FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavcodec/qdm2.c
Date: 2022-07-07 01:21:54
Exec Total Coverage
Lines: 471 850 55.4%
Branches: 277 669 41.4%

Line Branch Exec Source
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 #include "libavutil/mem_internal.h"
40 #include "libavutil/thread.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 "internal.h"
48 #include "mpegaudio.h"
49 #include "mpegaudiodsp.h"
50 #include "rdft.h"
51
52 #include "qdm2_tablegen.h"
53
54 #define QDM2_LIST_ADD(list, size, packet) \
55 do { \
56 if (size > 0) { \
57 list[size - 1].next = &list[size]; \
58 } \
59 list[size].packet = packet; \
60 list[size].next = NULL; \
61 size++; \
62 } while(0)
63
64 // Result is 8, 16 or 30
65 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
66
67 #define FIX_NOISE_IDX(noise_idx) \
68 if ((noise_idx) >= 3840) \
69 (noise_idx) -= 3840; \
70
71 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
72
73 #define SAMPLES_NEEDED \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
75
76 #define SAMPLES_NEEDED_2(why) \
77 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
78
79 #define QDM2_MAX_FRAME_SIZE 512
80
81 typedef int8_t sb_int8_array[2][30][64];
82
83 /**
84 * Subpacket
85 */
86 typedef struct QDM2SubPacket {
87 int type; ///< subpacket type
88 unsigned int size; ///< subpacket size
89 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
90 } QDM2SubPacket;
91
92 /**
93 * A node in the subpacket list
94 */
95 typedef struct QDM2SubPNode {
96 QDM2SubPacket *packet; ///< packet
97 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
98 } QDM2SubPNode;
99
100 typedef struct QDM2Complex {
101 float re;
102 float im;
103 } QDM2Complex;
104
105 typedef struct FFTTone {
106 float level;
107 QDM2Complex *complex;
108 const float *table;
109 int phase;
110 int phase_shift;
111 int duration;
112 short time_index;
113 short cutoff;
114 } FFTTone;
115
116 typedef struct FFTCoefficient {
117 int16_t sub_packet;
118 uint8_t channel;
119 int16_t offset;
120 int16_t exp;
121 uint8_t phase;
122 } FFTCoefficient;
123
124 typedef struct QDM2FFT {
125 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
126 } QDM2FFT;
127
128 /**
129 * QDM2 decoder context
130 */
131 typedef struct QDM2Context {
132 /// Parameters from codec header, do not change during playback
133 int nb_channels; ///< number of channels
134 int channels; ///< number of channels
135 int group_size; ///< size of frame group (16 frames per group)
136 int fft_size; ///< size of FFT, in complex numbers
137 int checksum_size; ///< size of data block, used also for checksum
138
139 /// Parameters built from header parameters, do not change during playback
140 int group_order; ///< order of frame group
141 int fft_order; ///< order of FFT (actually fftorder+1)
142 int frame_size; ///< size of data frame
143 int frequency_range;
144 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
145 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
146 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
147
148 /// Packets and packet lists
149 QDM2SubPacket sub_packets[16]; ///< the packets themselves
150 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
151 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
152 int sub_packets_B; ///< number of packets on 'B' list
153 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
154 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
155
156 /// FFT and tones
157 FFTTone fft_tones[1000];
158 int fft_tone_start;
159 int fft_tone_end;
160 FFTCoefficient fft_coefs[1000];
161 int fft_coefs_index;
162 int fft_coefs_min_index[5];
163 int fft_coefs_max_index[5];
164 int fft_level_exp[6];
165 RDFTContext rdft_ctx;
166 QDM2FFT fft;
167
168 /// I/O data
169 const uint8_t *compressed_data;
170 int compressed_size;
171 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
172
173 /// Synthesis filter
174 MPADSPContext mpadsp;
175 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
176 int synth_buf_offset[MPA_MAX_CHANNELS];
177 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
178 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
179
180 /// Mixed temporary data used in decoding
181 float tone_level[MPA_MAX_CHANNELS][30][64];
182 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
183 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
184 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
185 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
186 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
187 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
188 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
189 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
190
191 // Flags
192 int has_errors; ///< packet has errors
193 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
194 int do_synth_filter; ///< used to perform or skip synthesis filter
195
196 int sub_packet;
197 int noise_idx; ///< index for dithering noise table
198 } QDM2Context;
199
200 static const int switchtable[23] = {
201 0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
202 };
203
204 67959 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
205 {
206 int value;
207
208 67959 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
209
210 /* stage-2, 3 bits exponent escape sequence */
211
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67959 if (value < 0)
212 558 value = get_bits(gb, get_bits(gb, 3) + 1);
213
214 /* stage-3, optional */
215
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67959 if (flag) {
216 int tmp;
217
218
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18368 if (value >= 60) {
219 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
220 return 0;
221 }
222
223 18368 tmp= vlc_stage3_values[value];
224
225
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18368 if ((value & ~3) > 0)
226 14255 tmp += get_bits(gb, (value >> 2));
227 18368 value = tmp;
228 }
229
230 67959 return value;
231 }
232
233 11196 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
234 {
235 11196 int value = qdm2_get_vlc(gb, vlc, 0, depth);
236
237
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11196 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
238 }
239
240 /**
241 * QDM2 checksum
242 *
243 * @param data pointer to data to be checksummed
244 * @param length data length
245 * @param value checksum value
246 *
247 * @return 0 if checksum is OK
248 */
249 141 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
250 {
251 int i;
252
253
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52311 for (i = 0; i < length; i++)
254 52170 value -= data[i];
255
256 141 return (uint16_t)(value & 0xffff);
257 }
258
259 /**
260 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
261 *
262 * @param gb bitreader context
263 * @param sub_packet packet under analysis
264 */
265 1128 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
266 QDM2SubPacket *sub_packet)
267 {
268 1128 sub_packet->type = get_bits(gb, 8);
269
270
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1128 if (sub_packet->type == 0) {
271 141 sub_packet->size = 0;
272 141 sub_packet->data = NULL;
273 } else {
274 987 sub_packet->size = get_bits(gb, 8);
275
276
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987 if (sub_packet->type & 0x80) {
277 141 sub_packet->size <<= 8;
278 141 sub_packet->size |= get_bits(gb, 8);
279 141 sub_packet->type &= 0x7f;
280 }
281
282
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987 if (sub_packet->type == 0x7f)
283 sub_packet->type |= (get_bits(gb, 8) << 8);
284
285 // FIXME: this depends on bitreader-internal data
286 987 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
287 }
288
289 1128 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
290 1128 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
291 1128 }
292
293 /**
294 * Return node pointer to first packet of requested type in list.
