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1/*
2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
7 *
8 * This file is part of FFmpeg.
9 *
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
14 *
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
19 *
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 */
24
25/**
26 * @file
27 * QDM2 decoder
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29 *
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
32 */
33
34#include <math.h>
35#include <stddef.h>
36#include <stdio.h>
37
38#include "libavutil/channel_layout.h"
39
40#define BITSTREAM_READER_LE
41#include "avcodec.h"
42#include "get_bits.h"
43#include "bytestream.h"
44#include "internal.h"
45#include "mpegaudio.h"
46#include "mpegaudiodsp.h"
47#include "rdft.h"
48
49#include "qdm2_tablegen.h"
50
51#define QDM2_LIST_ADD(list, size, packet) \
52do { \
53 if (size > 0) { \
54 list[size - 1].next = &list[size]; \
55 } \
56 list[size].packet = packet; \
57 list[size].next = NULL; \
58 size++; \
59} while(0)
60
61// Result is 8, 16 or 30
62#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
63
64#define FIX_NOISE_IDX(noise_idx) \
65 if ((noise_idx) >= 3840) \
66 (noise_idx) -= 3840; \
67
68#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
69
70#define SAMPLES_NEEDED \
71 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
72
73#define SAMPLES_NEEDED_2(why) \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
75
76#define QDM2_MAX_FRAME_SIZE 512
77
78typedef int8_t sb_int8_array[2][30][64];
79
80/**
81 * Subpacket
82 */
83typedef struct QDM2SubPacket {
84 int type; ///< subpacket type
85 unsigned int size; ///< subpacket size
86 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
87} QDM2SubPacket;
88
89/**
90 * A node in the subpacket list
91 */
92typedef struct QDM2SubPNode {
93 QDM2SubPacket *packet; ///< packet
94 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
95} QDM2SubPNode;
96
97typedef struct QDM2Complex {
98 float re;
99 float im;
100} QDM2Complex;
101
102typedef struct FFTTone {
103 float level;
104 QDM2Complex *complex;
105 const float *table;
106 int phase;
107 int phase_shift;
108 int duration;
109 short time_index;
110 short cutoff;
111} FFTTone;
112
113typedef struct FFTCoefficient {
114 int16_t sub_packet;
115 uint8_t channel;
116 int16_t offset;
117 int16_t exp;
118 uint8_t phase;
119} FFTCoefficient;
120
121typedef struct QDM2FFT {
122 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
123} QDM2FFT;
124
125/**
126 * QDM2 decoder context
127 */
128typedef struct QDM2Context {
129 /// Parameters from codec header, do not change during playback
130 int nb_channels; ///< number of channels
131 int channels; ///< number of channels
132 int group_size; ///< size of frame group (16 frames per group)
133 int fft_size; ///< size of FFT, in complex numbers
134 int checksum_size; ///< size of data block, used also for checksum
135
136 /// Parameters built from header parameters, do not change during playback
137 int group_order; ///< order of frame group
138 int fft_order; ///< order of FFT (actually fftorder+1)
139 int frame_size; ///< size of data frame
140 int frequency_range;
141 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
142 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
143 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
144
145 /// Packets and packet lists
146 QDM2SubPacket sub_packets[16]; ///< the packets themselves
147 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
148 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
149 int sub_packets_B; ///< number of packets on 'B' list
150 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
151 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
152
153 /// FFT and tones
154 FFTTone fft_tones[1000];
155 int fft_tone_start;
156 int fft_tone_end;
157 FFTCoefficient fft_coefs[1000];
158 int fft_coefs_index;
159 int fft_coefs_min_index[5];
160 int fft_coefs_max_index[5];
161 int fft_level_exp[6];
162 RDFTContext rdft_ctx;
163 QDM2FFT fft;
164
165 /// I/O data
166 const uint8_t *compressed_data;
167 int compressed_size;
168 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
169
170 /// Synthesis filter
171 MPADSPContext mpadsp;
172 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
173 int synth_buf_offset[MPA_MAX_CHANNELS];
174 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
175 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
176
177 /// Mixed temporary data used in decoding
178 float tone_level[MPA_MAX_CHANNELS][30][64];
179 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
180 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
181 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
182 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
183 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
184 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
185 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
186 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
187
188 // Flags
189 int has_errors; ///< packet has errors
190 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
191 int do_synth_filter; ///< used to perform or skip synthesis filter
192
193 int sub_packet;
194 int noise_idx; ///< index for dithering noise table
195} QDM2Context;
196
197static const int switchtable[23] = {
198 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
199};
200
201static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
202{
203 int value;
204
205 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
206
207 /* stage-2, 3 bits exponent escape sequence */
208 if (value-- == 0)
209 value = get_bits(gb, get_bits(gb, 3) + 1);
210
211 /* stage-3, optional */
212 if (flag) {
213 int tmp;
214
215 if (value >= 60) {
216 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
217 return 0;
218 }
219
220 tmp= vlc_stage3_values[value];
221
222 if ((value & ~3) > 0)
223 tmp += get_bits(gb, (value >> 2));
224 value = tmp;
225 }
226
227 return value;
228}
229
230static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
231{
232 int value = qdm2_get_vlc(gb, vlc, 0, depth);
233
234 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
235}
236
237/**
238 * QDM2 checksum
239 *
240 * @param data pointer to data to be checksummed
241 * @param length data length
242 * @param value checksum value
243 *
244 * @return 0 if checksum is OK
245 */
246static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
247{
248 int i;
249
250 for (i = 0; i < length; i++)
251 value -= data[i];
252
253 return (uint16_t)(value & 0xffff);
254}
255
256/**
257 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
258 *
259 * @param gb bitreader context
260 * @param sub_packet packet under analysis
261 */
262static void qdm2_decode_sub_packet_header(GetBitContext *gb,
263 QDM2SubPacket *sub_packet)
264{
265 sub_packet->type = get_bits(gb, 8);
266
267 if (sub_packet->type == 0) {
268 sub_packet->size = 0;
269 sub_packet->data = NULL;
270 } else {
271 sub_packet->size = get_bits(gb, 8);
272
273 if (sub_packet->type & 0x80) {
274 sub_packet->size <<= 8;
275 sub_packet->size |= get_bits(gb, 8);
276 sub_packet->type &= 0x7f;
277 }
278
279 if (sub_packet->type == 0x7f)
280 sub_packet->type |= (get_bits(gb, 8) << 8);
281
282 // FIXME: this depends on bitreader-internal data
283 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
284 }
285
286 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
287 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
288}
289
290/**
291 * Return node pointer to first packet of requested type in list.
