1 /*
2 * QDM2 compatible decoder
3 * Copyright (c) 2003 Ewald Snel
4 * Copyright (c) 2005 Benjamin Larsson
5 * Copyright (c) 2005 Alex Beregszaszi
6 * Copyright (c) 2005 Roberto Togni
7 *
8 * This file is part of FFmpeg.
9 *
10 * FFmpeg is free software; you can redistribute it and/or
11 * modify it under the terms of the GNU Lesser General Public
12 * License as published by the Free Software Foundation; either
13 * version 2.1 of the License, or (at your option) any later version.
14 *
15 * FFmpeg is distributed in the hope that it will be useful,
16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
18 * Lesser General Public License for more details.
19 *
20 * You should have received a copy of the GNU Lesser General Public
21 * License along with FFmpeg; if not, write to the Free Software
22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
23 */
24
25 /**
26 * @file
27 * QDM2 decoder
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29 *
30 * The decoder is not perfect yet, there are still some distortions
31 * especially on files encoded with 16 or 8 subbands.
32 */
33
34 #include <math.h>
35 #include <stddef.h>
36 #include <stdio.h>
37
38 #include "libavutil/channel_layout.h"
39
40 #define BITSTREAM_READER_LE
41 #include "avcodec.h"
42 #include "get_bits.h"
43 #include "bytestream.h"
44 #include "internal.h"
45 #include "mpegaudio.h"
46 #include "mpegaudiodsp.h"
47 #include "rdft.h"
48
49 #include "qdm2_tablegen.h"
50
51 #define QDM2_LIST_ADD(list, size, packet) \
52 do { \
53 if (size > 0) { \
54 list[size - 1].next = &list[size]; \
55 } \
56 list[size].packet = packet; \
57 list[size].next = NULL; \
58 size++; \
59 } while(0)
60
61 // Result is 8, 16 or 30
62 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
63
64 #define FIX_NOISE_IDX(noise_idx) \
65 if ((noise_idx) >= 3840) \
66 (noise_idx) -= 3840; \
67
68 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
69
70 #define SAMPLES_NEEDED \
71 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
72
73 #define SAMPLES_NEEDED_2(why) \
74 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
75
76 #define QDM2_MAX_FRAME_SIZE 512
77
78 typedef int8_t sb_int8_array[2][30][64];
79
80 /**
81 * Subpacket
82 */
83 typedef struct QDM2SubPacket {
84 int type; ///< subpacket type
85 unsigned int size; ///< subpacket size
86 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
87 } QDM2SubPacket;
88
89 /**
90 * A node in the subpacket list
91 */
92 typedef struct QDM2SubPNode {
93 QDM2SubPacket *packet; ///< packet
94 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
95 } QDM2SubPNode;
96
97 typedef struct QDM2Complex {
98 float re;
99 float im;
100 } QDM2Complex;
101
102 typedef struct FFTTone {
103 float level;
104 QDM2Complex *complex;
105 const float *table;
106 int phase;
107 int phase_shift;
108 int duration;
109 short time_index;
110 short cutoff;
111 } FFTTone;
112
113 typedef struct FFTCoefficient {
114 int16_t sub_packet;
115 uint8_t channel;
116 int16_t offset;
117 int16_t exp;
118 uint8_t phase;
119 } FFTCoefficient;
120
121 typedef struct QDM2FFT {
122 DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
123 } QDM2FFT;
124
125 /**
126 * QDM2 decoder context
127 */
128 typedef struct QDM2Context {
129 /// Parameters from codec header, do not change during playback
130 int nb_channels; ///< number of channels
131 int channels; ///< number of channels
132 int group_size; ///< size of frame group (16 frames per group)
133 int fft_size; ///< size of FFT, in complex numbers
134 int checksum_size; ///< size of data block, used also for checksum
135
136 /// Parameters built from header parameters, do not change during playback
137 int group_order; ///< order of frame group
138 int fft_order; ///< order of FFT (actually fftorder+1)
139 int frame_size; ///< size of data frame
140 int frequency_range;
141 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
142 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
143 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
144
145 /// Packets and packet lists
146 QDM2SubPacket sub_packets[16]; ///< the packets themselves
147 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
148 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
149 int sub_packets_B; ///< number of packets on 'B' list
150 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
151 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
152
153 /// FFT and tones
154 FFTTone fft_tones[1000];
155 int fft_tone_start;
156 int fft_tone_end;
157 FFTCoefficient fft_coefs[1000];
158 int fft_coefs_index;
159 int fft_coefs_min_index[5];
160 int fft_coefs_max_index[5];
161 int fft_level_exp[6];
162 RDFTContext rdft_ctx;
163 QDM2FFT fft;
164
165 /// I/O data
166 const uint8_t *compressed_data;
167 int compressed_size;
168 float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
169
170 /// Synthesis filter
171 MPADSPContext mpadsp;
172 DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
173 int synth_buf_offset[MPA_MAX_CHANNELS];
174 DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
175 DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
176
177 /// Mixed temporary data used in decoding
178 float tone_level[MPA_MAX_CHANNELS][30][64];
179 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
180 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
181 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
182 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
183 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
184 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
185 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
186 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
187
188 // Flags
189 int has_errors; ///< packet has errors
190 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
191 int do_synth_filter; ///< used to perform or skip synthesis filter
192
193 int sub_packet;
194 int noise_idx; ///< index for dithering noise table
195 } QDM2Context;
196
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
qdm2_get_vlc(GetBitContext * gb,const VLC * vlc,int flag,int depth)201 static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
202 {
203 int value;
204
205 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
206
207 /* stage-2, 3 bits exponent escape sequence */
208 if (value-- == 0)
209 value = get_bits(gb, get_bits(gb, 3) + 1);
210
211 /* stage-3, optional */
212 if (flag) {
213 int tmp;
214
215 if (value >= 60) {
216 av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
217 return 0;
218 }
219
220 tmp= vlc_stage3_values[value];
221
222 if ((value & ~3) > 0)
223 tmp += get_bits(gb, (value >> 2));
224 value = tmp;
225 }
226
227 return value;
228 }
229
qdm2_get_se_vlc(const VLC * vlc,GetBitContext * gb,int depth)230 static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
231 {
232 int value = qdm2_get_vlc(gb, vlc, 0, depth);
233
234 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
235 }
236
237 /**
238 * QDM2 checksum
239 *
240 * @param data pointer to data to be checksummed
241 * @param length data length
242 * @param value checksum value
243 *
244 * @return 0 if checksum is OK
245 */
qdm2_packet_checksum(const uint8_t * data,int length,int value)246 static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
247 {
248 int i;
249
250 for (i = 0; i < length; i++)
251 value -= data[i];
252
253 return (uint16_t)(value & 0xffff);
254 }
255
256 /**
257 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
258 *
259 * @param gb bitreader context
260 * @param sub_packet packet under analysis
261 */
qdm2_decode_sub_packet_header(GetBitContext * gb,QDM2SubPacket * sub_packet)262 static void qdm2_decode_sub_packet_header(GetBitContext *gb,
263 QDM2SubPacket *sub_packet)
264 {
265 sub_packet->type = get_bits(gb, 8);
266
267 if (sub_packet->type == 0) {
268 sub_packet->size = 0;
269 sub_packet->data = NULL;
270 } else {
271 sub_packet->size = get_bits(gb, 8);
272
273 if (sub_packet->type & 0x80) {
274 sub_packet->size <<= 8;
275 sub_packet->size |= get_bits(gb, 8);
276 sub_packet->type &= 0x7f;
277 }
278
279 if (sub_packet->type == 0x7f)
280 sub_packet->type |= (get_bits(gb, 8) << 8);
281
282 // FIXME: this depends on bitreader-internal data
283 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
284 }
285
286 av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
287 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
288 }
289
290 /**
291 * Return node pointer to first packet of requested type in list.