295 *
296 * @param list list of subpackets to be scanned
297 * @param type type of searched subpacket
298 * @return node pointer for subpacket if found, else NULL
299 */
300 564 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
301 int type)
302 {
303
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987 while (list && list->packet) {
304
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564 if (list->packet->type == type)
305 141 return list;
306 423 list = list->next;
307 }
308 423 return NULL;
309 }
310
311 /**
312 * Replace 8 elements with their average value.
313 * Called by qdm2_decode_superblock before starting subblock decoding.
314 *
315 * @param q context
316 */
317 141 static void average_quantized_coeffs(QDM2Context *q)
318 {
319 int i, j, n, ch, sum;
320
321
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141 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
322
323
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423 for (ch = 0; ch < q->nb_channels; ch++)
324
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3102 for (i = 0; i < n; i++) {
325 2820 sum = 0;
326
327
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25380 for (j = 0; j < 8; j++)
328 22560 sum += q->quantized_coeffs[ch][i][j];
329
330 2820 sum /= 8;
331
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2820 if (sum > 0)
332 2466 sum--;
333
334
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25380 for (j = 0; j < 8; j++)
335 22560 q->quantized_coeffs[ch][i][j] = sum;
336 }
337 141 }
338
339 /**
340 * Build subband samples with noise weighted by q->tone_level.
341 * Called by synthfilt_build_sb_samples.
342 *
343 * @param q context
344 * @param sb subband index
345 */
346 4230 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
347 {
348 int ch, j;
349
350
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4230 FIX_NOISE_IDX(q->noise_idx);
351
352
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4230 if (!q->nb_channels)
353 return;
354
355
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12690 for (ch = 0; ch < q->nb_channels; ch++) {
356
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549900 for (j = 0; j < 64; j++) {
357 541440 q->sb_samples[ch][j * 2][sb] =
358 541440 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
359 541440 q->sb_samples[ch][j * 2 + 1][sb] =
360 541440 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
361 }
362 }
363 }
364
365 /**
366 * Called while processing data from subpackets 11 and 12.
367 * Used after making changes to coding_method array.
368 *
369 * @param sb subband index
370 * @param channels number of channels
371 * @param coding_method q->coding_method[0][0][0]
372 */
373 static int fix_coding_method_array(int sb, int channels,
374 sb_int8_array coding_method)
375 {
376 int j, k;
377 int ch;
378 int run, case_val;
379
380 for (ch = 0; ch < channels; ch++) {
381 for (j = 0; j < 64; ) {
382 if (coding_method[ch][sb][j] < 8)
383 return -1;
384 if ((coding_method[ch][sb][j] - 8) > 22) {
385 run = 1;
386 case_val = 8;
387 } else {
388 switch (switchtable[coding_method[ch][sb][j] - 8]) {
389 case 0: run = 10;
390 case_val = 10;
391 break;
392 case 1: run = 1;
393 case_val = 16;
394 break;
395 case 2: run = 5;
396 case_val = 24;
397 break;
398 case 3: run = 3;
399 case_val = 30;
400 break;
401 case 4: run = 1;
402 case_val = 30;
403 break;
404 case 5: run = 1;
405 case_val = 8;
406 break;
407 default: run = 1;
408 case_val = 8;
409 break;
410 }
411 }
412 for (k = 0; k < run; k++) {
413 if (j + k < 128) {
414 int sbjk = sb + (j + k) / 64;
415 if (sbjk > 29) {
416 SAMPLES_NEEDED
417 continue;
418 }
419 if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
420 if (k > 0) {
421 SAMPLES_NEEDED
422 //not debugged, almost never used
423 memset(&coding_method[ch][sb][j + k], case_val,
424 k *sizeof(int8_t));
425 memset(&coding_method[ch][sb][j + k], case_val,
426 3 * sizeof(int8_t));
427 }
428 }
429 }
430 }
431 j += run;
432 }
433 }
434 return 0;
435 }
436
437 /**
438 * Related to synthesis filter
439 * Called by process_subpacket_10
440 *
441 * @param q context
442 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
443 */
444 141 static void fill_tone_level_array(QDM2Context *q, int flag)
445 {
446 int i, sb, ch, sb_used;
447 int tmp, tab;
448
449
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423 for (ch = 0; ch < q->nb_channels; ch++)
450
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8742 for (sb = 0; sb < 30; sb++)
451
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76140 for (i = 0; i < 8; i++) {
452
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67680 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
453 54144 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
454 54144 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
455 else
456 13536 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
457
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67680 if(tmp < 0)
458 tmp += 0xff;
459 67680 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
460 }
461
462
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141 sb_used = QDM2_SB_USED(q->sub_sampling);
463
464
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141 if ((q->superblocktype_2_3 != 0) && !flag) {
465
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4371 for (sb = 0; sb < sb_used; sb++)
466
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467
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549900 for (i = 0; i < 64; i++) {
468 541440 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
469
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541440 if (q->tone_level_idx[ch][sb][i] < 0)
470 q->tone_level[ch][sb][i] = 0;
471 else
472 541440 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
473 }
474 } else {
475 tab = q->superblocktype_2_3 ? 0 : 1;
476 for (sb = 0; sb < sb_used; sb++) {
477 if ((sb >= 4) && (sb <= 23)) {
478 for (ch = 0; ch < q->nb_channels; ch++)
479 for (i = 0; i < 64; i++) {
480 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
481 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
482 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
483 q->tone_level_idx_hi2[ch][sb - 4];
484 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
485 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
486 q->tone_level[ch][sb][i] = 0;
487 else
488 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
489 }
490 } else {
491 if (sb > 4) {
492 for (ch = 0; ch < q->nb_channels; ch++)
493 for (i = 0; i < 64; i++) {
494 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
495 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
496 q->tone_level_idx_hi2[ch][sb - 4];
497 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
498 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
499 q->tone_level[ch][sb][i] = 0;
500 else
501 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
502 }
503 } else {
504 for (ch = 0; ch < q->nb_channels; ch++)
505 for (i = 0; i < 64; i++) {
506 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
507 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
508 q->tone_level[ch][sb][i] = 0;
509 else
510 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
511 }
512 }
513 }
514 }
515 }
516 141 }
517
518 /**
519 * Related to synthesis filter
520 * Called by process_subpacket_11
521 * c is built with data from subpacket 11
522 * Most of this function is used only if superblock_type_2_3 == 0,
523 * never seen it in samples.
