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avformat/mpeg: demux ivtv captions
[ffmpeg.git] / libavcodec / qdm2.c
blobe629e3b42ae5acc969b6eaed9ecc5acd4680a78e
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
37 #include "libavutil/channel_layout.h"
38 #include "libavutil/mem_internal.h"
39 #include "libavutil/thread.h"
40 #include "libavutil/tx.h"
41
42 #define BITSTREAM_READER_LE
43 #include "avcodec.h"
44 #include "get_bits.h"
45 #include "bytestream.h"
46 #include "codec_internal.h"
47 #include "decode.h"
48 #include "mpegaudio.h"
49 #include "mpegaudiodsp.h"
50
51 #include "qdm2_tablegen.h"
52
53 #define QDM2_LIST_ADD(list, size, packet) \
54 do { \
55 if (size > 0) { \
56 list[size - 1].next = &list[size]; \
57 } \
58 list[size].packet = packet; \
59 list[size].next = NULL; \
60 size++; \
61 } while(0)
62
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
65
66 #define FIX_NOISE_IDX(noise_idx) \
67 if ((noise_idx) >= 3840) \
68 (noise_idx) -= 3840; \
69
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
71
72 #define SAMPLES_NEEDED \
73 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
74
75 #define SAMPLES_NEEDED_2(why) \
76 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
77
78 #define QDM2_MAX_FRAME_SIZE 512
79
80 typedef int8_t sb_int8_array[2][30][64];
81
82 /**
83 * Subpacket
84 */
85 typedef struct QDM2SubPacket {
86 int type; ///< subpacket type
87 unsigned int size; ///< subpacket size
88 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
89 } QDM2SubPacket;
90
91 /**
92 * A node in the subpacket list
93 */
94 typedef struct QDM2SubPNode {
95 QDM2SubPacket *packet; ///< packet
96 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
97 } QDM2SubPNode;
98
99 typedef struct FFTTone {
100 float level;
101 AVComplexFloat *complex;
102 const float *table;
103 int phase;
104 int phase_shift;
105 int duration;
106 short time_index;
107 short cutoff;
108 } FFTTone;
109
110 typedef struct FFTCoefficient {
111 int16_t sub_packet;
112 uint8_t channel;
113 int16_t offset;
114 int16_t exp;
115 uint8_t phase;
116 } FFTCoefficient;
117
118 typedef struct QDM2FFT {
119 DECLARE_ALIGNED(32, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1];
120 DECLARE_ALIGNED(32, AVComplexFloat, temp)[MPA_MAX_CHANNELS][256];
121 } QDM2FFT;
122
123 /**
124 * QDM2 decoder context
125 */
126 typedef struct QDM2Context {
127 /// Parameters from codec header, do not change during playback
128 int nb_channels; ///< number of channels
129 int channels; ///< number of channels
130 int group_size; ///< size of frame group (16 frames per group)
131 int fft_size; ///< size of FFT, in complex numbers
132 int checksum_size; ///< size of data block, used also for checksum
133
134 /// Parameters built from header parameters, do not change during playback
135 int group_order; ///< order of frame group
136 int fft_order; ///< order of FFT (actually fftorder+1)
137 int frame_size; ///< size of data frame
138 int frequency_range;
139 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
140 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
141 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
142
143 /// Packets and packet lists
144 QDM2SubPacket sub_packets[16]; ///< the packets themselves
145 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
146 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
147 int sub_packets_B; ///< number of packets on 'B' list
148 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
149 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
150
151 /// FFT and tones
152 FFTTone fft_tones[1000];
153 int fft_tone_start;
154 int fft_tone_end;
155 FFTCoefficient fft_coefs[1000];
156 int fft_coefs_index;
157 int fft_coefs_min_index[5];
158 int fft_coefs_max_index[5];
159 int fft_level_exp[6];
160 AVTXContext *rdft_ctx;
161 av_tx_fn rdft_fn;
162 QDM2FFT fft;
163
164 /// I/O data
165 const uint8_t *compressed_data;
166 int compressed_size;
167 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
168
169 /// Synthesis filter
170 MPADSPContext mpadsp;
171 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
172 int synth_buf_offset[MPA_MAX_CHANNELS];
173 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
174 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
175
176 /// Mixed temporary data used in decoding
177 float tone_level[MPA_MAX_CHANNELS][30][64];
178 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
179 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
180 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
181 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
182 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
183 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
184 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
185 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
186
187 // Flags
188 int has_errors; ///< packet has errors
189 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
190 int do_synth_filter; ///< used to perform or skip synthesis filter
191
192 int sub_packet;
193 int noise_idx; ///< index for dithering noise table
194 } QDM2Context;
195
196 static const int switchtable[23] = {
197 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
198 };
199
200 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
201 {
202 int value;
203
204 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
205
206 /* stage-2, 3 bits exponent escape sequence */
207 if (value < 0)
208 value = get_bits(gb, get_bits(gb, 3) + 1);
209
210 /* stage-3, optional */
211 if (flag) {
212 int tmp;
213
214 if (value >= 60) {
215 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
216 return 0;
217 }
218
219 tmp= vlc_stage3_values[value];
220
221 if ((value & ~3) > 0)
222 tmp += get_bits(gb, (value >> 2));
223 value = tmp;
224 }
225
226 return value;
227 }
228
229 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
230 {
231 int value = qdm2_get_vlc(gb, vlc, 0, depth);
232
233 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
234 }
235
236 /**
237 * QDM2 checksum
238 *
239 * @param data pointer to data to be checksummed
240 * @param length data length
241 * @param value checksum value
242 *
243 * @return 0 if checksum is OK
244 */
245 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
246 {
247 int i;
248
249 for (i = 0; i < length; i++)
250 value -= data[i];
251
252 return (uint16_t)(value & 0xffff);
253 }
254
255 /**
256 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
257 *
258 * @param gb bitreader context
259 * @param sub_packet packet under analysis
260 */
261 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
262 QDM2SubPacket *sub_packet)
263 {
264 sub_packet->type = get_bits(gb, 8);
265
266 if (sub_packet->type == 0) {
267 sub_packet->size = 0;
268 sub_packet->data = NULL;
269 } else {
270 sub_packet->size = get_bits(gb, 8);
271
272 if (sub_packet->type & 0x80) {
273 sub_packet->size <<= 8;
274 sub_packet->size |= get_bits(gb, 8);
275 sub_packet->type &= 0x7f;
276 }
277
278 if (sub_packet->type == 0x7f)
279 sub_packet->type |= (get_bits(gb, 8) << 8);
280
281 // FIXME: this depends on bitreader-internal data
282 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
283 }
284
285 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
286 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
287 }
288
289 /**
290 * Return node pointer to first packet of requested type in list.