292 *
293 * @param list list of subpackets to be scanned
294 * @param type type of searched subpacket
295 * @return node pointer for subpacket if found, else NULL
296 */
qdm2_search_subpacket_type_in_list(QDM2SubPNode * list,int type)297 static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
298 int type)
299 {
300 while (list && list->packet) {
301 if (list->packet->type == type)
302 return list;
303 list = list->next;
304 }
305 return NULL;
306 }
307
308 /**
309 * Replace 8 elements with their average value.
310 * Called by qdm2_decode_superblock before starting subblock decoding.
311 *
312 * @param q context
313 */
average_quantized_coeffs(QDM2Context * q)314 static void average_quantized_coeffs(QDM2Context *q)
315 {
316 int i, j, n, ch, sum;
317
318 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
319
320 for (ch = 0; ch < q->nb_channels; ch++)
321 for (i = 0; i < n; i++) {
322 sum = 0;
323
324 for (j = 0; j < 8; j++)
325 sum += q->quantized_coeffs[ch][i][j];
326
327 sum /= 8;
328 if (sum > 0)
329 sum--;
330
331 for (j = 0; j < 8; j++)
332 q->quantized_coeffs[ch][i][j] = sum;
333 }
334 }
335
336 /**
337 * Build subband samples with noise weighted by q->tone_level.
338 * Called by synthfilt_build_sb_samples.
339 *
340 * @param q context
341 * @param sb subband index
342 */
build_sb_samples_from_noise(QDM2Context * q,int sb)343 static void build_sb_samples_from_noise(QDM2Context *q, int sb)
344 {
345 int ch, j;
346
347 FIX_NOISE_IDX(q->noise_idx);
348
349 if (!q->nb_channels)
350 return;
351
352 for (ch = 0; ch < q->nb_channels; ch++) {
353 for (j = 0; j < 64; j++) {
354 q->sb_samples[ch][j * 2][sb] =
355 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
356 q->sb_samples[ch][j * 2 + 1][sb] =
357 SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
358 }
359 }
360 }
361
362 /**
363 * Called while processing data from subpackets 11 and 12.
364 * Used after making changes to coding_method array.
365 *
366 * @param sb subband index
367 * @param channels number of channels
368 * @param coding_method q->coding_method[0][0][0]
369 */
fix_coding_method_array(int sb,int channels,sb_int8_array coding_method)370 static int fix_coding_method_array(int sb, int channels,
371 sb_int8_array coding_method)
372 {
373 int j, k;
374 int ch;
375 int run, case_val;
376
377 for (ch = 0; ch < channels; ch++) {
378 for (j = 0; j < 64; ) {
379 if (coding_method[ch][sb][j] < 8)
380 return -1;
381 if ((coding_method[ch][sb][j] - 8) > 22) {
382 run = 1;
383 case_val = 8;
384 } else {
385 switch (switchtable[coding_method[ch][sb][j] - 8]) {
386 case 0: run = 10;
387 case_val = 10;
388 break;
389 case 1: run = 1;
390 case_val = 16;
391 break;
392 case 2: run = 5;
393 case_val = 24;
394 break;
395 case 3: run = 3;
396 case_val = 30;
397 break;
398 case 4: run = 1;
399 case_val = 30;
400 break;
401 case 5: run = 1;
402 case_val = 8;
403 break;
404 default: run = 1;
405 case_val = 8;
406 break;
407 }
408 }
409 for (k = 0; k < run; k++) {
410 if (j + k < 128) {
411 int sbjk = sb + (j + k) / 64;
412 if (sbjk > 29) {
413 SAMPLES_NEEDED
414 continue;
415 }
416 if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
417 if (k > 0) {
418 SAMPLES_NEEDED
419 //not debugged, almost never used
420 memset(&coding_method[ch][sb][j + k], case_val,
421 k *sizeof(int8_t));
422 memset(&coding_method[ch][sb][j + k], case_val,
423 3 * sizeof(int8_t));
424 }
425 }
426 }
427 }
428 j += run;
429 }
430 }
431 return 0;
432 }
433
434 /**
435 * Related to synthesis filter
436 * Called by process_subpacket_10
437 *
438 * @param q context
439 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
440 */
fill_tone_level_array(QDM2Context * q,int flag)441 static void fill_tone_level_array(QDM2Context *q, int flag)
442 {
443 int i, sb, ch, sb_used;
444 int tmp, tab;
445
446 for (ch = 0; ch < q->nb_channels; ch++)
447 for (sb = 0; sb < 30; sb++)
448 for (i = 0; i < 8; i++) {
449 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
450 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
451 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
452 else
453 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
454 if(tmp < 0)
455 tmp += 0xff;
456 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
457 }
458
459 sb_used = QDM2_SB_USED(q->sub_sampling);
460
461 if ((q->superblocktype_2_3 != 0) && !flag) {
462 for (sb = 0; sb < sb_used; sb++)
463 for (ch = 0; ch < q->nb_channels; ch++)
464 for (i = 0; i < 64; i++) {
465 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
466 if (q->tone_level_idx[ch][sb][i] < 0)
467 q->tone_level[ch][sb][i] = 0;
468 else
469 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
470 }
471 } else {
472 tab = q->superblocktype_2_3 ? 0 : 1;
473 for (sb = 0; sb < sb_used; sb++) {
474 if ((sb >= 4) && (sb <= 23)) {
475 for (ch = 0; ch < q->nb_channels; ch++)
476 for (i = 0; i < 64; i++) {
477 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
478 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
479 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
