FFmpeg coverage

Directory: ../../../ffmpeg/
File: src/libavcodec/qdm2.c
Date: 2026-04-24 19:58:39
Exec Total Coverage
Lines: 491 906 54.2%
Functions: 27 31 87.1%
Branches: 290 715 40.6%
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
37 #include "libavutil/attributes.h"
38 #include "libavutil/channel_layout.h"
39 #include "libavutil/mem_internal.h"
40 #include "libavutil/thread.h"
41 #include "libavutil/tx.h"
42
43 #define BITSTREAM_READER_LE
44 #include "avcodec.h"
45 #include "get_bits.h"
46 #include "bytestream.h"
47 #include "codec_internal.h"
48 #include "decode.h"
49 #include "mpegaudio.h"
50 #include "mpegaudiodsp.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 FFTTone {
101 float level;
102 AVComplexFloat *complex;
103 const float *table;
104 int phase;
105 int phase_shift;
106 int duration;
107 short time_index;
108 short cutoff;
109 } FFTTone;
110
111 typedef struct FFTCoefficient {
112 int16_t sub_packet;
113 uint8_t channel;
114 int16_t offset;
115 int16_t exp;
116 uint8_t phase;
117 } FFTCoefficient;
118
119 typedef struct QDM2FFT {
120 DECLARE_ALIGNED(32, AVComplexFloat, complex)[MPA_MAX_CHANNELS][256 + 1];
121 DECLARE_ALIGNED(32, AVComplexFloat, temp)[MPA_MAX_CHANNELS][256];
122 } QDM2FFT;
123
124 /**
125 * QDM2 decoder context
126 */
127 typedef struct QDM2Context {
128 /// Parameters from codec header, do not change during playback
129 int nb_channels; ///< number of channels
130 int channels; ///< number of channels
131 int group_size; ///< size of frame group (16 frames per group)
132 int fft_size; ///< size of FFT, in complex numbers
133 int checksum_size; ///< size of data block, used also for checksum
134
135 /// Parameters built from header parameters, do not change during playback
136 int group_order; ///< order of frame group
137 int fft_order; ///< order of FFT (actually fftorder+1)
138 int frame_size; ///< size of data frame
139 int frequency_range;
140 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
141 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
142 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
143
144 /// Packets and packet lists
145 QDM2SubPacket sub_packets[16]; ///< the packets themselves
146 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
147 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
148 int sub_packets_B; ///< number of packets on 'B' list
149 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
150 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
151
152 /// FFT and tones
153 FFTTone fft_tones[1000];
154 int fft_tone_start;
155 int fft_tone_end;
156 FFTCoefficient fft_coefs[1000];
157 int fft_coefs_index;
158 int fft_coefs_min_index[5];
159 int fft_coefs_max_index[5];
160 int fft_level_exp[6];
161 AVTXContext *rdft_ctx;
162 av_tx_fn rdft_fn;
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
197 static 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
201 67959 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
202 {
203 67959 int value = get_vlc2(gb, vlc->table, vlc->bits,
204 av_builtin_constant_p(depth) ? depth : 2);
205
206 /* stage-2, 3 bits exponent escape sequence */
207
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 67959 if (value < 0)
208 558 value = get_bits(gb, get_bits(gb, 3) + 1);
209
210 /* stage-3, optional */
211
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 67959 if (flag) {
212 int tmp;
213
214
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 18368 if (value >= 60) {
215 ✗ av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
216 ✗ return 0;
217 }
218
219 18368 tmp= vlc_stage3_values[value];
220
221
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 18368 if ((value & ~3) > 0)
222 14255 tmp += get_bits(gb, (value >> 2));
223 18368 value = tmp;
224 }
225
226 67959 return value;
227 }
228
229 11196 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
230 {
231 11196 int value = qdm2_get_vlc(gb, vlc, 0, depth);
232
233
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 11196 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
234 }
235
236 /**
237 * QDM2 checksum
238 *
239 * @param data pointer to data to be checksummed
240 * @param length data length
241 * @param value checksum value
242 *
243 * @return 0 if checksum is OK
244 */
245 141 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
246 {
247 int i;
248
249
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 52311 for (i = 0; i < length; i++)
250 52170 value -= data[i];
251
252 141 return (uint16_t)(value & 0xffff);
253 }
254
255 /**
256 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
257 *
258 * @param gb bitreader context
259 * @param sub_packet packet under analysis
260 */
261 1128 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
262 QDM2SubPacket *sub_packet)
263 {
264 1128 sub_packet->type = get_bits(gb, 8);
265
266
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 1128 if (sub_packet->type == 0) {
267 141 sub_packet->size = 0;
268 141 sub_packet->data = NULL;
269 } else {
270 987 sub_packet->size = get_bits(gb, 8);
271
272
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 987 if (sub_packet->type & 0x80) {
273 141 sub_packet->size <<= 8;
274 141 sub_packet->size |= get_bits(gb, 8);
275 141 sub_packet->type &= 0x7f;
276 }
277
278
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 987 if (sub_packet->type == 0x7f)
279 ✗ sub_packet->type |= (get_bits(gb, 8) << 8);
280
281 // FIXME: this depends on bitreader-internal data
282 987 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
283 }
284
285 1128 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
286 1128 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
287 1128 }
288
289 /**
290 * Return node pointer to first packet of requested type in list.
291 *
292 * @param list list of subpackets to be scanned
293 * @param type type of searched subpacket
294 * @return node pointer for subpacket if found, else NULL
295 */
296 564 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
297 int type)
298 {
299
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 987 while (list && list->packet) {
300
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 564 if (list->packet->type == type)
301 141 return list;
302 423 list = list->next;
303 }
304 423 return NULL;
305 }
306
307 /**
308 * Replace 8 elements with their average value.
309 * Called by qdm2_decode_superblock before starting subblock decoding.
310 *
311 * @param q context
312 */
313 141 static void average_quantized_coeffs(QDM2Context *q)
314 {
315 int i, j, n, ch, sum;
316
317
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 141 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
318
319
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 423 for (ch = 0; ch < q->nb_channels; ch++)
320
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 3102 for (i = 0; i < n; i++) {
321 2820 sum = 0;
322
323
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 25380 for (j = 0; j < 8; j++)
324 22560 sum += q->quantized_coeffs[ch][i][j];
325
326 2820 sum /= 8;
327
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 2820 if (sum > 0)
328 2466 sum--;
329
330
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 25380 for (j = 0; j < 8; j++)
331 22560 q->quantized_coeffs[ch][i][j] = sum;
332 }
333 141 }
334
335 /**
336 * Build subband samples with noise weighted by q->tone_level.
337 * Called by synthfilt_build_sb_samples.
338 *
339 * @param q context
340 * @param sb subband index
341 */
342 4230 static int build_sb_samples_from_noise(QDM2Context *q, int sb)
343 {
344 int ch, j;
345
346
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 4230 FIX_NOISE_IDX(q->noise_idx);
347
348
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 4230 if (!q->nb_channels)
349 ✗ return AVERROR_INVALIDDATA;
350
351
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 12690 for (ch = 0; ch < q->nb_channels; ch++) {
352
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 549900 for (j = 0; j < 64; j++) {
353 541440 q->sb_samples[ch][j * 2][sb] =
354 541440 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
355 541440 q->sb_samples[ch][j * 2 + 1][sb] =
356 541440 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
357 }
358 }
359
360 4230 return 0;
361 }
362
363 /**
364 * Called while processing data from subpackets 11 and 12.
365 * Used after making changes to coding_method array.
