];
158  int fft_tone_start;
159  int fft_tone_end;
160  FFTCoefficient fft_coefs[1000];
161  int fft_coefs_index;
162  int fft_coefs_min_index[5];
163  int fft_coefs_max_index[5];
164  int fft_level_exp[6];
165  RDFTContext rdft_ctx;
166  QDM2FFT fft;
167 
168  /// I/O data
169  const uint8_t *compressed_data;
170  int compressed_size;
171  float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
172 
173  /// Synthesis filter
174  MPADSPContext mpadsp;
175  DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
176  int synth_buf_offset[MPA_MAX_CHANNELS];
177  DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
178  DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
179 
180  /// Mixed temporary data used in decoding
181  float tone_level[MPA_MAX_CHANNELS][30][64];
182  int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183  int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184  int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185  int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186  int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187  int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188  int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189  int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
190 
191  // Flags
192  int has_errors; ///< packet has errors
193  int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
194  int do_synth_filter; ///< used to perform or skip synthesis filter
195 
196  int sub_packet;
197  int noise_idx; ///< index for dithering noise table
198 } QDM2Context;
199 
200 
201 static VLC vlc_tab_level;
202 static VLC vlc_tab_diff;
203 static VLC vlc_tab_run;
204 static VLC fft_level_exp_alt_vlc;
205 static VLC fft_level_exp_vlc;
206 static VLC fft_stereo_exp_vlc;
207 static VLC fft_stereo_phase_vlc;
208 static VLC vlc_tab_tone_level_idx_hi1;
209 static VLC vlc_tab_tone_level_idx_mid;
210 static VLC vlc_tab_tone_level_idx_hi2;
211 static VLC vlc_tab_type30;
212 static VLC vlc_tab_type34;
213 static VLC vlc_tab_fft_tone_offset[5];
214 
215 static const uint16_t qdm2_vlc_offs[] = {
216  0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
217 };
218 
219 static const int switchtable[23] = {
220  0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
221 };
222 
223 static av_cold void qdm2_init_vlc(void)
224 {
225  static VLC_TYPE qdm2_table[3838][2];
226 
227  vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
228  vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
229  init_vlc(&vlc_tab_level, 8, 24,
230  vlc_tab_level_huffbits, 1, 1,
231  vlc_tab_level_huffcodes, 2, 2,
232  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
233 
234  vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
235  vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
236  init_vlc(&vlc_tab_diff, 8, 37,
237  vlc_tab_diff_huffbits, 1, 1,
238  vlc_tab_diff_huffcodes, 2, 2,
239  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
240 
241  vlc_tab_run.table = &qdm2_table[qdm2_vlc_offs[2]];
242  vlc_tab_run.table_allocated = qdm2_vlc_offs[3] - qdm2_vlc_offs[2];
243  init_vlc(&vlc_tab_run, 5, 6,
244  vlc_tab_run_huffbits, 1, 1,
245  vlc_tab_run_huffcodes, 1, 1,
246  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
247 
248  fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
249  fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] -
250  qdm2_vlc_offs[3];
251  init_vlc(&fft_level_exp_alt_vlc, 8, 28,
252  fft_level_exp_alt_huffbits, 1, 1,
253  fft_level_exp_alt_huffcodes, 2, 2,
254  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
255 
256  fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
257  fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
258  init_vlc(&fft_level_exp_vlc, 8, 20,
259  fft_level_exp_huffbits, 1, 1,
260  fft_level_exp_huffcodes, 2, 2,
261  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
262 
263  fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
264  fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] -
265  qdm2_vlc_offs[5];
266  init_vlc(&fft_stereo_exp_vlc, 6, 7,
267  fft_stereo_exp_huffbits, 1, 1,
268  fft_stereo_exp_huffcodes, 1, 1,
269  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
270 
271  fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
272  fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] -
273  qdm2_vlc_offs[6];
274  init_vlc(&fft_stereo_phase_vlc, 6, 9,
275  fft_stereo_phase_huffbits, 1, 1,
276  fft_stereo_phase_huffcodes, 1, 1,
277  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
278 
279  vlc_tab_tone_level_idx_hi1.table =
280  &qdm2_table[qdm2_vlc_offs[7]];
281  vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] -
282  qdm2_vlc_offs[7];
283  init_vlc(&vlc_tab_tone_level_idx_hi1, 8, 20,
284  vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
285  vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2,
286  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
287 
288  vlc_tab_tone_level_idx_mid.table =
289  &qdm2_table[qdm2_vlc_offs[8]];
290  vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] -
291  qdm2_vlc_offs[8];
292  init_vlc(&vlc_tab_tone_level_idx_mid, 8, 24,
293  vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
294  vlc_tab_tone_level_idx_mid_huffcodes, 2, 2,
295  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
296 
297  vlc_tab_tone_level_idx_hi2.table =
298  &qdm2_table[qdm2_vlc_offs[9]];
299  vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] -
300  qdm2_vlc_offs[9];
301  init_vlc(&vlc_tab_tone_level_idx_hi2, 8, 24,
302  vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
303  vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2,
304  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
305 
306  vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
307  vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
308  init_vlc(&vlc_tab_type30, 6, 9,
309  vlc_tab_type30_huffbits, 1, 1,
310  vlc_tab_type30_huffcodes, 1, 1,
311  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
312 
313  vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
314  vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
315  init_vlc(&vlc_tab_type34, 5, 10,
316  vlc_tab_type34_huffbits, 1, 1,
317  vlc_tab_type34_huffcodes, 1, 1,
318  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
319 
320  vlc_tab_fft_tone_offset[0].table =
321  &qdm2_table[qdm2_vlc_offs[12]];
322  vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] -
323  qdm2_vlc_offs[12];
324  init_vlc(&vlc_tab_fft_tone_offset[0], 8, 23,
325  vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
326  vlc_tab_fft_tone_offset_0_huffcodes, 2, 2,
327  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
328 
329  vlc_tab_fft_tone_offset[1].table =
330  &qdm2_table[qdm2_vlc_offs[13]];
331  vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] -
332  qdm2_vlc_offs[13];
333  init_vlc(&vlc_tab_fft_tone_offset[1], 8, 28,
334  vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
335  vlc_tab_fft_tone_offset_1_huffcodes, 2, 2,
336  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
337 
338  vlc_tab_fft_tone_offset[2].table =
339  &qdm2_table[qdm2_vlc_offs[14]];
340  vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] -
341  qdm2_vlc_offs[14];
342  init_vlc(&vlc_tab_fft_tone_offset[2], 8, 32,
343  vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
344  vlc_tab_fft_tone_offset_2_huffcodes, 2, 2,
345  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
346 
347  vlc_tab_fft_tone_offset[3].table =
348  &qdm2_table[qdm2_vlc_offs[15]];
349  vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] -
350  qdm2_vlc_offs[15];
351  init_vlc(&vlc_tab_fft_tone_offset[3], 8, 35,
352  vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
353  vlc_tab_fft_tone_offset_3_huffcodes, 2, 2,
354  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
355 
356  vlc_tab_fft_tone_offset[4].table =
357  &qdm2_table[qdm2_vlc_offs[16]];
358  vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] -
359  qdm2_vlc_offs[16];
360  init_vlc(&vlc_tab_fft_tone_offset[4], 8, 38,
361  vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
362  vlc_tab_fft_tone_offset_4_huffcodes, 2, 2,
363  INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
364 }
365 
366 static int qdm2_get_vlc(GetBitContext *gb, VLC *vlc, int flag, int depth)
367 {
368  int value;
369 