291 *
292 * @param list list of subpackets to be scanned
293 * @param type type of searched subpacket
294 * @return node pointer for subpacket if found, else NULL
295 */
296 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
297 int type)
298 {
299 while (list && list->packet) {
300 if (list->packet->type == type)
301 return list;
302 list = list->next;
303 }
304 return NULL;
305 }
306
307 /**
308 * Replace 8 elements with their average value.
309 * Called by qdm2_decode_superblock before starting subblock decoding.
310 *
311 * @param q context
312 */
313 static void average_quantized_coeffs(QDM2Context *q)
314 {
315 int i, j, n, ch, sum;
316
317 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
318
319 for (ch = 0; ch < q->nb_channels; ch++)
320 for (i = 0; i < n; i++) {
321 sum = 0;
322
323 for (j = 0; j < 8; j++)
324 sum += q->quantized_coeffs[ch][i][j];
325
326 sum /= 8;
327 if (sum > 0)
328 sum--;
329
330 for (j = 0; j < 8; j++)
331 q->quantized_coeffs[ch][i][j] = sum;
332 }
333 }
334
335 /**
336 * Build subband samples with noise weighted by q->tone_level.
337 * Called by synthfilt_build_sb_samples.
338 *
339 * @param q context
340 * @param sb subband index
341 */
342 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
343 {
344 int ch, j;
345
346 FIX_NOISE_IDX(q->noise_idx);
347
348 if (!q->nb_channels)
349 return;
350
351 for (ch = 0; ch < q->nb_channels; ch++) {
352 for (j = 0; j < 64; j++) {
353 q->sb_samples[ch][j * 2][sb] =
354 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
355 q->sb_samples[ch][j * 2 + 1][sb] =
356 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
357 }
358 }
359 }
360
361 /**
362 * Called while processing data from subpackets 11 and 12.
363 * Used after making changes to coding_method array.
364 *
365 * @param sb subband index
366 * @param channels number of channels
367 * @param coding_method q->coding_method[0][0][0]
368 */
369 static int fix_coding_method_array(int sb, int channels,
370 sb_int8_array coding_method)
371 {
372 int j, k;
373 int ch;
374 int run, case_val;
375
376 for (ch = 0; ch < channels; ch++) {
377 for (j = 0; j < 64; ) {
378 if (coding_method[ch][sb][j] < 8)
379 return -1;
380 if ((coding_method[ch][sb][j] - 8) > 22) {
381 run = 1;
382 case_val = 8;
383 } else {
384 switch (switchtable[coding_method[ch][sb][j] - 8]) {
385 case 0: run = 10;
386 case_val = 10;
387 break;
388 case 1: run = 1;
389 case_val = 16;
390 break;
391 case 2: run = 5;
392 case_val = 24;
393 break;
394 case 3: run = 3;
395 case_val = 30;
396 break;
397 case 4: run = 1;
398 case_val = 30;
399 break;
400 case 5: run = 1;
401 case_val = 8;
402 break;
403 default: run = 1;
404 case_val = 8;
405 break;
406 }
407 }
408 for (k = 0; k < run; k++) {
409 if (j + k < 128) {
410 int sbjk = sb + (j + k) / 64;
411 if (sbjk > 29) {
412 SAMPLES_NEEDED
413 continue;
414 }
415 if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
416 if (k > 0) {
417 SAMPLES_NEEDED
418 //not debugged, almost never used
419 memset(&coding_method[ch][sb][j + k], case_val,
420 k *sizeof(int8_t));
421 memset(&coding_method[ch][sb][j + k], case_val,
422 3 * sizeof(int8_t));
423 }
424 }
425 }
426 }
427 j += run;
428 }
429 }
430 return 0;
431 }
432
433 /**
434 * Related to synthesis filter
435 * Called by process_subpacket_10
436 *
437 * @param q context
438 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
439 */
440 static void fill_tone_level_array(QDM2Context *q, int flag)
441 {
442 int i, sb, ch, sb_used;
443 int tmp, tab;
444
445 for (ch = 0; ch < q->nb_channels; ch++)
446 for (sb = 0; sb < 30; sb++)
447 for (i = 0; i < 8; i++) {
448 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
449 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
450 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
451 else
452 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
453 if(tmp < 0)
454 tmp += 0xff;
455 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
456 }
457
458 sb_used = QDM2_SB_USED(q->sub_sampling);
459
460 if ((q->superblocktype_2_3 != 0) && !flag) {
461 for (sb = 0; sb < sb_used; sb++)
462 for (ch = 0; ch < q->nb_channels; ch++)
463 for (i = 0; i < 64; i++) {
464 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
465 if (q->tone_level_idx[ch][sb][i] < 0)
466 q->tone_level[ch][sb][i] = 0;
467 else
468 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
469 }
470 } else {
471 tab = q->superblocktype_2_3 ? 0 : 1;
472 for (sb = 0; sb < sb_used; sb++) {
473 if ((sb >= 4) && (sb <= 23)) {
474 for (ch = 0; ch < q->nb_channels; ch++)
475 for (i = 0; i < 64; i++) {
476 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
477 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
478 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
479 q->tone_level_idx_hi2[ch][sb - 4];
480 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
481 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
482 q->tone_level[ch][sb][i] = 0;
483 else
484 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
485 }
486 } else {
487 if (sb > 4) {
488 for (ch = 0; ch < q->nb_channels; ch++)
489 for (i = 0; i < 64; i++) {
490 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
491 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
492 q->tone_level_idx_hi2[ch][sb - 4];
493 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
494 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
495 q->tone_level[ch][sb][i] = 0;
496 else
497 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
498 }
499 } else {
500 for (ch = 0; ch < q->nb_channels; ch++)
501 for (i = 0; i < 64; i++) {
502 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
503 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
504 q->tone_level[ch][sb][i] = 0;
505 else
506 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
507 }
508 }
509 }
510 }
511 }
512 }
513
514 /**
515 * Related to synthesis filter
516 * Called by process_subpacket_11
517 * c is built with data from subpacket 11
518 * Most of this function is used only if superblock_type_2_3 == 0,
519 * never seen it in samples.