366 *
367 * @param sb subband index
368 * @param channels number of channels
369 * @param coding_method q->coding_method[0][0][0]
370 */
371 ✗ static int fix_coding_method_array(int sb, int channels,
372 sb_int8_array coding_method)
373 {
374 int j, k;
375 int ch;
376 int run, case_val;
377
378 ✗ for (ch = 0; ch < channels; ch++) {
379 ✗ for (j = 0; j < 64; ) {
380 ✗ if (coding_method[ch][sb][j] < 8)
381 ✗ return -1;
382 ✗ if ((coding_method[ch][sb][j] - 8) > 22) {
383 ✗ run = 1;
384 ✗ case_val = 8;
385 } else {
386 ✗ switch (switchtable[coding_method[ch][sb][j] - 8]) {
387 ✗ case 0: run = 10;
388 ✗ case_val = 10;
389 ✗ break;
390 ✗ case 1: run = 1;
391 ✗ case_val = 16;
392 ✗ break;
393 ✗ case 2: run = 5;
394 ✗ case_val = 24;
395 ✗ break;
396 ✗ case 3: run = 3;
397 ✗ case_val = 30;
398 ✗ break;
399 ✗ case 4: run = 1;
400 ✗ case_val = 30;
401 ✗ break;
402 ✗ case 5: run = 1;
403 ✗ case_val = 8;
404 ✗ break;
405 ✗ default: run = 1;
406 ✗ case_val = 8;
407 ✗ break;
408 }
409 }
410 ✗ for (k = 0; k < run; k++) {
411 ✗ if (j + k < 128) {
412 ✗ int sbjk = sb + (j + k) / 64;
413 ✗ if (sbjk > 29) {
414 ✗ SAMPLES_NEEDED
415 ✗ continue;
416 }
417 ✗ if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
418 ✗ if (k > 0) {
419 ✗ SAMPLES_NEEDED
420 //not debugged, almost never used
421 ✗ memset(&coding_method[ch][sb][j + k], case_val,
422 k *sizeof(int8_t));
423 ✗ memset(&coding_method[ch][sb][j + k], case_val,
424 3 * sizeof(int8_t));
425 }
426 }
427 }
428 }
429 ✗ j += run;
430 }
431 }
432 ✗ return 0;
433 }
434
435 /**
436 * Related to synthesis filter
437 * Called by process_subpacket_10
438 *
439 * @param q context
440 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
441 */
442 141 static void fill_tone_level_array(QDM2Context *q, int flag)
443 {
444 int i, sb, ch, sb_used;
445 int tmp, tab;
446
447
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 423 for (ch = 0; ch < q->nb_channels; ch++)
448
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 8742 for (sb = 0; sb < 30; sb++)
449
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 76140 for (i = 0; i < 8; i++) {
450
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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))
451 54144 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
452 54144 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
453 else
454 13536 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
455
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 67680 if(tmp < 0)
456 ✗ tmp += 0xff;
457 67680 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
458 }
459
460
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 141 sb_used = QDM2_SB_USED(q->sub_sampling);
461
462
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 141 if ((q->superblocktype_2_3 != 0) && !flag) {
463
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 4371 for (sb = 0; sb < sb_used; sb++)
464
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465
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466 541440 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
467
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 541440 if (q->tone_level_idx[ch][sb][i] < 0)
468 ✗ q->tone_level[ch][sb][i] = 0;
469 else
470 541440 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
471 }
472 } else {
473 ✗ tab = q->superblocktype_2_3 ? 0 : 1;
474 ✗ for (sb = 0; sb < sb_used; sb++) {
475 ✗ if ((sb >= 4) && (sb <= 23)) {
476 ✗ for (ch = 0; ch < q->nb_channels; ch++)
477 ✗ for (i = 0; i < 64; i++) {
478 ✗ tmp = q->tone_level_idx_base[ch][sb][i / 8] -
479 ✗ q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
480 ✗ q->tone_level_idx_mid[ch][sb - 4][i / 8] -
481 ✗ q->tone_level_idx_hi2[ch][sb - 4];
482 ✗ q->tone_level_idx[ch][sb][i] = tmp & 0xff;
483 ✗ if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
484 ✗ q->tone_level[ch][sb][i] = 0;
485 else
486 ✗ q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
487 }
488 } else {
489 ✗ if (sb > 4) {
490 ✗ for (ch = 0; ch < q->nb_channels; ch++)
491 ✗ for (i = 0; i < 64; i++) {
492 ✗ tmp = q->tone_level_idx_base[ch][sb][i / 8] -
493 ✗ q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
494 ✗ q->tone_level_idx_hi2[ch][sb - 4];
495 ✗ q->tone_level_idx[ch][sb][i] = tmp & 0xff;
496 ✗ if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
497 ✗ q->tone_level[ch][sb][i] = 0;
498 else
499 ✗ q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
500 }
501 } else {
502 ✗ for (ch = 0; ch < q->nb_channels; ch++)
503 ✗ for (i = 0; i < 64; i++) {
504 ✗ tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
505 ✗ if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
506 ✗ q->tone_level[ch][sb][i] = 0;
507 else
508 ✗ q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
509 }
510 }
511 }
512 }
513 }
514 141 }
515
516 /**
517 * Related to synthesis filter
518 * Called by process_subpacket_11
519 * c is built with data from subpacket 11
520 * Most of this function is used only if superblock_type_2_3 == 0,
521 * never seen it in samples.
522 *
523 * @param tone_level_idx
524 * @param tone_level_idx_temp
525 * @param coding_method q->coding_method[0][0][0]
526 * @param nb_channels number of channels
527 * @param c coming from subpacket 11, passed as 8*c
528 * @param superblocktype_2_3 flag based on superblock packet type
529 * @param cm_table_select q->cm_table_select
530 */
531 ✗ static int fill_coding_method_array(sb_int8_array tone_level_idx,
532 sb_int8_array tone_level_idx_temp,
533 sb_int8_array coding_method,
534 int nb_channels,
535 int c, int superblocktype_2_3,
536 int cm_table_select)
537 {
538 int ch, sb, j;
539 #if 0
540 int tmp, acc, esp_40, comp;
541 int add1, add2, add3, add4;
542 int64_t multres;
543 #endif
544
545 ✗ if (!superblocktype_2_3) {
546 /* This case is untested, no samples available */
547 ✗ avpriv_request_sample(NULL, "!superblocktype_2_3");
548 ✗ return AVERROR_PATCHWELCOME;
549 #if 0
550 for (ch = 0; ch < nb_channels; ch++) {
551 for (sb = 0; sb < 30; sb++) {
552 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
553 add1 = tone_level_idx[ch][sb][j] - 10;
554 if (add1 < 0)
555 add1 = 0;
556 add2 = add3 = add4 = 0;
557 if (sb > 1) {
558 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
559 if (add2 < 0)
560 add2 = 0;
561 }
562 if (sb > 0) {
563 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
564 if (add3 < 0)
565 add3 = 0;
566 }
567 if (sb < 29) {
568 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
569 if (add4 < 0)
570 add4 = 0;
571 }
572 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
573 if (tmp < 0)
574 tmp = 0;
575 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
576 }
577 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
578 }
579 }
580 acc = 0;
581 for (ch = 0; ch < nb_channels; ch++)
582 for (sb = 0; sb < 30; sb++)
583 for (j = 0; j < 64; j++)
584 acc += tone_level_idx_temp[ch][sb][j];
585
586 multres = 0x66666667LL * (acc * 10);
587 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
588 for (ch = 0; ch < nb_channels; ch++)
589 for (sb = 0; sb < 30; sb++)
590 for (j = 0; j < 64; j++) {
591 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
592 if (comp < 0)
593 comp += 0xff;
594 comp /= 256; // signed shift
595 switch(sb) {
596 case 0:
597 if (comp < 30)
598 comp = 30;
599 comp += 15;
600 break;
601 case 1:
602 if (comp < 24)
603 comp = 24;
604 comp += 10;
605 break;
606 case 2:
607 case 3:
608 case 4:
609 if (comp < 16)
610 comp = 16;
611 }
612 if (comp <= 5)
613 tmp = 0;
614 else if (comp <= 10)
615 tmp = 10;
616 else if (comp <= 16)
617 tmp = 16;
618 else if (comp <= 24)
619 tmp = -1;
620 else
621 tmp = 0;
622 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
623 }
624 for (sb = 0; sb < 30; sb++)
625 fix_coding_method_array(sb, nb_channels, coding_method);
626 for (ch = 0; ch < nb_channels; ch++)
627 for (sb = 0; sb < 30; sb++)
628 for (j = 0; j < 64; j++)
629 if (sb >= 10) {
630 if (coding_method[ch][sb][j] < 10)
631 coding_method[ch][sb][j] = 10;
632 } else {
633 if (sb >= 2) {
634 if (coding_method[ch][sb][j] < 16)
635 coding_method[ch][sb][j] = 16;
636 } else {
637 if (coding_method[ch][sb][j] < 30)
638 coding_method[ch][sb][j] = 30;
639 }
640 }
641 #endif
642 } else { // superblocktype_2_3 != 0
643 ✗ for (ch = 0; ch < nb_channels; ch++)
644 ✗ for (sb = 0; sb < 30; sb++)
645 ✗ for (j = 0; j < 64; j++)
646 ✗ coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
647 }
648 ✗ return 0;
649 }
650
651 /**
652 * Called by process_subpacket_11 to process more data from subpacket 11
653 * with sb 0-8.