370  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
371 
372  /* stage-2, 3 bits exponent escape sequence */
373  if (value-- == 0)
374  value = get_bits(gb, get_bits(gb, 3) + 1);
375 
376  /* stage-3, optional */
377  if (flag) {
378  int tmp;
379 
380  if (value >= 60) {
381  av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
382  return 0;
383  }
384 
385  tmp= vlc_stage3_values[value];
386 
387  if ((value & ~3) > 0)
388  tmp += get_bits(gb, (value >> 2));
389  value = tmp;
390  }
391 
392  return value;
393 }
394 
395 static int qdm2_get_se_vlc(VLC *vlc, GetBitContext *gb, int depth)
396 {
397  int value = qdm2_get_vlc(gb, vlc, 0, depth);
398 
399  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
400 }
401 
402 /**
403  * QDM2 checksum
404  *
405  * @param data pointer to data to be checksum'ed
406  * @param length data length
407  * @param value checksum value
408  *
409  * @return 0 if checksum is OK
410  */
411 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
412 {
413  int i;
414 
415  for (i = 0; i < length; i++)
416  value -= data[i];
417 
418  return (uint16_t)(value & 0xffff);
419 }
420 
421 /**
422  * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
423  *
424  * @param gb bitreader context
425  * @param sub_packet packet under analysis
426  */
427 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
428  QDM2SubPacket *sub_packet)
429 {
430  sub_packet->type = get_bits(gb, 8);
431 
432  if (sub_packet->type == 0) {
433  sub_packet->size = 0;
434  sub_packet->data = NULL;
435  } else {
436  sub_packet->size = get_bits(gb, 8);
437 
438  if (sub_packet->type & 0x80) {
439  sub_packet->size <<= 8;
440  sub_packet->size |= get_bits(gb, 8);
441  sub_packet->type &= 0x7f;
442  }
443 
444  if (sub_packet->type == 0x7f)
445  sub_packet->type |= (get_bits(gb, 8) << 8);
446 
447  // FIXME: this depends on bitreader-internal data
448  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
449  }
450 
451  av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
452  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
453 }
454 
455 /**
456  * Return node pointer to first packet of requested type in list.
457  *
458  * @param list list of subpackets to be scanned
459  * @param type type of searched subpacket
460  * @return node pointer for subpacket if found, else NULL
461  */
462 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
463  int type)
464 {
465  while (list != NULL && list->packet != NULL) {
466  if (list->packet->type == type)
467  return list;
468  list = list->next;
469  }
470  return NULL;
471 }
472 
473 /**
474  * Replace 8 elements with their average value.
475  * Called by qdm2_decode_superblock before starting subblock decoding.
476  *
477  * @param q context
478  */
479 static void average_quantized_coeffs(QDM2Context *q)
480 {
481  int i, j, n, ch, sum;
482 
483  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
484 
485  for (ch = 0; ch < q->nb_channels; ch++)
486  for (i = 0; i < n; i++) {
487  sum = 0;
488 
489  for (j = 0; j < 8; j++)
490  sum += q->quantized_coeffs[ch][i][j];
491 
492  sum /= 8;
493  if (sum > 0)
494  sum--;
495 
496  for (j = 0; j < 8; j++)
497  q->quantized_coeffs[ch][i][j] = sum;
498  }
499 }
500 
501 /**
502  * Build subband samples with noise weighted by q->tone_level.
503  * Called by synthfilt_build_sb_samples.
504  *
505  * @param q context
506  * @param sb subband index
507  */
508 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
509 {
510  int ch, j;
511 
512  FIX_NOISE_IDX(q->noise_idx);
513 
514  if (!q->nb_channels)
515  return;
516 
517  for (ch = 0; ch < q->nb_channels; ch++) {
518  for (j = 0; j < 64; j++) {
519  q->sb_samples[ch][j * 2][sb] =
520  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
521  q->sb_samples[ch][j * 2 + 1][sb] =
522  SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
523  }
524  }
525 }
526 
527 /**
528  * Called while processing data from subpackets 11 and 12.
529  * Used after making changes to coding_method array.
530  *
531  * @param sb subband index
532  * @param channels number of channels
533  * @param coding_method q->coding_method[0][0][0]
534  */
535 static int fix_coding_method_array(int sb, int channels,
536  sb_int8_array coding_method)
537 {
538  int j, k;
539  int ch;
540  int run, case_val;
541 
542  for (ch = 0; ch < channels; ch++) {
543  for (j = 0; j < 64; ) {
544  if (coding_method[ch][sb][j] < 8)
545  return -1;
546  if ((coding_method[ch][sb][j] - 8) > 22) {
547  run = 1;
548  case_val = 8;
549  } else {
550  switch (switchtable[coding_method[ch][sb][j] - 8]) {
551  case 0: run = 10;
552  case_val = 10;
553  break;
554  case 1: run = 1;
555  case_val = 16;
556  break;
557  case 2: run = 5;
558  case_val = 24;
559  break;
560  case 3: run = 3;
561  case_val = 30;
562  break;
563  case 4: run = 1;
564  case_val = 30;
565  break;
566  case 5: run = 1;
567  case_val = 8;
568  break;
569  default: run = 1;
570  case_val = 8;
571  break;
572  }
573  }
574  for (k = 0; k < run; k++) {
575  if (j + k < 128) {
576  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
577  if (k > 0) {
578  SAMPLES_NEEDED
579  //not debugged, almost never used
580  memset(&coding_method[ch][sb][j + k], case_val,
581  k *sizeof(int8_t));
582  memset(&coding_method[ch][sb][j + k], case_val,
583  3 * sizeof(int8_t));
584  }
585  }
586  }
587  }
588  j += run;
589  }
590  }
591  return 0;
592 }
593 
594 /**
595  * Related to synthesis filter
596  * Called by process_subpacket_10
597  *
598  * @param q context
599  * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
600  */
601 static void fill_tone_level_array(QDM2Context *q, int flag)
602 {
603  int i, sb, ch, sb_used;
604  int tmp, tab;
605 
606  for (ch = 0; ch < q->nb_channels; ch++)
607  for (sb = 0; sb < 30; sb++)
608  for (i = 0; i < 8; i++) {
609  if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
610  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
611  q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
612  else
613  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
614  if(tmp < 0)
615  tmp += 0xff;
616  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
617  }
618 
619  sb_used = QDM2_SB_USED(q->sub_sampling);
620 
621  if ((q->superblocktype_2_3 != 0) && !flag) {
622  for (sb = 0; sb < sb_used; sb++)
623  for (ch = 0; ch < q->nb_channels; ch++)
624  for (i = 0; i < 64; i++) {
625  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
626  if (q->tone_level_idx[ch][sb][i] < 0)
627  q->tone_level[ch][sb][i] = 0;
628  else
629  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
630  }
631  } else {
632  tab = q->superblocktype_2_3 ? 0 : 1;
633  for (sb = 0; sb < sb_used; sb++) {
634  if ((sb >= 4) && (sb <= 23)) {
635  for (ch = 0; ch < q->nb_channels; ch++)
636  for (i = 0; i < 64; i++) {
637  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
638  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
639  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
640  q->tone_level_idx_hi2[ch][sb - 4];
641  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
642  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
643  q->tone_level[ch][sb][i] = 0;
644  else
645  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
646  }
647  } else {
648  if (sb > 4) {
649  for (ch = 0; ch < q->nb_channels; ch++)
650  for (i = 0; i < 64; i++) {
651  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
652  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
653  q->tone_level_idx_hi2[ch][sb - 4];
654  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
655  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
656  q->tone_level[ch][sb][i] = 0;
657  else
658  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
659  }
660  } else {
661  for (ch = 0; ch < q->nb_channels; ch++)
662  for (i = 0; i < 64; i++) {
663  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
664  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
665  q->tone_level[ch][sb][i] = 0;
666  else
667  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
668  }
669  }
670  }
671  }
672  }
673 }
674 
675 /**
676  * Related to synthesis filter
677  * Called by process_subpacket_11
678  * c is built with data from subpacket 11
679  * Most of this function is used only if superblock_type_2_3 == 0,
680  * never seen it in samples.