524 *
525 * @param tone_level_idx
526 * @param tone_level_idx_temp
527 * @param coding_method q->coding_method[0][0][0]
528 * @param nb_channels number of channels
529 * @param c coming from subpacket 11, passed as 8*c
530 * @param superblocktype_2_3 flag based on superblock packet type
531 * @param cm_table_select q->cm_table_select
532 */
533 static void fill_coding_method_array(sb_int8_array tone_level_idx,
534 sb_int8_array tone_level_idx_temp,
535 sb_int8_array coding_method,
536 int nb_channels,
537 int c, int superblocktype_2_3,
538 int cm_table_select)
539 {
540 int ch, sb, j;
541 int tmp, acc, esp_40, comp;
542 int add1, add2, add3, add4;
543 int64_t multres;
544
545 if (!superblocktype_2_3) {
546 /* This case is untested, no samples available */
547 avpriv_request_sample(NULL, "!superblocktype_2_3");
548 return;
549 for (ch = 0; ch < nb_channels; ch++) {
550 for (sb = 0; sb < 30; sb++) {
551 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
552 add1 = tone_level_idx[ch][sb][j] - 10;
553 if (add1 < 0)
554 add1 = 0;
555 add2 = add3 = add4 = 0;
556 if (sb > 1) {
557 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
558 if (add2 < 0)
559 add2 = 0;
560 }
561 if (sb > 0) {
562 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
563 if (add3 < 0)
564 add3 = 0;
565 }
566 if (sb < 29) {
567 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
568 if (add4 < 0)
569 add4 = 0;
570 }
571 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
572 if (tmp < 0)
573 tmp = 0;
574 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
575 }
576 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
577 }
578 }
579 acc = 0;
580 for (ch = 0; ch < nb_channels; ch++)
581 for (sb = 0; sb < 30; sb++)
582 for (j = 0; j < 64; j++)
583 acc += tone_level_idx_temp[ch][sb][j];
584
585 multres = 0x66666667LL * (acc * 10);
586 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
587 for (ch = 0; ch < nb_channels; ch++)
588 for (sb = 0; sb < 30; sb++)
589 for (j = 0; j < 64; j++) {
590 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
591 if (comp < 0)
592 comp += 0xff;
593 comp /= 256; // signed shift
594 switch(sb) {
595 case 0:
596 if (comp < 30)
597 comp = 30;
598 comp += 15;
599 break;
600 case 1:
601 if (comp < 24)
602 comp = 24;
603 comp += 10;
604 break;
605 case 2:
606 case 3:
607 case 4:
608 if (comp < 16)
609 comp = 16;
610 }
611 if (comp <= 5)
612 tmp = 0;
613 else if (comp <= 10)
614 tmp = 10;
615 else if (comp <= 16)
616 tmp = 16;
617 else if (comp <= 24)
618 tmp = -1;
619 else
620 tmp = 0;
621 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
622 }
623 for (sb = 0; sb < 30; sb++)
624 fix_coding_method_array(sb, nb_channels, coding_method);
625 for (ch = 0; ch < nb_channels; ch++)
626 for (sb = 0; sb < 30; sb++)
627 for (j = 0; j < 64; j++)
628 if (sb >= 10) {
629 if (coding_method[ch][sb][j] < 10)
630 coding_method[ch][sb][j] = 10;
631 } else {
632 if (sb >= 2) {
633 if (coding_method[ch][sb][j] < 16)
634 coding_method[ch][sb][j] = 16;
635 } else {
636 if (coding_method[ch][sb][j] < 30)
637 coding_method[ch][sb][j] = 30;
638 }
639 }
640 } else { // superblocktype_2_3 != 0
641 for (ch = 0; ch < nb_channels; ch++)
642 for (sb = 0; sb < 30; sb++)
643 for (j = 0; j < 64; j++)
644 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
645 }
646 }
647
648 /**
649 * Called by process_subpacket_11 to process more data from subpacket 11
650 * with sb 0-8.
651 * Called by process_subpacket_12 to process data from subpacket 12 with
652 * sb 8-sb_used.
653 *
654 * @param q context
655 * @param gb bitreader context
656 * @param length packet length in bits
657 * @param sb_min lower subband processed (sb_min included)
658 * @param sb_max higher subband processed (sb_max excluded)
659 */
660 282 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
661 int length, int sb_min, int sb_max)
662 {
663 int sb, j, k, n, ch, run, channels;
664 int joined_stereo, zero_encoding;
665 int type34_first;
666 282 float type34_div = 0;
667 float type34_predictor;
668 float samples[10];
669 282 int sign_bits[16] = {0};
670
671
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282 if (length == 0) {
672 // If no data use noise
673
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4512 for (sb=sb_min; sb < sb_max; sb++)
674 4230 build_sb_samples_from_noise(q, sb);
675
676 282 return 0;
677 }
678
679 for (sb = sb_min; sb < sb_max; sb++) {
680 channels = q->nb_channels;
681
682 if (q->nb_channels <= 1 || sb < 12)
683 joined_stereo = 0;
684 else if (sb >= 24)
685 joined_stereo = 1;
686 else
687 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
688
689 if (joined_stereo) {
690 if (get_bits_left(gb) >= 16)
691 for (j = 0; j < 16; j++)
692 sign_bits[j] = get_bits1(gb);
693
694 for (j = 0; j < 64; j++)
695 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
696 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
697
698 if (fix_coding_method_array(sb, q->nb_channels,
699 q->coding_method)) {
700 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
701 build_sb_samples_from_noise(q, sb);
702 continue;
703 }
704 channels = 1;
705 }
706
707 for (ch = 0; ch < channels; ch++) {
708 FIX_NOISE_IDX(q->noise_idx);
709 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
710 type34_predictor = 0.0;
711 type34_first = 1;
712
713 for (j = 0; j < 128; ) {
714 switch (q->coding_method[ch][sb][j / 2]) {
715 case 8:
716 if (get_bits_left(gb) >= 10) {
717 if (zero_encoding) {
718 for (k = 0; k < 5; k++) {
719 if ((j + 2 * k) >= 128)
720 break;
721 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
722 }
723 } else {
724 n = get_bits(gb, 8);
725 if (n >= 243) {
726 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
727 return AVERROR_INVALIDDATA;
728 }
729
730 for (k = 0; k < 5; k++)
731 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
732 }
733 for (k = 0; k < 5; k++)
734 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
735 } else {
736 for (k = 0; k < 10; k++)
737 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
738 }
739 run = 10;
740 break;
741
742 case 10:
743 if (get_bits_left(gb) >= 1) {
744 float f = 0.81;
745
746 if (get_bits1(gb))
747 f = -f;
748 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
749 samples[0] = f;
750 } else {
751 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
752 }
753 run = 1;
754 break;
755
756 case 16:
757 if (get_bits_left(gb) >= 10) {
758 if (zero_encoding) {
759 for (k = 0; k < 5; k++) {
760 if ((j + k) >= 128)
761 break;
762 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
763 }
764 } else {
765 n = get_bits (gb, 8);
766 if (n >= 243) {
767 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
768 return AVERROR_INVALIDDATA;
769 }
770
771 for (k = 0; k < 5; k++)
772 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
773 }
774 } else {
775 for (k = 0; k < 5; k++)
776 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
777 }
778 run = 5;
779 break;
780
781 case 24:
782 if (get_bits_left(gb) >= 7) {
783 n = get_bits(gb, 7);
784 if (n >= 125) {
785 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
786 return AVERROR_INVALIDDATA;
787 }
788
789 for (k = 0; k < 3; k++)
790 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
791 } else {
792 for (k = 0; k < 3; k++)
793 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
794 }
795 run = 3;
796 break;
797
798 case 30:
799 if (get_bits_left(gb) >= 4) {
800 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
801 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
802 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
803 return AVERROR_INVALIDDATA;
804 }
805 samples[0] = type30_dequant[index];
806 } else
807 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
808
809 run = 1;
810 break;
811
812 case 34:
813 if (get_bits_left(gb) >= 7) {
814 if (type34_first) {
815 type34_div = (float)(1 << get_bits(gb, 2));
816 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
817 type34_predictor = samples[0];
818 type34_first = 0;
819 } else {
820 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
821 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
822 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
823 return AVERROR_INVALIDDATA;
824 }
825 samples[0] = type34_delta[index] / type34_div + type34_predictor;
826 type34_predictor = samples[0];
827 }
828 } else {
829 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