292 *
293 * @param list list of subpackets to be scanned
294 * @param type type of searched subpacket
295 * @return node pointer for subpacket if found, else NULL
296 */
297static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
298 int type)
299{
300 while (list && list->packet) {
301 if (list->packet->type == type)
302 return list;
303 list = list->next;
304 }
305 return NULL;
306}
307
308/**
309 * Replace 8 elements with their average value.
310 * Called by qdm2_decode_superblock before starting subblock decoding.
311 *
312 * @param q context
313 */
314static void average_quantized_coeffs(QDM2Context *q)
315{
316 int i, j, n, ch, sum;
317
318 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
319
320 for (ch = 0; ch < q->nb_channels; ch++)
321 for (i = 0; i < n; i++) {
322 sum = 0;
323
324 for (j = 0; j < 8; j++)
325 sum += q->quantized_coeffs[ch][i][j];
326
327 sum /= 8;
328 if (sum > 0)
329 sum--;
330
331 for (j = 0; j < 8; j++)
332 q->quantized_coeffs[ch][i][j] = sum;
333 }
334}
335
336/**
337 * Build subband samples with noise weighted by q->tone_level.
338 * Called by synthfilt_build_sb_samples.
339 *
340 * @param q context
341 * @param sb subband index
342 */
343static void build_sb_samples_from_noise(QDM2Context *q, int sb)
344{
345 int ch, j;
346
347 FIX_NOISE_IDX(q->noise_idx);
348
349 if (!q->nb_channels)
350 return;
351
352 for (ch = 0; ch < q->nb_channels; ch++) {
353 for (j = 0; j < 64; j++) {
354 q->sb_samples[ch][j * 2][sb] =
355 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
356 q->sb_samples[ch][j * 2 + 1][sb] =
357 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
358 }
359 }
360}
361
362/**
363 * Called while processing data from subpackets 11 and 12.
364 * Used after making changes to coding_method array.
365 *
366 * @param sb subband index
367 * @param channels number of channels
368 * @param coding_method q->coding_method[0][0][0]
369 */
370static int fix_coding_method_array(int sb, int channels,
371 sb_int8_array coding_method)
372{
373 int j, k;
374 int ch;
375 int run, case_val;
376
377 for (ch = 0; ch < channels; ch++) {
378 for (j = 0; j < 64; ) {
379 if (coding_method[ch][sb][j] < 8)
380 return -1;
381 if ((coding_method[ch][sb][j] - 8) > 22) {
382 run = 1;
383 case_val = 8;
384 } else {
385 switch (switchtable[coding_method[ch][sb][j] - 8]) {
386 case 0: run = 10;
387 case_val = 10;
388 break;
389 case 1: run = 1;
390 case_val = 16;
391 break;
392 case 2: run = 5;
393 case_val = 24;
394 break;
395 case 3: run = 3;
396 case_val = 30;
397 break;
398 case 4: run = 1;
399 case_val = 30;
400 break;
401 case 5: run = 1;
402 case_val = 8;
403 break;
404 default: run = 1;
405 case_val = 8;
406 break;
407 }
408 }
409 for (k = 0; k < run; k++) {
410 if (j + k < 128) {
411 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j]) {
412 if (k > 0) {
413 SAMPLES_NEEDED
414 //not debugged, almost never used
415 memset(&coding_method[ch][sb][j + k], case_val,
416 k *sizeof(int8_t));
417 memset(&coding_method[ch][sb][j + k], case_val,
418 3 * sizeof(int8_t));
419 }
420 }
421 }
422 }
423 j += run;
424 }
425 }
426 return 0;
427}
428
429/**
430 * Related to synthesis filter
431 * Called by process_subpacket_10
432 *
433 * @param q context
434 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
435 */
436static void fill_tone_level_array(QDM2Context *q, int flag)
437{
438 int i, sb, ch, sb_used;
439 int tmp, tab;
440
441 for (ch = 0; ch < q->nb_channels; ch++)
442 for (sb = 0; sb < 30; sb++)
443 for (i = 0; i < 8; i++) {
444 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
445 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
446 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
447 else
448 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
449 if(tmp < 0)
450 tmp += 0xff;
451 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
452 }
453
454 sb_used = QDM2_SB_USED(q->sub_sampling);
455
456 if ((q->superblocktype_2_3 != 0) && !flag) {
457 for (sb = 0; sb < sb_used; sb++)
458 for (ch = 0; ch < q->nb_channels; ch++)
459 for (i = 0; i < 64; i++) {
460 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
461 if (q->tone_level_idx[ch][sb][i] < 0)
462 q->tone_level[ch][sb][i] = 0;
463 else
464 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
465 }
466 } else {
467 tab = q->superblocktype_2_3 ? 0 : 1;
468 for (sb = 0; sb < sb_used; sb++) {
469 if ((sb >= 4) && (sb <= 23)) {
470 for (ch = 0; ch < q->nb_channels; ch++)
471 for (i = 0; i < 64; i++) {
472 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
473 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
474 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
475 q->tone_level_idx_hi2[ch][sb - 4];
476 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
477 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
478 q->tone_level[ch][sb][i] = 0;
479 else
480 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
481 }
482 } else {
483 if (sb > 4) {
484 for (ch = 0; ch < q->nb_channels; ch++)
485 for (i = 0; i < 64; i++) {
486 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
487 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
488 q->tone_level_idx_hi2[ch][sb - 4];
489 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
490 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
491 q->tone_level[ch][sb][i] = 0;
492 else
493 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
494 }
495 } else {
496 for (ch = 0; ch < q->nb_channels; ch++)
497 for (i = 0; i < 64; i++) {
498 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
499 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
500 q->tone_level[ch][sb][i] = 0;
501 else
502 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
503 }
504 }
505 }
506 }
507 }
508}
509
510/**
511 * Related to synthesis filter
512 * Called by process_subpacket_11
513 * c is built with data from subpacket 11
514 * Most of this function is used only if superblock_type_2_3 == 0,
515 * never seen it in samples.