480 q->tone_level_idx_hi2[ch][sb - 4];
481 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
482 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
483 q->tone_level[ch][sb][i] = 0;
484 else
485 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
486 }
487 } else {
488 if (sb > 4) {
489 for (ch = 0; ch < q->nb_channels; ch++)
490 for (i = 0; i < 64; i++) {
491 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
492 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
493 q->tone_level_idx_hi2[ch][sb - 4];
494 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
495 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
496 q->tone_level[ch][sb][i] = 0;
497 else
498 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
499 }
500 } else {
501 for (ch = 0; ch < q->nb_channels; ch++)
502 for (i = 0; i < 64; i++) {
503 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
504 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
505 q->tone_level[ch][sb][i] = 0;
506 else
507 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
508 }
509 }
510 }
511 }
512 }
513 }
514
515 /**
516 * Related to synthesis filter
517 * Called by process_subpacket_11
518 * c is built with data from subpacket 11
519 * Most of this function is used only if superblock_type_2_3 == 0,
520 * never seen it in samples.
521 *
522 * @param tone_level_idx
523 * @param tone_level_idx_temp
524 * @param coding_method q->coding_method[0][0][0]
525 * @param nb_channels number of channels
526 * @param c coming from subpacket 11, passed as 8*c
527 * @param superblocktype_2_3 flag based on superblock packet type
528 * @param cm_table_select q->cm_table_select
529 */
fill_coding_method_array(sb_int8_array tone_level_idx,sb_int8_array tone_level_idx_temp,sb_int8_array coding_method,int nb_channels,int c,int superblocktype_2_3,int cm_table_select)530 static void fill_coding_method_array(sb_int8_array tone_level_idx,
531 sb_int8_array tone_level_idx_temp,
532 sb_int8_array coding_method,
533 int nb_channels,
534 int c, int superblocktype_2_3,
535 int cm_table_select)
536 {
537 int ch, sb, j;
538 int tmp, acc, esp_40, comp;
539 int add1, add2, add3, add4;
540 int64_t multres;
541
542 if (!superblocktype_2_3) {
543 /* This case is untested, no samples available */
544 avpriv_request_sample(NULL, "!superblocktype_2_3");
545 return;
546 for (ch = 0; ch < nb_channels; ch++) {
547 for (sb = 0; sb < 30; sb++) {
548 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
549 add1 = tone_level_idx[ch][sb][j] - 10;
550 if (add1 < 0)
551 add1 = 0;
552 add2 = add3 = add4 = 0;
553 if (sb > 1) {
554 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
555 if (add2 < 0)
556 add2 = 0;
557 }
558 if (sb > 0) {
559 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
560 if (add3 < 0)
561 add3 = 0;
562 }
563 if (sb < 29) {
564 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
565 if (add4 < 0)
566 add4 = 0;
567 }
568 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
569 if (tmp < 0)
570 tmp = 0;
571 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
572 }
573 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
574 }
575 }
576 acc = 0;
577 for (ch = 0; ch < nb_channels; ch++)
578 for (sb = 0; sb < 30; sb++)
579 for (j = 0; j < 64; j++)
580 acc += tone_level_idx_temp[ch][sb][j];
581
582 multres = 0x66666667LL * (acc * 10);
583 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
584 for (ch = 0; ch < nb_channels; ch++)
585 for (sb = 0; sb < 30; sb++)
586 for (j = 0; j < 64; j++) {
587 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
588 if (comp < 0)
589 comp += 0xff;
590 comp /= 256; // signed shift
591 switch(sb) {
592 case 0:
593 if (comp < 30)
594 comp = 30;
595 comp += 15;
596 break;
597 case 1:
598 if (comp < 24)
599 comp = 24;
600 comp += 10;
601 break;
602 case 2:
603 case 3:
604 case 4:
605 if (comp < 16)
606 comp = 16;
607 }
608 if (comp <= 5)
609 tmp = 0;
610 else if (comp <= 10)
611 tmp = 10;
612 else if (comp <= 16)
613 tmp = 16;
614 else if (comp <= 24)
615 tmp = -1;
616 else
617 tmp = 0;
618 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
619 }
620 for (sb = 0; sb < 30; sb++)
621 fix_coding_method_array(sb, nb_channels, coding_method);
622 for (ch = 0; ch < nb_channels; ch++)
623 for (sb = 0; sb < 30; sb++)
624 for (j = 0; j < 64; j++)
625 if (sb >= 10) {
626 if (coding_method[ch][sb][j] < 10)
627 coding_method[ch][sb][j] = 10;
628 } else {
629 if (sb >= 2) {
630 if (coding_method[ch][sb][j] < 16)
631 coding_method[ch][sb][j] = 16;
632 } else {
633 if (coding_method[ch][sb][j] < 30)
634 coding_method[ch][sb][j] = 30;
635 }
636 }
637 } else { // superblocktype_2_3 != 0
638 for (ch = 0; ch < nb_channels; ch++)
639 for (sb = 0; sb < 30; sb++)
640 for (j = 0; j < 64; j++)
641 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
642 }
643 }
644
645 /**
646 * Called by process_subpacket_11 to process more data from subpacket 11
647 * with sb 0-8.
648 * Called by process_subpacket_12 to process data from subpacket 12 with
649 * sb 8-sb_used.