654 * Called by process_subpacket_12 to process data from subpacket 12 with
655 * sb 8-sb_used.
656 *
657 * @param q context
658 * @param gb bitreader context
659 * @param length packet length in bits
660 * @param sb_min lower subband processed (sb_min included)
661 * @param sb_max higher subband processed (sb_max excluded)
662 */
663 282 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
664 int length, int sb_min, int sb_max)
665 {
666 int sb, j, k, n, ch, run, channels;
667 int joined_stereo, zero_encoding;
668 int type34_first;
669 282 float type34_div = 0;
670 float type34_predictor;
671 float samples[10];
672 282 int sign_bits[16] = {0};
673
674
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 282 if (length == 0) {
675 // If no data use noise
676
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 4512 for (sb=sb_min; sb < sb_max; sb++) {
677 4230 int ret = build_sb_samples_from_noise(q, sb);
678
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 4230 if (ret < 0)
679 ✗ return ret;
680 }
681
682 282 return 0;
683 }
684
685 ✗ for (sb = sb_min; sb < sb_max; sb++) {
686 ✗ channels = q->nb_channels;
687
688 ✗ if (q->nb_channels <= 1 || sb < 12)
689 ✗ joined_stereo = 0;
690 ✗ else if (sb >= 24)
691 ✗ joined_stereo = 1;
692 else
693 ✗ joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
694
695 ✗ if (joined_stereo) {
696 ✗ if (get_bits_left(gb) >= 16)
697 ✗ for (j = 0; j < 16; j++)
698 ✗ sign_bits[j] = get_bits1(gb);
699
700 ✗ for (j = 0; j < 64; j++)
701 ✗ if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
702 ✗ q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
703
704 ✗ if (fix_coding_method_array(sb, q->nb_channels,
705 ✗ q->coding_method)) {
706 ✗ av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
707 ✗ int ret = build_sb_samples_from_noise(q, sb);
708 ✗ if (ret < 0)
709 ✗ return ret;
710 ✗ continue;
711 }
712 ✗ channels = 1;
713 }
714
715 ✗ for (ch = 0; ch < channels; ch++) {
716 ✗ FIX_NOISE_IDX(q->noise_idx);
717 ✗ zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
718 ✗ type34_predictor = 0.0;
719 ✗ type34_first = 1;
720
721 ✗ for (j = 0; j < 128; ) {
722 ✗ switch (q->coding_method[ch][sb][j / 2]) {
723 ✗ case 8:
724 ✗ if (get_bits_left(gb) >= 10) {
725 ✗ if (zero_encoding) {
726 ✗ for (k = 0; k < 5; k++) {
727 ✗ if ((j + 2 * k) >= 128)
728 ✗ break;
729 ✗ samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
730 }
731 } else {
732 ✗ n = get_bits(gb, 8);
733 ✗ if (n >= 243) {
734 ✗ av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
735 ✗ return AVERROR_INVALIDDATA;
736 }
737
738 ✗ for (k = 0; k < 5; k++)
739 ✗ samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
740 }
741 ✗ for (k = 0; k < 5; k++)
742 ✗ samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
743 } else {
744 ✗ for (k = 0; k < 10; k++)
745 ✗ samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
746 }
747 ✗ run = 10;
748 ✗ break;
749
750 ✗ case 10:
751 ✗ if (get_bits_left(gb) >= 1) {
752 ✗ float f = 0.81;
753
754 ✗ if (get_bits1(gb))
755 ✗ f = -f;
756 ✗ f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
757 ✗ samples[0] = f;
758 } else {
759 ✗ samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
760 }
761 ✗ run = 1;
762 ✗ break;
763
764 ✗ case 16:
765 ✗ if (get_bits_left(gb) >= 10) {
766 ✗ if (zero_encoding) {
767 ✗ for (k = 0; k < 5; k++) {
768 ✗ if ((j + k) >= 128)
769 ✗ break;
770 ✗ samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
771 }
772 } else {
773 ✗ n = get_bits (gb, 8);
774 ✗ if (n >= 243) {
775 ✗ av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
776 ✗ return AVERROR_INVALIDDATA;
777 }
778
779 ✗ for (k = 0; k < 5; k++)
780 ✗ samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
781 }
782 } else {
783 ✗ for (k = 0; k < 5; k++)
784 ✗ samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
785 }
786 ✗ run = 5;
787 ✗ break;
788
789 ✗ case 24:
790 ✗ if (get_bits_left(gb) >= 7) {
791 ✗ n = get_bits(gb, 7);
792 ✗ if (n >= 125) {
793 ✗ av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
794 ✗ return AVERROR_INVALIDDATA;
795 }
796
797 ✗ for (k = 0; k < 3; k++)
798 ✗ samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
799 } else {
800 ✗ for (k = 0; k < 3; k++)
801 ✗ samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
802 }
803 ✗ run = 3;
804 ✗ break;
805
806 ✗ case 30:
807 ✗ if (get_bits_left(gb) >= 4) {
808 ✗ unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
809 ✗ if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
810 ✗ av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
811 ✗ return AVERROR_INVALIDDATA;
812 }
813 ✗ samples[0] = type30_dequant[index];
814 } else
815 ✗ samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
816
817 ✗ run = 1;
818 ✗ break;
819
820 ✗ case 34:
821 ✗ if (get_bits_left(gb) >= 7) {
822 ✗ if (type34_first) {
823 ✗ type34_div = (float)(1 << get_bits(gb, 2));
824 ✗ samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
825 ✗ type34_predictor = samples[0];
826 ✗ type34_first = 0;
827 } else {
828 ✗ unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
829 ✗ if (index >= FF_ARRAY_ELEMS(type34_delta)) {
830 ✗ av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
831 ✗ return AVERROR_INVALIDDATA;
832 }
833 ✗ samples[0] = type34_delta[index] / type34_div + type34_predictor;
834 ✗ type34_predictor = samples[0];
835 }
836 } else {
837 ✗ samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
838 }
839 ✗ run = 1;
840 ✗ break;
841
842 ✗ default:
843 ✗ samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
844 ✗ run = 1;
845 ✗ break;
846 }
847
848 ✗ if (joined_stereo) {
849 ✗ for (k = 0; k < run && j + k < 128; k++) {
850 ✗ q->sb_samples[0][j + k][sb] =
851 ✗ q->tone_level[0][sb][(j + k) / 2] * samples[k];
852 ✗ if (q->nb_channels == 2) {
853 ✗ if (sign_bits[(j + k) / 8])
854 ✗ q->sb_samples[1][j + k][sb] =
855 ✗ q->tone_level[1][sb][(j + k) / 2] * -samples[k];
856 else
857 ✗ q->sb_samples[1][j + k][sb] =
858 ✗ q->tone_level[1][sb][(j + k) / 2] * samples[k];
859 }
860 }
861 } else {
862 ✗ for (k = 0; k < run; k++)
863 ✗ if ((j + k) < 128)
864 ✗ q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
865 }
866
867 ✗ j += run;
868 } // j loop
869 } // channel loop
870 } // subband loop
871 ✗ return 0;
872 }
873
874 /**
875 * Init the first element of a channel in quantized_coeffs with data
876 * from packet 10 (quantized_coeffs[ch][0]).
