qdm2.c
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1 /*
2  * QDM2 compatible decoder
3  * Copyright (c) 2003 Ewald Snel
4  * Copyright (c) 2005 Benjamin Larsson
5  * Copyright (c) 2005 Alex Beregszaszi
6  * Copyright (c) 2005 Roberto Togni
7  *
8  * This file is part of Libav.
9  *
10  * Libav is free software; you can redistribute it and/or
11  * modify it under the terms of the GNU Lesser General Public
12  * License as published by the Free Software Foundation; either
13  * version 2.1 of the License, or (at your option) any later version.
14  *
15  * Libav is distributed in the hope that it will be useful,
16  * but WITHOUT ANY WARRANTY; without even the implied warranty of
17  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18  * Lesser General Public License for more details.
19  *
20  * You should have received a copy of the GNU Lesser General Public
21  * License along with Libav; if not, write to the Free Software
22  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23  */
24 
34 #include <math.h>
35 #include <stddef.h>
36 #include <stdio.h>
37 
38 #define BITSTREAM_READER_LE
39 #include "libavutil/channel_layout.h"
40 #include "avcodec.h"
41 #include "get_bits.h"
42 #include "dsputil.h"
43 #include "internal.h"
44 #include "rdft.h"
45 #include "mpegaudiodsp.h"
46 #include "mpegaudio.h"
47 
48 #include "qdm2data.h"
49 #include "qdm2_tablegen.h"
50 
51 #undef NDEBUG
52 #include <assert.h>
53 
54 
55 #define QDM2_LIST_ADD(list, size, packet) \
56 do { \
57  if (size > 0) { \
58  list[size - 1].next = &list[size]; \
59  } \
60  list[size].packet = packet; \
61  list[size].next = NULL; \
62  size++; \
63 } while(0)
64 
65 // Result is 8, 16 or 30
66 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
67 
68 #define FIX_NOISE_IDX(noise_idx) \
69  if ((noise_idx) >= 3840) \
70  (noise_idx) -= 3840; \
71 
72 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
73 
74 #define SAMPLES_NEEDED \
75  av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
76 
77 #define SAMPLES_NEEDED_2(why) \
78  av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
79 
80 #define QDM2_MAX_FRAME_SIZE 512
81 
82 typedef int8_t sb_int8_array[2][30][64];
83 
87 typedef struct {
88  int type;
89  unsigned int size;
90  const uint8_t *data;
91 } QDM2SubPacket;
92 
96 typedef struct QDM2SubPNode {
97  QDM2SubPacket *packet;
98  struct QDM2SubPNode *next;
99 } QDM2SubPNode;
100 
101 typedef struct {
102  float re;
103  float im;
104 } QDM2Complex;
105 
106 typedef struct {
107  float level;
108  QDM2Complex *complex;
109  const float *table;
110  int phase;
111  int phase_shift;
112  int duration;
113  short time_index;
114  short cutoff;
115 } FFTTone;
116 
117 typedef struct {
118  int16_t sub_packet;
119  uint8_t channel;
120  int16_t offset;
121  int16_t exp;
122  uint8_t phase;
123 } FFTCoefficient;
124 
125 typedef struct {
126  DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
127 } QDM2FFT;
128 
132 typedef struct {
133  AVFrame frame;
134 
136  int nb_channels;
137  int channels;
138  int group_size;
139  int fft_size;
140  int checksum_size;
141 
143  int group_order;
144  int fft_order;
145  int frame_size;
146  int frequency_range;
147  int sub_sampling;
148  int coeff_per_sb_select;
149  int cm_table_select;
150 
152  QDM2SubPacket sub_packets[16];
153  QDM2SubPNode sub_packet_list_A[16];
154  QDM2SubPNode sub_packet_list_B[16];
155  int sub_packets_B;
156  QDM2SubPNode sub_packet_list_C[16];
157  QDM2SubPNode sub_packet_list_D[16];
158 
160  FFTTone fft_tones[1000];
161  int fft_tone_start;
162  int fft_tone_end;
163  FFTCoefficient fft_coefs[1000];
164  int fft_coefs_index;
165  int fft_coefs_min_index[5];
166  int fft_coefs_max_index[5];
167  int fft_level_exp[6];
168  RDFTContext rdft_ctx;
169  QDM2FFT fft;
170 
172  const uint8_t *compressed_data;
173  int compressed_size;
174  float output_buffer[QDM2_MAX_FRAME_SIZE * 2];
175 
177  MPADSPContext mpadsp;
178  DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
179  int synth_buf_offset[MPA_MAX_CHANNELS];
180  DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
181  DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
182 
184  float tone_level[MPA_MAX_CHANNELS][30][64];
185  int8_t coding_method[MPA_MAX_CHANNELS][30][64];
186  int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
187  int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
188  int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
189  int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
190  int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
191  int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
192  int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
193 
194  // Flags
195  int has_errors;
196  int superblocktype_2_3;
197  int do_synth_filter;
198 
199  int sub_packet;
200  int noise_idx;
201 } QDM2Context;
202 
203 
204 static VLC vlc_tab_level;
205 static VLC vlc_tab_diff;
206 static VLC vlc_tab_run;
207 static VLC fft_level_exp_alt_vlc;
208 static VLC fft_level_exp_vlc;
209 static VLC fft_stereo_exp_vlc;
210 static VLC fft_stereo_phase_vlc;
211 static VLC vlc_tab_tone_level_idx_hi1;
212 static VLC vlc_tab_tone_level_idx_mid;
213 static VLC vlc_tab_tone_level_idx_hi2;
214 static VLC vlc_tab_type30;
215 static VLC vlc_tab_type34;
216 static VLC vlc_tab_fft_tone_offset[5];
217 
218 static const uint16_t qdm2_vlc_offs[] = {
219  0,260,566,598,894,1166,1230,1294,1678,1950,2214,2278,2310,2570,2834,3124,3448,3838,
220 };
221 
222 static av_cold void qdm2_init_vlc(void)
223 {
224  static int vlcs_initialized = 0;
225  static VLC_TYPE qdm2_table[3838][2];
226 
227  if (!vlcs_initialized) {
228 
229  vlc_tab_level.table = &qdm2_table[qdm2_vlc_offs[0]];
230  vlc_tab_level.table_allocated = qdm2_vlc_offs[1] - qdm2_vlc_offs[0];
231  init_vlc (&vlc_tab_level, 8, 24,
232  vlc_tab_level_huffbits, 1, 1,
233  vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
234 
235  vlc_tab_diff.table = &qdm2_table[qdm2_vlc_offs[1]];
236  vlc_tab_diff.table_allocated = qdm2_vlc_offs[2] - qdm2_vlc_offs[1];
237  init_vlc (&vlc_tab_diff, 8, 37,
238  vlc_tab_diff_huffbits, 1, 1,
239  vlc_tab_diff_huffcodes, 2, 2, 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, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
246 
247  fft_level_exp_alt_vlc.table = &qdm2_table[qdm2_vlc_offs[3]];
248  fft_level_exp_alt_vlc.table_allocated = qdm2_vlc_offs[4] - qdm2_vlc_offs[3];
249  init_vlc (&fft_level_exp_alt_vlc, 8, 28,
250  fft_level_exp_alt_huffbits, 1, 1,
251  fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
252 
253 
254  fft_level_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[4]];