681  *
682  * @param tone_level_idx
683  * @param tone_level_idx_temp
684  * @param coding_method q->coding_method[0][0][0]
685  * @param nb_channels number of channels
686  * @param c coming from subpacket 11, passed as 8*c
687  * @param superblocktype_2_3 flag based on superblock packet type
688  * @param cm_table_select q->cm_table_select
689  */
690 static void fill_coding_method_array(sb_int8_array tone_level_idx,
691  sb_int8_array tone_level_idx_temp,
692  sb_int8_array coding_method,
693  int nb_channels,
694  int c, int superblocktype_2_3,
695  int cm_table_select)
696 {
697  int ch, sb, j;
698  int tmp, acc, esp_40, comp;
699  int add1, add2, add3, add4;
700  int64_t multres;
701 
702  if (!superblocktype_2_3) {
703  /* This case is untested, no samples available */
704  avpriv_request_sample(NULL, "!superblocktype_2_3");
705  return;
706  for (ch = 0; ch < nb_channels; ch++)
707  for (sb = 0; sb < 30; sb++) {
708  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
709  add1 = tone_level_idx[ch][sb][j] - 10;
710  if (add1 < 0)
711  add1 = 0;
712  add2 = add3 = add4 = 0;
713  if (sb > 1) {
714  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
715  if (add2 < 0)
716  add2 = 0;
717  }
718  if (sb > 0) {
719  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
720  if (add3 < 0)
721  add3 = 0;
722  }
723  if (sb < 29) {
724  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
725  if (add4 < 0)
726  add4 = 0;
727  }
728  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
729  if (tmp < 0)
730  tmp = 0;
731  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
732  }
733  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
734  }
735  acc = 0;
736  for (ch = 0; ch < nb_channels; ch++)
737  for (sb = 0; sb < 30; sb++)
738  for (j = 0; j < 64; j++)
739  acc += tone_level_idx_temp[ch][sb][j];
740 
741  multres = 0x66666667LL * (acc * 10);
742  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
743  for (ch = 0; ch < nb_channels; ch++)
744  for (sb = 0; sb < 30; sb++)
745  for (j = 0; j < 64; j++) {
746  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
747  if (comp < 0)
748  comp += 0xff;
749  comp /= 256; // signed shift
750  switch(sb) {
751  case 0:
752  if (comp < 30)
753  comp = 30;
754  comp += 15;
755  break;
756  case 1:
757  if (comp < 24)
758  comp = 24;
759  comp += 10;
760  break;
761  case 2:
762  case 3:
763  case 4:
764  if (comp < 16)
765  comp = 16;
766  }
767  if (comp <= 5)
768  tmp = 0;
769  else if (comp <= 10)
770  tmp = 10;
771  else if (comp <= 16)
772  tmp = 16;
773  else if (comp <= 24)
774  tmp = -1;
775  else
776  tmp = 0;
777  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
778  }
779  for (sb = 0; sb < 30; sb++)
780  fix_coding_method_array(sb, nb_channels, coding_method);
781  for (ch = 0; ch < nb_channels; ch++)
782  for (sb = 0; sb < 30; sb++)
783  for (j = 0; j < 64; j++)
784  if (sb >= 10) {
785  if (coding_method[ch][sb][j] < 10)
786  coding_method[ch][sb][j] = 10;
787  } else {
788  if (sb >= 2) {
789  if (coding_method[ch][sb][j] < 16)
790  coding_method[ch][sb][j] = 16;
791  } else {
792  if (coding_method[ch][sb][j] < 30)
793  coding_method[ch][sb][j] = 30;
794  }
795  }
796  } else { // superblocktype_2_3 != 0
797  for (ch = 0; ch < nb_channels; ch++)
798  for (sb = 0; sb < 30; sb++)
799  for (j = 0; j < 64; j++)
800  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
801  }
802 }
803 
804 /**
805  *
806  * Called by process_subpacket_11 to process more data from subpacket 11
807  * with sb 0-8.
808  * Called by process_subpacket_12 to process data from subpacket 12 with
809  * sb 8-sb_used.