520 *
521 * @param tone_level_idx
522 * @param tone_level_idx_temp
523 * @param coding_method q->coding_method[0][0][0]
524 * @param nb_channels number of channels
525 * @param c coming from subpacket 11, passed as 8*c
526 * @param superblocktype_2_3 flag based on superblock packet type
527 * @param cm_table_select q->cm_table_select
528 */
529 static void fill_coding_method_array(sb_int8_array tone_level_idx,
530 sb_int8_array tone_level_idx_temp,
531 sb_int8_array coding_method,
532 int nb_channels,
533 int c, int superblocktype_2_3,
534 int cm_table_select)
535 {
536 int ch, sb, j;
537 int tmp, acc, esp_40, comp;
538 int add1, add2, add3, add4;
539 int64_t multres;
540
541 if (!superblocktype_2_3) {
542 /* This case is untested, no samples available */
543 avpriv_request_sample(NULL, "!superblocktype_2_3");
544 return;
545 for (ch = 0; ch < nb_channels; ch++) {
546 for (sb = 0; sb < 30; sb++) {
547 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
548 add1 = tone_level_idx[ch][sb][j] - 10;
549 if (add1 < 0)
550 add1 = 0;
551 add2 = add3 = add4 = 0;
552 if (sb > 1) {
553 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
554 if (add2 < 0)
555 add2 = 0;
556 }
557 if (sb > 0) {
558 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
559 if (add3 < 0)
560 add3 = 0;
561 }
562 if (sb < 29) {
563 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
564 if (add4 < 0)
565 add4 = 0;
566 }
567 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
568 if (tmp < 0)
569 tmp = 0;
570 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
571 }
572 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
573 }
574 }
575 acc = 0;
576 for (ch = 0; ch < nb_channels; ch++)
577 for (sb = 0; sb < 30; sb++)
578 for (j = 0; j < 64; j++)
579 acc += tone_level_idx_temp[ch][sb][j];
580
581 multres = 0x66666667LL * (acc * 10);
582 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
583 for (ch = 0; ch < nb_channels; ch++)
584 for (sb = 0; sb < 30; sb++)
585 for (j = 0; j < 64; j++) {
586 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
587 if (comp < 0)
588 comp += 0xff;
589 comp /= 256; // signed shift
590 switch(sb) {
591 case 0:
592 if (comp < 30)
593 comp = 30;
594 comp += 15;
595 break;
596 case 1:
597 if (comp < 24)
598 comp = 24;
599 comp += 10;
600 break;
601 case 2:
602 case 3:
603 case 4:
604 if (comp < 16)
605 comp = 16;
606 }
607 if (comp <= 5)
608 tmp = 0;
609 else if (comp <= 10)
610 tmp = 10;
611 else if (comp <= 16)
612 tmp = 16;
613 else if (comp <= 24)
614 tmp = -1;
615 else
616 tmp = 0;
617 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
618 }
619 for (sb = 0; sb < 30; sb++)
620 fix_coding_method_array(sb, nb_channels, coding_method);
621 for (ch = 0; ch < nb_channels; ch++)
622 for (sb = 0; sb < 30; sb++)
623 for (j = 0; j < 64; j++)
624 if (sb >= 10) {
625 if (coding_method[ch][sb][j] < 10)
626 coding_method[ch][sb][j] = 10;
627 } else {
628 if (sb >= 2) {
629 if (coding_method[ch][sb][j] < 16)
630 coding_method[ch][sb][j] = 16;
631 } else {
632 if (coding_method[ch][sb][j] < 30)
633 coding_method[ch][sb][j] = 30;
634 }
635 }
636 } else { // superblocktype_2_3 != 0
637 for (ch = 0; ch < nb_channels; ch++)
638 for (sb = 0; sb < 30; sb++)
639 for (j = 0; j < 64; j++)
640 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
641 }
642 }
643
644 /**
645 * Called by process_subpacket_11 to process more data from subpacket 11
646 * with sb 0-8.
647 * Called by process_subpacket_12 to process data from subpacket 12 with
648 * sb 8-sb_used.
649 *
650 * @param q context
651 * @param gb bitreader context
652 * @param length packet length in bits
653 * @param sb_min lower subband processed (sb_min included)
654 * @param sb_max higher subband processed (sb_max excluded)
655 */
656 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
657 int length, int sb_min, int sb_max)
658 {
659 int sb, j, k, n, ch, run, channels;
660 int joined_stereo, zero_encoding;
661 int type34_first;
662 float type34_div = 0;
663 float type34_predictor;
664 float samples[10];
665 int sign_bits[16] = {0};
666
667 if (length == 0) {
668 // If no data use noise
669 for (sb=sb_min; sb < sb_max; sb++)
670 build_sb_samples_from_noise(q, sb);
671
672 return 0;
673 }
674
675 for (sb = sb_min; sb < sb_max; sb++) {
676 channels = q->nb_channels;
677
678 if (q->nb_channels <= 1 || sb < 12)
679 joined_stereo = 0;
680 else if (sb >= 24)
681 joined_stereo = 1;
682 else
683 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
684
685 if (joined_stereo) {
686 if (get_bits_left(gb) >= 16)
687 for (j = 0; j < 16; j++)
688 sign_bits[j] = get_bits1(gb);
689
690 for (j = 0; j < 64; j++)
691 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
692 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
693
694 if (fix_coding_method_array(sb, q->nb_channels,
695 q->coding_method)) {
696 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
697 build_sb_samples_from_noise(q, sb);
698 continue;
699 }
700 channels = 1;
701 }
702
703 for (ch = 0; ch < channels; ch++) {
704 FIX_NOISE_IDX(q->noise_idx);
705 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
706 type34_predictor = 0.0;
707 type34_first = 1;
708
709 for (j = 0; j < 128; ) {
710 switch (q->coding_method[ch][sb][j / 2]) {
711 case 8:
712 if (get_bits_left(gb) >= 10) {
713 if (zero_encoding) {
714 for (k = 0; k < 5; k++) {
715 if ((j + 2 * k) >= 128)
716 break;
717 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
718 }
719 } else {
720 n = get_bits(gb, 8);
721 if (n >= 243) {
722 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
723 return AVERROR_INVALIDDATA;
724 }
725
726 for (k = 0; k < 5; k++)
727 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
728 }
729 for (k = 0; k < 5; k++)
730 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
731 } else {
732 for (k = 0; k < 10; k++)
733 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
734 }
735 run = 10;
736 break;
737
738 case 10:
739 if (get_bits_left(gb) >= 1) {
740 float f = 0.81;
741
742 if (get_bits1(gb))
743 f = -f;
744 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
745 samples[0] = f;
746 } else {
747 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
748 }
749 run = 1;
750 break;
751
752 case 16:
753 if (get_bits_left(gb) >= 10) {
754 if (zero_encoding) {
755 for (k = 0; k < 5; k++) {