650 *
651 * @param q context
652 * @param gb bitreader context
653 * @param length packet length in bits
654 * @param sb_min lower subband processed (sb_min included)
655 * @param sb_max higher subband processed (sb_max excluded)
656 */
synthfilt_build_sb_samples(QDM2Context * q,GetBitContext * gb,int length,int sb_min,int sb_max)657 static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
658 int length, int sb_min, int sb_max)
659 {
660 int sb, j, k, n, ch, run, channels;
661 int joined_stereo, zero_encoding;
662 int type34_first;
663 float type34_div = 0;
664 float type34_predictor;
665 float samples[10];
666 int sign_bits[16] = {0};
667
668 if (length == 0) {
669 // If no data use noise
670 for (sb=sb_min; sb < sb_max; sb++)
671 build_sb_samples_from_noise(q, sb);
672
673 return 0;
674 }
675
676 for (sb = sb_min; sb < sb_max; sb++) {
677 channels = q->nb_channels;
678
679 if (q->nb_channels <= 1 || sb < 12)
680 joined_stereo = 0;
681 else if (sb >= 24)
682 joined_stereo = 1;
683 else
684 joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
685
686 if (joined_stereo) {
687 if (get_bits_left(gb) >= 16)
688 for (j = 0; j < 16; j++)
689 sign_bits[j] = get_bits1(gb);
690
691 for (j = 0; j < 64; j++)
692 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
693 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
694
695 if (fix_coding_method_array(sb, q->nb_channels,
696 q->coding_method)) {
697 av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
698 build_sb_samples_from_noise(q, sb);
699 continue;
700 }
701 channels = 1;
702 }
703
704 for (ch = 0; ch < channels; ch++) {
705 FIX_NOISE_IDX(q->noise_idx);
706 zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
707 type34_predictor = 0.0;
708 type34_first = 1;
709
710 for (j = 0; j < 128; ) {
711 switch (q->coding_method[ch][sb][j / 2]) {
712 case 8:
713 if (get_bits_left(gb) >= 10) {
714 if (zero_encoding) {
715 for (k = 0; k < 5; k++) {
716 if ((j + 2 * k) >= 128)
717 break;
718 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
719 }
720 } else {
721 n = get_bits(gb, 8);
722 if (n >= 243) {
723 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
724 return AVERROR_INVALIDDATA;
725 }
726
727 for (k = 0; k < 5; k++)
728 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
729 }
730 for (k = 0; k < 5; k++)
731 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
732 } else {
733 for (k = 0; k < 10; k++)
734 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
735 }
736 run = 10;
737 break;
738
739 case 10:
740 if (get_bits_left(gb) >= 1) {
741 float f = 0.81;
742
743 if (get_bits1(gb))
744 f = -f;
745 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
746 samples[0] = f;
747 } else {
748 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
749 }
750 run = 1;
751 break;
752
753 case 16:
754 if (get_bits_left(gb) >= 10) {
755 if (zero_encoding) {
756 for (k = 0; k < 5; k++) {
757 if ((j + k) >= 128)
758 break;
759 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
760 }
761 } else {
762 n = get_bits (gb, 8);
763 if (n >= 243) {
764 av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
765 return AVERROR_INVALIDDATA;
766 }
767
768 for (k = 0; k < 5; k++)
769 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
770 }
771 } else {
772 for (k = 0; k < 5; k++)
773 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
774 }
775 run = 5;
776 break;
777
778 case 24:
779 if (get_bits_left(gb) >= 7) {
780 n = get_bits(gb, 7);
781 if (n >= 125) {
782 av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
783 return AVERROR_INVALIDDATA;
784 }
785
786 for (k = 0; k < 3; k++)
787 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
788 } else {
789 for (k = 0; k < 3; k++)
790 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
791 }
792 run = 3;
793 break;
794
795 case 30:
796 if (get_bits_left(gb) >= 4) {
797 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
798 if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
799 av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
800 return AVERROR_INVALIDDATA;
801 }
802 samples[0] = type30_dequant[index];
803 } else
804 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
805
806 run = 1;
807 break;
808
809 case 34:
810 if (get_bits_left(gb) >= 7) {
811 if (type34_first) {
812 type34_div = (float)(1 << get_bits(gb, 2));
813 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
814 type34_predictor = samples[0];
815 type34_first = 0;
816 } else {
817 unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
818 if (index >= FF_ARRAY_ELEMS(type34_delta)) {
819 av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
820 return AVERROR_INVALIDDATA;
821 }
822 samples[0] = type34_delta[index] / type34_div + type34_predictor;
823 type34_predictor = samples[0];
824 }
825 } else {
826 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
827 }
828 run = 1;
829 break;
830
831 default:
832 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
833 run = 1;
834 break;
835 }
836
837 if (joined_stereo) {
838 for (k = 0; k < run && j + k < 128; k++) {
839 q->sb_samples[0][j + k][sb] =
840 q->tone_level[0][sb][(j + k) / 2] * samples[k];
841 if (q->nb_channels == 2) {
842 if (sign_bits[(j + k) / 8])
843 q->sb_samples[1][j + k][sb] =
844 q->tone_level[1][sb][(j + k) / 2] * -samples[k];
845 else
846 q->sb_samples[1][j + k][sb] =
847 q->tone_level[1][sb][(j + k) / 2] * samples[k];
848 }
849 }
850 } else {
851 for (k = 0; k < run; k++)
852 if ((j + k) < 128)
853 q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
854 }
855
856 j += run;
857 } // j loop
858 } // channel loop
859 } // subband loop
860 return 0;
861 }
862
863 /**
864 * Init the first element of a channel in quantized_coeffs with data
865 * from packet 10 (quantized_coeffs[ch][0]).
866 * This is similar to process_subpacket_9, but for a single channel
867 * and for element [0]
868 * same VLC tables as process_subpacket_9 are used.