255  fft_level_exp_vlc.table_allocated = qdm2_vlc_offs[5] - qdm2_vlc_offs[4];
256  init_vlc (&fft_level_exp_vlc, 8, 20,
257  fft_level_exp_huffbits, 1, 1,
258  fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
259 
260  fft_stereo_exp_vlc.table = &qdm2_table[qdm2_vlc_offs[5]];
261  fft_stereo_exp_vlc.table_allocated = qdm2_vlc_offs[6] - qdm2_vlc_offs[5];
262  init_vlc (&fft_stereo_exp_vlc, 6, 7,
263  fft_stereo_exp_huffbits, 1, 1,
264  fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
265 
266  fft_stereo_phase_vlc.table = &qdm2_table[qdm2_vlc_offs[6]];
267  fft_stereo_phase_vlc.table_allocated = qdm2_vlc_offs[7] - qdm2_vlc_offs[6];
268  init_vlc (&fft_stereo_phase_vlc, 6, 9,
269  fft_stereo_phase_huffbits, 1, 1,
270  fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
271 
272  vlc_tab_tone_level_idx_hi1.table = &qdm2_table[qdm2_vlc_offs[7]];
273  vlc_tab_tone_level_idx_hi1.table_allocated = qdm2_vlc_offs[8] - qdm2_vlc_offs[7];
274  init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
275  vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
276  vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
277 
278  vlc_tab_tone_level_idx_mid.table = &qdm2_table[qdm2_vlc_offs[8]];
279  vlc_tab_tone_level_idx_mid.table_allocated = qdm2_vlc_offs[9] - qdm2_vlc_offs[8];
280  init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
281  vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
282  vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
283 
284  vlc_tab_tone_level_idx_hi2.table = &qdm2_table[qdm2_vlc_offs[9]];
285  vlc_tab_tone_level_idx_hi2.table_allocated = qdm2_vlc_offs[10] - qdm2_vlc_offs[9];
286  init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
287  vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
288  vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
289 
290  vlc_tab_type30.table = &qdm2_table[qdm2_vlc_offs[10]];
291  vlc_tab_type30.table_allocated = qdm2_vlc_offs[11] - qdm2_vlc_offs[10];
292  init_vlc (&vlc_tab_type30, 6, 9,
293  vlc_tab_type30_huffbits, 1, 1,
294  vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
295 
296  vlc_tab_type34.table = &qdm2_table[qdm2_vlc_offs[11]];
297  vlc_tab_type34.table_allocated = qdm2_vlc_offs[12] - qdm2_vlc_offs[11];
298  init_vlc (&vlc_tab_type34, 5, 10,
299  vlc_tab_type34_huffbits, 1, 1,
300  vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
301 
302  vlc_tab_fft_tone_offset[0].table = &qdm2_table[qdm2_vlc_offs[12]];
303  vlc_tab_fft_tone_offset[0].table_allocated = qdm2_vlc_offs[13] - qdm2_vlc_offs[12];
304  init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
305  vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
306  vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
307 
308  vlc_tab_fft_tone_offset[1].table = &qdm2_table[qdm2_vlc_offs[13]];
309  vlc_tab_fft_tone_offset[1].table_allocated = qdm2_vlc_offs[14] - qdm2_vlc_offs[13];
310  init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
311  vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
312  vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
313 
314  vlc_tab_fft_tone_offset[2].table = &qdm2_table[qdm2_vlc_offs[14]];
315  vlc_tab_fft_tone_offset[2].table_allocated = qdm2_vlc_offs[15] - qdm2_vlc_offs[14];
316  init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
317  vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
318  vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
319 
320  vlc_tab_fft_tone_offset[3].table = &qdm2_table[qdm2_vlc_offs[15]];
321  vlc_tab_fft_tone_offset[3].table_allocated = qdm2_vlc_offs[16] - qdm2_vlc_offs[15];
322  init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
323  vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
324  vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
325 
326  vlc_tab_fft_tone_offset[4].table = &qdm2_table[qdm2_vlc_offs[16]];
327  vlc_tab_fft_tone_offset[4].table_allocated = qdm2_vlc_offs[17] - qdm2_vlc_offs[16];
328  init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
329  vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
330  vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_NEW_STATIC | INIT_VLC_LE);
331 
332  vlcs_initialized=1;
333  }
334 }
335 
336 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
337 {
338  int value;
339 
340  value = get_vlc2(gb, vlc->table, vlc->bits, depth);
341 
342  /* stage-2, 3 bits exponent escape sequence */
343  if (value-- == 0)
344  value = get_bits (gb, get_bits (gb, 3) + 1);
345 
346  /* stage-3, optional */
347  if (flag) {
348  int tmp = vlc_stage3_values[value];
349 
350  if ((value & ~3) > 0)
351  tmp += get_bits (gb, (value >> 2));
352  value = tmp;
353  }
354 
355  return value;
356 }
357 
358 
359 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
360 {
361  int value = qdm2_get_vlc (gb, vlc, 0, depth);
362 
363  return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
364 }
365 
366 
376 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
377  int i;
378 
379  for (i=0; i < length; i++)
380  value -= data[i];
381 
382  return (uint16_t)(value & 0xffff);
383 }
384 
385 
392 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
393 {
394  sub_packet->type = get_bits (gb, 8);
395 
396  if (sub_packet->type == 0) {
397  sub_packet->size = 0;
398  sub_packet->data = NULL;
399  } else {
400  sub_packet->size = get_bits (gb, 8);
401 
402  if (sub_packet->type & 0x80) {
403  sub_packet->size <<= 8;
404  sub_packet->size |= get_bits (gb, 8);
405  sub_packet->type &= 0x7f;
406  }
407 
408  if (sub_packet->type == 0x7f)
409  sub_packet->type |= (get_bits (gb, 8) << 8);
410 
411  sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
412  }
413 
414  av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
415  sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
416 }
417 
418 
426 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
427 {
428  while (list != NULL && list->packet != NULL) {
429  if (list->packet->type == type)
430  return list;
431  list = list->next;
432  }
433  return NULL;
434 }
435 
436 
443 static void average_quantized_coeffs (QDM2Context *q)
444 {
445  int i, j, n, ch, sum;
446 
447  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
448 
449  for (ch = 0; ch < q->nb_channels; ch++)
450  for (i = 0; i < n; i++) {
451  sum = 0;
452 
453  for (j = 0; j < 8; j++)
454  sum += q->quantized_coeffs[ch][i][j];
455 
456  sum /= 8;
457  if (sum > 0)
458  sum--;
459 
460  for (j=0; j < 8; j++)
461  q->quantized_coeffs[ch][i][j] = sum;
462  }
463 }
464 
465 
473 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
474 {
475  int ch, j;
476 