810  *
811  * @param q context
812  * @param gb bitreader context
813  * @param length packet length in bits
814  * @param sb_min lower subband processed (sb_min included)
815  * @param sb_max higher subband processed (sb_max excluded)
816  */
817 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
818  int length, int sb_min, int sb_max)
819 {
820  int sb, j, k, n, ch, run, channels;
821  int joined_stereo, zero_encoding;
822  int type34_first;
823  float type34_div = 0;
824  float type34_predictor;
825  float samples[10];
826  int sign_bits[16] = {0};
827 
828  if (length == 0) {
829  // If no data use noise
830  for (sb=sb_min; sb < sb_max; sb++)
831  build_sb_samples_from_noise(q, sb);
832 
833  return 0;
834  }
835 
836  for (sb = sb_min; sb < sb_max; sb++) {
837  channels = q->nb_channels;
838 
839  if (q->nb_channels <= 1 || sb < 12)
840  joined_stereo = 0;
841  else if (sb >= 24)
842  joined_stereo = 1;
843  else
844  joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
845 
846  if (joined_stereo) {
847  if (get_bits_left(gb) >= 16)
848  for (j = 0; j < 16; j++)
849  sign_bits[j] = get_bits1(gb);
850 
851  for (j = 0; j < 64; j++)
852  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
853  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
854 
855  if (fix_coding_method_array(sb, q->nb_channels,
856  q->coding_method)) {
857  av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
858  build_sb_samples_from_noise(q, sb);
859  continue;
860  }
861  channels = 1;
862  }
863 
864  for (ch = 0; ch < channels; ch++) {
865  FIX_NOISE_IDX(q->noise_idx);
866  zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
867  type34_predictor = 0.0;
868  type34_first = 1;
869 
870  for (j = 0; j < 128; ) {
871  switch (q->coding_method[ch][sb][j / 2]) {
872  case 8:
873  if (get_bits_left(gb) >= 10) {
874  if (zero_encoding) {
875  for (k = 0; k < 5; k++) {
876  if ((j + 2 * k) >= 128)
877  break;
878  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
879  }
880  } else {
881  n = get_bits(gb, 8);
882  if (n >= 243) {
883  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
884  return AVERROR_INVALIDDATA;
885  }
886 
887  for (k = 0; k < 5; k++)
888  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
889  }
890  for (k = 0; k < 5; k++)
891  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
892  } else {
893  for (k = 0; k < 10; k++)
894  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
895  }
896  run = 10;
897  break;
898 
899  case 10:
900  if (get_bits_left(gb) >= 1) {
901  float f = 0.81;
902 
903  if (get_bits1(gb))
904  f = -f;
905  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
906  samples[0] = f;
907  } else {
908  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
909  }
910  run = 1;
911  break;
912 
913  case 16:
914  if (get_bits_left(gb) >= 10) {
915  if (zero_encoding) {
916  for (k = 0; k < 5; k++) {
917  if ((j + k) >= 128)
918  break;
919  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
920  }
921  } else {
922  n = get_bits (gb, 8);
923  if (n >= 243) {
924  av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
925  return AVERROR_INVALIDDATA;
926  }
927 
928  for (k = 0; k < 5; k++)
929  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
930  }
931  } else {
932  for (k = 0; k < 5; k++)
933  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
934  }
935  run = 5;
936  break;
937 
938  case 24:
939  if (get_bits_left(gb) >= 7) {
940  n = get_bits(gb, 7);
941  if (n >= 125) {
942  av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
943  return AVERROR_INVALIDDATA;
944  }
945 
946  for (k = 0; k < 3; k++)
947  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
948  } else {
949  for (k = 0; k < 3; k++)
950  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
951  }
952  run = 3;
953  break;
954 
955  case 30:
956  if (get_bits_left(gb) >= 4) {
957  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
958  if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
959  av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
960  return AVERROR_INVALIDDATA;
961  }
962  samples[0] = type30_dequant[index];
963  } else
964  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
965 
966  run = 1;
967  break;
968 
969  case 34:
970  if (get_bits_left(gb) >= 7) {
971  if (type34_first) {
972  type34_div = (float)(1 << get_bits(gb, 2));
973  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
974  type34_predictor = samples[0];
975  type34_first = 0;
976  } else {
977  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
978  if (index >= FF_ARRAY_ELEMS(type34_delta)) {
979  av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
980  return AVERROR_INVALIDDATA;
981  }
982  samples[0] = type34_delta[index] / type34_div + type34_predictor;
983  type34_predictor = samples[0];
984  }
985  } else {
986  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
987  }
988  run = 1;
989  break;
990 
991  default:
992  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
993  run = 1;
994  break;
995  }
996 
997  if (joined_stereo) {
998  for (k = 0; k < run && j + k < 128; k++) {
999  q->sb_samples[0][j + k][sb] =
1000  q->tone_level[0][sb][(j + k) / 2] * samples[k];
1001  if (q->nb_channels == 2) {
1002  if (sign_bits[(j + k) / 8])
1003  q->sb_samples[1][j + k][sb] =
1004  q->tone_level[1][sb][(j + k) / 2] * -samples[k];
1005  else
1006  q->sb_samples[1][j + k][sb] =
1007  q->tone_level[1][sb][(j + k) / 2] * samples[k];
1008  }
1009  }
1010  } else {
1011  for (k = 0; k < run; k++)
1012  if ((j + k) < 128)
1013  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
1014  }
1015 
1016  j += run;
1017  } // j loop
1018  } // channel loop
1019  } // subband loop
1020  return 0;
1021 }
1022 
1023 /**
1024  * Init the first element of a channel in quantized_coeffs with data
1025  * from packet 10 (quantized_coeffs[ch][0]).
1026  * This is similar to process_subpacket_9, but for a single channel
1027  * and for element [0]
1028  * same VLC tables as process_subpacket_9 are used.