869 *
870 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
871 * @param gb bitreader context
872 */
init_quantized_coeffs_elem0(int8_t * quantized_coeffs,GetBitContext * gb)873 static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
874 GetBitContext *gb)
875 {
876 int i, k, run, level, diff;
877
878 if (get_bits_left(gb) < 16)
879 return -1;
880 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
881
882 quantized_coeffs[0] = level;
883
884 for (i = 0; i < 7; ) {
885 if (get_bits_left(gb) < 16)
886 return -1;
887 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
888
889 if (i + run >= 8)
890 return -1;
891
892 if (get_bits_left(gb) < 16)
893 return -1;
894 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
895
896 for (k = 1; k <= run; k++)
897 quantized_coeffs[i + k] = (level + ((k * diff) / run));
898
899 level += diff;
900 i += run;
901 }
902 return 0;
903 }
904
905 /**
906 * Related to synthesis filter, process data from packet 10
907 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
908 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
909 * data from packet 10
910 *
911 * @param q context
912 * @param gb bitreader context
913 */
init_tone_level_dequantization(QDM2Context * q,GetBitContext * gb)914 static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
915 {
916 int sb, j, k, n, ch;
917
918 for (ch = 0; ch < q->nb_channels; ch++) {
919 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
920
921 if (get_bits_left(gb) < 16) {
922 memset(q->quantized_coeffs[ch][0], 0, 8);
923 break;
924 }
925 }
926
927 n = q->sub_sampling + 1;
928
929 for (sb = 0; sb < n; sb++)
930 for (ch = 0; ch < q->nb_channels; ch++)
931 for (j = 0; j < 8; j++) {
932 if (get_bits_left(gb) < 1)
933 break;
934 if (get_bits1(gb)) {
935 for (k=0; k < 8; k++) {
936 if (get_bits_left(gb) < 16)
937 break;
938 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
939 }
940 } else {
941 for (k=0; k < 8; k++)
942 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
943 }
944 }
945
946 n = QDM2_SB_USED(q->sub_sampling) - 4;
947
948 for (sb = 0; sb < n; sb++)
949 for (ch = 0; ch < q->nb_channels; ch++) {
950 if (get_bits_left(gb) < 16)
951 break;
952 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
953 if (sb > 19)
954 q->tone_level_idx_hi2[ch][sb] -= 16;
955 else
956 for (j = 0; j < 8; j++)
957 q->tone_level_idx_mid[ch][sb][j] = -16;
958 }
959
960 n = QDM2_SB_USED(q->sub_sampling) - 5;
961
962 for (sb = 0; sb < n; sb++)
963 for (ch = 0; ch < q->nb_channels; ch++)
964 for (j = 0; j < 8; j++) {
965 if (get_bits_left(gb) < 16)
966 break;
967 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
968 }
969 }
970
971 /**
972 * Process subpacket 9, init quantized_coeffs with data from it
973 *
974 * @param q context
975 * @param node pointer to node with packet
976 */
process_subpacket_9(QDM2Context * q,QDM2SubPNode * node)977 static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
978 {
979 GetBitContext gb;
980 int i, j, k, n, ch, run, level, diff;
981
982 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
983
984 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
985
986 for (i = 1; i < n; i++)
987 for (ch = 0; ch < q->nb_channels; ch++) {
988 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
989 q->quantized_coeffs[ch][i][0] = level;
990
991 for (j = 0; j < (8 - 1); ) {
992 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
993 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
994
995 if (j + run >= 8)
996 return -1;
997
998 for (k = 1; k <= run; k++)
999 q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1000
1001 level += diff;
1002 j += run;
1003 }
1004 }
1005
1006 for (ch = 0; ch < q->nb_channels; ch++)
1007 for (i = 0; i < 8; i++)
1008 q->quantized_coeffs[ch][0][i] = 0;
1009
1010 return 0;
1011 }
1012
1013 /**
1014 * Process subpacket 10 if not null, else
1015 *
1016 * @param q context
1017 * @param node pointer to node with packet
1018 */
process_subpacket_10(QDM2Context * q,QDM2SubPNode * node)1019 static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1020 {
1021 GetBitContext gb;
1022
1023 if (node) {
1024 init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1025 init_tone_level_dequantization(q, &gb);
1026 fill_tone_level_array(q, 1);
1027 } else {
1028 fill_tone_level_array(q, 0);
1029 }
1030 }
1031
1032 /**
1033 * Process subpacket 11
1034 *
1035 * @param q context
1036 * @param node pointer to node with packet
1037 */
process_subpacket_11(QDM2Context * q,QDM2SubPNode * node)1038 static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1039 {
1040 GetBitContext gb;
1041 int length = 0;
1042
1043 if (node) {
1044 length = node->packet->size * 8;
1045 init_get_bits(&gb, node->packet->data, length);
1046 }
1047
1048 if (length >= 32) {
1049 int c = get_bits(&gb, 13);
1050
1051 if (c > 3)
1052 fill_coding_method_array(q->tone_level_idx,
1053 q->tone_level_idx_temp, q->coding_method,
1054 q->nb_channels, 8 * c,
1055 q->superblocktype_2_3, q->cm_table_select);
1056 }
1057
1058 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1059 }
1060
1061 /**
1062 * Process subpacket 12
1063 *
1064 * @param q context
1065 * @param node pointer to node with packet
1066 */
process_subpacket_12(QDM2Context * q,QDM2SubPNode * node)1067 static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1068 {
1069 GetBitContext gb;
1070 int length = 0;
1071
1072 if (node) {
1073 length = node->packet->size * 8;
1074 init_get_bits(&gb, node->packet->data, length);
1075 }
1076
1077 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1078 }
1079
1080 /**
1081 * Process new subpackets for synthesis filter
1082 *
1083 * @param q context
1084 * @param list list with synthesis filter packets (list D)
1085 */
process_synthesis_subpackets(QDM2Context * q,QDM2SubPNode * list)1086 static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1087 {
1088 QDM2SubPNode *nodes[4];
1089
1090 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1091 if (nodes[0])
1092 process_subpacket_9(q, nodes[0]);
1093
1094 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1095 if (nodes[1])
1096 process_subpacket_10(q, nodes[1]);
1097 else
1098 process_subpacket_10(q, NULL);
1099
1100 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1101 if (nodes[0] && nodes[1] && nodes[2])
1102 process_subpacket_11(q, nodes[2]);
1103 else
1104 process_subpacket_11(q, NULL);
1105
1106 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1107 if (nodes[0] && nodes[1] && nodes[3])
1108 process_subpacket_12(q, nodes[3]);
1109 else
1110 process_subpacket_12(q, NULL);
1111 }
1112
1113 /**
1114 * Decode superblock, fill packet lists.