516 *
517 * @param tone_level_idx
518 * @param tone_level_idx_temp
519 * @param coding_method q->coding_method[0][0][0]
520 * @param nb_channels number of channels
521 * @param c coming from subpacket 11, passed as 8*c
522 * @param superblocktype_2_3 flag based on superblock packet type
523 * @param cm_table_select q->cm_table_select
524 */
525static void fill_coding_method_array(sb_int8_array tone_level_idx,
526 sb_int8_array tone_level_idx_temp,
527 sb_int8_array coding_method,
528 int nb_channels,
529 int c, int superblocktype_2_3,
530 int cm_table_select)
531{
532 int ch, sb, j;
533 int tmp, acc, esp_40, comp;
534 int add1, add2, add3, add4;
535 int64_t multres;
536
537 if (!superblocktype_2_3) {
538 /* This case is untested, no samples available */
539 avpriv_request_sample(NULL, "!superblocktype_2_3");
540 return;
541 for (ch = 0; ch < nb_channels; ch++) {
542 for (sb = 0; sb < 30; sb++) {
543 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
544 add1 = tone_level_idx[ch][sb][j] - 10;
545 if (add1 < 0)
546 add1 = 0;
547 add2 = add3 = add4 = 0;
548 if (sb > 1) {
549 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
550 if (add2 < 0)
551 add2 = 0;
552 }
553 if (sb > 0) {
554 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
555 if (add3 < 0)
556 add3 = 0;
557 }
558 if (sb < 29) {
559 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
560 if (add4 < 0)
561 add4 = 0;
562 }
563 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
564 if (tmp < 0)
565 tmp = 0;
566 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
567 }
568 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
569 }
570 }
571 acc = 0;
572 for (ch = 0; ch < nb_channels; ch++)
573 for (sb = 0; sb < 30; sb++)
574 for (j = 0; j < 64; j++)
575 acc += tone_level_idx_temp[ch][sb][j];
576
577 multres = 0x66666667LL * (acc * 10);
578 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
579 for (ch = 0; ch < nb_channels; ch++)
580 for (sb = 0; sb < 30; sb++)
581 for (j = 0; j < 64; j++) {
582 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
583 if (comp < 0)
584 comp += 0xff;
585 comp /= 256; // signed shift
586 switch(sb) {
587 case 0:
588 if (comp < 30)
589 comp = 30;
590 comp += 15;
591 break;
592 case 1:
593 if (comp < 24)
594 comp = 24;
595 comp += 10;
596 break;
597 case 2:
598 case 3:
599 case 4:
600 if (comp < 16)
601 comp = 16;
602 }
603 if (comp <= 5)
604 tmp = 0;
605 else if (comp <= 10)
606 tmp = 10;
607 else if (comp <= 16)
608 tmp = 16;
609 else if (comp <= 24)
610 tmp = -1;
611 else
612 tmp = 0;
613 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
614 }
615 for (sb = 0; sb < 30; sb++)
616 fix_coding_method_array(sb, nb_channels, coding_method);
617 for (ch = 0; ch < nb_channels; ch++)
618 for (sb = 0; sb < 30; sb++)
619 for (j = 0; j < 64; j++)
620 if (sb >= 10) {
621 if (coding_method[ch][sb][j] < 10)
622 coding_method[ch][sb][j] = 10;
623 } else {
624 if (sb >= 2) {
625 if (coding_method[ch][sb][j] < 16)
626 coding_method[ch][sb][j] = 16;
627 } else {
628 if (coding_method[ch][sb][j] < 30)
629 coding_method[ch][sb][j] = 30;
630 }
631 }
632 } else { // superblocktype_2_3 != 0
633 for (ch = 0; ch < nb_channels; ch++)
634 for (sb = 0; sb < 30; sb++)
635 for (j = 0; j < 64; j++)
636 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
637 }
638}
639
640/**
641 * Called by process_subpacket_11 to process more data from subpacket 11
642 * with sb 0-8.
643 * Called by process_subpacket_12 to process data from subpacket 12 with
644 * sb 8-sb_used.
645 *
646 * @param q context
647 * @param gb bitreader context
648 * @param length packet length in bits
649 * @param sb_min lower subband processed (sb_min included)
650 * @param sb_max higher subband processed (sb_max excluded)
651 */
652static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
653 int length, int sb_min, int sb_max)
654{
655 int sb, j, k, n, ch, run, channels;
656 int joined_stereo, zero_encoding;
657 int type34_first;
658 float type34_div = 0;
659 float type34_predictor;
660 float samples[10];
661 int sign_bits[16] = {0};
662
663 if (length == 0) {
664 // If no data use noise
665 for (sb=sb_min; sb < sb_max; sb++)
666 build_sb_samples_from_noise(q, sb);
667
668 return 0;
669 }
670
671 for (sb = sb_min; sb < sb_max; sb++) {
672 channels = q->nb_channels;
673
674 if (q->nb_channels <= 1 || sb < 12)
675 joined_stereo = 0;
676 else if (sb >= 24)
677 joined_stereo = 1;
678 else
679 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
680
681 if (joined_stereo) {
682 if (get_bits_left(gb) >= 16)
683 for (j = 0; j < 16; j++)
684 sign_bits[j] = get_bits1(gb);
685
686 for (j = 0; j < 64; j++)
687 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
688 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
689
690 if (fix_coding_method_array(sb, q->nb_channels,
691 q->coding_method)) {
692 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
693 build_sb_samples_from_noise(q, sb);
694 continue;
695 }
696 channels = 1;
697 }
698
699 for (ch = 0; ch < channels; ch++) {
700 FIX_NOISE_IDX(q->noise_idx);
701 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
702 type34_predictor = 0.0;
703 type34_first = 1;
704
705 for (j = 0; j < 128; ) {
706 switch (q->coding_method[ch][sb][j / 2]) {
707 case 8:
708 if (get_bits_left(gb) >= 10) {
709 if (zero_encoding) {
710 for (k = 0; k < 5; k++) {
711 if ((j + 2 * k) >= 128)
712 break;
713 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
714 }
715 } else {
716 n = get_bits(gb, 8);
717 if (n >= 243) {
718 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
719 return AVERROR_INVALIDDATA;
720 }
721
722 for (k = 0; k < 5; k++)
723 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
724 }
725 for (k = 0; k < 5; k++)
726 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
727 } else {
728 for (k = 0; k < 10; k++)
729 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
730 }
731 run = 10;
732 break;
733
734 case 10:
735 if (get_bits_left(gb) >= 1) {
736 float f = 0.81;
737
738 if (get_bits1(gb))
739 f = -f;
740 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
741 samples[0] = f;
742 } else {
743 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
744 }
745 run = 1;
746 break;
747
748 case 16:
749 if (get_bits_left(gb) >= 10) {
750 if (zero_encoding) {