756 if ((j + k) >= 128)
757 break;
758 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
759 }
760 } else {
761 n = get_bits (gb, 8);
762 if (n >= 243) {
763 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
764 return AVERROR_INVALIDDATA;
765 }
766
767 for (k = 0; k < 5; k++)
768 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
769 }
770 } else {
771 for (k = 0; k < 5; k++)
772 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
773 }
774 run = 5;
775 break;
776
777 case 24:
778 if (get_bits_left(gb) >= 7) {
779 n = get_bits(gb, 7);
780 if (n >= 125) {
781 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
782 return AVERROR_INVALIDDATA;
783 }
784
785 for (k = 0; k < 3; k++)
786 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
787 } else {
788 for (k = 0; k < 3; k++)
789 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
790 }
791 run = 3;
792 break;
793
794 case 30:
795 if (get_bits_left(gb) >= 4) {
796 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
797 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
798 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
799 return AVERROR_INVALIDDATA;
800 }
801 samples[0] = type30_dequant[index];
802 } else
803 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
804
805 run = 1;
806 break;
807
808 case 34:
809 if (get_bits_left(gb) >= 7) {
810 if (type34_first) {
811 type34_div = (float)(1 << get_bits(gb, 2));
812 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
813 type34_predictor = samples[0];
814 type34_first = 0;
815 } else {
816 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
817 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
818 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
819 return AVERROR_INVALIDDATA;
820 }
821 samples[0] = type34_delta[index] / type34_div + type34_predictor;
822 type34_predictor = samples[0];
823 }
824 } else {
825 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
826 }
827 run = 1;
828 break;
829
830 default:
831 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
832 run = 1;
833 break;
834 }
835
836 if (joined_stereo) {
837 for (k = 0; k < run && j + k < 128; k++) {
838 q->sb_samples[0][j + k][sb] =
839 q->tone_level[0][sb][(j + k) / 2] * samples[k];
840 if (q->nb_channels == 2) {
841 if (sign_bits[(j + k) / 8])
842 q->sb_samples[1][j + k][sb] =
843 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
844 else
845 q->sb_samples[1][j + k][sb] =
846 q->tone_level[1][sb][(j + k) / 2] * samples[k];
847 }
848 }
849 } else {
850 for (k = 0; k < run; k++)
851 if ((j + k) < 128)
852 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
853 }
854
855 j += run;
856 } // j loop
857 } // channel loop
858 } // subband loop
859 return 0;
860 }
861
862 /**
863 * Init the first element of a channel in quantized_coeffs with data
864 * from packet 10 (quantized_coeffs[ch][0]).
865 * This is similar to process_subpacket_9, but for a single channel
866 * and for element [0]
867 * same VLC tables as process_subpacket_9 are used.
868 *
869 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
870 * @param gb bitreader context
871 */
872 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
873 GetBitContext *gb)
874 {
875 int i, k, run, level, diff;
876
877 if (get_bits_left(gb) < 16)
878 return -1;
879 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
880
881 quantized_coeffs[0] = level;
882
883 for (i = 0; i < 7; ) {
884 if (get_bits_left(gb) < 16)
885 return -1;
886 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
887
888 if (i + run >= 8)
889 return -1;
890
891 if (get_bits_left(gb) < 16)
892 return -1;
893 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
894
895 for (k = 1; k <= run; k++)
896 quantized_coeffs[i + k] = (level + ((k * diff) / run));
897
898 level += diff;
899 i += run;
900 }
901 return 0;
902 }
903
904 /**
905 * Related to synthesis filter, process data from packet 10
906 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
907 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
908 * data from packet 10
909 *
910 * @param q context
911 * @param gb bitreader context
912 */
913 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
914 {
915 int sb, j, k, n, ch;
916
917 for (ch = 0; ch < q->nb_channels; ch++) {
918 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
919
920 if (get_bits_left(gb) < 16) {
921 memset(q->quantized_coeffs[ch][0], 0, 8);
922 break;
923 }
924 }
925
926 n = q->sub_sampling + 1;
927
928 for (sb = 0; sb < n; sb++)
929 for (ch = 0; ch < q->nb_channels; ch++)
930 for (j = 0; j < 8; j++) {
931 if (get_bits_left(gb) < 1)
932 break;
933 if (get_bits1(gb)) {
934 for (k=0; k < 8; k++) {
935 if (get_bits_left(gb) < 16)
936 break;
937 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
938 }
939 } else {
940 for (k=0; k < 8; k++)
941 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
942 }
943 }
944
945 n = QDM2_SB_USED(q->sub_sampling) - 4;
946
947 for (sb = 0; sb < n; sb++)
948 for (ch = 0; ch < q->nb_channels; ch++) {
949 if (get_bits_left(gb) < 16)
950 break;
951 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
952 if (sb > 19)
953 q->tone_level_idx_hi2[ch][sb] -= 16;
954 else
955 for (j = 0; j < 8; j++)
956 q->tone_level_idx_mid[ch][sb][j] = -16;
957 }
958
959 n = QDM2_SB_USED(q->sub_sampling) - 5;
960
961 for (sb = 0; sb < n; sb++)
962 for (ch = 0; ch < q->nb_channels; ch++)
963 for (j = 0; j < 8; j++) {
964 if (get_bits_left(gb) < 16)
965 break;
966 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
967 }
968 }
969
970 /**
971 * Process subpacket 9, init quantized_coeffs with data from it
972 *
973 * @param q context
974 * @param node pointer to node with packet
975 */
976 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
977 {
978 GetBitContext gb;
979 int i, j, k, n, ch, run, level, diff;
980
981 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
982
983 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
984
985 for (i = 1; i < n; i++)
986 for (ch = 0; ch < q->nb_channels; ch++) {
987 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
988 q->quantized_coeffs[ch][i][0] = level;
989
990 for (j = 0; j < (8 - 1); ) {
991 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
992 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
993
994 if (j + run >= 8)
995 return -1;
996
997 for (k = 1; k <= run; k++)