1029  *
1030  * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
1031  * @param gb bitreader context
1032  */
1033 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
1034  GetBitContext *gb)
1035 {
1036  int i, k, run, level, diff;
1037 
1038  if (get_bits_left(gb) < 16)
1039  return -1;
1040  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
1041 
1042  quantized_coeffs[0] = level;
1043 
1044  for (i = 0; i < 7; ) {
1045  if (get_bits_left(gb) < 16)
1046  return -1;
1047  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
1048 
1049  if (i + run >= 8)
1050  return -1;
1051 
1052  if (get_bits_left(gb) < 16)
1053  return -1;
1054  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
1055 
1056  for (k = 1; k <= run; k++)
1057  quantized_coeffs[i + k] = (level + ((k * diff) / run));
1058 
1059  level += diff;
1060  i += run;
1061  }
1062  return 0;
1063 }
1064 
1065 /**
1066  * Related to synthesis filter, process data from packet 10
1067  * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1068  * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
1069  * data from packet 10
1070  *
1071  * @param q context
1072  * @param gb bitreader context
1073  */
1074 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
1075 {
1076  int sb, j, k, n, ch;
1077 
1078  for (ch = 0; ch < q->nb_channels; ch++) {
1079  init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
1080 
1081  if (get_bits_left(gb) < 16) {
1082  memset(q->quantized_coeffs[ch][0], 0, 8);
1083  break;
1084  }
1085  }
1086 
1087  n = q->sub_sampling + 1;
1088 
1089  for (sb = 0; sb < n; sb++)
1090  for (ch = 0; ch < q->nb_channels; ch++)
1091  for (j = 0; j < 8; j++) {
1092  if (get_bits_left(gb) < 1)
1093  break;
1094  if (get_bits1(gb)) {
1095  for (k=0; k < 8; k++) {
1096  if (get_bits_left(gb) < 16)
1097  break;
1098  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1099  }
1100  } else {
1101  for (k=0; k < 8; k++)
1102  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1103  }
1104  }
1105 
1106  n = QDM2_SB_USED(q->sub_sampling) - 4;
1107 
1108  for (sb = 0; sb < n; sb++)
1109  for (ch = 0; ch < q->nb_channels; ch++) {
1110  if (get_bits_left(gb) < 16)
1111  break;
1112  q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1113  if (sb > 19)
1114  q->tone_level_idx_hi2[ch][sb] -= 16;
1115  else
1116  for (j = 0; j < 8; j++)
1117  q->tone_level_idx_mid[ch][sb][j] = -16;
1118  }
1119 
1120  n = QDM2_SB_USED(q->sub_sampling) - 5;
1121 
1122  for (sb = 0; sb < n; sb++)
1123  for (ch = 0; ch < q->nb_channels; ch++)
1124  for (j = 0; j < 8; j++) {
1125  if (get_bits_left(gb) < 16)
1126  break;
1127  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1128  }
1129 }
1130 
1131 /**
1132  * Process subpacket 9, init quantized_coeffs with data from it
1133  *
1134  * @param q context
1135  * @param node pointer to node with packet
1136  */
1137 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
1138 {
1139  GetBitContext gb;
1140  int i, j, k, n, ch, run, level, diff;
1141 
1142  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1143 
1144  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
1145 
1146  for (i = 1; i < n; i++)
1147  for (ch = 0; ch < q->nb_channels; ch++) {
1148  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1149  q->quantized_coeffs[ch][i][0] = level;
1150 
1151  for (j = 0; j < (8 - 1); ) {
1152  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1153  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1154 
1155  if (j + run >= 8)
1156  return -1;
1157 
1158  for (k = 1; k <= run; k++)
1159  q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1160 
1161  level += diff;
1162  j += run;
1163  }
1164  }
1165 
1166  for (ch = 0; ch < q->nb_channels; ch++)
1167  for (i = 0; i < 8; i++)
1168  q->quantized_coeffs[ch][0][i] = 0;
1169 
1170  return 0;
1171 }
1172 
1173 /**
1174  * Process subpacket 10 if not null, else
1175  *
1176  * @param q context
1177  * @param node pointer to node with packet
1178  */
1179 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1180 {
1181  GetBitContext gb;
1182 
1183  if (node) {
1184  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1185  init_tone_level_dequantization(q, &gb);
1186  fill_tone_level_array(q, 1);
1187  } else {
1188  fill_tone_level_array(q, 0);
1189  }
1190 }
1191 
1192 /**
1193  * Process subpacket 11
1194  *
1195  * @param q context
1196  * @param node pointer to node with packet
1197  */
1198 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1199 {
1200  GetBitContext gb;
1201  int length = 0;
1202 
1203  if (node) {
1204  length = node->packet->size * 8;
1205  init_get_bits(&gb, node->packet->data, length);
1206  }
1207 
1208  if (length >= 32) {
1209  int c = get_bits(&gb, 13);
1210 
1211  if (c > 3)
1212  fill_coding_method_array(q->tone_level_idx,
1213  q->tone_level_idx_temp, q->coding_method,
1214  q->nb_channels, 8 * c,
1215  q->superblocktype_2_3, q->cm_table_select);
1216  }
1217 
1218  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1219 }
1220 
1221 /**
1222  * Process subpacket 12
1223  *
1224  * @param q context
1225  * @param node pointer to node with packet
1226  */
1227 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1228 {
1229  GetBitContext gb;
1230  int length = 0;
1231 
1232  if (node) {
1233  length = node->packet->size * 8;
1234  init_get_bits(&gb, node->packet->data, length);
1235  }
1236 
1237  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1238 }
1239 
1240 /**
1241  * Process new subpackets for synthesis filter
1242  *
1243  * @param q context
1244  * @param list list with synthesis filter packets (list D)
1245  */
1246 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1247 {
1248  QDM2SubPNode *nodes[4];
1249 
1250  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1251  if (nodes[0] != NULL)
1252  process_subpacket_9(q, nodes[0]);
1253 
1254  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1255  if (nodes[1] != NULL)
1256  process_subpacket_10(q, nodes[1]);
1257  else
1258  process_subpacket_10(q, NULL);
1259 
1260  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1261  if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1262  process_subpacket_11(q, nodes[2]);
1263  else
1264  process_subpacket_11(q, NULL);
1265 
1266  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1267  if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1268  process_subpacket_12(q, nodes[3]);
1269  else
1270  process_subpacket_12(q, NULL);
1271 }
1272 
1273 /**
1274  * Decode superblock, fill packet lists.
1275  *
1276  * @param q context
1277  */
1278 static void qdm2_decode_super_block(QDM2Context *q)
1279 {
1280  GetBitContext gb;
1281  QDM2SubPacket header, *packet;
1282  int i, packet_bytes, sub_packet_size, sub_packets_D;
1283  unsigned int next_index = 0;
1284 
1285  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1286  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1287  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1288 
1289  q->sub_packets_B = 0;
1290  sub_packets_D = 0;
1291 
1292  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1293 
1294  init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1295  qdm2_decode_sub_packet_header(&gb, &header);
1296 
1297  if (header.type < 2 || header.type >= 8) {
1298  q->has_errors = 1;
1299  av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1300  return;
1301  }
1302 
1303  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1304  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1305 
1306  init_get_bits(&gb, header.data, header.size * 8);
1307 