877 * This is similar to process_subpacket_9, but for a single channel
878 * and for element [0]
879 * same VLC tables as process_subpacket_9 are used.
880 *
881 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
882 * @param gb bitreader context
883 */
884 ✗ static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
885 GetBitContext *gb)
886 {
887 int i, k, run, level, diff;
888
889 ✗ if (get_bits_left(gb) < 16)
890 ✗ return AVERROR_INVALIDDATA;
891 ✗ level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
892
893 ✗ quantized_coeffs[0] = level;
894
895 ✗ for (i = 0; i < 7; ) {
896 ✗ if (get_bits_left(gb) < 16)
897 ✗ return AVERROR_INVALIDDATA;
898 ✗ run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
899
900 ✗ if (i + run >= 8)
901 ✗ return AVERROR_INVALIDDATA;
902
903 ✗ if (get_bits_left(gb) < 16)
904 ✗ return AVERROR_INVALIDDATA;
905 ✗ diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
906
907 ✗ for (k = 1; k <= run; k++)
908 ✗ quantized_coeffs[i + k] = (level + ((k * diff) / run));
909
910 ✗ level += diff;
911 ✗ i += run;
912 }
913 ✗ return 0;
914 }
915
916 /**
917 * Related to synthesis filter, process data from packet 10
918 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
919 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
920 * data from packet 10
921 *
922 * @param q context
923 * @param gb bitreader context
924 */
925 ✗ static int init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
926 {
927 int sb, j, k, n, ch;
928
929 ✗ for (ch = 0; ch < q->nb_channels; ch++) {
930 ✗ int ret = init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
931
932 ✗ if (ret < 0)
933 ✗ return ret;
934
935 ✗ if (get_bits_left(gb) < 16) {
936 ✗ memset(q->quantized_coeffs[ch][0], 0, 8);
937 ✗ break;
938 }
939 }
940
941 ✗ n = q->sub_sampling + 1;
942
943 ✗ for (sb = 0; sb < n; sb++)
944 ✗ for (ch = 0; ch < q->nb_channels; ch++)
945 ✗ for (j = 0; j < 8; j++) {
946 ✗ if (get_bits_left(gb) < 1)
947 ✗ break;
948 ✗ if (get_bits1(gb)) {
949 ✗ for (k=0; k < 8; k++) {
950 ✗ if (get_bits_left(gb) < 16)
951 ✗ break;
952 ✗ q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
953 }
954 } else {
955 ✗ for (k=0; k < 8; k++)
956 ✗ q->tone_level_idx_hi1[ch][sb][j][k] = 0;
957 }
958 }
959
960 ✗ n = QDM2_SB_USED(q->sub_sampling) - 4;
961
962 ✗ for (sb = 0; sb < n; sb++)
963 ✗ for (ch = 0; ch < q->nb_channels; ch++) {
964 ✗ if (get_bits_left(gb) < 16)
965 ✗ break;
966 ✗ q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
967 ✗ if (sb > 19)
968 ✗ q->tone_level_idx_hi2[ch][sb] -= 16;
969 else
970 ✗ for (j = 0; j < 8; j++)
971 ✗ q->tone_level_idx_mid[ch][sb][j] = -16;
972 }
973
974 ✗ n = QDM2_SB_USED(q->sub_sampling) - 5;
975
976 ✗ for (sb = 0; sb < n; sb++)
977 ✗ for (ch = 0; ch < q->nb_channels; ch++)
978 ✗ for (j = 0; j < 8; j++) {
979 ✗ if (get_bits_left(gb) < 16)
980 ✗ break;
981 ✗ q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
982 }
983
984 ✗ return 0;
985 }
986
987 /**
988 * Process subpacket 9, init quantized_coeffs with data from it
989 *
990 * @param q context
991 * @param node pointer to node with packet
992 */
993 141 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
994 {
995 GetBitContext gb;
996 int i, j, k, n, ch, run, level, diff;
997
998 141 int ret = init_get_bits8(&gb, node->packet->data, node->packet->size);
999
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 141 if (ret < 0)
1000 ✗ return ret;
1001
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 141 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
1003
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 1410 for (i = 1; i < n; i++)
1005
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 3807 for (ch = 0; ch < q->nb_channels; ch++) {
1006 2538 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1007 2538 q->quantized_coeffs[ch][i][0] = level;
1008
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 13734 for (j = 0; j < (8 - 1); ) {
1010 11196 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1011 11196 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1012
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 11196 if (j + run >= 8)
1014 ✗ return -1;
1015
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 28962 for (k = 1; k <= run; k++)
1017 17766 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1018
1019 11196 level += diff;
1020 11196 j += run;
1021 }
1022 }
1023
1024
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 423 for (ch = 0; ch < q->nb_channels; ch++)
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 2538 for (i = 0; i < 8; i++)
1026 2256 q->quantized_coeffs[ch][0][i] = 0;
1027
1028 141 return 0;
1029 }
1030
1031 /**
1032 * Process subpacket 10 if not null, else
1033 *
1034 * @param q context
1035 * @param node pointer to node with packet
1036 */
1037 141 static int process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1038 {
1039 GetBitContext gb;
1040
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 141 if (node) {
1042 ✗ int ret = init_get_bits8(&gb, node->packet->data, node->packet->size);
1043 ✗ if (ret < 0)
1044 ✗ return ret;
1045 ✗ ret = init_tone_level_dequantization(q, &gb);
1046 ✗ if (ret < 0)
1047 ✗ return ret;
1048 ✗ fill_tone_level_array(q, 1);
1049 } else {
1050 141 fill_tone_level_array(q, 0);
1051 }
1052 141 return 0;
1053 }
1054
1055 /**
1056 * Process subpacket 11
1057 *
1058 * @param q context
1059 * @param node pointer to node with packet
1060 */
1061 141 static int process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1062 {
1063 GetBitContext gb;
1064 141 int ret, length = 0;
1065
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 141 if (node) {
1067 ✗ ret = init_get_bits8(&gb, node->packet->data, node->packet->size);
1068 ✗ if (ret < 0)
1069 ✗ return ret;
1070 ✗ length = node->packet->size * 8;
1071 }
1072
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 141 if (length >= 32) {
1074 ✗ int c = get_bits(&gb, 13);
1075
1076 ✗ if (c > 3) {
1077 ✗ ret = fill_coding_method_array(q->tone_level_idx,
1078 ✗ q->tone_level_idx_temp, q->coding_method,
1079 q->nb_channels, 8 * c,
1080 q->superblocktype_2_3, q->cm_table_select);
1081 ✗ if (ret < 0)
1082 ✗ return ret;
1083 }
1084 }
1085
1086 141 return synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1087 }
1088
1089 /**
1090 * Process subpacket 12
1091 *
1092 * @param q context
1093 * @param node pointer to node with packet
1094 */
1095 141 static int process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1096 {
1097 GetBitContext gb;
1098 141 int length = 0;
1099
1100
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 141 if (node) {
1101 ✗ int ret = init_get_bits8(&gb, node->packet->data, length);
1102 ✗ if (ret < 0)
1103 ✗ return ret;
1104 ✗ length = node->packet->size * 8;
1105 }
1106
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 141 return synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1108 }
1109
1110 /**
1111 * Process new subpackets for synthesis filter
1112 *
1113 * @param q context
1114 * @param list list with synthesis filter packets (list D)
1115 */
1116 141 static int process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1117 {
1118 QDM2SubPNode *nodes[4];
1119 141 int ret = 0;
1120
1121 141 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1122
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 141 if (nodes[0])
1123 141 ret = process_subpacket_9(q, nodes[0]);
1124
1125
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 141 if (ret < 0)
1126 ✗ return ret;
1127
1128 141 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1129
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 141 if (nodes[1])
1130 ✗ ret = process_subpacket_10(q, nodes[1]);
1131 else
1132 141 ret = process_subpacket_10(q, NULL);
1133
1134
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 141 if (ret < 0)
1135 ✗ return ret;
1136
1137 141 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1138
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 141 if (nodes[0] && nodes[1] && nodes[2])
1139 ✗ ret = process_subpacket_11(q, nodes[2]);
1140 else
1141 141 ret = process_subpacket_11(q, NULL);
1142
1143
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 141 if (ret < 0)
1144 ✗ return ret;
1145
1146 141 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1147
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 141 if (nodes[0] && nodes[1] && nodes[3])
1148 ✗ ret = process_subpacket_12(q, nodes[3]);
1149 else
1150 141 ret = process_subpacket_12(q, NULL);
1151
1152 141 return ret;
1153 }
1154
1155 /**
1156 * Decode superblock, fill packet lists.