830 }
831 run = 1;
832 break;
833
834 default:
835 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
836 run = 1;
837 break;
838 }
839
840 if (joined_stereo) {
841 for (k = 0; k < run && j + k < 128; k++) {
842 q->sb_samples[0][j + k][sb] =
843 q->tone_level[0][sb][(j + k) / 2] * samples[k];
844 if (q->nb_channels == 2) {
845 if (sign_bits[(j + k) / 8])
846 q->sb_samples[1][j + k][sb] =
847 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
848 else
849 q->sb_samples[1][j + k][sb] =
850 q->tone_level[1][sb][(j + k) / 2] * samples[k];
851 }
852 }
853 } else {
854 for (k = 0; k < run; k++)
855 if ((j + k) < 128)
856 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
857 }
858
859 j += run;
860 } // j loop
861 } // channel loop
862 } // subband loop
863 return 0;
864 }
865
866 /**
867 * Init the first element of a channel in quantized_coeffs with data
868 * from packet 10 (quantized_coeffs[ch][0]).
869 * This is similar to process_subpacket_9, but for a single channel
870 * and for element [0]
871 * same VLC tables as process_subpacket_9 are used.
872 *
873 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
874 * @param gb bitreader context
875 */
876 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
877 GetBitContext *gb)
878 {
879 int i, k, run, level, diff;
880
881 if (get_bits_left(gb) < 16)
882 return -1;
883 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
884
885 quantized_coeffs[0] = level;
886
887 for (i = 0; i < 7; ) {
888 if (get_bits_left(gb) < 16)
889 return -1;
890 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
891
892 if (i + run >= 8)
893 return -1;
894
895 if (get_bits_left(gb) < 16)
896 return -1;
897 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
898
899 for (k = 1; k <= run; k++)
900 quantized_coeffs[i + k] = (level + ((k * diff) / run));
901
902 level += diff;
903 i += run;
904 }
905 return 0;
906 }
907
908 /**
909 * Related to synthesis filter, process data from packet 10
910 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
911 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
912 * data from packet 10
913 *
914 * @param q context
915 * @param gb bitreader context
916 */
917 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
918 {
919 int sb, j, k, n, ch;
920
921 for (ch = 0; ch < q->nb_channels; ch++) {
922 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
923
924 if (get_bits_left(gb) < 16) {
925 memset(q->quantized_coeffs[ch][0], 0, 8);
926 break;
927 }
928 }
929
930 n = q->sub_sampling + 1;
931
932 for (sb = 0; sb < n; sb++)
933 for (ch = 0; ch < q->nb_channels; ch++)
934 for (j = 0; j < 8; j++) {
935 if (get_bits_left(gb) < 1)
936 break;
937 if (get_bits1(gb)) {
938 for (k=0; k < 8; k++) {
939 if (get_bits_left(gb) < 16)
940 break;
941 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
942 }
943 } else {
944 for (k=0; k < 8; k++)
945 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
946 }
947 }
948
949 n = QDM2_SB_USED(q->sub_sampling) - 4;
950
951 for (sb = 0; sb < n; sb++)
952 for (ch = 0; ch < q->nb_channels; ch++) {
953 if (get_bits_left(gb) < 16)
954 break;
955 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
956 if (sb > 19)
957 q->tone_level_idx_hi2[ch][sb] -= 16;
958 else
959 for (j = 0; j < 8; j++)
960 q->tone_level_idx_mid[ch][sb][j] = -16;
961 }
962
963 n = QDM2_SB_USED(q->sub_sampling) - 5;
964
965 for (sb = 0; sb < n; sb++)
966 for (ch = 0; ch < q->nb_channels; ch++)
967 for (j = 0; j < 8; j++) {
968 if (get_bits_left(gb) < 16)
969 break;
970 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
971 }
972 }
973
974 /**
975 * Process subpacket 9, init quantized_coeffs with data from it
976 *
977 * @param q context
978 * @param node pointer to node with packet
979 */
980 141 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
981 {
982 GetBitContext gb;
983 int i, j, k, n, ch, run, level, diff;
984
985 141 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
986
987
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141 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
988
989
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1410 for (i = 1; i < n; i++)
990
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3807 for (ch = 0; ch < q->nb_channels; ch++) {
991 2538 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
992 2538 q->quantized_coeffs[ch][i][0] = level;
993
994
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13734 for (j = 0; j < (8 - 1); ) {
995 11196 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
996 11196 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
997
998
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11196 if (j + run >= 8)
999 return -1;
1000
1001
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28962 for (k = 1; k <= run; k++)
1002 17766 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1003
1004 11196 level += diff;
1005 11196 j += run;
1006 }
1007 }
1008
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423 for (ch = 0; ch < q->nb_channels; ch++)
1010
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2538 for (i = 0; i < 8; i++)
1011 2256 q->quantized_coeffs[ch][0][i] = 0;
1012
1013 141 return 0;
1014 }
1015
1016 /**
1017 * Process subpacket 10 if not null, else
1018 *
1019 * @param q context
1020 * @param node pointer to node with packet
1021 */
1022 141 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1023 {
1024 GetBitContext gb;
1025
1026
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141 if (node) {
1027 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1028 init_tone_level_dequantization(q, &gb);
1029 fill_tone_level_array(q, 1);
1030 } else {
1031 141 fill_tone_level_array(q, 0);
1032 }
1033 141 }
1034
1035 /**
1036 * Process subpacket 11
1037 *
1038 * @param q context
1039 * @param node pointer to node with packet
1040 */
1041 141 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1042 {
1043 GetBitContext gb;
1044 141 int length = 0;
1045
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141 if (node) {
1047 length = node->packet->size * 8;
1048 init_get_bits(&gb, node->packet->data, length);
1049 }
1050
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141 if (length >= 32) {
1052 int c = get_bits(&gb, 13);
1053
1054 if (c > 3)
1055 fill_coding_method_array(q->tone_level_idx,
1056 q->tone_level_idx_temp, q->coding_method,
1057 q->nb_channels, 8 * c,
1058 q->superblocktype_2_3, q->cm_table_select);
1059 }
1060
1061 141 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1062 141 }
1063
1064 /**
1065 * Process subpacket 12
1066 *
1067 * @param q context
1068 * @param node pointer to node with packet
1069 */
1070 141 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1071 {
1072 GetBitContext gb;
1073 141 int length = 0;
1074
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141 if (node) {
1076 length = node->packet->size * 8;
1077 init_get_bits(&gb, node->packet->data, length);
1078 }
1079
1080
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141 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1081 141 }
1082
1083 /**
1084 * Process new subpackets for synthesis filter
1085 *
1086 * @param q context
1087 * @param list list with synthesis filter packets (list D)
1088 */
1089 141 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1090 {
1091 QDM2SubPNode *nodes[4];
1092
1093 141 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1094
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141 if (nodes[0])
1095 141 process_subpacket_9(q, nodes[0]);
1096
1097 141 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1098
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141 if (nodes[1])
1099 process_subpacket_10(q, nodes[1]);
1100 else
1101 141 process_subpacket_10(q, NULL);
1102
1103 141 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1104
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141 if (nodes[0] && nodes[1] && nodes[2])
1105 process_subpacket_11(q, nodes[2]);
1106 else
1107 141 process_subpacket_11(q, NULL);
1108
1109 141 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1110
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141 if (nodes[0] && nodes[1] && nodes[3])
1111 process_subpacket_12(q, nodes[3]);
1112 else
1113 141 process_subpacket_12(q, NULL);
1114 141 }
1115
1116 /**
1117 * Decode superblock, fill packet lists.