1308  if (header.type == 2 || header.type == 4 || header.type == 5) {
1309  int csum = 257 * get_bits(&gb, 8);
1310  csum += 2 * get_bits(&gb, 8);
1311 
1312  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1313 
1314  if (csum != 0) {
1315  q->has_errors = 1;
1316  av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1317  return;
1318  }
1319  }
1320 
1321  q->sub_packet_list_B[0].packet = NULL;
1322  q->sub_packet_list_D[0].packet = NULL;
1323 
1324  for (i = 0; i < 6; i++)
1325  if (--q->fft_level_exp[i] < 0)
1326  q->fft_level_exp[i] = 0;
1327 
1328  for (i = 0; packet_bytes > 0; i++) {
1329  int j;
1330 
1331  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1332  SAMPLES_NEEDED_2("too many packet bytes");
1333  return;
1334  }
1335 
1336  q->sub_packet_list_A[i].next = NULL;
1337 
1338  if (i > 0) {
1339  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1340 
1341  /* seek to next block */
1342  init_get_bits(&gb, header.data, header.size * 8);
1343  skip_bits(&gb, next_index * 8);
1344 
1345  if (next_index >= header.size)
1346  break;
1347  }
1348 
1349  /* decode subpacket */
1350  packet = &q->sub_packets[i];
1351  qdm2_decode_sub_packet_header(&gb, packet);
1352  next_index = packet->size + get_bits_count(&gb) / 8;
1353  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1354 
1355  if (packet->type == 0)
1356  break;
1357 
1358  if (sub_packet_size > packet_bytes) {
1359  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1360  break;
1361  packet->size += packet_bytes - sub_packet_size;
1362  }
1363 
1364  packet_bytes -= sub_packet_size;
1365 
1366  /* add subpacket to 'all subpackets' list */
1367  q->sub_packet_list_A[i].packet = packet;
1368 
1369  /* add subpacket to related list */
1370  if (packet->type == 8) {
1371  SAMPLES_NEEDED_2("packet type 8");
1372  return;
1373  } else if (packet->type >= 9 && packet->type <= 12) {
1374  /* packets for MPEG Audio like Synthesis Filter */
1375  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1376  } else if (packet->type == 13) {
1377  for (j = 0; j < 6; j++)
1378  q->fft_level_exp[j] = get_bits(&gb, 6);
1379  } else if (packet->type == 14) {
1380  for (j = 0; j < 6; j++)
1381  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1382  } else if (packet->type == 15) {
1383  SAMPLES_NEEDED_2("packet type 15")
1384  return;
1385  } else if (packet->type >= 16 && packet->type < 48 &&
1386  !fft_subpackets[packet->type - 16]) {
1387  /* packets for FFT */
1388  QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1389  }
1390  } // Packet bytes loop
1391 
1392  if (q->sub_packet_list_D[0].packet != NULL) {
1393  process_synthesis_subpackets(q, q->sub_packet_list_D);
1394  q->do_synth_filter = 1;
1395  } else if (q->do_synth_filter) {
1396  process_subpacket_10(q, NULL);
1397  process_subpacket_11(q, NULL);
1398  process_subpacket_12(q, NULL);
1399  }
1400 }
1401 
1402 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1403  int offset, int duration, int channel,
1404  int exp, int phase)
1405 {
1406  if (q->fft_coefs_min_index[duration] < 0)
1407  q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1408 
1409  q->fft_coefs[q->fft_coefs_index].sub_packet =
1410  ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1411  q->fft_coefs[q->fft_coefs_index].channel = channel;
1412  q->fft_coefs[q->fft_coefs_index].offset = offset;
1413  q->fft_coefs[q->fft_coefs_index].exp = exp;
1414  q->fft_coefs[q->fft_coefs_index].phase = phase;
1415  q->fft_coefs_index++;
1416 }
1417 
1418 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1419  GetBitContext *gb, int b)
1420 {
1421  int channel, stereo, phase, exp;
1422  int local_int_4, local_int_8, stereo_phase, local_int_10;
1423  int local_int_14, stereo_exp, local_int_20, local_int_28;
1424  int n, offset;
1425 
1426  local_int_4 = 0;
1427  local_int_28 = 0;
1428  local_int_20 = 2;
1429  local_int_8 = (4 - duration);
1430  local_int_10 = 1 << (q->group_order - duration - 1);
1431  offset = 1;
1432 
1433  while (get_bits_left(gb)>0) {
1434  if (q->superblocktype_2_3) {
1435  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1436  if (get_bits_left(gb)<0) {
1437  if(local_int_4 < q->group_size)
1438  av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1439  return;
1440  }
1441  offset = 1;
1442  if (n == 0) {
1443  local_int_4 += local_int_10;
1444  local_int_28 += (1 << local_int_8);
1445  } else {
1446  local_int_4 += 8 * local_int_10;
1447  local_int_28 += (8 << local_int_8);
1448  }
1449  }
1450  offset += (n - 2);
1451  } else {
1452  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1453  while (offset >= (local_int_10 - 1)) {
1454  offset += (1 - (local_int_10 - 1));
1455  local_int_4 += local_int_10;
1456  local_int_28 += (1 << local_int_8);
1457  }
1458  }
1459 
1460  if (local_int_4 >= q->group_size)
1461  return;
1462 
1463  local_int_14 = (offset >> local_int_8);
1464  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1465  return;
1466 
1467  if (q->nb_channels > 1) {
1468  channel = get_bits1(gb);
1469  stereo = get_bits1(gb);
1470  } else {
1471  channel = 0;
1472  stereo = 0;
1473  }
1474 
1475  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1476  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1477  exp = (exp < 0) ? 0 : exp;
1478 
1479  phase = get_bits(gb, 3);
1480  stereo_exp = 0;
1481  stereo_phase = 0;
1482 
1483  if (stereo) {
1484  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1485  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1486  if (stereo_phase < 0)
1487  stereo_phase += 8;
1488  }
1489 
1490  if (q->frequency_range > (local_int_14 + 1)) {
1491  int sub_packet = (local_int_20 + local_int_28);
1492 
1493  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1494  channel, exp, phase);
1495  if (stereo)
1496  qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1497  1 - channel,
1498  stereo_exp, stereo_phase);
1499  }
1500  offset++;
1501  }
1502 }
1503 
1504 static void qdm2_decode_fft_packets(QDM2Context *q)
1505 {
1506  int i, j, min, max, value, type, unknown_flag;
1507  GetBitContext gb;
1508 
1509  if (q->sub_packet_list_B[0].packet == NULL)
1510  return;
1511 
1512  /* reset minimum indexes for FFT coefficients */
1513  q->fft_coefs_index = 0;
1514  for (i = 0; i < 5; i++)
1515  q->fft_coefs_min_index[i] = -1;
1516 
1517  /* process subpackets ordered by type, largest type first */
1518  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1519  QDM2SubPacket *packet = NULL;
1520 
1521  /* find subpacket with largest type less than max */
1522  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1523  value = q->sub_packet_list_B[j].packet->type;
1524  if (value > min && value < max) {
1525  min = value;
1526  packet = q->sub_packet_list_B[j].packet;
1527  }
1528  }
1529 
1530  max = min;
1531 
1532  /* check for errors (?) */
1533  if (!packet)
1534  return;
1535 
1536  if (i == 0 &&
1537  (packet->type < 16 || packet->type >= 48 ||
1538  fft_subpackets[packet->type - 16]))
1539  return;
1540 
1541  /* decode FFT tones */
1542  init_get_bits(&gb, packet->data, packet->size * 8);
1543 
1544  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1545  unknown_flag = 1;
1546  else
1547  unknown_flag = 0;
1548 
1549  type = packet->type;
1550 
1551  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1552  int duration = q->sub_sampling + 5 - (type & 15);
1553 
1554  if (duration >= 0 && duration < 4)
1555  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1556  } else if (type == 31) {
1557  for (j = 0; j < 4; j++)
1558  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1559  } else if (type == 46) {