1157 *
1158 * @param q context
1159 */
1160 141 static int qdm2_decode_super_block(QDM2Context *q)
1161 {
1162 GetBitContext gb;
1163 QDM2SubPacket header, *packet;
1164 int i, packet_bytes, sub_packet_size, sub_packets_D;
1165 int ret;
1166 141 unsigned int next_index = 0;
1167
1168 141 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1169 141 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1170 141 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1171
1172 141 q->sub_packets_B = 0;
1173 141 sub_packets_D = 0;
1174
1175 141 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1176
1177 141 ret = init_get_bits8(&gb, q->compressed_data, q->compressed_size);
1178
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 141 if (ret < 0)
1179 ✗ return ret;
1180
1181 141 qdm2_decode_sub_packet_header(&gb, &header);
1182
1183
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 141 if (header.type < 2 || header.type >= 8) {
1184 ✗ q->has_errors = 1;
1185 ✗ av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1186 ✗ return AVERROR_INVALIDDATA;
1187 }
1188
1189
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 141 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1190 141 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1191
1192 141 ret = init_get_bits8(&gb, header.data, header.size);
1193
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 141 if (ret < 0)
1194 ✗ return ret;
1195
1196
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 141 if (header.type == 2 || header.type == 4 || header.type == 5) {
1197 141 int csum = 257 * get_bits(&gb, 8);
1198 141 csum += 2 * get_bits(&gb, 8);
1199
1200 141 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1201
1202
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 141 if (csum != 0) {
1203 ✗ q->has_errors = 1;
1204 ✗ av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1205 ✗ return AVERROR_INVALIDDATA;
1206 }
1207 }
1208
1209 141 q->sub_packet_list_B[0].packet = NULL;
1210 141 q->sub_packet_list_D[0].packet = NULL;
1211
1212
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 987 for (i = 0; i < 6; i++)
1213
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 846 if (--q->fft_level_exp[i] < 0)
1214 846 q->fft_level_exp[i] = 0;
1215
1216
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 987 for (i = 0; packet_bytes > 0; i++) {
1217 int j;
1218
1219
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 987 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1220 ✗ SAMPLES_NEEDED_2("too many packet bytes");
1221 ✗ return AVERROR_PATCHWELCOME;
1222 }
1223
1224 987 q->sub_packet_list_A[i].next = NULL;
1225
1226
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 987 if (i > 0) {
1227 846 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1228
1229 /* seek to next block */
1230 846 ret = init_get_bits8(&gb, header.data, header.size);
1231
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 846 if (ret < 0)
1232 ✗ return ret;
1233
1234 846 skip_bits(&gb, next_index * 8);
1235
1236
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 846 if (next_index >= header.size)
1237 ✗ break;
1238 }
1239
1240 /* decode subpacket */
1241 987 packet = &q->sub_packets[i];
1242 987 qdm2_decode_sub_packet_header(&gb, packet);
1243 987 next_index = packet->size + get_bits_count(&gb) / 8;
1244
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 987 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1245
1246
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 987 if (packet->type == 0)
1247 141 break;
1248
1249
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 846 if (sub_packet_size > packet_bytes) {
1250 ✗ if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1251 ✗ break;
1252 ✗ packet->size += packet_bytes - sub_packet_size;
1253 }
1254
1255 846 packet_bytes -= sub_packet_size;
1256
1257 /* add subpacket to 'all subpackets' list */
1258 846 q->sub_packet_list_A[i].packet = packet;
1259
1260 /* add subpacket to related list */
1261
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 846 if (packet->type == 8) {
1262 ✗ SAMPLES_NEEDED_2("packet type 8");
1263 ✗ return AVERROR_PATCHWELCOME;
1264
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 846 } else if (packet->type >= 9 && packet->type <= 12) {
1265 /* packets for MPEG Audio like Synthesis Filter */
1266
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 141 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1267
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 705 } else if (packet->type == 13) {
1268 ✗ for (j = 0; j < 6; j++)
1269 ✗ q->fft_level_exp[j] = get_bits(&gb, 6);
1270
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 705 } else if (packet->type == 14) {
1271 ✗ for (j = 0; j < 6; j++)
1272 ✗ q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1273
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 705 } else if (packet->type == 15) {
1274 ✗ SAMPLES_NEEDED_2("packet type 15")
1275 ✗ return AVERROR_PATCHWELCOME;
1276
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 705 } else if (packet->type >= 16 && packet->type < 48 &&
1277
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 705 !fft_subpackets[packet->type - 16]) {
1278 /* packets for FFT */
1279
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 705 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1280 }
1281 } // Packet bytes loop
1282
1283
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 141 if (q->sub_packet_list_D[0].packet) {
1284 141 ret = process_synthesis_subpackets(q, q->sub_packet_list_D);
1285
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 141 if (ret < 0)
1286 ✗ return ret;
1287 141 q->do_synth_filter = 1;
1288 ✗ } else if (q->do_synth_filter) {
1289 ✗ ret = process_subpacket_10(q, NULL);
1290 ✗ if (ret < 0)
1291 ✗ return ret;
1292 ✗ ret = process_subpacket_11(q, NULL);
1293 ✗ if (ret < 0)
1294 ✗ return ret;
1295 ✗ ret = process_subpacket_12(q, NULL);
1296 ✗ if (ret < 0)
1297 ✗ return ret;
1298 }
1299 141 return 0;
1300 }
1301
1302 20217 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1303 int offset, int duration, int channel,
1304 int exp, int phase)
1305 {
1306
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 20217 if (q->fft_coefs_min_index[duration] < 0)
1307 551 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1308
1309
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 20217 q->fft_coefs[q->fft_coefs_index].sub_packet =
1310 296 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1311 20217 q->fft_coefs[q->fft_coefs_index].channel = channel;
1312 20217 q->fft_coefs[q->fft_coefs_index].offset = offset;
1313 20217 q->fft_coefs[q->fft_coefs_index].exp = exp;
1314 20217 q->fft_coefs[q->fft_coefs_index].phase = phase;
1315 20217 q->fft_coefs_index++;
1316 20217 }
1317
1318 564 static int qdm2_fft_decode_tones(QDM2Context *q, int duration,
1319 GetBitContext *gb, int b)
1320 {
1321 int channel, stereo, phase, exp;
1322 int local_int_4, local_int_8, stereo_phase, local_int_10;
1323 int local_int_14, stereo_exp, local_int_20, local_int_28;
1324 int n, offset;
1325
1326 564 local_int_4 = 0;
1327 564 local_int_28 = 0;
1328 564 local_int_20 = 2;
1329 564 local_int_8 = (4 - duration);
1330 564 local_int_10 = 1 << (q->group_order - duration - 1);
1331 564 offset = 1;
1332
1333
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 16337 while (get_bits_left(gb)>0) {
1334
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 16337 if (q->superblocktype_2_3) {
1335
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 18368 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1336
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 2098 if (get_bits_left(gb)<0) {