998 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
999
1000 level += diff;
1001 j += run;
1002 }
1003 }
1004
1005 for (ch = 0; ch < q->nb_channels; ch++)
1006 for (i = 0; i < 8; i++)
1007 q->quantized_coeffs[ch][0][i] = 0;
1008
1009 return 0;
1010 }
1011
1012 /**
1013 * Process subpacket 10 if not null, else
1014 *
1015 * @param q context
1016 * @param node pointer to node with packet
1017 */
1018 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1019 {
1020 GetBitContext gb;
1021
1022 if (node) {
1023 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1024 init_tone_level_dequantization(q, &gb);
1025 fill_tone_level_array(q, 1);
1026 } else {
1027 fill_tone_level_array(q, 0);
1028 }
1029 }
1030
1031 /**
1032 * Process subpacket 11
1033 *
1034 * @param q context
1035 * @param node pointer to node with packet
1036 */
1037 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1038 {
1039 GetBitContext gb;
1040 int length = 0;
1041
1042 if (node) {
1043 length = node->packet->size * 8;
1044 init_get_bits(&gb, node->packet->data, length);
1045 }
1046
1047 if (length >= 32) {
1048 int c = get_bits(&gb, 13);
1049
1050 if (c > 3)
1051 fill_coding_method_array(q->tone_level_idx,
1052 q->tone_level_idx_temp, q->coding_method,
1053 q->nb_channels, 8 * c,
1054 q->superblocktype_2_3, q->cm_table_select);
1055 }
1056
1057 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1058 }
1059
1060 /**
1061 * Process subpacket 12
1062 *
1063 * @param q context
1064 * @param node pointer to node with packet
1065 */
1066 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1067 {
1068 GetBitContext gb;
1069 int length = 0;
1070
1071 if (node) {
1072 length = node->packet->size * 8;
1073 init_get_bits(&gb, node->packet->data, length);
1074 }
1075
1076 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1077 }
1078
1079 /**
1080 * Process new subpackets for synthesis filter
1081 *
1082 * @param q context
1083 * @param list list with synthesis filter packets (list D)
1084 */
1085 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1086 {
1087 QDM2SubPNode *nodes[4];
1088
1089 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1090 if (nodes[0])
1091 process_subpacket_9(q, nodes[0]);
1092
1093 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1094 if (nodes[1])
1095 process_subpacket_10(q, nodes[1]);
1096 else
1097 process_subpacket_10(q, NULL);
1098
1099 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1100 if (nodes[0] && nodes[1] && nodes[2])
1101 process_subpacket_11(q, nodes[2]);
1102 else
1103 process_subpacket_11(q, NULL);
1104
1105 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1106 if (nodes[0] && nodes[1] && nodes[3])
1107 process_subpacket_12(q, nodes[3]);
1108 else
1109 process_subpacket_12(q, NULL);
1110 }
1111
1112 /**
1113 * Decode superblock, fill packet lists.
1114 *
1115 * @param q context
1116 */
1117 static void qdm2_decode_super_block(QDM2Context *q)
1118 {
1119 GetBitContext gb;
1120 QDM2SubPacket header, *packet;
1121 int i, packet_bytes, sub_packet_size, sub_packets_D;
1122 unsigned int next_index = 0;
1123
1124 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1125 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1126 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1127
1128 q->sub_packets_B = 0;
1129 sub_packets_D = 0;
1130
1131 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1132
1133 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1134 qdm2_decode_sub_packet_header(&gb, &header);
1135
1136 if (header.type < 2 || header.type >= 8) {
1137 q->has_errors = 1;
1138 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1139 return;
1140 }
1141
1142 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1143 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1144
1145 init_get_bits(&gb, header.data, header.size * 8);
1146
1147 if (header.type == 2 || header.type == 4 || header.type == 5) {
1148 int csum = 257 * get_bits(&gb, 8);
1149 csum += 2 * get_bits(&gb, 8);
1150
1151 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1152
1153 if (csum != 0) {
1154 q->has_errors = 1;
1155 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1156 return;
1157 }
1158 }
1159
1160 q->sub_packet_list_B[0].packet = NULL;
1161 q->sub_packet_list_D[0].packet = NULL;
1162
1163 for (i = 0; i < 6; i++)
1164 if (--q->fft_level_exp[i] < 0)
1165 q->fft_level_exp[i] = 0;
1166
1167 for (i = 0; packet_bytes > 0; i++) {
1168 int j;
1169
1170 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1171 SAMPLES_NEEDED_2("too many packet bytes");
1172 return;
1173 }
1174
1175 q->sub_packet_list_A[i].next = NULL;
1176
1177 if (i > 0) {
1178 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1179
1180 /* seek to next block */
1181 init_get_bits(&gb, header.data, header.size * 8);
1182 skip_bits(&gb, next_index * 8);
1183
1184 if (next_index >= header.size)
1185 break;
1186 }
1187
1188 /* decode subpacket */
1189 packet = &q->sub_packets[i];
1190 qdm2_decode_sub_packet_header(&gb, packet);
1191 next_index = packet->size + get_bits_count(&gb) / 8;
1192 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1193
1194 if (packet->type == 0)
1195 break;
1196
1197 if (sub_packet_size > packet_bytes) {
1198 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1199 break;
1200 packet->size += packet_bytes - sub_packet_size;
1201 }
1202
1203 packet_bytes -= sub_packet_size;
1204
1205 /* add subpacket to 'all subpackets' list */
1206 q->sub_packet_list_A[i].packet = packet;
1207
1208 /* add subpacket to related list */
1209 if (packet->type == 8) {
1210 SAMPLES_NEEDED_2("packet type 8");
1211 return;
1212 } else if (packet->type >= 9 && packet->type <= 12) {
1213 /* packets for MPEG Audio like Synthesis Filter */
1214 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1215 } else if (packet->type == 13) {
1216 for (j = 0; j < 6; j++)
1217 q->fft_level_exp[j] = get_bits(&gb, 6);
1218 } else if (packet->type == 14) {
1219 for (j = 0; j < 6; j++)
1220 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1221 } else if (packet->type == 15) {
1222 SAMPLES_NEEDED_2("packet type 15")
1223 return;
1224 } else if (packet->type >= 16 && packet->type < 48 &&
1225 !fft_subpackets[packet->type - 16]) {
1226 /* packets for FFT */
1227 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1228 }