477  FIX_NOISE_IDX(q->noise_idx);
478 
479  if (!q->nb_channels)
480  return;
481 
482  for (ch = 0; ch < q->nb_channels; ch++)
483  for (j = 0; j < 64; j++) {
484  q->sb_samples[ch][j * 2][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
485  q->sb_samples[ch][j * 2 + 1][sb] = SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j];
486  }
487 }
488 
489 
498 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
499 {
500  int j,k;
501  int ch;
502  int run, case_val;
503  static const int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
504 
505  for (ch = 0; ch < channels; ch++) {
506  for (j = 0; j < 64; ) {
507  if((coding_method[ch][sb][j] - 8) > 22) {
508  run = 1;
509  case_val = 8;
510  } else {
511  switch (switchtable[coding_method[ch][sb][j]-8]) {
512  case 0: run = 10; case_val = 10; break;
513  case 1: run = 1; case_val = 16; break;
514  case 2: run = 5; case_val = 24; break;
515  case 3: run = 3; case_val = 30; break;
516  case 4: run = 1; case_val = 30; break;
517  case 5: run = 1; case_val = 8; break;
518  default: run = 1; case_val = 8; break;
519  }
520  }
521  for (k = 0; k < run; k++)
522  if (j + k < 128)
523  if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
524  if (k > 0) {
525  SAMPLES_NEEDED
526  //not debugged, almost never used
527  memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
528  memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
529  }
530  j += run;
531  }
532  }
533 }
534 
535 
543 static void fill_tone_level_array (QDM2Context *q, int flag)
544 {
545  int i, sb, ch, sb_used;
546  int tmp, tab;
547 
548  for (ch = 0; ch < q->nb_channels; ch++)
549  for (sb = 0; sb < 30; sb++)
550  for (i = 0; i < 8; i++) {
551  if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
552  tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
553  q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
554  else
555  tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
556  if(tmp < 0)
557  tmp += 0xff;
558  q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
559  }
560 
561  sb_used = QDM2_SB_USED(q->sub_sampling);
562 
563  if ((q->superblocktype_2_3 != 0) && !flag) {
564  for (sb = 0; sb < sb_used; sb++)
565  for (ch = 0; ch < q->nb_channels; ch++)
566  for (i = 0; i < 64; i++) {
567  q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
568  if (q->tone_level_idx[ch][sb][i] < 0)
569  q->tone_level[ch][sb][i] = 0;
570  else
571  q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
572  }
573  } else {
574  tab = q->superblocktype_2_3 ? 0 : 1;
575  for (sb = 0; sb < sb_used; sb++) {
576  if ((sb >= 4) && (sb <= 23)) {
577  for (ch = 0; ch < q->nb_channels; ch++)
578  for (i = 0; i < 64; i++) {
579  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
580  q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
581  q->tone_level_idx_mid[ch][sb - 4][i / 8] -
582  q->tone_level_idx_hi2[ch][sb - 4];
583  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
584  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
585  q->tone_level[ch][sb][i] = 0;
586  else
587  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
588  }
589  } else {
590  if (sb > 4) {
591  for (ch = 0; ch < q->nb_channels; ch++)
592  for (i = 0; i < 64; i++) {
593  tmp = q->tone_level_idx_base[ch][sb][i / 8] -
594  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
595  q->tone_level_idx_hi2[ch][sb - 4];
596  q->tone_level_idx[ch][sb][i] = tmp & 0xff;
597  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
598  q->tone_level[ch][sb][i] = 0;
599  else
600  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
601  }
602  } else {
603  for (ch = 0; ch < q->nb_channels; ch++)
604  for (i = 0; i < 64; i++) {
605  tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
606  if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
607  q->tone_level[ch][sb][i] = 0;
608  else
609  q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
610  }
611  }
612  }
613  }
614  }
615 
616  return;
617 }
618 
619 
634 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
635  sb_int8_array coding_method, int nb_channels,
636  int c, int superblocktype_2_3, int cm_table_select)
637 {
638  int ch, sb, j;
639  int tmp, acc, esp_40, comp;
640  int add1, add2, add3, add4;
641  int64_t multres;
642 
643  if (!superblocktype_2_3) {
644  /* This case is untested, no samples available */
645  SAMPLES_NEEDED
646  for (ch = 0; ch < nb_channels; ch++)
647  for (sb = 0; sb < 30; sb++) {
648  for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
649  add1 = tone_level_idx[ch][sb][j] - 10;
650  if (add1 < 0)
651  add1 = 0;
652  add2 = add3 = add4 = 0;
653  if (sb > 1) {
654  add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
655  if (add2 < 0)
656  add2 = 0;
657  }
658  if (sb > 0) {
659  add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
660  if (add3 < 0)
661  add3 = 0;
662  }
663  if (sb < 29) {
664  add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
665  if (add4 < 0)
666  add4 = 0;
667  }
668  tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
669  if (tmp < 0)
670  tmp = 0;
671  tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
672  }
673  tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
674  }
675  acc = 0;
676  for (ch = 0; ch < nb_channels; ch++)
677  for (sb = 0; sb < 30; sb++)
678  for (j = 0; j < 64; j++)
679  acc += tone_level_idx_temp[ch][sb][j];
680 
681  multres = 0x66666667 * (acc * 10);
682  esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
683  for (ch = 0; ch < nb_channels; ch++)
684  for (sb = 0; sb < 30; sb++)
685  for (j = 0; j < 64; j++) {
686  comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
687  if (comp < 0)
688  comp += 0xff;
689  comp /= 256; // signed shift
690  switch(sb) {
691  case 0:
692  if (comp < 30)
693  comp = 30;
694  comp += 15;
695  break;
696  case 1:
697  if (comp < 24)
698  comp = 24;
699  comp += 10;
700  break;
701  case 2:
702  case 3:
703  case 4:
704  if (comp < 16)
705  comp = 16;
706  }
707  if (comp <= 5)
708  tmp = 0;
709  else if (comp <= 10)
710  tmp = 10;
711  else if (comp <= 16)
712  tmp = 16;
713  else if (comp <= 24)
714  tmp = -1;
715  else
716  tmp = 0;
717  coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
718  }
719  for (sb = 0; sb < 30; sb++)
720  fix_coding_method_array(sb, nb_channels, coding_method);
721  for (ch = 0; ch < nb_channels; ch++)
722  for (sb = 0; sb < 30; sb++)
723  for (j = 0; j < 64; j++)
724  if (sb >= 10) {
725  if (coding_method[ch][sb][j] < 10)
726  coding_method[ch][sb][j] = 10;
727  } else {
728  if (sb >= 2) {
729  if (coding_method[ch][sb][j] < 16)
730  coding_method[ch][sb][j] = 16;