751 for (k = 0; k < 5; k++) {
752 if ((j + k) >= 128)
753 break;
754 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
755 }
756 } else {
757 n = get_bits (gb, 8);
758 if (n >= 243) {
759 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
760 return AVERROR_INVALIDDATA;
761 }
762
763 for (k = 0; k < 5; k++)
764 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
765 }
766 } else {
767 for (k = 0; k < 5; k++)
768 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
769 }
770 run = 5;
771 break;
772
773 case 24:
774 if (get_bits_left(gb) >= 7) {
775 n = get_bits(gb, 7);
776 if (n >= 125) {
777 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
778 return AVERROR_INVALIDDATA;
779 }
780
781 for (k = 0; k < 3; k++)
782 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
783 } else {
784 for (k = 0; k < 3; k++)
785 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
786 }
787 run = 3;
788 break;
789
790 case 30:
791 if (get_bits_left(gb) >= 4) {
792 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
793 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
794 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
795 return AVERROR_INVALIDDATA;
796 }
797 samples[0] = type30_dequant[index];
798 } else
799 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
800
801 run = 1;
802 break;
803
804 case 34:
805 if (get_bits_left(gb) >= 7) {
806 if (type34_first) {
807 type34_div = (float)(1 << get_bits(gb, 2));
808 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
809 type34_predictor = samples[0];
810 type34_first = 0;
811 } else {
812 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
813 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
814 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
815 return AVERROR_INVALIDDATA;
816 }
817 samples[0] = type34_delta[index] / type34_div + type34_predictor;
818 type34_predictor = samples[0];
819 }
820 } else {
821 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
822 }
823 run = 1;
824 break;
825
826 default:
827 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
828 run = 1;
829 break;
830 }
831
832 if (joined_stereo) {
833 for (k = 0; k < run && j + k < 128; k++) {
834 q->sb_samples[0][j + k][sb] =
835 q->tone_level[0][sb][(j + k) / 2] * samples[k];
836 if (q->nb_channels == 2) {
837 if (sign_bits[(j + k) / 8])
838 q->sb_samples[1][j + k][sb] =
839 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
840 else
841 q->sb_samples[1][j + k][sb] =
842 q->tone_level[1][sb][(j + k) / 2] * samples[k];
843 }
844 }
845 } else {
846 for (k = 0; k < run; k++)
847 if ((j + k) < 128)
848 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
849 }
850
851 j += run;
852 } // j loop
853 } // channel loop
854 } // subband loop
855 return 0;
856}
857
858/**
859 * Init the first element of a channel in quantized_coeffs with data
860 * from packet 10 (quantized_coeffs[ch][0]).
861 * This is similar to process_subpacket_9, but for a single channel
862 * and for element [0]
863 * same VLC tables as process_subpacket_9 are used.
864 *
865 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
866 * @param gb bitreader context
867 */
868static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
869 GetBitContext *gb)
870{
871 int i, k, run, level, diff;
872
873 if (get_bits_left(gb) < 16)
874 return -1;
875 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
876
877 quantized_coeffs[0] = level;
878
879 for (i = 0; i < 7; ) {
880 if (get_bits_left(gb) < 16)
881 return -1;
882 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
883
884 if (i + run >= 8)
885 return -1;
886
887 if (get_bits_left(gb) < 16)
888 return -1;
889 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
890
891 for (k = 1; k <= run; k++)
892 quantized_coeffs[i + k] = (level + ((k * diff) / run));
893
894 level += diff;
895 i += run;
896 }
897 return 0;
898}
899
900/**
901 * Related to synthesis filter, process data from packet 10
902 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
903 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
904 * data from packet 10
905 *
906 * @param q context
907 * @param gb bitreader context
908 */
909static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
910{
911 int sb, j, k, n, ch;
912
913 for (ch = 0; ch < q->nb_channels; ch++) {
914 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
915
916 if (get_bits_left(gb) < 16) {
917 memset(q->quantized_coeffs[ch][0], 0, 8);
918 break;
919 }
920 }
921
922 n = q->sub_sampling + 1;
923
924 for (sb = 0; sb < n; sb++)
925 for (ch = 0; ch < q->nb_channels; ch++)
926 for (j = 0; j < 8; j++) {
927 if (get_bits_left(gb) < 1)
928 break;
929 if (get_bits1(gb)) {
930 for (k=0; k < 8; k++) {
931 if (get_bits_left(gb) < 16)
932 break;
933 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
934 }
935 } else {
936 for (k=0; k < 8; k++)
937 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
938 }
939 }
940
941 n = QDM2_SB_USED(q->sub_sampling) - 4;
942
943 for (sb = 0; sb < n; sb++)
944 for (ch = 0; ch < q->nb_channels; ch++) {
945 if (get_bits_left(gb) < 16)
946 break;
947 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
948 if (sb > 19)
949 q->tone_level_idx_hi2[ch][sb] -= 16;
950 else
951 for (j = 0; j < 8; j++)
952 q->tone_level_idx_mid[ch][sb][j] = -16;
953 }
954
955 n = QDM2_SB_USED(q->sub_sampling) - 5;
956
957 for (sb = 0; sb < n; sb++)
958 for (ch = 0; ch < q->nb_channels; ch++)
959 for (j = 0; j < 8; j++) {
960 if (get_bits_left(gb) < 16)
961 break;
962 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
963 }
964}
965
966/**
967 * Process subpacket 9, init quantized_coeffs with data from it
968 *
969 * @param q context
970 * @param node pointer to node with packet
971 */
972static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
973{
974 GetBitContext gb;
975 int i, j, k, n, ch, run, level, diff;
976
977 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
978
979 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
980
981 for (i = 1; i < n; i++)
982 for (ch = 0; ch < q->nb_channels; ch++) {
983 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
984 q->quantized_coeffs[ch][i][0] = level;
985
986 for (j = 0; j < (8 - 1); ) {
987 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
988 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