1337
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 67 if(local_int_4 < q->group_size)
1338 ✗ av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1339 67 return AVERROR_INVALIDDATA;
1340 }
1341 2031 offset = 1;
1342
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 2031 if (n == 0) {
1343 2019 local_int_4 += local_int_10;
1344 2019 local_int_28 += (1 << local_int_8);
1345 } else {
1346 12 local_int_4 += 8 * local_int_10;
1347 12 local_int_28 += (8 << local_int_8);
1348 }
1349 }
1350 16270 offset += (n - 2);
1351 } else {
1352 ✗ if (local_int_10 <= 2) {
1353 ✗ av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
1354 ✗ return AVERROR_INVALIDDATA;
1355 }
1356 ✗ offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1357 ✗ while (offset >= (local_int_10 - 1)) {
1358 ✗ offset += (1 - (local_int_10 - 1));
1359 ✗ local_int_4 += local_int_10;
1360 ✗ local_int_28 += (1 << local_int_8);
1361 }
1362 }
1363
1364
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 16270 if (local_int_4 >= q->group_size)
1365 497 return AVERROR_INVALIDDATA;
1366
1367 15773 local_int_14 = (offset >> local_int_8);
1368
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 15773 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1369 ✗ return AVERROR_INVALIDDATA;
1370
1371
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 15773 if (q->nb_channels > 1) {
1372 15773 channel = get_bits1(gb);
1373 15773 stereo = get_bits1(gb);
1374 } else {
1375 ✗ channel = 0;
1376 ✗ stereo = 0;
1377 }
1378
1379
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 15773 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1380 15773 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1381 15773 exp = (exp < 0) ? 0 : exp;
1382
1383 15773 phase = get_bits(gb, 3);
1384 15773 stereo_exp = 0;
1385 15773 stereo_phase = 0;
1386
1387
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 15773 if (stereo) {
1388 4444 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1389 4444 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1390
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 4444 if (stereo_phase < 0)
1391 497 stereo_phase += 8;
1392 }
1393
1394
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 15773 if (q->frequency_range > (local_int_14 + 1)) {
1395 15773 int sub_packet = (local_int_20 + local_int_28);
1396
1397
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 15773 if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
1398 ✗ return AVERROR_INVALIDDATA;
1399
1400 15773 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1401 channel, exp, phase);
1402
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 15773 if (stereo)
1403 4444 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1404 1 - channel,
1405 stereo_exp, stereo_phase);
1406 }
1407 15773 offset++;
1408 }
1409
1410 ✗ return 0;
1411 }
1412
1413 141 static int qdm2_decode_fft_packets(QDM2Context *q)
1414 {
1415 int i, j, min, max, value, type, unknown_flag;
1416 GetBitContext gb;
1417
1418
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 141 if (!q->sub_packet_list_B[0].packet)
1419 ✗ return AVERROR_INVALIDDATA;
1420
1421 /* reset minimum indexes for FFT coefficients */
1422 141 q->fft_coefs_index = 0;
1423
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 846 for (i = 0; i < 5; i++)
1424 705 q->fft_coefs_min_index[i] = -1;
1425
1426 /* process subpackets ordered by type, largest type first */
1427
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 846 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1428 705 QDM2SubPacket *packet = NULL;
1429
1430 /* find subpacket with largest type less than max */
1431
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 4230 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1432 3525 value = q->sub_packet_list_B[j].packet->type;
1433
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 3525 if (value > min && value < max) {
1434 705 min = value;
1435 705 packet = q->sub_packet_list_B[j].packet;
1436 }
1437 }
1438
1439 705 max = min;
1440
1441 /* check for errors (?) */
1442
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 705 if (!packet)
1443 ✗ return AVERROR_INVALIDDATA;
1444
1445
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 705 if (i == 0 &&
1446
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 141 (packet->type < 16 || packet->type >= 48 ||
1447
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 141 fft_subpackets[packet->type - 16]))
1448 ✗ return AVERROR_INVALIDDATA;
1449
1450 /* decode FFT tones */
1451 705 int ret = init_get_bits8(&gb, packet->data, packet->size);
1452
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 705 if (ret < 0)
1453 ✗ return ret;
1454
1455
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 705 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1456 ✗ unknown_flag = 1;
1457 else
1458 705 unknown_flag = 0;
1459
1460 705 type = packet->type;
1461
1462
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 1410 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1463 705 int duration = q->sub_sampling + 5 - (type & 15);
1464
1465
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 705 if (duration >= 0 && duration < 4)
1466 564 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1467 ✗ } else if (type == 31) {
1468 ✗ for (j = 0; j < 4; j++)
1469 ✗ qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1470 ✗ } else if (type == 46) {
1471 ✗ for (j = 0; j < 6; j++)
1472 ✗ q->fft_level_exp[j] = get_bits(&gb, 6);
1473 ✗ for (j = 0; j < 4; j++)
1474 ✗ qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1475 }
1476 } // Loop on B packets
1477
1478 /* calculate maximum indexes for FFT coefficients */
1479
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 846 for (i = 0, j = -1; i < 5; i++)
1480
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 705 if (q->fft_coefs_min_index[i] >= 0) {
1481
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 551 if (j >= 0)
1482 410 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1483 551 j = i;
1484 }
1485
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 141 if (j >= 0)
1486 141 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1487
1488 141 return 0;
1489 }
1490
1491 341696 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1492 {
1493 float level, f[6];
1494 int i;
1495 AVComplexFloat c;
1496 341696 const double iscale = 2.0 * M_PI / 512.0;
1497
1498 341696 tone->phase += tone->phase_shift;
1499
1500 /* calculate current level (maximum amplitude) of tone */
1501 341696 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1502 341696 c.im = level * sin(tone->phase * iscale);
1503 341696 c.re = level * cos(tone->phase * iscale);
1504
1505 /* generate FFT coefficients for tone */
1506
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 341696 if (tone->duration >= 3 || tone->cutoff >= 3) {
1507 46655 tone->complex[0].im += c.im;
1508 46655 tone->complex[0].re += c.re;
1509 46655 tone->complex[1].im -= c.im;
1510 46655 tone->complex[1].re -= c.re;
1511 } else {
1512 295041 f[1] = -tone->table[4];
1513 295041 f[0] = tone->table[3] - tone->table[0];
1514 295041 f[2] = 1.0 - tone->table[2] - tone->table[3];
1515 295041 f[3] = tone->table[1] + tone->table[4] - 1.0;
1516 295041 f[4] = tone->table[0] - tone->table[1];
1517 295041 f[5] = tone->table[2];
1518
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 885123 for (i = 0; i < 2; i++) {
1519 590082 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1520 590082 c.re * f[i];
1521 1180164 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1522
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 590082 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1523 }