1229 } // Packet bytes loop
1230
1231 if (q->sub_packet_list_D[0].packet) {
1232 process_synthesis_subpackets(q, q->sub_packet_list_D);
1233 q->do_synth_filter = 1;
1234 } else if (q->do_synth_filter) {
1235 process_subpacket_10(q, NULL);
1236 process_subpacket_11(q, NULL);
1237 process_subpacket_12(q, NULL);
1238 }
1239 }
1240
1241 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1242 int offset, int duration, int channel,
1243 int exp, int phase)
1244 {
1245 if (q->fft_coefs_min_index[duration] < 0)
1246 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1247
1248 q->fft_coefs[q->fft_coefs_index].sub_packet =
1249 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1250 q->fft_coefs[q->fft_coefs_index].channel = channel;
1251 q->fft_coefs[q->fft_coefs_index].offset = offset;
1252 q->fft_coefs[q->fft_coefs_index].exp = exp;
1253 q->fft_coefs[q->fft_coefs_index].phase = phase;
1254 q->fft_coefs_index++;
1255 }
1256
1257 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1258 GetBitContext *gb, int b)
1259 {
1260 int channel, stereo, phase, exp;
1261 int local_int_4, local_int_8, stereo_phase, local_int_10;
1262 int local_int_14, stereo_exp, local_int_20, local_int_28;
1263 int n, offset;
1264
1265 local_int_4 = 0;
1266 local_int_28 = 0;
1267 local_int_20 = 2;
1268 local_int_8 = (4 - duration);
1269 local_int_10 = 1 << (q->group_order - duration - 1);
1270 offset = 1;
1271
1272 while (get_bits_left(gb)>0) {
1273 if (q->superblocktype_2_3) {
1274 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1275 if (get_bits_left(gb)<0) {
1276 if(local_int_4 < q->group_size)
1277 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1278 return;
1279 }
1280 offset = 1;
1281 if (n == 0) {
1282 local_int_4 += local_int_10;
1283 local_int_28 += (1 << local_int_8);
1284 } else {
1285 local_int_4 += 8 * local_int_10;
1286 local_int_28 += (8 << local_int_8);
1287 }
1288 }
1289 offset += (n - 2);
1290 } else {
1291 if (local_int_10 <= 2) {
1292 av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
1293 return;
1294 }
1295 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1296 while (offset >= (local_int_10 - 1)) {
1297 offset += (1 - (local_int_10 - 1));
1298 local_int_4 += local_int_10;
1299 local_int_28 += (1 << local_int_8);
1300 }
1301 }
1302
1303 if (local_int_4 >= q->group_size)
1304 return;
1305
1306 local_int_14 = (offset >> local_int_8);
1307 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1308 return;
1309
1310 if (q->nb_channels > 1) {
1311 channel = get_bits1(gb);
1312 stereo = get_bits1(gb);
1313 } else {
1314 channel = 0;
1315 stereo = 0;
1316 }
1317
1318 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1319 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1320 exp = (exp < 0) ? 0 : exp;
1321
1322 phase = get_bits(gb, 3);
1323 stereo_exp = 0;
1324 stereo_phase = 0;
1325
1326 if (stereo) {
1327 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1328 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1329 if (stereo_phase < 0)
1330 stereo_phase += 8;
1331 }
1332
1333 if (q->frequency_range > (local_int_14 + 1)) {
1334 int sub_packet = (local_int_20 + local_int_28);
1335
1336 if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
1337 return;
1338
1339 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1340 channel, exp, phase);
1341 if (stereo)
1342 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1343 1 - channel,
1344 stereo_exp, stereo_phase);
1345 }
1346 offset++;
1347 }
1348 }
1349
1350 static void qdm2_decode_fft_packets(QDM2Context *q)
1351 {
1352 int i, j, min, max, value, type, unknown_flag;
1353 GetBitContext gb;
1354
1355 if (!q->sub_packet_list_B[0].packet)
1356 return;
1357
1358 /* reset minimum indexes for FFT coefficients */
1359 q->fft_coefs_index = 0;
1360 for (i = 0; i < 5; i++)
1361 q->fft_coefs_min_index[i] = -1;
1362
1363 /* process subpackets ordered by type, largest type first */
1364 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1365 QDM2SubPacket *packet = NULL;
1366
1367 /* find subpacket with largest type less than max */
1368 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1369 value = q->sub_packet_list_B[j].packet->type;
1370 if (value > min && value < max) {
1371 min = value;
1372 packet = q->sub_packet_list_B[j].packet;
1373 }
1374 }
1375
1376 max = min;
1377
1378 /* check for errors (?) */
1379 if (!packet)
1380 return;
1381
1382 if (i == 0 &&
1383 (packet->type < 16 || packet->type >= 48 ||
1384 fft_subpackets[packet->type - 16]))
1385 return;
1386
1387 /* decode FFT tones */
1388 init_get_bits(&gb, packet->data, packet->size * 8);
1389
1390 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1391 unknown_flag = 1;
1392 else
1393 unknown_flag = 0;
1394
1395 type = packet->type;
1396
1397 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1398 int duration = q->sub_sampling + 5 - (type & 15);
1399
1400 if (duration >= 0 && duration < 4)
1401 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1402 } else if (type == 31) {
1403 for (j = 0; j < 4; j++)
1404 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1405 } else if (type == 46) {
1406 for (j = 0; j < 6; j++)
1407 q->fft_level_exp[j] = get_bits(&gb, 6);
1408 for (j = 0; j < 4; j++)
1409 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1410 }
1411 } // Loop on B packets
1412
1413 /* calculate maximum indexes for FFT coefficients */
1414 for (i = 0, j = -1; i < 5; i++)
1415 if (q->fft_coefs_min_index[i] >= 0) {
1416 if (j >= 0)
1417 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1418 j = i;
1419 }
1420 if (j >= 0)
1421 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1422 }
1423
1424 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1425 {
1426 float level, f[6];
1427 int i;
1428 AVComplexFloat c;
1429 const double iscale = 2.0 * M_PI / 512.0;
1430
1431 tone->phase += tone->phase_shift;
1432
1433 /* calculate current level (maximum amplitude) of tone */
1434 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1435 c.im = level * sin(tone->phase * iscale);
1436 c.re = level * cos(tone->phase * iscale);
1437
1438 /* generate FFT coefficients for tone */
1439 if (tone->duration >= 3 || tone->cutoff >= 3) {
1440 tone->complex[0].im += c.im;
1441 tone->complex[0].re += c.re;
1442 tone->complex[1].im -= c.im;
1443 tone->complex[1].re -= c.re;