1560  for (j = 0; j < 6; j++)
1561  q->fft_level_exp[j] = get_bits(&gb, 6);
1562  for (j = 0; j < 4; j++)
1563  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1564  }
1565  } // Loop on B packets
1566 
1567  /* calculate maximum indexes for FFT coefficients */
1568  for (i = 0, j = -1; i < 5; i++)
1569  if (q->fft_coefs_min_index[i] >= 0) {
1570  if (j >= 0)
1571  q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1572  j = i;
1573  }
1574  if (j >= 0)
1575  q->fft_coefs_max_index[j] = q->fft_coefs_index;
1576 }
1577 
1578 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1579 {
1580  float level, f[6];
1581  int i;
1582  QDM2Complex c;
1583  const double iscale = 2.0 * M_PI / 512.0;
1584 
1585  tone->phase += tone->phase_shift;
1586 
1587  /* calculate current level (maximum amplitude) of tone */
1588  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1589  c.im = level * sin(tone->phase * iscale);
1590  c.re = level * cos(tone->phase * iscale);
1591 
1592  /* generate FFT coefficients for tone */
1593  if (tone->duration >= 3 || tone->cutoff >= 3) {
1594  tone->complex[0].im += c.im;
1595  tone->complex[0].re += c.re;
1596  tone->complex[1].im -= c.im;
1597  tone->complex[1].re -= c.re;
1598  } else {
1599  f[1] = -tone->table[4];
1600  f[0] = tone->table[3] - tone->table[0];
1601  f[2] = 1.0 - tone->table[2] - tone->table[3];
1602  f[3] = tone->table[1] + tone->table[4] - 1.0;
1603  f[4] = tone->table[0] - tone->table[1];
1604  f[5] = tone->table[2];
1605  for (i = 0; i < 2; i++) {
1606  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1607  c.re * f[i];
1608  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1609  c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1610  }
1611  for (i = 0; i < 4; i++) {
1612  tone->complex[i].re += c.re * f[i + 2];
1613  tone->complex[i].im += c.im * f[i + 2];
1614  }
1615  }
1616 
1617  /* copy the tone if it has not yet died out */
1618  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1619  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1620  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1621  }
1622 }
1623 
1624 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1625 {
1626  int i, j, ch;
1627  const double iscale = 0.25 * M_PI;
1628 
1629  for (ch = 0; ch < q->channels; ch++) {
1630  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1631  }
1632 
1633 
1634  /* apply FFT tones with duration 4 (1 FFT period) */
1635  if (q->fft_coefs_min_index[4] >= 0)
1636  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1637  float level;
1638  QDM2Complex c;
1639 
1640  if (q->fft_coefs[i].sub_packet != sub_packet)
1641  break;
1642 
1643  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1644  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];
1645 
1646  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1647  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1648  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1649  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1650  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1651  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1652  }
1653 
1654  /* generate existing FFT tones */
1655  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1656  qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1657  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1658  }
1659 
1660  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1661  for (i = 0; i < 4; i++)
1662  if (q->fft_coefs_min_index[i] >= 0) {
1663  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1664  int offset, four_i;
1665  FFTTone tone;
1666 
1667  if (q->fft_coefs[j].sub_packet != sub_packet)
1668  break;
1669 
1670  four_i = (4 - i);
1671  offset = q->fft_coefs[j].offset >> four_i;
1672  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1673 
1674  if (offset < q->frequency_range) {
1675  if (offset < 2)
1676  tone.cutoff = offset;
1677  else
1678  tone.cutoff = (offset >= 60) ? 3 : 2;
1679 
1680  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];
1681  tone.complex = &q->fft.complex[ch][offset];
1682  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1683  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1684  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1685  tone.duration = i;
1686  tone.time_index = 0;
1687 
1688  qdm2_fft_generate_tone(q, &tone);
1689  }
1690  }
1691  q->fft_coefs_min_index[i] = j;
1692  }
1693 }
1694 
1695 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1696 {
1697  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1698  float *out = q->output_buffer + channel;
1699  int i;
1700  q->fft.complex[channel][0].re *= 2.0f;
1701  q->fft.complex[channel][0].im = 0.0f;
1702  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1703  /* add samples to output buffer */
1704  for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1705  out[0] += q->fft.complex[channel][i].re * gain;
1706  out[q->channels] += q->fft.complex[channel][i].im * gain;
1707  out += 2 * q->channels;
1708  }
1709 }
1710 
1711 /**
1712  * @param q context
1713  * @param index subpacket number
1714  */
1715 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1716 {
1717  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1718 
1719  /* copy sb_samples */
1720  sb_used = QDM2_SB_USED(q->sub_sampling);
1721 
1722  for (ch = 0; ch < q->channels; ch++)
1723  for (i = 0; i < 8; i++)
1724  for (k = sb_used; k < SBLIMIT; k++)
1725  q->sb_samples[ch][(8 * index) + i][k] = 0;
1726 
1727  for (ch = 0; ch < q->nb_channels; ch++) {
1728  float *samples_ptr = q->samples + ch;
1729 
1730  for (i = 0; i < 8; i++) {
1731  ff_mpa_synth_filter_float(&q->mpadsp,
1732  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1733  ff_mpa_synth_window_float, &dither_state,
1734  samples_ptr, q->nb_channels,
1735  q->sb_samples[ch][(8 * index) + i]);
1736  samples_ptr += 32 * q->nb_channels;
1737  }
1738  }
1739 
1740  /* add samples to output buffer */
1741  sub_sampling = (4 >> q->sub_sampling);
1742 
1743  for (ch = 0; ch < q->channels; ch++)
1744  for (i = 0; i < q->frame_size; i++)
1745  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1746 }
1747 
1748 /**
1749  * Init static data (does not depend on specific file)
1750  *
1751  * @param q context
1752  */
1753 static av_cold void qdm2_init_static_data(AVCodec *codec) {
1754  qdm2_init_vlc();
1755  ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1756  softclip_table_init();
1757  rnd_table_init();
1758  init_noise_samples();
1759 }
1760 
1761 /**
1762  * Init parameters from codec extradata
1763  */
1764 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1765 {
1766  QDM2Context *s = avctx->priv_data;
1767  uint8_t *extradata;
1768  int extradata_size;
1769  int tmp_val, tmp, size;
1770 
1771  /* extradata parsing
1772 
1773  Structure:
1774  wave {
1775  frma (QDM2)
1776  QDCA
1777  QDCP
1778  }
1779 
1780  32 size (including this field)
1781  32 tag (=frma)
1782  32 type (=QDM2 or QDMC)
1783 
1784  32 size (including this field, in bytes)
1785  32 tag (=QDCA) // maybe mandatory parameters
1786  32 unknown (=1)
1787  32 channels (=2)
1788  32 samplerate (=44100)
1789  32 bitrate (=96000)
1790  32 block size (=4096)
1791  32 frame size (=256) (for one channel)
1792  32 packet size (=1300)
1793 
1794  32 size (including this field, in bytes)
1795  32 tag (=QDCP) // maybe some tuneable parameters
1796  32 float1 (=1.0)
1797  32 zero ?
1798  32 float2 (=1.0)
1799  32 float3 (=1.0)
1800  32 unknown (27)
1801  32 unknown (8)
1802  32 zero ?