989
990 if (j + run >= 8)
991 return -1;
992
993 for (k = 1; k <= run; k++)
994 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
995
996 level += diff;
997 j += run;
998 }
999 }
1000
1001 for (ch = 0; ch < q->nb_channels; ch++)
1002 for (i = 0; i < 8; i++)
1003 q->quantized_coeffs[ch][0][i] = 0;
1004
1005 return 0;
1006}
1007
1008/**
1009 * Process subpacket 10 if not null, else
1010 *
1011 * @param q context
1012 * @param node pointer to node with packet
1013 */
1014static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1015{
1016 GetBitContext gb;
1017
1018 if (node) {
1019 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1020 init_tone_level_dequantization(q, &gb);
1021 fill_tone_level_array(q, 1);
1022 } else {
1023 fill_tone_level_array(q, 0);
1024 }
1025}
1026
1027/**
1028 * Process subpacket 11
1029 *
1030 * @param q context
1031 * @param node pointer to node with packet
1032 */
1033static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1034{
1035 GetBitContext gb;
1036 int length = 0;
1037
1038 if (node) {
1039 length = node->packet->size * 8;
1040 init_get_bits(&gb, node->packet->data, length);
1041 }
1042
1043 if (length >= 32) {
1044 int c = get_bits(&gb, 13);
1045
1046 if (c > 3)
1047 fill_coding_method_array(q->tone_level_idx,
1048 q->tone_level_idx_temp, q->coding_method,
1049 q->nb_channels, 8 * c,
1050 q->superblocktype_2_3, q->cm_table_select);
1051 }
1052
1053 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1054}
1055
1056/**
1057 * Process subpacket 12
1058 *
1059 * @param q context
1060 * @param node pointer to node with packet
1061 */
1062static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1063{
1064 GetBitContext gb;
1065 int length = 0;
1066
1067 if (node) {
1068 length = node->packet->size * 8;
1069 init_get_bits(&gb, node->packet->data, length);
1070 }
1071
1072 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1073}
1074
1075/**
1076 * Process new subpackets for synthesis filter
1077 *
1078 * @param q context
1079 * @param list list with synthesis filter packets (list D)
1080 */
1081static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1082{
1083 QDM2SubPNode *nodes[4];
1084
1085 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1086 if (nodes[0])
1087 process_subpacket_9(q, nodes[0]);
1088
1089 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1090 if (nodes[1])
1091 process_subpacket_10(q, nodes[1]);
1092 else
1093 process_subpacket_10(q, NULL);
1094
1095 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1096 if (nodes[0] && nodes[1] && nodes[2])
1097 process_subpacket_11(q, nodes[2]);
1098 else
1099 process_subpacket_11(q, NULL);
1100
1101 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1102 if (nodes[0] && nodes[1] && nodes[3])
1103 process_subpacket_12(q, nodes[3]);
1104 else
1105 process_subpacket_12(q, NULL);
1106}
1107
1108/**
1109 * Decode superblock, fill packet lists.
1110 *
1111 * @param q context
1112 */
1113static void qdm2_decode_super_block(QDM2Context *q)
1114{
1115 GetBitContext gb;
1116 QDM2SubPacket header, *packet;
1117 int i, packet_bytes, sub_packet_size, sub_packets_D;
1118 unsigned int next_index = 0;
1119
1120 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1121 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1122 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1123
1124 q->sub_packets_B = 0;
1125 sub_packets_D = 0;
1126
1127 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1128
1129 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1130 qdm2_decode_sub_packet_header(&gb, &header);
1131
1132 if (header.type < 2 || header.type >= 8) {
1133 q->has_errors = 1;
1134 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1135 return;
1136 }
1137
1138 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1139 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1140
1141 init_get_bits(&gb, header.data, header.size * 8);
1142
1143 if (header.type == 2 || header.type == 4 || header.type == 5) {
1144 int csum = 257 * get_bits(&gb, 8);
1145 csum += 2 * get_bits(&gb, 8);
1146
1147 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1148
1149 if (csum != 0) {
1150 q->has_errors = 1;
1151 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1152 return;
1153 }
1154 }
1155
1156 q->sub_packet_list_B[0].packet = NULL;
1157 q->sub_packet_list_D[0].packet = NULL;
1158
1159 for (i = 0; i < 6; i++)
1160 if (--q->fft_level_exp[i] < 0)
1161 q->fft_level_exp[i] = 0;
1162
1163 for (i = 0; packet_bytes > 0; i++) {
1164 int j;
1165
1166 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1167 SAMPLES_NEEDED_2("too many packet bytes");
1168 return;
1169 }
1170
1171 q->sub_packet_list_A[i].next = NULL;
1172
1173 if (i > 0) {
1174 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1175
1176 /* seek to next block */
1177 init_get_bits(&gb, header.data, header.size * 8);
1178 skip_bits(&gb, next_index * 8);
1179
1180 if (next_index >= header.size)
1181 break;
1182 }
1183
1184 /* decode subpacket */
1185 packet = &q->sub_packets[i];
1186 qdm2_decode_sub_packet_header(&gb, packet);
1187 next_index = packet->size + get_bits_count(&gb) / 8;
1188 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1189
1190 if (packet->type == 0)
1191 break;
1192
1193 if (sub_packet_size > packet_bytes) {
1194 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1195 break;
1196 packet->size += packet_bytes - sub_packet_size;
1197 }
1198
1199 packet_bytes -= sub_packet_size;
1200
1201 /* add subpacket to 'all subpackets' list */
1202 q->sub_packet_list_A[i].packet = packet;
1203
1204 /* add subpacket to related list */
1205 if (packet->type == 8) {
1206 SAMPLES_NEEDED_2("packet type 8");
1207 return;
1208 } else if (packet->type >= 9 && packet->type <= 12) {
1209 /* packets for MPEG Audio like Synthesis Filter */
1210 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1211 } else if (packet->type == 13) {
1212 for (j = 0; j < 6; j++)
1213 q->fft_level_exp[j] = get_bits(&gb, 6);
1214 } else if (packet->type == 14) {
1215 for (j = 0; j < 6; j++)
1216 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1217 } else if (packet->type == 15) {
1218 SAMPLES_NEEDED_2("packet type 15")
1219 return;
1220 } else if (packet->type >= 16 && packet->type < 48 &&
1221 !fft_subpackets[packet->type - 16]) {