731  } else {
732  if (coding_method[ch][sb][j] < 30)
733  coding_method[ch][sb][j] = 30;
734  }
735  }
736  } else { // superblocktype_2_3 != 0
737  for (ch = 0; ch < nb_channels; ch++)
738  for (sb = 0; sb < 30; sb++)
739  for (j = 0; j < 64; j++)
740  coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
741  }
742 
743  return;
744 }
745 
746 
758 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
759 {
760  int sb, j, k, n, ch, run, channels;
761  int joined_stereo, zero_encoding, chs;
762  int type34_first;
763  float type34_div = 0;
764  float type34_predictor;
765  float samples[10], sign_bits[16];
766 
767  if (length == 0) {
768  // If no data use noise
769  for (sb=sb_min; sb < sb_max; sb++)
770  build_sb_samples_from_noise (q, sb);
771 
772  return;
773  }
774 
775  for (sb = sb_min; sb < sb_max; sb++) {
776  FIX_NOISE_IDX(q->noise_idx);
777 
778  channels = q->nb_channels;
779 
780  if (q->nb_channels <= 1 || sb < 12)
781  joined_stereo = 0;
782  else if (sb >= 24)
783  joined_stereo = 1;
784  else
785  joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1 (gb) : 0;
786 
787  if (joined_stereo) {
788  if (get_bits_left(gb) >= 16)
789  for (j = 0; j < 16; j++)
790  sign_bits[j] = get_bits1 (gb);
791 
792  for (j = 0; j < 64; j++)
793  if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
794  q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
795 
796  fix_coding_method_array(sb, q->nb_channels, q->coding_method);
797  channels = 1;
798  }
799 
800  for (ch = 0; ch < channels; ch++) {
801  zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
802  type34_predictor = 0.0;
803  type34_first = 1;
804 
805  for (j = 0; j < 128; ) {
806  switch (q->coding_method[ch][sb][j / 2]) {
807  case 8:
808  if (get_bits_left(gb) >= 10) {
809  if (zero_encoding) {
810  for (k = 0; k < 5; k++) {
811  if ((j + 2 * k) >= 128)
812  break;
813  samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
814  }
815  } else {
816  n = get_bits(gb, 8);
817  for (k = 0; k < 5; k++)
818  samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
819  }
820  for (k = 0; k < 5; k++)
821  samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
822  } else {
823  for (k = 0; k < 10; k++)
824  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
825  }
826  run = 10;
827  break;
828 
829  case 10:
830  if (get_bits_left(gb) >= 1) {
831  float f = 0.81;
832 
833  if (get_bits1(gb))
834  f = -f;
835  f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
836  samples[0] = f;
837  } else {
838  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
839  }
840  run = 1;
841  break;
842 
843  case 16:
844  if (get_bits_left(gb) >= 10) {
845  if (zero_encoding) {
846  for (k = 0; k < 5; k++) {
847  if ((j + k) >= 128)
848  break;
849  samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
850  }
851  } else {
852  n = get_bits (gb, 8);
853  for (k = 0; k < 5; k++)
854  samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
855  }
856  } else {
857  for (k = 0; k < 5; k++)
858  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
859  }
860  run = 5;
861  break;
862 
863  case 24:
864  if (get_bits_left(gb) >= 7) {
865  n = get_bits(gb, 7);
866  for (k = 0; k < 3; k++)
867  samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
868  } else {
869  for (k = 0; k < 3; k++)
870  samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
871  }
872  run = 3;
873  break;
874 
875  case 30:
876  if (get_bits_left(gb) >= 4) {
877  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
878  if (index < FF_ARRAY_ELEMS(type30_dequant)) {
879  samples[0] = type30_dequant[index];
880  } else
881  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
882  } else
883  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
884 
885  run = 1;
886  break;
887 
888  case 34:
889  if (get_bits_left(gb) >= 7) {
890  if (type34_first) {
891  type34_div = (float)(1 << get_bits(gb, 2));
892  samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
893  type34_predictor = samples[0];
894  type34_first = 0;
895  } else {
896  unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
897  if (index < FF_ARRAY_ELEMS(type34_delta)) {
898  samples[0] = type34_delta[index] / type34_div + type34_predictor;
899  type34_predictor = samples[0];
900  } else
901  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
902  }
903  } else {
904  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
905  }
906  run = 1;
907  break;
908 
909  default:
910  samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
911  run = 1;
912  break;
913  }
914 
915  if (joined_stereo) {
916  float tmp[10][MPA_MAX_CHANNELS];
917 
918  for (k = 0; k < run; k++) {
919  tmp[k][0] = samples[k];
920  tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
921  }
922  for (chs = 0; chs < q->nb_channels; chs++)
923  for (k = 0; k < run; k++)
924  if ((j + k) < 128)
925  q->sb_samples[chs][j + k][sb] = q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs];
926  } else {
927  for (k = 0; k < run; k++)
928  if ((j + k) < 128)
929  q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
930  }
931 
932  j += run;
933  } // j loop
934  } // channel loop
935  } // subband loop
936 }
937 
938 
947 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb)
948 {
949  int i, k, run, level, diff;
950 
951  if (get_bits_left(gb) < 16)
952  return;
953  level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
954 
955  quantized_coeffs[0] = level;
956 
957  for (i = 0; i < 7; ) {
958  if (get_bits_left(gb) < 16)
959  break;
960  run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
961 
962  if (get_bits_left(gb) < 16)
963  break;
964  diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
965 
966  for (k = 1; k <= run; k++)
967  quantized_coeffs[i + k] = (level + ((k * diff) / run));
968 
969  level += diff;
970  i += run;
971  }
972 }
973 
974 
983 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb)
984 {
985  int sb, j, k, n, ch;
986 
987  for (ch = 0; ch < q->nb_channels; ch++) {
988  init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
989 
990  if (get_bits_left(gb) < 16) {
991  memset(q->quantized_coeffs[ch][0], 0, 8);
992  break;
993  }
994  }
995 
996  n = q->sub_sampling + 1;
997 
998  for (sb = 0; sb < n; sb++)
999  for (ch = 0; ch < q->nb_channels; ch++)
1000  for (j = 0; j < 8; j++) {
1001  if (get_bits_left(gb) < 1)
1002  break;
1003  if (get_bits1(gb)) {
1004  for (k=0; k < 8; k++) {
1005  if (get_bits_left(gb) < 16)
1006  break;
1007  q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1008  }
1009  } else {
1010  for (k=0; k < 8; k++)