1444 } else {
1445 f[1] = -tone->table[4];
1446 f[0] = tone->table[3] - tone->table[0];
1447 f[2] = 1.0 - tone->table[2] - tone->table[3];
1448 f[3] = tone->table[1] + tone->table[4] - 1.0;
1449 f[4] = tone->table[0] - tone->table[1];
1450 f[5] = tone->table[2];
1451 for (i = 0; i < 2; i++) {
1452 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1453 c.re * f[i];
1454 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1455 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1456 }
1457 for (i = 0; i < 4; i++) {
1458 tone->complex[i].re += c.re * f[i + 2];
1459 tone->complex[i].im += c.im * f[i + 2];
1460 }
1461 }
1462
1463 /* copy the tone if it has not yet died out */
1464 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1465 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1466 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1467 }
1468 }
1469
1470 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1471 {
1472 int i, j, ch;
1473 const double iscale = 0.25 * M_PI;
1474
1475 for (ch = 0; ch < q->channels; ch++) {
1476 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(AVComplexFloat));
1477 }
1478
1479
1480 /* apply FFT tones with duration 4 (1 FFT period) */
1481 if (q->fft_coefs_min_index[4] >= 0)
1482 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1483 float level;
1484 AVComplexFloat c;
1485
1486 if (q->fft_coefs[i].sub_packet != sub_packet)
1487 break;
1488
1489 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1490 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];
1491
1492 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1493 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1494 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1495 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1496 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1497 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1498 }
1499
1500 /* generate existing FFT tones */
1501 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1502 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1503 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1504 }
1505
1506 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1507 for (i = 0; i < 4; i++)
1508 if (q->fft_coefs_min_index[i] >= 0) {
1509 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1510 int offset, four_i;
1511 FFTTone tone;
1512
1513 if (q->fft_coefs[j].sub_packet != sub_packet)
1514 break;
1515
1516 four_i = (4 - i);
1517 offset = q->fft_coefs[j].offset >> four_i;
1518 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1519
1520 if (offset < q->frequency_range) {
1521 if (offset < 2)
1522 tone.cutoff = offset;
1523 else
1524 tone.cutoff = (offset >= 60) ? 3 : 2;
1525
1526 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];
1527 tone.complex = &q->fft.complex[ch][offset];
1528 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1529 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1530 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1531 tone.duration = i;
1532 tone.time_index = 0;
1533
1534 qdm2_fft_generate_tone(q, &tone);
1535 }
1536 }
1537 q->fft_coefs_min_index[i] = j;
1538 }
1539 }
1540
1541 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1542 {
1543 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1544 float *out = q->output_buffer + channel;
1545
1546 q->fft.complex[channel][0].re *= 2.0f;
1547 q->fft.complex[channel][0].im = 0.0f;
1548 q->fft.complex[channel][q->fft_size].re = 0.0f;
1549 q->fft.complex[channel][q->fft_size].im = 0.0f;
1550
1551 q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel],
1552 sizeof(AVComplexFloat));
1553
1554 /* add samples to output buffer */
1555 for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1556 out[0] += q->fft.temp[channel][i].re * gain;
1557 out[q->channels] += q->fft.temp[channel][i].im * gain;
1558 out += 2 * q->channels;
1559 }
1560 }
1561
1562 /**
1563 * @param q context
1564 * @param index subpacket number
1565 */
1566 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1567 {
1568 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1569
1570 /* copy sb_samples */
1571 sb_used = QDM2_SB_USED(q->sub_sampling);
1572
1573 for (ch = 0; ch < q->channels; ch++)
1574 for (i = 0; i < 8; i++)
1575 for (k = sb_used; k < SBLIMIT; k++)
1576 q->sb_samples[ch][(8 * index) + i][k] = 0;
1577
1578 for (ch = 0; ch < q->nb_channels; ch++) {
1579 float *samples_ptr = q->samples + ch;
1580
1581 for (i = 0; i < 8; i++) {
1582 ff_mpa_synth_filter_float(&q->mpadsp,
1583 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1584 ff_mpa_synth_window_float, &dither_state,
1585 samples_ptr, q->nb_channels,
1586 q->sb_samples[ch][(8 * index) + i]);
1587 samples_ptr += 32 * q->nb_channels;
1588 }
1589 }
1590
1591 /* add samples to output buffer */
1592 sub_sampling = (4 >> q->sub_sampling);
1593
1594 for (ch = 0; ch < q->channels; ch++)
1595 for (i = 0; i < q->frame_size; i++)
1596 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1597 }
1598
1599 /**
1600 * Init static data (does not depend on specific file)
1601 */
1602 static av_cold void qdm2_init_static_data(void) {
1603 qdm2_init_vlc();
1604 softclip_table_init();
1605 rnd_table_init();
1606 init_noise_samples();
1607
1608 ff_mpa_synth_init_float();
1609 }
1610
1611 /**
1612 * Init parameters from codec extradata
1613 */
1614 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1615 {
1616 static AVOnce init_static_once = AV_ONCE_INIT;
1617 QDM2Context *s = avctx->priv_data;
1618 int ret, tmp_val, tmp, size;
1619 float scale = 1.0f / 2.0f;
1620 GetByteContext gb;
1621
1622 /* extradata parsing
1623
1624 Structure:
1625 wave {
1626 frma (QDM2)
1627 QDCA
1628 QDCP
1629 }
1630
1631 32 size (including this field)
1632 32 tag (=frma)
1633 32 type (=QDM2 or QDMC)
1634
1635 32 size (including this field, in bytes)
1636 32 tag (=QDCA) // maybe mandatory parameters
1637 32 unknown (=1)
1638 32 channels (=2)
1639 32 samplerate (=44100)
1640 32 bitrate (=96000)
1641 32 block size (=4096)
1642 32 frame size (=256) (for one channel)
1643 32 packet size (=1300)
1644
1645 32 size (including this field, in bytes)
1646 32 tag (=QDCP) // maybe some tuneable parameters
1647 32 float1 (=1.0)
1648 32 zero ?
1649 32 float2 (=1.0)
1650 32 float3 (=1.0)
1651 32 unknown (27)
1652 32 unknown (8)
1653 32 zero ?