1115 *
1116 * @param q context
1117 */
qdm2_decode_super_block(QDM2Context * q)1118 static void qdm2_decode_super_block(QDM2Context *q)
1119 {
1120 GetBitContext gb;
1121 QDM2SubPacket header, *packet;
1122 int i, packet_bytes, sub_packet_size, sub_packets_D;
1123 unsigned int next_index = 0;
1124
1125 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1126 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1127 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1128
1129 q->sub_packets_B = 0;
1130 sub_packets_D = 0;
1131
1132 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1133
1134 init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1135 qdm2_decode_sub_packet_header(&gb, &header);
1136
1137 if (header.type < 2 || header.type >= 8) {
1138 q->has_errors = 1;
1139 av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1140 return;
1141 }
1142
1143 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1144 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1145
1146 init_get_bits(&gb, header.data, header.size * 8);
1147
1148 if (header.type == 2 || header.type == 4 || header.type == 5) {
1149 int csum = 257 * get_bits(&gb, 8);
1150 csum += 2 * get_bits(&gb, 8);
1151
1152 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1153
1154 if (csum != 0) {
1155 q->has_errors = 1;
1156 av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1157 return;
1158 }
1159 }
1160
1161 q->sub_packet_list_B[0].packet = NULL;
1162 q->sub_packet_list_D[0].packet = NULL;
1163
1164 for (i = 0; i < 6; i++)
1165 if (--q->fft_level_exp[i] < 0)
1166 q->fft_level_exp[i] = 0;
1167
1168 for (i = 0; packet_bytes > 0; i++) {
1169 int j;
1170
1171 if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1172 SAMPLES_NEEDED_2("too many packet bytes");
1173 return;
1174 }
1175
1176 q->sub_packet_list_A[i].next = NULL;
1177
1178 if (i > 0) {
1179 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1180
1181 /* seek to next block */
1182 init_get_bits(&gb, header.data, header.size * 8);
1183 skip_bits(&gb, next_index * 8);
1184
1185 if (next_index >= header.size)
1186 break;
1187 }
1188
1189 /* decode subpacket */
1190 packet = &q->sub_packets[i];
1191 qdm2_decode_sub_packet_header(&gb, packet);
1192 next_index = packet->size + get_bits_count(&gb) / 8;
1193 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1194
1195 if (packet->type == 0)
1196 break;
1197
1198 if (sub_packet_size > packet_bytes) {
1199 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1200 break;
1201 packet->size += packet_bytes - sub_packet_size;
1202 }
1203
1204 packet_bytes -= sub_packet_size;
1205
1206 /* add subpacket to 'all subpackets' list */
1207 q->sub_packet_list_A[i].packet = packet;
1208
1209 /* add subpacket to related list */
1210 if (packet->type == 8) {
1211 SAMPLES_NEEDED_2("packet type 8");
1212 return;
1213 } else if (packet->type >= 9 && packet->type <= 12) {
1214 /* packets for MPEG Audio like Synthesis Filter */
1215 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1216 } else if (packet->type == 13) {
1217 for (j = 0; j < 6; j++)
1218 q->fft_level_exp[j] = get_bits(&gb, 6);
1219 } else if (packet->type == 14) {
1220 for (j = 0; j < 6; j++)
1221 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1222 } else if (packet->type == 15) {
1223 SAMPLES_NEEDED_2("packet type 15")
1224 return;
1225 } else if (packet->type >= 16 && packet->type < 48 &&
1226 !fft_subpackets[packet->type - 16]) {
1227 /* packets for FFT */
1228 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1229 }
1230 } // Packet bytes loop
1231
1232 if (q->sub_packet_list_D[0].packet) {
1233 process_synthesis_subpackets(q, q->sub_packet_list_D);
1234 q->do_synth_filter = 1;
1235 } else if (q->do_synth_filter) {
1236 process_subpacket_10(q, NULL);
1237 process_subpacket_11(q, NULL);
1238 process_subpacket_12(q, NULL);
1239 }
1240 }
1241
qdm2_fft_init_coefficient(QDM2Context * q,int sub_packet,int offset,int duration,int channel,int exp,int phase)1242 static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1243 int offset, int duration, int channel,
1244 int exp, int phase)
1245 {
1246 if (q->fft_coefs_min_index[duration] < 0)
1247 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1248
1249 q->fft_coefs[q->fft_coefs_index].sub_packet =
1250 ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1251 q->fft_coefs[q->fft_coefs_index].channel = channel;
1252 q->fft_coefs[q->fft_coefs_index].offset = offset;
1253 q->fft_coefs[q->fft_coefs_index].exp = exp;
1254 q->fft_coefs[q->fft_coefs_index].phase = phase;
1255 q->fft_coefs_index++;
1256 }
1257
qdm2_fft_decode_tones(QDM2Context * q,int duration,GetBitContext * gb,int b)1258 static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1259 GetBitContext *gb, int b)
1260 {
1261 int channel, stereo, phase, exp;
1262 int local_int_4, local_int_8, stereo_phase, local_int_10;
1263 int local_int_14, stereo_exp, local_int_20, local_int_28;
1264 int n, offset;
1265
1266 local_int_4 = 0;
1267 local_int_28 = 0;
1268 local_int_20 = 2;
1269 local_int_8 = (4 - duration);
1270 local_int_10 = 1 << (q->group_order - duration - 1);
1271 offset = 1;
1272
1273 while (get_bits_left(gb)>0) {
1274 if (q->superblocktype_2_3) {
1275 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1276 if (get_bits_left(gb)<0) {
1277 if(local_int_4 < q->group_size)
1278 av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1279 return;
1280 }
1281 offset = 1;
1282 if (n == 0) {
1283 local_int_4 += local_int_10;
1284 local_int_28 += (1 << local_int_8);
1285 } else {
1286 local_int_4 += 8 * local_int_10;
1287 local_int_28 += (8 << local_int_8);
1288 }
1289 }
1290 offset += (n - 2);
1291 } else {
1292 if (local_int_10 <= 2) {
1293 av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
1294 return;
1295 }
1296 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1297 while (offset >= (local_int_10 - 1)) {
1298 offset += (1 - (local_int_10 - 1));
1299 local_int_4 += local_int_10;
1300 local_int_28 += (1 << local_int_8);
1301 }
1302 }
1303
1304 if (local_int_4 >= q->group_size)
1305 return;
1306
1307 local_int_14 = (offset >> local_int_8);