1011  q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1012  }
1013  }
1014 
1015  n = QDM2_SB_USED(q->sub_sampling) - 4;
1016 
1017  for (sb = 0; sb < n; sb++)
1018  for (ch = 0; ch < q->nb_channels; ch++) {
1019  if (get_bits_left(gb) < 16)
1020  break;
1021  q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1022  if (sb > 19)
1023  q->tone_level_idx_hi2[ch][sb] -= 16;
1024  else
1025  for (j = 0; j < 8; j++)
1026  q->tone_level_idx_mid[ch][sb][j] = -16;
1027  }
1028 
1029  n = QDM2_SB_USED(q->sub_sampling) - 5;
1030 
1031  for (sb = 0; sb < n; sb++)
1032  for (ch = 0; ch < q->nb_channels; ch++)
1033  for (j = 0; j < 8; j++) {
1034  if (get_bits_left(gb) < 16)
1035  break;
1036  q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1037  }
1038 }
1039 
1046 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1047 {
1048  GetBitContext gb;
1049  int i, j, k, n, ch, run, level, diff;
1050 
1051  init_get_bits(&gb, node->packet->data, node->packet->size*8);
1052 
1053  n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1054 
1055  for (i = 1; i < n; i++)
1056  for (ch=0; ch < q->nb_channels; ch++) {
1057  level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1058  q->quantized_coeffs[ch][i][0] = level;
1059 
1060  for (j = 0; j < (8 - 1); ) {
1061  run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1062  diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1063 
1064  for (k = 1; k <= run; k++)
1065  q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1066 
1067  level += diff;
1068  j += run;
1069  }
1070  }
1071 
1072  for (ch = 0; ch < q->nb_channels; ch++)
1073  for (i = 0; i < 8; i++)
1074  q->quantized_coeffs[ch][0][i] = 0;
1075 }
1076 
1077 
1084 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node)
1085 {
1086  GetBitContext gb;
1087 
1088  if (node) {
1089  init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1090  init_tone_level_dequantization(q, &gb);
1091  fill_tone_level_array(q, 1);
1092  } else {
1093  fill_tone_level_array(q, 0);
1094  }
1095 }
1096 
1097 
1104 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node)
1105 {
1106  GetBitContext gb;
1107  int length = 0;
1108 
1109  if (node) {
1110  length = node->packet->size * 8;
1111  init_get_bits(&gb, node->packet->data, length);
1112  }
1113 
1114  if (length >= 32) {
1115  int c = get_bits (&gb, 13);
1116 
1117  if (c > 3)
1118  fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1119  q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1120  }
1121 
1122  synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1123 }
1124 
1125 
1132 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node)
1133 {
1134  GetBitContext gb;
1135  int length = 0;
1136 
1137  if (node) {
1138  length = node->packet->size * 8;
1139  init_get_bits(&gb, node->packet->data, length);
1140  }
1141 
1142  synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1143 }
1144 
1145 /*
1146  * Process new subpackets for synthesis filter
1147  *
1148  * @param q context
1149  * @param list list with synthesis filter packets (list D)
1150  */
1151 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1152 {
1153  QDM2SubPNode *nodes[4];
1154 
1155  nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1156  if (nodes[0] != NULL)
1157  process_subpacket_9(q, nodes[0]);
1158 
1159  nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1160  if (nodes[1] != NULL)
1161  process_subpacket_10(q, nodes[1]);
1162  else
1163  process_subpacket_10(q, NULL);
1164 
1165  nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1166  if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1167  process_subpacket_11(q, nodes[2]);
1168  else
1169  process_subpacket_11(q, NULL);
1170 
1171  nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1172  if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1173  process_subpacket_12(q, nodes[3]);
1174  else
1175  process_subpacket_12(q, NULL);
1176 }
1177 
1178 
1179 /*
1180  * Decode superblock, fill packet lists.
1181  *
1182  * @param q context
1183  */
1184 static void qdm2_decode_super_block (QDM2Context *q)
1185 {
1186  GetBitContext gb;
1187  QDM2SubPacket header, *packet;
1188  int i, packet_bytes, sub_packet_size, sub_packets_D;
1189  unsigned int next_index = 0;
1190 
1191  memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1192  memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1193  memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1194 
1195  q->sub_packets_B = 0;
1196  sub_packets_D = 0;
1197 
1198  average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1199 
1200  init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1201  qdm2_decode_sub_packet_header(&gb, &header);
1202 
1203  if (header.type < 2 || header.type >= 8) {
1204  q->has_errors = 1;
1205  av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1206  return;
1207  }
1208 
1209  q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1210  packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1211 
1212  init_get_bits(&gb, header.data, header.size*8);
1213 
1214  if (header.type == 2 || header.type == 4 || header.type == 5) {
1215  int csum = 257 * get_bits(&gb, 8);
1216  csum += 2 * get_bits(&gb, 8);
1217 
1218  csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1219 
1220  if (csum != 0) {
1221  q->has_errors = 1;
1222  av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1223  return;
1224  }
1225  }
1226 
1227  q->sub_packet_list_B[0].packet = NULL;
1228  q->sub_packet_list_D[0].packet = NULL;
1229 
1230  for (i = 0; i < 6; i++)
1231  if (--q->fft_level_exp[i] < 0)
1232  q->fft_level_exp[i] = 0;
1233 
1234  for (i = 0; packet_bytes > 0; i++) {
1235  int j;
1236 
1237  if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1238  SAMPLES_NEEDED_2("too many packet bytes");
1239  return;
1240  }
1241 
1242  q->sub_packet_list_A[i].next = NULL;
1243 
1244  if (i > 0) {
1245  q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1246 
1247  /* seek to next block */
1248  init_get_bits(&gb, header.data, header.size*8);
1249  skip_bits(&gb, next_index*8);
1250 
1251  if (next_index >= header.size)
1252  break;
1253  }
1254 
1255  /* decode subpacket */
1256  packet = &q->sub_packets[i];
1257  qdm2_decode_sub_packet_header(&gb, packet);
1258  next_index = packet->size + get_bits_count(&gb) / 8;
1259  sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1260 
1261  if (packet->type == 0)
1262  break;
1263 
1264  if (sub_packet_size > packet_bytes) {
1265  if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1266  break;
1267  packet->size += packet_bytes - sub_packet_size;
1268  }
1269 
1270  packet_bytes -= sub_packet_size;