1803  */
1804 
1805  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1806  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1807  return -1;
1808  }
1809 
1810  extradata = avctx->extradata;
1811  extradata_size = avctx->extradata_size;
1812 
1813  while (extradata_size > 7) {
1814  if (!memcmp(extradata, "frmaQDM", 7))
1815  break;
1816  extradata++;
1817  extradata_size--;
1818  }
1819 
1820  if (extradata_size < 12) {
1821  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1822  extradata_size);
1823  return -1;
1824  }
1825 
1826  if (memcmp(extradata, "frmaQDM", 7)) {
1827  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1828  return -1;
1829  }
1830 
1831  if (extradata[7] == 'C') {
1832 // s->is_qdmc = 1;
1833  av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1834  return -1;
1835  }
1836 
1837  extradata += 8;
1838  extradata_size -= 8;
1839 
1840  size = AV_RB32(extradata);
1841 
1842  if(size > extradata_size){
1843  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1844  extradata_size, size);
1845  return -1;
1846  }
1847 
1848  extradata += 4;
1849  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1850  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1851  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1852  return -1;
1853  }
1854 
1855  extradata += 8;
1856 
1857  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1858  extradata += 4;
1859  if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1860  av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1861  return AVERROR_INVALIDDATA;
1862  }
1863  avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1864  AV_CH_LAYOUT_MONO;
1865 
1866  avctx->sample_rate = AV_RB32(extradata);
1867  extradata += 4;
1868 
1869  avctx->bit_rate = AV_RB32(extradata);
1870  extradata += 4;
1871 
1872  s->group_size = AV_RB32(extradata);
1873  extradata += 4;
1874 
1875  s->fft_size = AV_RB32(extradata);
1876  extradata += 4;
1877 
1878  s->checksum_size = AV_RB32(extradata);
1879  if (s->checksum_size >= 1U << 28) {
1880  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1881  return AVERROR_INVALIDDATA;
1882  }
1883 
1884  s->fft_order = av_log2(s->fft_size) + 1;
1885 
1886  // something like max decodable tones
1887  s->group_order = av_log2(s->group_size) + 1;
1888  s->frame_size = s->group_size / 16; // 16 iterations per super block
1889 
1890  if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1891  return AVERROR_INVALIDDATA;
1892 
1893  s->sub_sampling = s->fft_order - 7;
1894  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1895 
1896  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1897  case 0: tmp = 40; break;
1898  case 1: tmp = 48; break;
1899  case 2: tmp = 56; break;
1900  case 3: tmp = 72; break;
1901  case 4: tmp = 80; break;
1902  case 5: tmp = 100;break;
1903  default: tmp=s->sub_sampling; break;
1904  }
1905  tmp_val = 0;
1906  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1907  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1908  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1909  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1910  s->cm_table_select = tmp_val;
1911 
1912  if (avctx->bit_rate <= 8000)
1913  s->coeff_per_sb_select = 0;
1914  else if (avctx->bit_rate < 16000)
1915  s->coeff_per_sb_select = 1;
1916  else
1917  s->coeff_per_sb_select = 2;
1918 
1919  // Fail on unknown fft order
1920  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1921  av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1922  return -1;
1923  }
1924  if (s->fft_size != (1 << (s->fft_order - 1))) {
1925  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1926  return AVERROR_INVALIDDATA;
1927  }
1928 
1929  ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1930  ff_mpadsp_init(&s->mpadsp);
1931 
1932  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1933 
1934  return 0;
1935 }
1936 
1937 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1938 {
1939  QDM2Context *s = avctx->priv_data;
1940 
1941  ff_rdft_end(&s->rdft_ctx);
1942 
1943  return 0;
1944 }
1945 
1946 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1947 {
1948  int ch, i;
1949  const int frame_size = (q->frame_size * q->channels);
1950 
1951  if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1952  return -1;
1953 
1954  /* select input buffer */
1955  q->compressed_data = in;
1956  q->compressed_size = q->checksum_size;
1957 
1958  /* copy old block, clear new block of output samples */
1959  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1960  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1961 
1962  /* decode block of QDM2 compressed data */
1963  if (q->sub_packet == 0) {
1964  q->has_errors = 0; // zero it for a new super block
1965  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1966  qdm2_decode_super_block(q);
1967  }
1968 
1969  /* parse subpackets */
1970  if (!q->has_errors) {
1971  if (q->sub_packet == 2)
1972  qdm2_decode_fft_packets(q);
1973 
1974  qdm2_fft_tone_synthesizer(q, q->sub_packet);
1975  }
1976 
1977  /* sound synthesis stage 1 (FFT) */
1978  for (ch = 0; ch < q->channels; ch++) {
1979  qdm2_calculate_fft(q, ch, q->sub_packet);
1980 
1981  if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1982  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1983  return -1;
1984  }
1985  }
1986 
1987  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1988  if (!q->has_errors && q->do_synth_filter)
1989  qdm2_synthesis_filter(q, q->sub_packet);
1990 
1991  q->sub_packet = (q->sub_packet + 1) % 16;
1992 
1993  /* clip and convert output float[] to 16bit signed samples */
1994  for (i = 0; i < frame_size; i++) {
1995  int value = (int)q->output_buffer[i];
1996 
1997  if (value > SOFTCLIP_THRESHOLD)
1998  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1999  else if (value < -SOFTCLIP_THRESHOLD)
2000  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2001 
2002  out[i] = value;
2003  }
2004 
2005  return 0;
2006 }
2007 
2008 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
2009  int *got_frame_ptr, AVPacket *avpkt)
2010 {
2011  AVFrame *frame = data;
2012  const uint8_t *buf = avpkt->data;
2013  int buf_size = avpkt->size;
2014  QDM2Context *s = avctx->priv_data;
2015  int16_t *out;
2016  int i, ret;
2017 
2018  if(!buf)
2019  return 0;
2020  if(buf_size < s->checksum_size)
2021  return -1;
2022 
2023  /* get output buffer */
2024  frame->nb_samples = 16 * s->frame_size;
2025  if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
2026  return ret;
2027  out = (int16_t *)frame->data[0];
2028 
2029  for (i = 0; i < 16; i++) {
2030  if (qdm2_decode(s, buf, out) < 0)
2031  return -1;
2032  out += s->channels * s->frame_size;
2033  }
2034 
2035  *got_frame_ptr = 1;
2036 
2037  return s->checksum_size;
2038 }
2039 
2040 AVCodec ff_qdm2_decoder = {
2041  .name = "qdm2",
2042  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2043  .type = AVMEDIA_TYPE_AUDIO,
2044  .id = AV_CODEC_ID_QDM2,
2045  .priv_data_size = sizeof(QDM2Context),
2046  .init = qdm2_decode_init,
2047  .init_static_data = qdm2_init_static_data,
2048  .close = qdm2_decode_close,
2049  .decode = qdm2_decode_frame,
2050  .capabilities = CODEC_CAP_DR1,
2051 };
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