1118 *
1119 * @param q context
1120 */
1121 141 static void qdm2_decode_super_block(QDM2Context *q)
1122 {
1123 GetBitContext gb;
1124 QDM2SubPacket header, *packet;
1125 int i, packet_bytes, sub_packet_size, sub_packets_D;
1126 141 unsigned int next_index = 0;
1127
1128 141 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1129 141 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1130 141 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1131
1132 141 q->sub_packets_B = 0;
1133 141 sub_packets_D = 0;
1134
1135 141 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1136
1137 141 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1138 141 qdm2_decode_sub_packet_header(&gb, &header);
1139
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141 if (header.type < 2 || header.type >= 8) {
1141 q->has_errors = 1;
1142 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1143 return;
1144 }
1145
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141 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1147 141 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1148
1149 141 init_get_bits(&gb, header.data, header.size * 8);
1150
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141 if (header.type == 2 || header.type == 4 || header.type == 5) {
1152 141 int csum = 257 * get_bits(&gb, 8);
1153 141 csum += 2 * get_bits(&gb, 8);
1154
1155 141 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1156
1157
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141 if (csum != 0) {
1158 q->has_errors = 1;
1159 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1160 return;
1161 }
1162 }
1163
1164 141 q->sub_packet_list_B[0].packet = NULL;
1165 141 q->sub_packet_list_D[0].packet = NULL;
1166
1167
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987 for (i = 0; i < 6; i++)
1168
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846 if (--q->fft_level_exp[i] < 0)
1169 846 q->fft_level_exp[i] = 0;
1170
1171
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987 for (i = 0; packet_bytes > 0; i++) {
1172 int j;
1173
1174
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987 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1175 SAMPLES_NEEDED_2("too many packet bytes");
1176 return;
1177 }
1178
1179 987 q->sub_packet_list_A[i].next = NULL;
1180
1181
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987 if (i > 0) {
1182 846 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1183
1184 /* seek to next block */
1185 846 init_get_bits(&gb, header.data, header.size * 8);
1186 846 skip_bits(&gb, next_index * 8);
1187
1188
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846 if (next_index >= header.size)
1189 break;
1190 }
1191
1192 /* decode subpacket */
1193 987 packet = &q->sub_packets[i];
1194 987 qdm2_decode_sub_packet_header(&gb, packet);
1195 987 next_index = packet->size + get_bits_count(&gb) / 8;
1196
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987 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1197
1198
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987 if (packet->type == 0)
1199 141 break;
1200
1201
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846 if (sub_packet_size > packet_bytes) {
1202 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1203 break;
1204 packet->size += packet_bytes - sub_packet_size;
1205 }
1206
1207 846 packet_bytes -= sub_packet_size;
1208
1209 /* add subpacket to 'all subpackets' list */
1210 846 q->sub_packet_list_A[i].packet = packet;
1211
1212 /* add subpacket to related list */
1213
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846 if (packet->type == 8) {
1214 SAMPLES_NEEDED_2("packet type 8");
1215 return;
1216
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846 } else if (packet->type >= 9 && packet->type <= 12) {
1217 /* packets for MPEG Audio like Synthesis Filter */
1218
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141 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1219
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705 } else if (packet->type == 13) {
1220 for (j = 0; j < 6; j++)
1221 q->fft_level_exp[j] = get_bits(&gb, 6);
1222
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705 } else if (packet->type == 14) {
1223 for (j = 0; j < 6; j++)
1224 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1225
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705 } else if (packet->type == 15) {
1226 SAMPLES_NEEDED_2("packet type 15")
1227 return;
1228
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705 } else if (packet->type >= 16 && packet->type < 48 &&
1229
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705 !fft_subpackets[packet->type - 16]) {
1230 /* packets for FFT */
1231
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705 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1232 }
1233 } // Packet bytes loop
1234
1235
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141 if (q->sub_packet_list_D[0].packet) {
1236 141 process_synthesis_subpackets(q, q->sub_packet_list_D);
1237 141 q->do_synth_filter = 1;
1238 } else if (q->do_synth_filter) {
1239 process_subpacket_10(q, NULL);
1240 process_subpacket_11(q, NULL);
1241 process_subpacket_12(q, NULL);
1242 }
1243 }
1244
1245 20217 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1246 int offset, int duration, int channel,
1247 int exp, int phase)
1248 {
1249
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20217 if (q->fft_coefs_min_index[duration] < 0)
1250 551 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1251
1252
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20217 q->fft_coefs[q->fft_coefs_index].sub_packet =
1253 296 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1254 20217 q->fft_coefs[q->fft_coefs_index].channel = channel;
1255 20217 q->fft_coefs[q->fft_coefs_index].offset = offset;
1256 20217 q->fft_coefs[q->fft_coefs_index].exp = exp;
1257 20217 q->fft_coefs[q->fft_coefs_index].phase = phase;
1258 20217 q->fft_coefs_index++;
1259 20217 }
1260
1261 564 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1262 GetBitContext *gb, int b)
1263 {
1264 int channel, stereo, phase, exp;
1265 int local_int_4, local_int_8, stereo_phase, local_int_10;
1266 int local_int_14, stereo_exp, local_int_20, local_int_28;
1267 int n, offset;
1268
1269 564 local_int_4 = 0;
1270 564 local_int_28 = 0;
1271 564 local_int_20 = 2;
1272 564 local_int_8 = (4 - duration);
1273 564 local_int_10 = 1 << (q->group_order - duration - 1);
1274 564 offset = 1;
1275
1276
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16337 while (get_bits_left(gb)>0) {
1277
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16337 if (q->superblocktype_2_3) {
1278
2/2
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18368 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1279
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2098 if (get_bits_left(gb)<0) {
1280
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67 if(local_int_4 < q->group_size)
1281 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1282 67 return;
1283 }
1284 2031 offset = 1;
1285
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2031 if (n == 0) {
1286 2019 local_int_4 += local_int_10;
1287 2019 local_int_28 += (1 << local_int_8);
1288 } else {
1289 12 local_int_4 += 8 * local_int_10;
1290 12 local_int_28 += (8 << local_int_8);
1291 }
1292 }
1293 16270 offset += (n - 2);
1294 } else {
1295 if (local_int_10 <= 2) {
1296 av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
1297 return;
1298 }
1299 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1300 while (offset >= (local_int_10 - 1)) {
1301 offset += (1 - (local_int_10 - 1));
1302 local_int_4 += local_int_10;
1303 local_int_28 += (1 << local_int_8);
1304 }
1305 }
1306
1307
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16270 if (local_int_4 >= q->group_size)
1308 497 return;
1309
1310 15773 local_int_14 = (offset >> local_int_8);
1311
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15773 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1312 return;
1313
1314
1/2
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15773 if (q->nb_channels > 1) {
1315 15773 channel = get_bits1(gb);
1316 15773 stereo = get_bits1(gb);