1271 
1272  /* add subpacket to 'all subpackets' list */
1273  q->sub_packet_list_A[i].packet = packet;
1274 
1275  /* add subpacket to related list */
1276  if (packet->type == 8) {
1277  SAMPLES_NEEDED_2("packet type 8");
1278  return;
1279  } else if (packet->type >= 9 && packet->type <= 12) {
1280  /* packets for MPEG Audio like Synthesis Filter */
1281  QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1282  } else if (packet->type == 13) {
1283  for (j = 0; j < 6; j++)
1284  q->fft_level_exp[j] = get_bits(&gb, 6);
1285  } else if (packet->type == 14) {
1286  for (j = 0; j < 6; j++)
1287  q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1288  } else if (packet->type == 15) {
1289  SAMPLES_NEEDED_2("packet type 15")
1290  return;
1291  } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1292  /* packets for FFT */
1293  QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1294  }
1295  } // Packet bytes loop
1296 
1297 /* **************************************************************** */
1298  if (q->sub_packet_list_D[0].packet != NULL) {
1299  process_synthesis_subpackets(q, q->sub_packet_list_D);
1300  q->do_synth_filter = 1;
1301  } else if (q->do_synth_filter) {
1302  process_subpacket_10(q, NULL);
1303  process_subpacket_11(q, NULL);
1304  process_subpacket_12(q, NULL);
1305  }
1306 /* **************************************************************** */
1307 }
1308 
1309 
1310 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1311  int offset, int duration, int channel,
1312  int exp, int phase)
1313 {
1314  if (q->fft_coefs_min_index[duration] < 0)
1315  q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1316 
1317  q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1318  q->fft_coefs[q->fft_coefs_index].channel = channel;
1319  q->fft_coefs[q->fft_coefs_index].offset = offset;
1320  q->fft_coefs[q->fft_coefs_index].exp = exp;
1321  q->fft_coefs[q->fft_coefs_index].phase = phase;
1322  q->fft_coefs_index++;
1323 }
1324 
1325 
1326 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1327 {
1328  int channel, stereo, phase, exp;
1329  int local_int_4, local_int_8, stereo_phase, local_int_10;
1330  int local_int_14, stereo_exp, local_int_20, local_int_28;
1331  int n, offset;
1332 
1333  local_int_4 = 0;
1334  local_int_28 = 0;
1335  local_int_20 = 2;
1336  local_int_8 = (4 - duration);
1337  local_int_10 = 1 << (q->group_order - duration - 1);
1338  offset = 1;
1339 
1340  while (1) {
1341  if (q->superblocktype_2_3) {
1342  while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1343  offset = 1;
1344  if (n == 0) {
1345  local_int_4 += local_int_10;
1346  local_int_28 += (1 << local_int_8);
1347  } else {
1348  local_int_4 += 8*local_int_10;
1349  local_int_28 += (8 << local_int_8);
1350  }
1351  }
1352  offset += (n - 2);
1353  } else {
1354  offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1355  while (offset >= (local_int_10 - 1)) {
1356  offset += (1 - (local_int_10 - 1));
1357  local_int_4 += local_int_10;
1358  local_int_28 += (1 << local_int_8);
1359  }
1360  }
1361 
1362  if (local_int_4 >= q->group_size)
1363  return;
1364 
1365  local_int_14 = (offset >> local_int_8);
1366  if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1367  return;
1368 
1369  if (q->nb_channels > 1) {
1370  channel = get_bits1(gb);
1371  stereo = get_bits1(gb);
1372  } else {
1373  channel = 0;
1374  stereo = 0;
1375  }
1376 
1377  exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1378  exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1379  exp = (exp < 0) ? 0 : exp;
1380 
1381  phase = get_bits(gb, 3);
1382  stereo_exp = 0;
1383  stereo_phase = 0;
1384 
1385  if (stereo) {
1386  stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1387  stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1388  if (stereo_phase < 0)
1389  stereo_phase += 8;
1390  }
1391 
1392  if (q->frequency_range > (local_int_14 + 1)) {
1393  int sub_packet = (local_int_20 + local_int_28);
1394 
1395  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1396  if (stereo)
1397  qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1398  }
1399 
1400  offset++;
1401  }
1402 }
1403 
1404 
1405 static void qdm2_decode_fft_packets (QDM2Context *q)
1406 {
1407  int i, j, min, max, value, type, unknown_flag;
1408  GetBitContext gb;
1409 
1410  if (q->sub_packet_list_B[0].packet == NULL)
1411  return;
1412 
1413  /* reset minimum indexes for FFT coefficients */
1414  q->fft_coefs_index = 0;
1415  for (i=0; i < 5; i++)
1416  q->fft_coefs_min_index[i] = -1;
1417 
1418  /* process subpackets ordered by type, largest type first */
1419  for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1420  QDM2SubPacket *packet= NULL;
1421 
1422  /* find subpacket with largest type less than max */
1423  for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1424  value = q->sub_packet_list_B[j].packet->type;
1425  if (value > min && value < max) {
1426  min = value;
1427  packet = q->sub_packet_list_B[j].packet;
1428  }
1429  }
1430 
1431  max = min;
1432 
1433  /* check for errors (?) */
1434  if (!packet)
1435  return;
1436 
1437  if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1438  return;
1439 
1440  /* decode FFT tones */
1441  init_get_bits (&gb, packet->data, packet->size*8);
1442 
1443  if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1444  unknown_flag = 1;
1445  else
1446  unknown_flag = 0;
1447 
1448  type = packet->type;
1449 
1450  if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1451  int duration = q->sub_sampling + 5 - (type & 15);
1452 
1453  if (duration >= 0 && duration < 4)
1454  qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1455  } else if (type == 31) {
1456  for (j=0; j < 4; j++)
1457  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1458  } else if (type == 46) {
1459  for (j=0; j < 6; j++)
1460  q->fft_level_exp[j] = get_bits(&gb, 6);
1461  for (j=0; j < 4; j++)
1462  qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1463  }
1464  } // Loop on B packets
1465 
1466  /* calculate maximum indexes for FFT coefficients */
1467  for (i = 0, j = -1; i < 5; i++)
1468  if (q->fft_coefs_min_index[i] >= 0) {
1469  if (j >= 0)
1470  q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1471  j = i;
1472  }
1473  if (j >= 0)
1474  q->fft_coefs_max_index[j] = q->fft_coefs_index;
1475 }
1476 
1477 
1478 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1479 {
1480  float level, f[6];
1481  int i;
1482  QDM2Complex c;
1483  const double iscale = 2.0*M_PI / 512.0;
1484 
1485  tone->phase += tone->phase_shift;
1486 
1487  /* calculate current level (maximum amplitude) of tone */