1654 */
1655
1656 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1657 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1658 return AVERROR_INVALIDDATA;
1659 }
1660
1661 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1662
1663 while (bytestream2_get_bytes_left(&gb) > 8) {
1664 if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1665 (uint64_t)MKBETAG('Q','D','M','2')))
1666 break;
1667 bytestream2_skip(&gb, 1);
1668 }
1669
1670 if (bytestream2_get_bytes_left(&gb) < 12) {
1671 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1672 bytestream2_get_bytes_left(&gb));
1673 return AVERROR_INVALIDDATA;
1674 }
1675
1676 bytestream2_skip(&gb, 8);
1677 size = bytestream2_get_be32(&gb);
1678
1679 if (size > bytestream2_get_bytes_left(&gb)) {
1680 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1681 bytestream2_get_bytes_left(&gb), size);
1682 return AVERROR_INVALIDDATA;
1683 }
1684
1685 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1686 if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1687 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1688 return AVERROR_INVALIDDATA;
1689 }
1690
1691 bytestream2_skip(&gb, 4);
1692
1693 s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1694 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1695 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1696 return AVERROR_INVALIDDATA;
1697 }
1698 av_channel_layout_uninit(&avctx->ch_layout);
1699 av_channel_layout_default(&avctx->ch_layout, s->channels);
1700
1701 avctx->sample_rate = bytestream2_get_be32(&gb);
1702 avctx->bit_rate = bytestream2_get_be32(&gb);
1703 s->group_size = bytestream2_get_be32(&gb);
1704 s->fft_size = bytestream2_get_be32(&gb);
1705 s->checksum_size = bytestream2_get_be32(&gb);
1706 if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
1707 av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
1708 return AVERROR_INVALIDDATA;
1709 }
1710
1711 s->fft_order = av_log2(s->fft_size) + 1;
1712
1713 // Fail on unknown fft order
1714 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1715 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1716 return AVERROR_PATCHWELCOME;
1717 }
1718
1719 // something like max decodable tones
1720 s->group_order = av_log2(s->group_size) + 1;
1721 s->frame_size = s->group_size / 16; // 16 iterations per super block
1722
1723 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1724 return AVERROR_INVALIDDATA;
1725
1726 s->sub_sampling = s->fft_order - 7;
1727 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1728
1729 if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
1730 avpriv_request_sample(avctx, "large frames");
1731 return AVERROR_PATCHWELCOME;
1732 }
1733
1734 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1735 case 0: tmp = 40; break;
1736 case 1: tmp = 48; break;
1737 case 2: tmp = 56; break;
1738 case 3: tmp = 72; break;
1739 case 4: tmp = 80; break;
1740 case 5: tmp = 100;break;
1741 default: tmp=s->sub_sampling; break;
1742 }
1743 tmp_val = 0;
1744 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1745 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1746 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1747 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1748 s->cm_table_select = tmp_val;
1749
1750 if (avctx->bit_rate <= 8000)
1751 s->coeff_per_sb_select = 0;
1752 else if (avctx->bit_rate < 16000)
1753 s->coeff_per_sb_select = 1;
1754 else
1755 s->coeff_per_sb_select = 2;
1756
1757 if (s->fft_size != (1 << (s->fft_order - 1))) {
1758 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1759 return AVERROR_INVALIDDATA;
1760 }
1761
1762 ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0);
1763 if (ret < 0)
1764 return ret;
1765
1766 ff_mpadsp_init(&s->mpadsp);
1767
1768 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1769
1770 ff_thread_once(&init_static_once, qdm2_init_static_data);
1771
1772 return 0;
1773 }
1774
1775 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1776 {
1777 QDM2Context *s = avctx->priv_data;
1778
1779 av_tx_uninit(&s->rdft_ctx);
1780
1781 return 0;
1782 }
1783
1784 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1785 {
1786 int ch, i;
1787 const int frame_size = (q->frame_size * q->channels);
1788
1789 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1790 return -1;
1791
1792 /* select input buffer */
1793 q->compressed_data = in;
1794 q->compressed_size = q->checksum_size;
1795
1796 /* copy old block, clear new block of output samples */
1797 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1798 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1799
1800 /* decode block of QDM2 compressed data */
1801 if (q->sub_packet == 0) {
1802 q->has_errors = 0; // zero it for a new super block
1803 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1804 qdm2_decode_super_block(q);
1805 }
1806
1807 /* parse subpackets */
1808 if (!q->has_errors) {
1809 if (q->sub_packet == 2)
1810 qdm2_decode_fft_packets(q);
1811
1812 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1813 }
1814
1815 /* sound synthesis stage 1 (FFT) */
1816 for (ch = 0; ch < q->channels; ch++) {
1817 qdm2_calculate_fft(q, ch, q->sub_packet);
1818
1819 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1820 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1821 return -1;
1822 }
1823 }
1824
1825 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1826 if (!q->has_errors && q->do_synth_filter)
1827 qdm2_synthesis_filter(q, q->sub_packet);
1828
1829 q->sub_packet = (q->sub_packet + 1) % 16;
1830
1831 /* clip and convert output float[] to 16-bit signed samples */
1832 for (i = 0; i < frame_size; i++) {
1833 int value = (int)q->output_buffer[i];
1834
1835 if (value > SOFTCLIP_THRESHOLD)
1836 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1837 else if (value < -SOFTCLIP_THRESHOLD)
1838 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1839
1840 out[i] = value;
1841 }
1842
1843 return 0;
1844 }
1845
1846 static int qdm2_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1847 int *got_frame_ptr, AVPacket *avpkt)
1848 {
1849 const uint8_t *buf = avpkt->data;
1850 int buf_size = avpkt->size;
1851 QDM2Context *s = avctx->priv_data;
1852 int16_t *out;
1853 int i, ret;
1854
1855 if(!buf)
1856 return 0;
1857 if(buf_size < s->checksum_size)
1858 return -1;
1859
1860 /* get output buffer */
1861 frame->nb_samples = 16 * s->frame_size;
1862 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1863 return ret;
1864 out = (int16_t *)frame->data[0];
1865
1866 for (i = 0; i < 16; i++) {
1867 if ((ret = qdm2_decode(s, buf, out)) < 0)
1868 return ret;
1869 out += s->channels * s->frame_size;
1870 }
1871
1872 *got_frame_ptr = 1;
1873
1874 return s->checksum_size;
1875 }
1876
1877 const FFCodec ff_qdm2_decoder = {
1878 .p.name = "qdm2",
1879 CODEC_LONG_NAME("QDesign Music Codec 2"),
1880 .p.type = AVMEDIA_TYPE_AUDIO,
1881 .p.id = AV_CODEC_ID_QDM2,
1882 .priv_data_size = sizeof(QDM2Context),
1883 .init = qdm2_decode_init,
1884 .close = qdm2_decode_close,
1885 FF_CODEC_DECODE_CB(qdm2_decode_frame),
1886 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1887 };