1222 /* packets for FFT */
1223 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1224 }
1225 } // Packet bytes loop
1226
1227 if (q->sub_packet_list_D[0].packet) {
1228 process_synthesis_subpackets(q, q->sub_packet_list_D);
1229 q->do_synth_filter = 1;
1230 } else if (q->do_synth_filter) {
1231 process_subpacket_10(q, NULL);
1232 process_subpacket_11(q, NULL);
1233 process_subpacket_12(q, NULL);
1234 }
1235}
1236
1237static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1238 int offset, int duration, int channel,
1239 int exp, int phase)
1240{
1241 if (q->fft_coefs_min_index[duration] < 0)
1242 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1243
1244 q->fft_coefs[q->fft_coefs_index].sub_packet =
1245 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1246 q->fft_coefs[q->fft_coefs_index].channel = channel;
1247 q->fft_coefs[q->fft_coefs_index].offset = offset;
1248 q->fft_coefs[q->fft_coefs_index].exp = exp;
1249 q->fft_coefs[q->fft_coefs_index].phase = phase;
1250 q->fft_coefs_index++;
1251}
1252
1253static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1254 GetBitContext *gb, int b)
1255{
1256 int channel, stereo, phase, exp;
1257 int local_int_4, local_int_8, stereo_phase, local_int_10;
1258 int local_int_14, stereo_exp, local_int_20, local_int_28;
1259 int n, offset;
1260
1261 local_int_4 = 0;
1262 local_int_28 = 0;
1263 local_int_20 = 2;
1264 local_int_8 = (4 - duration);
1265 local_int_10 = 1 << (q->group_order - duration - 1);
1266 offset = 1;
1267
1268 while (get_bits_left(gb)>0) {
1269 if (q->superblocktype_2_3) {
1270 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1271 if (get_bits_left(gb)<0) {
1272 if(local_int_4 < q->group_size)
1273 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1274 return;
1275 }
1276 offset = 1;
1277 if (n == 0) {
1278 local_int_4 += local_int_10;
1279 local_int_28 += (1 << local_int_8);
1280 } else {
1281 local_int_4 += 8 * local_int_10;
1282 local_int_28 += (8 << local_int_8);
1283 }
1284 }
1285 offset += (n - 2);
1286 } else {
1287 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1288 while (offset >= (local_int_10 - 1)) {
1289 offset += (1 - (local_int_10 - 1));
1290 local_int_4 += local_int_10;
1291 local_int_28 += (1 << local_int_8);
1292 }
1293 }
1294
1295 if (local_int_4 >= q->group_size)
1296 return;
1297
1298 local_int_14 = (offset >> local_int_8);
1299 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1300 return;
1301
1302 if (q->nb_channels > 1) {
1303 channel = get_bits1(gb);
1304 stereo = get_bits1(gb);
1305 } else {
1306 channel = 0;
1307 stereo = 0;
1308 }
1309
1310 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1311 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1312 exp = (exp < 0) ? 0 : exp;
1313
1314 phase = get_bits(gb, 3);
1315 stereo_exp = 0;
1316 stereo_phase = 0;
1317
1318 if (stereo) {
1319 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1320 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1321 if (stereo_phase < 0)
1322 stereo_phase += 8;
1323 }
1324
1325 if (q->frequency_range > (local_int_14 + 1)) {
1326 int sub_packet = (local_int_20 + local_int_28);
1327
1328 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1329 channel, exp, phase);
1330 if (stereo)
1331 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1332 1 - channel,
1333 stereo_exp, stereo_phase);
1334 }
1335 offset++;
1336 }
1337}
1338
1339static void qdm2_decode_fft_packets(QDM2Context *q)
1340{
1341 int i, j, min, max, value, type, unknown_flag;
1342 GetBitContext gb;
1343
1344 if (!q->sub_packet_list_B[0].packet)
1345 return;
1346
1347 /* reset minimum indexes for FFT coefficients */
1348 q->fft_coefs_index = 0;
1349 for (i = 0; i < 5; i++)
1350 q->fft_coefs_min_index[i] = -1;
1351
1352 /* process subpackets ordered by type, largest type first */
1353 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1354 QDM2SubPacket *packet = NULL;
1355
1356 /* find subpacket with largest type less than max */
1357 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1358 value = q->sub_packet_list_B[j].packet->type;
1359 if (value > min && value < max) {
1360 min = value;
1361 packet = q->sub_packet_list_B[j].packet;
1362 }
1363 }
1364
1365 max = min;
1366
1367 /* check for errors (?) */
1368 if (!packet)
1369 return;
1370
1371 if (i == 0 &&
1372 (packet->type < 16 || packet->type >= 48 ||
1373 fft_subpackets[packet->type - 16]))
1374 return;
1375
1376 /* decode FFT tones */
1377 init_get_bits(&gb, packet->data, packet->size * 8);
1378
1379 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1380 unknown_flag = 1;
1381 else
1382 unknown_flag = 0;
1383
1384 type = packet->type;
1385
1386 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1387 int duration = q->sub_sampling + 5 - (type & 15);
1388
1389 if (duration >= 0 && duration < 4)
1390 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1391 } else if (type == 31) {
1392 for (j = 0; j < 4; j++)
1393 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1394 } else if (type == 46) {
1395 for (j = 0; j < 6; j++)
1396 q->fft_level_exp[j] = get_bits(&gb, 6);
1397 for (j = 0; j < 4; j++)
1398 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1399 }
1400 } // Loop on B packets
1401
1402 /* calculate maximum indexes for FFT coefficients */
1403 for (i = 0, j = -1; i < 5; i++)
1404 if (q->fft_coefs_min_index[i] >= 0) {
1405 if (j >= 0)
1406 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1407 j = i;
1408 }
1409 if (j >= 0)
1410 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1411}
1412
1413static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1414{
1415 float level, f[6];
1416 int i;
1417 QDM2Complex c;
1418 const double iscale = 2.0 * M_PI / 512.0;
1419
1420 tone->phase += tone->phase_shift;
1421
1422 /* calculate current level (maximum amplitude) of tone */
1423 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1424 c.im = level * sin(tone->phase * iscale);
1425 c.re = level * cos(tone->phase * iscale);
1426
1427 /* generate FFT coefficients for tone */
1428 if (tone->duration >= 3 || tone->cutoff >= 3) {
1429 tone->complex[0].im += c.im;
1430 tone->complex[0].re += c.re;
1431 tone->complex[1].im -= c.im;
1432 tone->complex[1].re -= c.re;
1433 } else {
1434 f[1] = -tone->table[4];