1317 } else {
1318 channel = 0;
1319 stereo = 0;
1320 }
1321
1322
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15773 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1323 15773 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1324 15773 exp = (exp < 0) ? 0 : exp;
1325
1326 15773 phase = get_bits(gb, 3);
1327 15773 stereo_exp = 0;
1328 15773 stereo_phase = 0;
1329
1330
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15773 if (stereo) {
1331 4444 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1332 4444 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1333
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4444 if (stereo_phase < 0)
1334 497 stereo_phase += 8;
1335 }
1336
1337
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15773 if (q->frequency_range > (local_int_14 + 1)) {
1338 15773 int sub_packet = (local_int_20 + local_int_28);
1339
1340
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15773 if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
1341 return;
1342
1343 15773 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1344 channel, exp, phase);
1345
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15773 if (stereo)
1346 4444 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1347 1 - channel,
1348 stereo_exp, stereo_phase);
1349 }
1350 15773 offset++;
1351 }
1352 }
1353
1354 141 static void qdm2_decode_fft_packets(QDM2Context *q)
1355 {
1356 int i, j, min, max, value, type, unknown_flag;
1357 GetBitContext gb;
1358
1359
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141 if (!q->sub_packet_list_B[0].packet)
1360 return;
1361
1362 /* reset minimum indexes for FFT coefficients */
1363 141 q->fft_coefs_index = 0;
1364
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846 for (i = 0; i < 5; i++)
1365 705 q->fft_coefs_min_index[i] = -1;
1366
1367 /* process subpackets ordered by type, largest type first */
1368
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846 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1369 705 QDM2SubPacket *packet = NULL;
1370
1371 /* find subpacket with largest type less than max */
1372
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4230 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1373 3525 value = q->sub_packet_list_B[j].packet->type;
1374
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3525 if (value > min && value < max) {
1375 705 min = value;
1376 705 packet = q->sub_packet_list_B[j].packet;
1377 }
1378 }
1379
1380 705 max = min;
1381
1382 /* check for errors (?) */
1383
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705 if (!packet)
1384 return;
1385
1386
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705 if (i == 0 &&
1387
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141 (packet->type < 16 || packet->type >= 48 ||
1388
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141 fft_subpackets[packet->type - 16]))
1389 return;
1390
1391 /* decode FFT tones */
1392 705 init_get_bits(&gb, packet->data, packet->size * 8);
1393
1394
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705 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1395 unknown_flag = 1;
1396 else
1397 705 unknown_flag = 0;
1398
1399 705 type = packet->type;
1400
1401
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1410 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1402 705 int duration = q->sub_sampling + 5 - (type & 15);
1403
1404
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705 if (duration >= 0 && duration < 4)
1405 564 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1406 } else if (type == 31) {
1407 for (j = 0; j < 4; j++)
1408 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1409 } else if (type == 46) {
1410 for (j = 0; j < 6; j++)
1411 q->fft_level_exp[j] = get_bits(&gb, 6);
1412 for (j = 0; j < 4; j++)
1413 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1414 }
1415 } // Loop on B packets
1416
1417 /* calculate maximum indexes for FFT coefficients */
1418
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846 for (i = 0, j = -1; i < 5; i++)
1419
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705 if (q->fft_coefs_min_index[i] >= 0) {
1420
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551 if (j >= 0)
1421 410 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1422 551 j = i;
1423 }
1424
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141 if (j >= 0)
1425 141 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1426 }
1427
1428 341696 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1429 {
1430 float level, f[6];
1431 int i;
1432 QDM2Complex c;
1433 341696 const double iscale = 2.0 * M_PI / 512.0;
1434
1435 341696 tone->phase += tone->phase_shift;
1436
1437 /* calculate current level (maximum amplitude) of tone */
1438 341696 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1439 341696 c.im = level * sin(tone->phase * iscale);
1440 341696 c.re = level * cos(tone->phase * iscale);
1441
1442 /* generate FFT coefficients for tone */
1443
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341696 if (tone->duration >= 3 || tone->cutoff >= 3) {
1444 46655 tone->complex[0].im += c.im;
1445 46655 tone->complex[0].re += c.re;
1446 46655 tone->complex[1].im -= c.im;
1447 46655 tone->complex[1].re -= c.re;
1448 } else {
1449 295041 f[1] = -tone->table[4];
1450 295041 f[0] = tone->table[3] - tone->table[0];
1451 295041 f[2] = 1.0 - tone->table[2] - tone->table[3];
1452 295041 f[3] = tone->table[1] + tone->table[4] - 1.0;
1453 295041 f[4] = tone->table[0] - tone->table[1];
1454 295041 f[5] = tone->table[2];
1455
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885123 for (i = 0; i < 2; i++) {
1456 590082 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1457 590082 c.re * f[i];
1458 1180164 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1459
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590082 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1460 }
1461
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1475205 for (i = 0; i < 4; i++) {
1462 1180164 tone->complex[i].re += c.re * f[i + 2];
1463 1180164 tone->complex[i].im += c.im * f[i + 2];
1464 }
1465 }
1466
1467 /* copy the tone if it has not yet died out */
1468
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341696 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1469 321878 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1470 321878 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1471 }
1472 341696 }
1473
1474 2256 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1475 {
1476 int i, j, ch;
1477 2256 const double iscale = 0.25 * M_PI;
1478
1479
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6768 for (ch = 0; ch < q->channels; ch++) {
1480 4512 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1481 }
1482
1483
1484 /* apply FFT tones with duration 4 (1 FFT period) */
1485
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2256 if (q->fft_coefs_min_index[4] >= 0)
1486
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8 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1487 float level;
1488 QDM2Complex c;
1489
1490 if (q->fft_coefs[i].sub_packet != sub_packet)
1491 break;
1492
1493 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1494 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];
1495
1496 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1497 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1498 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1499 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1500 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1501 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1502 }
1503
1504 /* generate existing FFT tones */
1505
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323747 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1506 321491 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1507 321491 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1508 }
1509
1510 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1511
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11280 for (i = 0; i < 4; i++)
1512
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✓ Branch 1 taken 200 times.