1308 if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1309 return;
1310
1311 if (q->nb_channels > 1) {
1312 channel = get_bits1(gb);
1313 stereo = get_bits1(gb);
1314 } else {
1315 channel = 0;
1316 stereo = 0;
1317 }
1318
1319 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1320 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1321 exp = (exp < 0) ? 0 : exp;
1322
1323 phase = get_bits(gb, 3);
1324 stereo_exp = 0;
1325 stereo_phase = 0;
1326
1327 if (stereo) {
1328 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1329 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1330 if (stereo_phase < 0)
1331 stereo_phase += 8;
1332 }
1333
1334 if (q->frequency_range > (local_int_14 + 1)) {
1335 int sub_packet = (local_int_20 + local_int_28);
1336
1337 if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
1338 return;
1339
1340 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1341 channel, exp, phase);
1342 if (stereo)
1343 qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1344 1 - channel,
1345 stereo_exp, stereo_phase);
1346 }
1347 offset++;
1348 }
1349 }
1350
qdm2_decode_fft_packets(QDM2Context * q)1351 static void qdm2_decode_fft_packets(QDM2Context *q)
1352 {
1353 int i, j, min, max, value, type, unknown_flag;
1354 GetBitContext gb;
1355
1356 if (!q->sub_packet_list_B[0].packet)
1357 return;
1358
1359 /* reset minimum indexes for FFT coefficients */
1360 q->fft_coefs_index = 0;
1361 for (i = 0; i < 5; i++)
1362 q->fft_coefs_min_index[i] = -1;
1363
1364 /* process subpackets ordered by type, largest type first */
1365 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1366 QDM2SubPacket *packet = NULL;
1367
1368 /* find subpacket with largest type less than max */
1369 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1370 value = q->sub_packet_list_B[j].packet->type;
1371 if (value > min && value < max) {
1372 min = value;
1373 packet = q->sub_packet_list_B[j].packet;
1374 }
1375 }
1376
1377 max = min;
1378
1379 /* check for errors (?) */
1380 if (!packet)
1381 return;
1382
1383 if (i == 0 &&
1384 (packet->type < 16 || packet->type >= 48 ||
1385 fft_subpackets[packet->type - 16]))
1386 return;
1387
1388 /* decode FFT tones */
1389 init_get_bits(&gb, packet->data, packet->size * 8);
1390
1391 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1392 unknown_flag = 1;
1393 else
1394 unknown_flag = 0;
1395
1396 type = packet->type;
1397
1398 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1399 int duration = q->sub_sampling + 5 - (type & 15);
1400
1401 if (duration >= 0 && duration < 4)
1402 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1403 } else if (type == 31) {
1404 for (j = 0; j < 4; j++)
1405 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1406 } else if (type == 46) {
1407 for (j = 0; j < 6; j++)
1408 q->fft_level_exp[j] = get_bits(&gb, 6);
1409 for (j = 0; j < 4; j++)
1410 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1411 }
1412 } // Loop on B packets
1413
1414 /* calculate maximum indexes for FFT coefficients */
1415 for (i = 0, j = -1; i < 5; i++)
1416 if (q->fft_coefs_min_index[i] >= 0) {
1417 if (j >= 0)
1418 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1419 j = i;
1420 }
1421 if (j >= 0)
1422 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1423 }
1424
qdm2_fft_generate_tone(QDM2Context * q,FFTTone * tone)1425 static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1426 {
1427 float level, f[6];
1428 int i;
1429 QDM2Complex c;
1430 const double iscale = 2.0 * M_PI / 512.0;
1431
1432 tone->phase += tone->phase_shift;
1433
1434 /* calculate current level (maximum amplitude) of tone */
1435 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1436 c.im = level * sin(tone->phase * iscale);
1437 c.re = level * cos(tone->phase * iscale);
1438
1439 /* generate FFT coefficients for tone */
1440 if (tone->duration >= 3 || tone->cutoff >= 3) {
1441 tone->complex[0].im += c.im;
1442 tone->complex[0].re += c.re;
1443 tone->complex[1].im -= c.im;
1444 tone->complex[1].re -= c.re;
1445 } else {
1446 f[1] = -tone->table[4];
1447 f[0] = tone->table[3] - tone->table[0];
1448 f[2] = 1.0 - tone->table[2] - tone->table[3];
1449 f[3] = tone->table[1] + tone->table[4] - 1.0;
1450 f[4] = tone->table[0] - tone->table[1];
1451 f[5] = tone->table[2];
1452 for (i = 0; i < 2; i++) {
1453 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1454 c.re * f[i];
1455 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1456 c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1457 }
1458 for (i = 0; i < 4; i++) {
1459 tone->complex[i].re += c.re * f[i + 2];
1460 tone->complex[i].im += c.im * f[i + 2];
1461 }
1462 }
1463
1464 /* copy the tone if it has not yet died out */
1465 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1466 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1467 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1468 }
1469 }
1470
qdm2_fft_tone_synthesizer(QDM2Context * q,int sub_packet)1471 static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1472 {
1473 int i, j, ch;
1474 const double iscale = 0.25 * M_PI;
1475
1476 for (ch = 0; ch < q->channels; ch++) {
1477 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1478 }
1479
1480
1481 /* apply FFT tones with duration 4 (1 FFT period) */
1482 if (q->fft_coefs_min_index[4] >= 0)
1483 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1484 float level;
1485 QDM2Complex c;
1486
1487 if (q->fft_coefs[i].sub_packet != sub_packet)
1488 break;
1489
1490 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1491 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];
1492
1493 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1494 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1495 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1496 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1497 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1498 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1499 }
1500
1501 /* generate existing FFT tones */
1502 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1503 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1504 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1505 }