1435 f[0] = tone->table[3] - tone->table[0];
1436 f[2] = 1.0 - tone->table[2] - tone->table[3];
1437 f[3] = tone->table[1] + tone->table[4] - 1.0;
1438 f[4] = tone->table[0] - tone->table[1];
1439 f[5] = tone->table[2];
1440 for (i = 0; i < 2; i++) {
1441 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1442 c.re * f[i];
1443 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1444 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1445 }
1446 for (i = 0; i < 4; i++) {
1447 tone->complex[i].re += c.re * f[i + 2];
1448 tone->complex[i].im += c.im * f[i + 2];
1449 }
1450 }
1451
1452 /* copy the tone if it has not yet died out */
1453 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1454 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1455 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1456 }
1457}
1458
1459static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1460{
1461 int i, j, ch;
1462 const double iscale = 0.25 * M_PI;
1463
1464 for (ch = 0; ch < q->channels; ch++) {
1465 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1466 }
1467
1468
1469 /* apply FFT tones with duration 4 (1 FFT period) */
1470 if (q->fft_coefs_min_index[4] >= 0)
1471 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1472 float level;
1473 QDM2Complex c;
1474
1475 if (q->fft_coefs[i].sub_packet != sub_packet)
1476 break;
1477
1478 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1479 level = (q->fft_coefs[i].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[i].exp & 63];
1480
1481 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1482 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1483 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1484 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1485 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1486 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1487 }
1488
1489 /* generate existing FFT tones */
1490 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1491 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1492 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1493 }
1494
1495 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1496 for (i = 0; i < 4; i++)
1497 if (q->fft_coefs_min_index[i] >= 0) {
1498 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1499 int offset, four_i;
1500 FFTTone tone;
1501
1502 if (q->fft_coefs[j].sub_packet != sub_packet)
1503 break;
1504
1505 four_i = (4 - i);
1506 offset = q->fft_coefs[j].offset >> four_i;
1507 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1508
1509 if (offset < q->frequency_range) {
1510 if (offset < 2)
1511 tone.cutoff = offset;
1512 else
1513 tone.cutoff = (offset >= 60) ? 3 : 2;
1514
1515 tone.level = (q->fft_coefs[j].exp < 0) ? 0.0 : fft_tone_level_table[q->superblocktype_2_3 ? 0 : 1][q->fft_coefs[j].exp & 63];
1516 tone.complex = &q->fft.complex[ch][offset];
1517 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1518 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1519 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1520 tone.duration = i;
1521 tone.time_index = 0;
1522
1523 qdm2_fft_generate_tone(q, &tone);
1524 }
1525 }
1526 q->fft_coefs_min_index[i] = j;
1527 }
1528}
1529
1530static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1531{
1532 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1533 float *out = q->output_buffer + channel;
1534 int i;
1535 q->fft.complex[channel][0].re *= 2.0f;
1536 q->fft.complex[channel][0].im = 0.0f;
1537 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1538 /* add samples to output buffer */
1539 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1540 out[0] += q->fft.complex[channel][i].re * gain;
1541 out[q->channels] += q->fft.complex[channel][i].im * gain;
1542 out += 2 * q->channels;
1543 }
1544}
1545
1546/**
1547 * @param q context
1548 * @param index subpacket number
1549 */
1550static void qdm2_synthesis_filter(QDM2Context *q, int index)
1551{
1552 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1553
1554 /* copy sb_samples */
1555 sb_used = QDM2_SB_USED(q->sub_sampling);
1556
1557 for (ch = 0; ch < q->channels; ch++)
1558 for (i = 0; i < 8; i++)
1559 for (k = sb_used; k < SBLIMIT; k++)
1560 q->sb_samples[ch][(8 * index) + i][k] = 0;
1561
1562 for (ch = 0; ch < q->nb_channels; ch++) {
1563 float *samples_ptr = q->samples + ch;
1564
1565 for (i = 0; i < 8; i++) {
1566 ff_mpa_synth_filter_float(&q->mpadsp,
1567 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1568 ff_mpa_synth_window_float, &dither_state,
1569 samples_ptr, q->nb_channels,
1570 q->sb_samples[ch][(8 * index) + i]);
1571 samples_ptr += 32 * q->nb_channels;
1572 }
1573 }
1574
1575 /* add samples to output buffer */
1576 sub_sampling = (4 >> q->sub_sampling);
1577
1578 for (ch = 0; ch < q->channels; ch++)
1579 for (i = 0; i < q->frame_size; i++)
1580 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1581}
1582
1583/**
1584 * Init static data (does not depend on specific file)
1585 *
1586 * @param q context
1587 */
1588static av_cold void qdm2_init_static_data(void) {
1589 static int done;
1590
1591 if(done)
1592 return;
1593
1594 qdm2_init_vlc();
1595 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1596 softclip_table_init();
1597 rnd_table_init();
1598 init_noise_samples();
1599
1600 done = 1;
1601}
1602
1603/**
1604 * Init parameters from codec extradata
1605 */
1606static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1607{
1608 QDM2Context *s = avctx->priv_data;
1609 int tmp_val, tmp, size;
1610 GetByteContext gb;
1611
1612 qdm2_init_static_data();
1613
1614 /* extradata parsing
1615
1616 Structure:
1617 wave {
1618 frma (QDM2)
1619 QDCA
1620 QDCP
1621 }
1622
1623 32 size (including this field)
1624 32 tag (=frma)
1625 32 type (=QDM2 or QDMC)
1626
1627 32 size (including this field, in bytes)
1628 32 tag (=QDCA) // maybe mandatory parameters
1629 32 unknown (=1)
1630 32 channels (=2)
1631 32 samplerate (=44100)
1632 32 bitrate (=96000)
1633 32 block size (=4096)
1634 32 frame size (=256) (for one channel)
1635 32 packet size (=1300)
1636
1637 32 size (including this field, in bytes)
1638 32 tag (=QDCP) // maybe some tuneable parameters
1639 32 float1 (=1.0)
1640 32 zero ?
1641 32 float2 (=1.0)
1642 32 float3 (=1.0)
1643 32 unknown (27)
1644 32 unknown (8)
1645 32 zero ?