9024 if (q->fft_coefs_min_index[i] >= 0) {
1513
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29029 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1514 int offset, four_i;
1515 FFTTone tone;
1516
1517
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24549 if (q->fft_coefs[j].sub_packet != sub_packet)
1518 4344 break;
1519
1520 20205 four_i = (4 - i);
1521 20205 offset = q->fft_coefs[j].offset >> four_i;
1522
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20205 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1523
1524
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20205 if (offset < q->frequency_range) {
1525
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20205 if (offset < 2)
1526 2795 tone.cutoff = offset;
1527 else
1528
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17410 tone.cutoff = (offset >= 60) ? 3 : 2;
1529
1530
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20205 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];
1531 20205 tone.complex = &q->fft.complex[ch][offset];
1532 20205 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1533 20205 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1534 20205 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1535 20205 tone.duration = i;
1536 20205 tone.time_index = 0;
1537
1538 20205 qdm2_fft_generate_tone(q, &tone);
1539 }
1540 }
1541 8824 q->fft_coefs_min_index[i] = j;
1542 }
1543 2256 }
1544
1545 4512 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1546 {
1547
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4512 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1548 4512 float *out = q->output_buffer + channel;
1549 int i;
1550 4512 q->fft.complex[channel][0].re *= 2.0f;
1551 4512 q->fft.complex[channel][0].im = 0.0f;
1552 4512 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1553 /* add samples to output buffer */
1554
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1159584 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1555 1155072 out[0] += q->fft.complex[channel][i].re * gain;
1556 1155072 out[q->channels] += q->fft.complex[channel][i].im * gain;
1557 1155072 out += 2 * q->channels;
1558 }
1559 4512 }
1560
1561 /**
1562 * @param q context
1563 * @param index subpacket number
1564 */
1565 2256 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1566 {
1567 2256 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1568
1569 /* copy sb_samples */
1570
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2256 sb_used = QDM2_SB_USED(q->sub_sampling);
1571
1572
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6768 for (ch = 0; ch < q->channels; ch++)
1573
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40608 for (i = 0; i < 8; i++)
1574
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108288 for (k = sb_used; k < SBLIMIT; k++)
1575 72192 q->sb_samples[ch][(8 * index) + i][k] = 0;
1576
1577
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6768 for (ch = 0; ch < q->nb_channels; ch++) {
1578 4512 float *samples_ptr = q->samples + ch;
1579
1580
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40608 for (i = 0; i < 8; i++) {
1581 36096 ff_mpa_synth_filter_float(&q->mpadsp,
1582 36096 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1583 ff_mpa_synth_window_float, &dither_state,
1584 36096 samples_ptr, q->nb_channels,
1585 36096 q->sb_samples[ch][(8 * index) + i]);
1586 36096 samples_ptr += 32 * q->nb_channels;
1587 }
1588 }
1589
1590 /* add samples to output buffer */
1591 2256 sub_sampling = (4 >> q->sub_sampling);
1592
1593
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6768 for (ch = 0; ch < q->channels; ch++)
1594
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1159584 for (i = 0; i < q->frame_size; i++)
1595 1155072 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1596 2256 }
1597
1598 /**
1599 * Init static data (does not depend on specific file)
1600 */
1601 3 static av_cold void qdm2_init_static_data(void) {
1602 3 qdm2_init_vlc();
1603 3 softclip_table_init();
1604 3 rnd_table_init();
1605 3 init_noise_samples();
1606
1607 3 ff_mpa_synth_init_float();
1608 3 }
1609
1610 /**
1611 * Init parameters from codec extradata
1612 */
1613 4 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1614 {
1615 static AVOnce init_static_once = AV_ONCE_INIT;
1616 4 QDM2Context *s = avctx->priv_data;
1617 int tmp_val, tmp, size;
1618 GetByteContext gb;
1619
1620 /* extradata parsing
1621
1622 Structure:
1623 wave {
1624 frma (QDM2)
1625 QDCA
1626 QDCP
1627 }
1628
1629 32 size (including this field)
1630 32 tag (=frma)
1631 32 type (=QDM2 or QDMC)
1632
1633 32 size (including this field, in bytes)
1634 32 tag (=QDCA) // maybe mandatory parameters
1635 32 unknown (=1)
1636 32 channels (=2)
1637 32 samplerate (=44100)
1638 32 bitrate (=96000)
1639 32 block size (=4096)
1640 32 frame size (=256) (for one channel)
1641 32 packet size (=1300)
1642
1643 32 size (including this field, in bytes)
1644 32 tag (=QDCP) // maybe some tuneable parameters
1645 32 float1 (=1.0)
1646 32 zero ?
1647 32 float2 (=1.0)
1648 32 float3 (=1.0)
1649 32 unknown (27)
1650 32 unknown (8)
1651 32 zero ?