1488  level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1489  c.im = level * sin(tone->phase*iscale);
1490  c.re = level * cos(tone->phase*iscale);
1491 
1492  /* generate FFT coefficients for tone */
1493  if (tone->duration >= 3 || tone->cutoff >= 3) {
1494  tone->complex[0].im += c.im;
1495  tone->complex[0].re += c.re;
1496  tone->complex[1].im -= c.im;
1497  tone->complex[1].re -= c.re;
1498  } else {
1499  f[1] = -tone->table[4];
1500  f[0] = tone->table[3] - tone->table[0];
1501  f[2] = 1.0 - tone->table[2] - tone->table[3];
1502  f[3] = tone->table[1] + tone->table[4] - 1.0;
1503  f[4] = tone->table[0] - tone->table[1];
1504  f[5] = tone->table[2];
1505  for (i = 0; i < 2; i++) {
1506  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1507  tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1508  }
1509  for (i = 0; i < 4; i++) {
1510  tone->complex[i].re += c.re * f[i+2];
1511  tone->complex[i].im += c.im * f[i+2];
1512  }
1513  }
1514 
1515  /* copy the tone if it has not yet died out */
1516  if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1517  memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1518  q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1519  }
1520 }
1521 
1522 
1523 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1524 {
1525  int i, j, ch;
1526  const double iscale = 0.25 * M_PI;
1527 
1528  for (ch = 0; ch < q->channels; ch++) {
1529  memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1530  }
1531 
1532 
1533  /* apply FFT tones with duration 4 (1 FFT period) */
1534  if (q->fft_coefs_min_index[4] >= 0)
1535  for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1536  float level;
1537  QDM2Complex c;
1538 
1539  if (q->fft_coefs[i].sub_packet != sub_packet)
1540  break;
1541 
1542  ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1543  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];
1544 
1545  c.re = level * cos(q->fft_coefs[i].phase * iscale);
1546  c.im = level * sin(q->fft_coefs[i].phase * iscale);
1547  q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1548  q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1549  q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1550  q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1551  }
1552 
1553  /* generate existing FFT tones */
1554  for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1555  qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1556  q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1557  }
1558 
1559  /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1560  for (i = 0; i < 4; i++)
1561  if (q->fft_coefs_min_index[i] >= 0) {
1562  for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1563  int offset, four_i;
1564  FFTTone tone;
1565 
1566  if (q->fft_coefs[j].sub_packet != sub_packet)
1567  break;
1568 
1569  four_i = (4 - i);
1570  offset = q->fft_coefs[j].offset >> four_i;
1571  ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1572 
1573  if (offset < q->frequency_range) {
1574  if (offset < 2)
1575  tone.cutoff = offset;
1576  else
1577  tone.cutoff = (offset >= 60) ? 3 : 2;
1578 
1579  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];
1580  tone.complex = &q->fft.complex[ch][offset];
1581  tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1582  tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1583  tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1584  tone.duration = i;
1585  tone.time_index = 0;
1586 
1587  qdm2_fft_generate_tone(q, &tone);
1588  }
1589  }
1590  q->fft_coefs_min_index[i] = j;
1591  }
1592 }
1593 
1594 
1595 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1596 {
1597  const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1598  float *out = q->output_buffer + channel;
1599  int i;
1600  q->fft.complex[channel][0].re *= 2.0f;
1601  q->fft.complex[channel][0].im = 0.0f;
1602  q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1603  /* add samples to output buffer */
1604  for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1605  out[0] += q->fft.complex[channel][i].re * gain;
1606  out[q->channels] += q->fft.complex[channel][i].im * gain;
1607  out += 2 * q->channels;
1608  }
1609 }
1610 
1611 
1616 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1617 {
1618  int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1619 
1620  /* copy sb_samples */
1621  sb_used = QDM2_SB_USED(q->sub_sampling);
1622 
1623  for (ch = 0; ch < q->channels; ch++)
1624  for (i = 0; i < 8; i++)
1625  for (k=sb_used; k < SBLIMIT; k++)
1626  q->sb_samples[ch][(8 * index) + i][k] = 0;
1627 
1628  for (ch = 0; ch < q->nb_channels; ch++) {
1629  float *samples_ptr = q->samples + ch;
1630 
1631  for (i = 0; i < 8; i++) {
1632  ff_mpa_synth_filter_float(&q->mpadsp,
1633  q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1634  ff_mpa_synth_window_float, &dither_state,
1635  samples_ptr, q->nb_channels,
1636  q->sb_samples[ch][(8 * index) + i]);
1637  samples_ptr += 32 * q->nb_channels;
1638  }
1639  }
1640 
1641  /* add samples to output buffer */
1642  sub_sampling = (4 >> q->sub_sampling);
1643 
1644  for (ch = 0; ch < q->channels; ch++)
1645  for (i = 0; i < q->frame_size; i++)
1646  q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1647 }
1648 
1649 
1655 static av_cold void qdm2_init(QDM2Context *q) {
1656  static int initialized = 0;
1657 
1658  if (initialized != 0)
1659  return;
1660  initialized = 1;
1661 
1662  qdm2_init_vlc();
1663  ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1664  softclip_table_init();
1665  rnd_table_init();
1666  init_noise_samples();
1667 
1668  av_log(NULL, AV_LOG_DEBUG, "init done\n");
1669 }
1670 
1671 
1675 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1676 {
1677  QDM2Context *s = avctx->priv_data;
1678  uint8_t *extradata;
1679  int extradata_size;
1680  int tmp_val, tmp, size;
1681 
1682  /* extradata parsing
1683 
1684  Structure:
1685  wave {
1686  frma (QDM2)
1687  QDCA
1688  QDCP
1689  }
1690 
1691  32 size (including this field)
1692  32 tag (=frma)
1693  32 type (=QDM2 or QDMC)
1694 
1695  32 size (including this field, in bytes)
1696  32 tag (=QDCA) // maybe mandatory parameters
1697  32 unknown (=1)
1698  32 channels (=2)
1699  32 samplerate (=44100)
1700  32 bitrate (=96000)
1701  32 block size (=4096)
1702  32 frame size (=256) (for one channel)
1703  32 packet size (=1300)
1704 
1705  32 size (including this field, in bytes)
1706  32 tag (=QDCP) // maybe some tuneable parameters
1707  32 float1 (=1.0)
1708  32 zero ?
1709  32 float2 (=1.0)
1710  32 float3 (=1.0)
1711  32 unknown (27)
1712  32 unknown (8)
1713  32 zero ?