1524
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 1475205 for (i = 0; i < 4; i++) {
1525 1180164 tone->complex[i].re += c.re * f[i + 2];
1526 1180164 tone->complex[i].im += c.im * f[i + 2];
1527 }
1528 }
1529
1530 /* copy the tone if it has not yet died out */
1531
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 341696 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1532 321878 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1533 321878 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1534 }
1535 341696 }
1536
1537 2256 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1538 {
1539 int i, j, ch;
1540 2256 const double iscale = 0.25 * M_PI;
1541
1542
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 6768 for (ch = 0; ch < q->channels; ch++) {
1543 4512 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(AVComplexFloat));
1544 }
1545
1546
1547 /* apply FFT tones with duration 4 (1 FFT period) */
1548
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 2256 if (q->fft_coefs_min_index[4] >= 0)
1549
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 8 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1550 float level;
1551 AVComplexFloat c;
1552
1553 ✗ if (q->fft_coefs[i].sub_packet != sub_packet)
1554 ✗ break;
1555
1556 ✗ ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1557 ✗ 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];
1558
1559 ✗ c.re = level * cos(q->fft_coefs[i].phase * iscale);
1560 ✗ c.im = level * sin(q->fft_coefs[i].phase * iscale);
1561 ✗ q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1562 ✗ q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1563 ✗ q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1564 ✗ q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1565 }
1566
1567 /* generate existing FFT tones */
1568
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 323747 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1569 321491 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1570 321491 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1571 }
1572
1573 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1574
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 11280 for (i = 0; i < 4; i++)
1575
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 9024 if (q->fft_coefs_min_index[i] >= 0) {
1576
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 29029 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1577 int offset, four_i;
1578 FFTTone tone;
1579
1580
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 24549 if (q->fft_coefs[j].sub_packet != sub_packet)
1581 4344 break;
1582
1583 20205 four_i = (4 - i);
1584 20205 offset = q->fft_coefs[j].offset >> four_i;
1585
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 20205 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1586
1587
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 20205 if (offset < q->frequency_range) {
1588
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 20205 if (offset < 2)
1589 2795 tone.cutoff = offset;
1590 else
1591
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 17410 tone.cutoff = (offset >= 60) ? 3 : 2;
1592
1593
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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];
1594 20205 tone.complex = &q->fft.complex[ch][offset];
1595 20205 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1596 20205 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1597 20205 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1598 20205 tone.duration = i;
1599 20205 tone.time_index = 0;
1600
1601 20205 qdm2_fft_generate_tone(q, &tone);
1602 }
1603 }
1604 8824 q->fft_coefs_min_index[i] = j;
1605 }
1606 2256 }
1607
1608 4512 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1609 {
1610
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 4512 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1611 4512 float *out = q->output_buffer + channel;
1612
1613 4512 q->fft.complex[channel][0].re *= 2.0f;
1614 4512 q->fft.complex[channel][0].im = 0.0f;
1615 4512 q->fft.complex[channel][q->fft_size].re = 0.0f;
1616 4512 q->fft.complex[channel][q->fft_size].im = 0.0f;
1617
1618 4512 q->rdft_fn(q->rdft_ctx, q->fft.temp[channel], q->fft.complex[channel],
1619 sizeof(AVComplexFloat));
1620
1621 /* add samples to output buffer */
1622
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 1159584 for (int i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1623 1155072 out[0] += q->fft.temp[channel][i].re * gain;
1624 1155072 out[q->channels] += q->fft.temp[channel][i].im * gain;
1625 1155072 out += 2 * q->channels;
1626 }
1627 4512 }
1628
1629 /**
1630 * @param q context
1631 * @param index subpacket number
1632 */
1633 2256 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1634 {
1635 2256 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1636
1637 /* copy sb_samples */
1638
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 2256 sb_used = QDM2_SB_USED(q->sub_sampling);
1639
1640
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 6768 for (ch = 0; ch < q->channels; ch++)
1641
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 40608 for (i = 0; i < 8; i++)
1642
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 108288 for (k = sb_used; k < SBLIMIT; k++)
1643 72192 q->sb_samples[ch][(8 * index) + i][k] = 0;
1644
1645
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 6768 for (ch = 0; ch < q->nb_channels; ch++) {
1646 4512 float *samples_ptr = q->samples + ch;
1647
1648
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 40608 for (i = 0; i < 8; i++) {
1649 36096 ff_mpa_synth_filter_float(&q->mpadsp,
1650 36096 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1651 ff_mpa_synth_window_float, &dither_state,
1652 36096 samples_ptr, q->nb_channels,
1653 36096 q->sb_samples[ch][(8 * index) + i]);
1654 36096 samples_ptr += 32 * q->nb_channels;
1655 }
1656 }
1657
1658 /* add samples to output buffer */
1659 2256 sub_sampling = (4 >> q->sub_sampling);
1660
1661
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 6768 for (ch = 0; ch < q->channels; ch++)
1662
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 1159584 for (i = 0; i < q->frame_size; i++)
1663 1155072 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1664 2256 }
1665
1666 /**
1667 * Init static data (does not depend on specific file)
1668 */
1669 3 static av_cold void qdm2_init_static_data(void) {
1670 3 qdm2_init_vlc();
1671 3 softclip_table_init();
1672 3 rnd_table_init();
1673 3 init_noise_samples();
1674
1675 3 ff_mpa_synth_init_float();
1676 3 }
1677
1678 /**
1679 * Init parameters from codec extradata
1680 */
1681 4 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1682 {
1683 static AVOnce init_static_once = AV_ONCE_INIT;
1684 4 QDM2Context *s = avctx->priv_data;
1685 int ret, tmp_val, tmp, size;
1686 4 float scale = 1.0f / 2.0f;
1687 GetByteContext gb;
1688
1689 /* extradata parsing
1690
1691 Structure:
1692 wave {
1693 frma (QDM2)
1694 QDCA
1695 QDCP
1696 }
1697
1698 32 size (including this field)
1699 32 tag (=frma)
1700 32 type (=QDM2 or QDMC)
1701
1702 32 size (including this field, in bytes)
1703 32 tag (=QDCA) // maybe mandatory parameters
1704 32 unknown (=1)
1705 32 channels (=2)
1706 32 samplerate (=44100)
1707 32 bitrate (=96000)
1708 32 block size (=4096)
1709 32 frame size (=256) (for one channel)
1710 32 packet size (=1300)
1711
1712 32 size (including this field, in bytes)
1713 32 tag (=QDCP) // maybe some tuneable parameters
1714 32 float1 (=1.0)
1715 32 zero ?
1716 32 float2 (=1.0)
1717 32 float3 (=1.0)
1718 32 unknown (27)
1719 32 unknown (8)
1720 32 zero ?