1652 */
1653
1654
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4 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1655 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1656 return AVERROR_INVALIDDATA;
1657 }
1658
1659 4 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1660
1661
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20 while (bytestream2_get_bytes_left(&gb) > 8) {
1662
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20 if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1663 (uint64_t)MKBETAG('Q','D','M','2')))
1664 4 break;
1665 16 bytestream2_skip(&gb, 1);
1666 }
1667
1668
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4 if (bytestream2_get_bytes_left(&gb) < 12) {
1669 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1670 bytestream2_get_bytes_left(&gb));
1671 return AVERROR_INVALIDDATA;
1672 }
1673
1674 4 bytestream2_skip(&gb, 8);
1675 4 size = bytestream2_get_be32(&gb);
1676
1677
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4 if (size > bytestream2_get_bytes_left(&gb)) {
1678 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1679 bytestream2_get_bytes_left(&gb), size);
1680 return AVERROR_INVALIDDATA;
1681 }
1682
1683 4 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1684
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4 if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1685 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1686 return AVERROR_INVALIDDATA;
1687 }
1688
1689 4 bytestream2_skip(&gb, 4);
1690
1691 4 s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1692
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4 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1693 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1694 return AVERROR_INVALIDDATA;
1695 }
1696 4 av_channel_layout_uninit(&avctx->ch_layout);
1697 4 av_channel_layout_default(&avctx->ch_layout, s->channels);
1698
1699 4 avctx->sample_rate = bytestream2_get_be32(&gb);
1700 4 avctx->bit_rate = bytestream2_get_be32(&gb);
1701 4 s->group_size = bytestream2_get_be32(&gb);
1702 4 s->fft_size = bytestream2_get_be32(&gb);
1703 4 s->checksum_size = bytestream2_get_be32(&gb);
1704
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4 if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
1705 av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
1706 return AVERROR_INVALIDDATA;
1707 }
1708
1709 4 s->fft_order = av_log2(s->fft_size) + 1;
1710
1711 // Fail on unknown fft order
1712
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4 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1713 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1714 return AVERROR_PATCHWELCOME;
1715 }
1716
1717 // something like max decodable tones
1718 4 s->group_order = av_log2(s->group_size) + 1;
1719 4 s->frame_size = s->group_size / 16; // 16 iterations per super block
1720
1721
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4 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1722 return AVERROR_INVALIDDATA;
1723
1724 4 s->sub_sampling = s->fft_order - 7;
1725 4 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1726
1727
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4 if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
1728 avpriv_request_sample(avctx, "large frames");
1729 return AVERROR_PATCHWELCOME;
1730 }
1731
1732
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4 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1733 case 0: tmp = 40; break;
1734 case 1: tmp = 48; break;
1735 case 2: tmp = 56; break;
1736 case 3: tmp = 72; break;
1737 case 4: tmp = 80; break;
1738 4 case 5: tmp = 100;break;
1739 default: tmp=s->sub_sampling; break;
1740 }
1741 4 tmp_val = 0;
1742
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4 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1743
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4 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1744
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4 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1745
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4 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1746 4 s->cm_table_select = tmp_val;
1747
1748
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4 if (avctx->bit_rate <= 8000)
1749 s->coeff_per_sb_select = 0;
1750
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4 else if (avctx->bit_rate < 16000)
1751 s->coeff_per_sb_select = 1;
1752 else
1753 4 s->coeff_per_sb_select = 2;
1754
1755
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4 if (s->fft_size != (1 << (s->fft_order - 1))) {
1756 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1757 return AVERROR_INVALIDDATA;
1758 }
1759
1760 4 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1761 4 ff_mpadsp_init(&s->mpadsp);
1762
1763 4 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1764
1765 4 ff_thread_once(&init_static_once, qdm2_init_static_data);
1766
1767 4 return 0;
1768 }
1769
1770 4 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1771 {
1772 4 QDM2Context *s = avctx->priv_data;
1773
1774 4 ff_rdft_end(&s->rdft_ctx);
1775
1776 4 return 0;
1777 }
1778
1779 2256 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1780 {
1781 int ch, i;
1782 2256 const int frame_size = (q->frame_size * q->channels);
1783
1784
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2256 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1785 return -1;
1786
1787 /* select input buffer */
1788 2256 q->compressed_data = in;
1789 2256 q->compressed_size = q->checksum_size;
1790
1791 /* copy old block, clear new block of output samples */
1792 2256 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1793 2256 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1794
1795 /* decode block of QDM2 compressed data */
1796
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2256 if (q->sub_packet == 0) {
1797 141 q->has_errors = 0; // zero it for a new super block
1798 141 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1799 141 qdm2_decode_super_block(q);
1800 }
1801
1802 /* parse subpackets */
1803
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2256 if (!q->has_errors) {
1804
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2256 if (q->sub_packet == 2)
1805 141 qdm2_decode_fft_packets(q);
1806
1807 2256 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1808 }
1809
1810 /* sound synthesis stage 1 (FFT) */
1811
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6768 for (ch = 0; ch < q->channels; ch++) {
1812 4512 qdm2_calculate_fft(q, ch, q->sub_packet);
1813
1814
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4512 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1815 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1816 return -1;
1817 }
1818 }
1819
1820 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1821
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2256 if (!q->has_errors && q->do_synth_filter)
1822 2256 qdm2_synthesis_filter(q, q->sub_packet);
1823
1824 2256 q->sub_packet = (q->sub_packet + 1) % 16;
1825
1826 /* clip and convert output float[] to 16-bit signed samples */
1827
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1157328 for (i = 0; i < frame_size; i++) {
1828 1155072 int value = (int)q->output_buffer[i];
1829
1830
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1155072 if (value > SOFTCLIP_THRESHOLD)
1831
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245 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1832
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1154827 else if (value < -SOFTCLIP_THRESHOLD)
1833
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839 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1834
1835 1155072 out[i] = value;
1836 }
1837
1838 2256 return 0;
1839 }
1840
1841 141 static int qdm2_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1842 int *got_frame_ptr, AVPacket *avpkt)
1843 {
1844 141 const uint8_t *buf = avpkt->data;
1845 141 int buf_size = avpkt->size;
1846 141 QDM2Context *s = avctx->priv_data;
1847 int16_t *out;
1848 int i, ret;
1849
1850
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141 if(!buf)
1851 return 0;
1852
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141 if(buf_size < s->checksum_size)
1853 return -1;
1854
1855 /* get output buffer */
1856 141 frame->nb_samples = 16 * s->frame_size;
1857
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141 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1858 return ret;
1859 141 out = (int16_t *)frame->data[0];
1860
1861
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2397 for (i = 0; i < 16; i++) {
1862
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2256 if ((ret = qdm2_decode(s, buf, out)) < 0)
1863 return ret;
1864 2256 out += s->channels * s->frame_size;
1865 }
1866
1867 141 *got_frame_ptr = 1;
1868
1869 141 return s->checksum_size;
1870 }
1871
1872 const FFCodec ff_qdm2_decoder = {
1873 .p.name = "qdm2",
1874 .p.long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1875 .p.type = AVMEDIA_TYPE_AUDIO,
1876 .p.id = AV_CODEC_ID_QDM2,
1877 .priv_data_size = sizeof(QDM2Context),
1878 .init = qdm2_decode_init,
1879 .close = qdm2_decode_close,
1880 FF_CODEC_DECODE_CB(qdm2_decode_frame),
1881 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1882 .caps_internal = FF_CODEC_CAP_INIT_THREADSAFE,
1883 };
1884