1714  */
1715 
1716  if (!avctx->extradata || (avctx->extradata_size < 48)) {
1717  av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1718  return -1;
1719  }
1720 
1721  extradata = avctx->extradata;
1722  extradata_size = avctx->extradata_size;
1723 
1724  while (extradata_size > 7) {
1725  if (!memcmp(extradata, "frmaQDM", 7))
1726  break;
1727  extradata++;
1728  extradata_size--;
1729  }
1730 
1731  if (extradata_size < 12) {
1732  av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1733  extradata_size);
1734  return -1;
1735  }
1736 
1737  if (memcmp(extradata, "frmaQDM", 7)) {
1738  av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1739  return -1;
1740  }
1741 
1742  if (extradata[7] == 'C') {
1743 // s->is_qdmc = 1;
1744  av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1745  return -1;
1746  }
1747 
1748  extradata += 8;
1749  extradata_size -= 8;
1750 
1751  size = AV_RB32(extradata);
1752 
1753  if(size > extradata_size){
1754  av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1755  extradata_size, size);
1756  return -1;
1757  }
1758 
1759  extradata += 4;
1760  av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1761  if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1762  av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1763  return -1;
1764  }
1765 
1766  extradata += 8;
1767 
1768  avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1769  extradata += 4;
1770  if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS)
1771  return AVERROR_INVALIDDATA;
1772  avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1773  AV_CH_LAYOUT_MONO;
1774 
1775  avctx->sample_rate = AV_RB32(extradata);
1776  extradata += 4;
1777 
1778  avctx->bit_rate = AV_RB32(extradata);
1779  extradata += 4;
1780 
1781  s->group_size = AV_RB32(extradata);
1782  extradata += 4;
1783 
1784  s->fft_size = AV_RB32(extradata);
1785  extradata += 4;
1786 
1787  s->checksum_size = AV_RB32(extradata);
1788  if (s->checksum_size >= 1U << 28) {
1789  av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1790  return AVERROR_INVALIDDATA;
1791  }
1792 
1793  s->fft_order = av_log2(s->fft_size) + 1;
1794 
1795  // something like max decodable tones
1796  s->group_order = av_log2(s->group_size) + 1;
1797  s->frame_size = s->group_size / 16; // 16 iterations per super block
1798  if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1799  return AVERROR_INVALIDDATA;
1800 
1801  s->sub_sampling = s->fft_order - 7;
1802  s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1803 
1804  switch ((s->sub_sampling * 2 + s->channels - 1)) {
1805  case 0: tmp = 40; break;
1806  case 1: tmp = 48; break;
1807  case 2: tmp = 56; break;
1808  case 3: tmp = 72; break;
1809  case 4: tmp = 80; break;
1810  case 5: tmp = 100;break;
1811  default: tmp=s->sub_sampling; break;
1812  }
1813  tmp_val = 0;
1814  if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1815  if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1816  if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1817  if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1818  s->cm_table_select = tmp_val;
1819 
1820  if (s->sub_sampling == 0)
1821  tmp = 7999;
1822  else
1823  tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1824  /*
1825  0: 7999 -> 0
1826  1: 20000 -> 2
1827  2: 28000 -> 2
1828  */
1829  if (tmp < 8000)
1830  s->coeff_per_sb_select = 0;
1831  else if (tmp <= 16000)
1832  s->coeff_per_sb_select = 1;
1833  else
1834  s->coeff_per_sb_select = 2;
1835 
1836  // Fail on unknown fft order
1837  if ((s->fft_order < 7) || (s->fft_order > 9)) {
1838  av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1839  return -1;
1840  }
1841  if (s->fft_size != (1 << (s->fft_order - 1))) {
1842  av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1843  return AVERROR_INVALIDDATA;
1844  }
1845 
1846  ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1847  ff_mpadsp_init(&s->mpadsp);
1848 
1849  qdm2_init(s);
1850 
1851  avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1852 
1853  avcodec_get_frame_defaults(&s->frame);
1854  avctx->coded_frame = &s->frame;
1855 
1856  return 0;
1857 }
1858 
1859 
1860 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1861 {
1862  QDM2Context *s = avctx->priv_data;
1863 
1864  ff_rdft_end(&s->rdft_ctx);
1865 
1866  return 0;
1867 }
1868 
1869 
1870 static int qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1871 {
1872  int ch, i;
1873  const int frame_size = (q->frame_size * q->channels);
1874 
1875  /* select input buffer */
1876  q->compressed_data = in;
1877  q->compressed_size = q->checksum_size;
1878 
1879  /* copy old block, clear new block of output samples */
1880  memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1881  memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1882 
1883  /* decode block of QDM2 compressed data */
1884  if (q->sub_packet == 0) {
1885  q->has_errors = 0; // zero it for a new super block
1886  av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1887  qdm2_decode_super_block(q);
1888  }
1889 
1890  /* parse subpackets */
1891  if (!q->has_errors) {
1892  if (q->sub_packet == 2)
1893  qdm2_decode_fft_packets(q);
1894 
1895  qdm2_fft_tone_synthesizer(q, q->sub_packet);
1896  }
1897 
1898  /* sound synthesis stage 1 (FFT) */
1899  for (ch = 0; ch < q->channels; ch++) {
1900  qdm2_calculate_fft(q, ch, q->sub_packet);
1901 
1902  if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1903  SAMPLES_NEEDED_2("has errors, and C list is not empty")
1904  return -1;
1905  }
1906  }
1907 
1908  /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1909  if (!q->has_errors && q->do_synth_filter)
1910  qdm2_synthesis_filter(q, q->sub_packet);
1911 
1912  q->sub_packet = (q->sub_packet + 1) % 16;
1913 
1914  /* clip and convert output float[] to 16bit signed samples */
1915  for (i = 0; i < frame_size; i++) {
1916  int value = (int)q->output_buffer[i];
1917 
1918  if (value > SOFTCLIP_THRESHOLD)
1919  value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1920  else if (value < -SOFTCLIP_THRESHOLD)
1921  value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1922 
1923  out[i] = value;
1924  }
1925 
1926  return 0;
1927 }
1928 
1929 
1930 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1931  int *got_frame_ptr, AVPacket *avpkt)
1932 {
1933  const uint8_t *buf = avpkt->data;
1934  int buf_size = avpkt->size;
1935  QDM2Context *s = avctx->priv_data;
1936  int16_t *out;
1937  int i, ret;
1938 
1939  if(!buf)
1940  return 0;
1941  if(buf_size < s->checksum_size)
1942  return -1;
1943 
1944  /* get output buffer */
1945  s->frame.nb_samples = 16 * s->frame_size;
1946  if ((ret = ff_get_buffer(avctx, &s->frame)) < 0) {
1947  av_log(avctx, AV_LOG_ERROR, "get_buffer() failed\n");
1948  return ret;
1949  }
1950  out = (int16_t *)s->frame.data[0];
1951 
1952  for (i = 0; i < 16; i++) {
1953  if (qdm2_decode(s, buf, out) < 0)
1954  return -1;
1955  out += s->channels * s->frame_size;
1956  }
1957 
1958  *got_frame_ptr = 1;
1959  *(AVFrame *)data = s->frame;
1960 
1961  return s->checksum_size;
1962 }
1963 
1964 AVCodec ff_qdm2_decoder =
1965 {
1966  .name = "qdm2",
1967  .type = AVMEDIA_TYPE_AUDIO,
1968  .id = AV_CODEC_ID_QDM2,
1969  .priv_data_size = sizeof(QDM2Context),
1970  .init = qdm2_decode_init,
1971  .close = qdm2_decode_close,
1972  .decode = qdm2_decode_frame,
1973  .capabilities = CODEC_CAP_DR1,
1974  .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1975 };