1721 */
1722
1723
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 4 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1724 ✗ av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1725 ✗ return AVERROR_INVALIDDATA;
1726 }
1727
1728 4 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1729
1730
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 20 while (bytestream2_get_bytes_left(&gb) > 8) {
1731
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 20 if (bytestream2_peek_be64u(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1732 (uint64_t)MKBETAG('Q','D','M','2')))
1733 4 break;
1734 16 bytestream2_skipu(&gb, 1);
1735 }
1736
1737
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 4 if (bytestream2_get_bytes_left(&gb) < 44) {
1738 ✗ av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1739 bytestream2_get_bytes_left(&gb));
1740 ✗ return AVERROR_INVALIDDATA;
1741 }
1742
1743 4 bytestream2_skipu(&gb, 8);
1744 4 size = bytestream2_get_be32u(&gb);
1745
1746
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 4 if (size > bytestream2_get_bytes_left(&gb)) {
1747 ✗ av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1748 bytestream2_get_bytes_left(&gb), size);
1749 ✗ return AVERROR_INVALIDDATA;
1750 }
1751
1752 4 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1753
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 4 if (bytestream2_get_be32u(&gb) != MKBETAG('Q','D','C','A')) {
1754 ✗ av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1755 ✗ return AVERROR_INVALIDDATA;
1756 }
1757
1758 4 bytestream2_skipu(&gb, 4);
1759
1760 4 s->nb_channels = s->channels = bytestream2_get_be32u(&gb);
1761
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 4 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1762 ✗ av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1763 ✗ return AVERROR_INVALIDDATA;
1764 }
1765 4 av_channel_layout_uninit(&avctx->ch_layout);
1766 4 av_channel_layout_default(&avctx->ch_layout, s->channels);
1767
1768 4 avctx->sample_rate = bytestream2_get_be32u(&gb);
1769 4 avctx->bit_rate = bytestream2_get_be32u(&gb);
1770 4 s->group_size = bytestream2_get_be32u(&gb);
1771 4 s->fft_size = bytestream2_get_be32u(&gb);
1772 4 s->checksum_size = bytestream2_get_be32u(&gb);
1773
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 4 if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
1774 ✗ av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
1775 ✗ return AVERROR_INVALIDDATA;
1776 }
1777
1778 4 s->fft_order = av_log2(s->fft_size) + 1;
1779
1780 // Fail on unknown fft order
1781
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 4 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1782 ✗ avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1783 ✗ return AVERROR_PATCHWELCOME;
1784 }
1785
1786 // something like max decodable tones
1787 4 s->group_order = av_log2(s->group_size) + 1;
1788 4 s->frame_size = s->group_size / 16; // 16 iterations per super block
1789
1790
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 4 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1791 ✗ return AVERROR_INVALIDDATA;
1792
1793 4 s->sub_sampling = s->fft_order - 7;
1794 4 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1795
1796
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 4 if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
1797 ✗ avpriv_request_sample(avctx, "large frames");
1798 ✗ return AVERROR_PATCHWELCOME;
1799 }
1800
1801
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 4 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1802 ✗ case 0: tmp = 40; break;
1803 ✗ case 1: tmp = 48; break;
1804 ✗ case 2: tmp = 56; break;
1805 ✗ case 3: tmp = 72; break;
1806 ✗ case 4: tmp = 80; break;
1807 4 case 5: tmp = 100;break;
1808 ✗ default: tmp=s->sub_sampling; break;
1809 }
1810 4 tmp_val = 0;
1811
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 4 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1812
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 4 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1813
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 4 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1814
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 4 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1815 4 s->cm_table_select = tmp_val;
1816
1817
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 4 if (avctx->bit_rate <= 8000)
1818 ✗ s->coeff_per_sb_select = 0;
1819
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 4 else if (avctx->bit_rate < 16000)
1820 ✗ s->coeff_per_sb_select = 1;
1821 else
1822 4 s->coeff_per_sb_select = 2;
1823
1824
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 4 if (s->fft_size != (1 << (s->fft_order - 1))) {
1825 ✗ av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1826 ✗ return AVERROR_INVALIDDATA;
1827 }
1828
1829 4 ret = av_tx_init(&s->rdft_ctx, &s->rdft_fn, AV_TX_FLOAT_RDFT, 1, 2*s->fft_size, &scale, 0);
1830
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 4 if (ret < 0)
1831 ✗ return ret;
1832
1833 4 ff_mpadsp_init(&s->mpadsp);
1834
1835 4 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1836
1837 4 ff_thread_once(&init_static_once, qdm2_init_static_data);
1838
1839 4 return 0;
1840 }
1841
1842 4 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1843 {
1844 4 QDM2Context *s = avctx->priv_data;
1845
1846 4 av_tx_uninit(&s->rdft_ctx);
1847
1848 4 return 0;
1849 }
1850
1851 2256 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1852 {
1853 int ch, i;
1854 2256 const int frame_size = (q->frame_size * q->channels);
1855
1856
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 2256 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1857 ✗ return AVERROR_INVALIDDATA;
1858
1859 /* select input buffer */
1860 2256 q->compressed_data = in;
1861 2256 q->compressed_size = q->checksum_size;
1862
1863 /* copy old block, clear new block of output samples */
1864 2256 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1865 2256 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1866
1867 /* decode block of QDM2 compressed data */
1868
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 2256 if (q->sub_packet == 0) {
1869 141 q->has_errors = 0; // zero it for a new super block
1870 141 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1871 141 int ret = qdm2_decode_super_block(q);
1872
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 141 if (ret < 0)
1873 ✗ return ret;
1874 }
1875
1876 /* parse subpackets */
1877
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 2256 if (!q->has_errors) {
1878 2256 int ret = 0;
1879
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 2256 if (q->sub_packet == 2)
1880 141 ret = qdm2_decode_fft_packets(q);
1881
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 2256 if (ret < 0)
1882 ✗ return ret;
1883
1884 2256 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1885 }
1886
1887 /* sound synthesis stage 1 (FFT) */
1888
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 6768 for (ch = 0; ch < q->channels; ch++) {
1889 4512 qdm2_calculate_fft(q, ch, q->sub_packet);
1890
1891
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 4512 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1892 ✗ SAMPLES_NEEDED_2("has errors, and C list is not empty")
1893 ✗ return AVERROR_PATCHWELCOME;
1894 }
1895 }
1896
1897 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1898
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 2256 if (!q->has_errors && q->do_synth_filter)
1899 2256 qdm2_synthesis_filter(q, q->sub_packet);
1900
1901 2256 q->sub_packet = (q->sub_packet + 1) % 16;
1902
1903 /* clip and convert output float[] to 16-bit signed samples */
1904
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 1157328 for (i = 0; i < frame_size; i++) {
1905 1155072 int value = (int)q->output_buffer[i];
1906
1907
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 1155072 if (value > SOFTCLIP_THRESHOLD)
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 245 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1909
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 1154827 else if (value < -SOFTCLIP_THRESHOLD)
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 839 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1911
1912 1155072 out[i] = value;
1913 }
1914
1915 2256 return 0;
1916 }
1917
1918 141 static int qdm2_decode_frame(AVCodecContext *avctx, AVFrame *frame,
1919 int *got_frame_ptr, AVPacket *avpkt)
1920 {
1921 141 const uint8_t *buf = avpkt->data;
1922 141 int buf_size = avpkt->size;
1923 141 QDM2Context *s = avctx->priv_data;
1924 int16_t *out;
1925 int i, ret;
1926
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 141 if(!buf)
1928 ✗ return 0;
1929
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 141 if(buf_size < s->checksum_size)
1930 ✗ return AVERROR_INVALIDDATA;
1931
1932 141 s->sub_packet = 0;
1933
1934 /* get output buffer */
1935 141 frame->nb_samples = 16 * s->frame_size;
1936
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 141 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1937 ✗ return ret;
1938 141 out = (int16_t *)frame->data[0];
1939
1940
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 2397 for (i = 0; i < 16; i++) {
1941
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 2256 if ((ret = qdm2_decode(s, buf, out)) < 0)
1942 ✗ return ret;
1943 2256 out += s->channels * s->frame_size;
1944 }
1945
1946 141 *got_frame_ptr = 1;
1947
1948 141 return s->checksum_size;
1949 }
1950
1951 const FFCodec ff_qdm2_decoder = {
1952 .p.name = "qdm2",
1953 CODEC_LONG_NAME("QDesign Music Codec 2"),
1954 .p.type = AVMEDIA_TYPE_AUDIO,
1955 .p.id = AV_CODEC_ID_QDM2,
1956 .priv_data_size = sizeof(QDM2Context),
1957 .init = qdm2_decode_init,
1958 .close = qdm2_decode_close,
1959 FF_CODEC_DECODE_CB(qdm2_decode_frame),
1960 .p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_CHANNEL_CONF,
1961 };
1962