1646 */
1647
1648 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1649 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1650 return AVERROR_INVALIDDATA;
1651 }
1652
1653 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1654
1655 while (bytestream2_get_bytes_left(&gb) > 8) {
1656 if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1657 (uint64_t)MKBETAG('Q','D','M','2')))
1658 break;
1659 bytestream2_skip(&gb, 1);
1660 }
1661
1662 if (bytestream2_get_bytes_left(&gb) < 12) {
1663 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1664 bytestream2_get_bytes_left(&gb));
1665 return AVERROR_INVALIDDATA;
1666 }
1667
1668 bytestream2_skip(&gb, 8);
1669 size = bytestream2_get_be32(&gb);
1670
1671 if (size > bytestream2_get_bytes_left(&gb)) {
1672 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1673 bytestream2_get_bytes_left(&gb), size);
1674 return AVERROR_INVALIDDATA;
1675 }
1676
1677 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1678 if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1679 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1680 return AVERROR_INVALIDDATA;
1681 }
1682
1683 bytestream2_skip(&gb, 4);
1684
1685 avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1686 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1687 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1688 return AVERROR_INVALIDDATA;
1689 }
1690 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1691 AV_CH_LAYOUT_MONO;
1692
1693 avctx->sample_rate = bytestream2_get_be32(&gb);
1694 avctx->bit_rate = bytestream2_get_be32(&gb);
1695 s->group_size = bytestream2_get_be32(&gb);
1696 s->fft_size = bytestream2_get_be32(&gb);
1697 s->checksum_size = bytestream2_get_be32(&gb);
1698 if (s->checksum_size >= 1U << 28) {
1699 av_log(avctx, AV_LOG_ERROR, "data block size too large (%u)\n", s->checksum_size);
1700 return AVERROR_INVALIDDATA;
1701 }
1702
1703 s->fft_order = av_log2(s->fft_size) + 1;
1704
1705 // something like max decodable tones
1706 s->group_order = av_log2(s->group_size) + 1;
1707 s->frame_size = s->group_size / 16; // 16 iterations per super block
1708
1709 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1710 return AVERROR_INVALIDDATA;
1711
1712 s->sub_sampling = s->fft_order - 7;
1713 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1714
1715 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1716 case 0: tmp = 40; break;
1717 case 1: tmp = 48; break;
1718 case 2: tmp = 56; break;
1719 case 3: tmp = 72; break;
1720 case 4: tmp = 80; break;
1721 case 5: tmp = 100;break;
1722 default: tmp=s->sub_sampling; break;
1723 }
1724 tmp_val = 0;
1725 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1726 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1727 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1728 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1729 s->cm_table_select = tmp_val;
1730
1731 if (avctx->bit_rate <= 8000)
1732 s->coeff_per_sb_select = 0;
1733 else if (avctx->bit_rate < 16000)
1734 s->coeff_per_sb_select = 1;
1735 else
1736 s->coeff_per_sb_select = 2;
1737
1738 // Fail on unknown fft order
1739 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1740 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1741 return AVERROR_PATCHWELCOME;
1742 }
1743 if (s->fft_size != (1 << (s->fft_order - 1))) {
1744 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1745 return AVERROR_INVALIDDATA;
1746 }
1747
1748 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1749 ff_mpadsp_init(&s->mpadsp);
1750
1751 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1752
1753 return 0;
1754}
1755
1756static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1757{
1758 QDM2Context *s = avctx->priv_data;
1759
1760 ff_rdft_end(&s->rdft_ctx);
1761
1762 return 0;
1763}
1764
1765static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1766{
1767 int ch, i;
1768 const int frame_size = (q->frame_size * q->channels);
1769
1770 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1771 return -1;
1772
1773 /* select input buffer */
1774 q->compressed_data = in;
1775 q->compressed_size = q->checksum_size;
1776
1777 /* copy old block, clear new block of output samples */
1778 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1779 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1780
1781 /* decode block of QDM2 compressed data */
1782 if (q->sub_packet == 0) {
1783 q->has_errors = 0; // zero it for a new super block
1784 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1785 qdm2_decode_super_block(q);
1786 }
1787
1788 /* parse subpackets */
1789 if (!q->has_errors) {
1790 if (q->sub_packet == 2)
1791 qdm2_decode_fft_packets(q);
1792
1793 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1794 }
1795
1796 /* sound synthesis stage 1 (FFT) */
1797 for (ch = 0; ch < q->channels; ch++) {
1798 qdm2_calculate_fft(q, ch, q->sub_packet);
1799
1800 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1801 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1802 return -1;
1803 }
1804 }
1805
1806 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1807 if (!q->has_errors && q->do_synth_filter)
1808 qdm2_synthesis_filter(q, q->sub_packet);
1809
1810 q->sub_packet = (q->sub_packet + 1) % 16;
1811
1812 /* clip and convert output float[] to 16-bit signed samples */
1813 for (i = 0; i < frame_size; i++) {
1814 int value = (int)q->output_buffer[i];
1815
1816 if (value > SOFTCLIP_THRESHOLD)
1817 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1818 else if (value < -SOFTCLIP_THRESHOLD)
1819 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1820
1821 out[i] = value;
1822 }
1823
1824 return 0;
1825}
1826
1827static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1828 int *got_frame_ptr, AVPacket *avpkt)
1829{
1830 AVFrame *frame = data;
1831 const uint8_t *buf = avpkt->data;
1832 int buf_size = avpkt->size;
1833 QDM2Context *s = avctx->priv_data;
1834 int16_t *out;
1835 int i, ret;
1836
1837 if(!buf)
1838 return 0;
1839 if(buf_size < s->checksum_size)
1840 return -1;
1841
1842 /* get output buffer */
1843 frame->nb_samples = 16 * s->frame_size;
1844 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1845 return ret;
1846 out = (int16_t *)frame->data[0];
1847
1848 for (i = 0; i < 16; i++) {
1849 if ((ret = qdm2_decode(s, buf, out)) < 0)
1850 return ret;
1851 out += s->channels * s->frame_size;
1852 }
1853
1854 *got_frame_ptr = 1;
1855
1856 return s->checksum_size;
1857}
1858
1859AVCodec ff_qdm2_decoder = {
1860 .name = "qdm2",
1861 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1862 .type = AVMEDIA_TYPE_AUDIO,
1863 .id = AV_CODEC_ID_QDM2,
1864 .priv_data_size = sizeof(QDM2Context),
1865 .init = qdm2_decode_init,
1866 .close = qdm2_decode_close,
1867 .decode = qdm2_decode_frame,
1868 .capabilities = AV_CODEC_CAP_DR1,
1869};
1870
generated by cgit at 2024-05-17 06:11:08 (GMT)