1506
1507 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1508 for (i = 0; i < 4; i++)
1509 if (q->fft_coefs_min_index[i] >= 0) {
1510 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1511 int offset, four_i;
1512 FFTTone tone;
1513
1514 if (q->fft_coefs[j].sub_packet != sub_packet)
1515 break;
1516
1517 four_i = (4 - i);
1518 offset = q->fft_coefs[j].offset >> four_i;
1519 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1520
1521 if (offset < q->frequency_range) {
1522 if (offset < 2)
1523 tone.cutoff = offset;
1524 else
1525 tone.cutoff = (offset >= 60) ? 3 : 2;
1526
1527 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];
1528 tone.complex = &q->fft.complex[ch][offset];
1529 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1530 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1531 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1532 tone.duration = i;
1533 tone.time_index = 0;
1534
1535 qdm2_fft_generate_tone(q, &tone);
1536 }
1537 }
1538 q->fft_coefs_min_index[i] = j;
1539 }
1540 }
1541
qdm2_calculate_fft(QDM2Context * q,int channel,int sub_packet)1542 static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1543 {
1544 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1545 float *out = q->output_buffer + channel;
1546 int i;
1547 q->fft.complex[channel][0].re *= 2.0f;
1548 q->fft.complex[channel][0].im = 0.0f;
1549 q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1550 /* add samples to output buffer */
1551 for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1552 out[0] += q->fft.complex[channel][i].re * gain;
1553 out[q->channels] += q->fft.complex[channel][i].im * gain;
1554 out += 2 * q->channels;
1555 }
1556 }
1557
1558 /**
1559 * @param q context
1560 * @param index subpacket number
1561 */
qdm2_synthesis_filter(QDM2Context * q,int index)1562 static void qdm2_synthesis_filter(QDM2Context *q, int index)
1563 {
1564 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1565
1566 /* copy sb_samples */
1567 sb_used = QDM2_SB_USED(q->sub_sampling);
1568
1569 for (ch = 0; ch < q->channels; ch++)
1570 for (i = 0; i < 8; i++)
1571 for (k = sb_used; k < SBLIMIT; k++)
1572 q->sb_samples[ch][(8 * index) + i][k] = 0;
1573
1574 for (ch = 0; ch < q->nb_channels; ch++) {
1575 float *samples_ptr = q->samples + ch;
1576
1577 for (i = 0; i < 8; i++) {
1578 ff_mpa_synth_filter_float(&q->mpadsp,
1579 q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1580 ff_mpa_synth_window_float, &dither_state,
1581 samples_ptr, q->nb_channels,
1582 q->sb_samples[ch][(8 * index) + i]);
1583 samples_ptr += 32 * q->nb_channels;
1584 }
1585 }
1586
1587 /* add samples to output buffer */
1588 sub_sampling = (4 >> q->sub_sampling);
1589
1590 for (ch = 0; ch < q->channels; ch++)
1591 for (i = 0; i < q->frame_size; i++)
1592 q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1593 }
1594
1595 /**
1596 * Init static data (does not depend on specific file)
1597 *
1598 * @param q context
1599 */
qdm2_init_static_data(void)1600 static av_cold void qdm2_init_static_data(void) {
1601 static int done;
1602
1603 if(done)
1604 return;
1605
1606 qdm2_init_vlc();
1607 ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1608 softclip_table_init();
1609 rnd_table_init();
1610 init_noise_samples();
1611
1612 done = 1;
1613 }
1614
1615 /**
1616 * Init parameters from codec extradata
1617 */
qdm2_decode_init(AVCodecContext * avctx)1618 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1619 {
1620 QDM2Context *s = avctx->priv_data;
1621 int tmp_val, tmp, size;
1622 GetByteContext gb;
1623
1624 qdm2_init_static_data();
1625
1626 /* extradata parsing
1627
1628 Structure:
1629 wave {
1630 frma (QDM2)
1631 QDCA
1632 QDCP
1633 }
1634
1635 32 size (including this field)
1636 32 tag (=frma)
1637 32 type (=QDM2 or QDMC)
1638
1639 32 size (including this field, in bytes)
1640 32 tag (=QDCA) // maybe mandatory parameters
1641 32 unknown (=1)
1642 32 channels (=2)
1643 32 samplerate (=44100)
1644 32 bitrate (=96000)
1645 32 block size (=4096)
1646 32 frame size (=256) (for one channel)
1647 32 packet size (=1300)
1648
1649 32 size (including this field, in bytes)
1650 32 tag (=QDCP) // maybe some tuneable parameters
1651 32 float1 (=1.0)
1652 32 zero ?
1653 32 float2 (=1.0)
1654 32 float3 (=1.0)
1655 32 unknown (27)
1656 32 unknown (8)
1657 32 zero ?
1658 */
1659
1660 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1661 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1662 return AVERROR_INVALIDDATA;
1663 }
1664
1665 bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1666
1667 while (bytestream2_get_bytes_left(&gb) > 8) {
1668 if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1669 (uint64_t)MKBETAG('Q','D','M','2')))
1670 break;
1671 bytestream2_skip(&gb, 1);
1672 }
1673
1674 if (bytestream2_get_bytes_left(&gb) < 12) {
1675 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1676 bytestream2_get_bytes_left(&gb));
1677 return AVERROR_INVALIDDATA;
1678 }
1679
1680 bytestream2_skip(&gb, 8);
1681 size = bytestream2_get_be32(&gb);
1682
1683 if (size > bytestream2_get_bytes_left(&gb)) {
1684 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1685 bytestream2_get_bytes_left(&gb), size);
1686 return AVERROR_INVALIDDATA;
1687 }
1688
1689 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1690 if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1691 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1692 return AVERROR_INVALIDDATA;
1693 }
1694
1695 bytestream2_skip(&gb, 4);
1696
1697 avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1698 if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1699 av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1700 return AVERROR_INVALIDDATA;
1701 }
1702 avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1703 AV_CH_LAYOUT_MONO;
1704
1705 avctx->sample_rate = bytestream2_get_be32(&gb);
1706 avctx->bit_rate = bytestream2_get_be32(&gb);
1707 s->group_size = bytestream2_get_be32(&gb);
1708 s->fft_size = bytestream2_get_be32(&gb);
1709 s->checksum_size = bytestream2_get_be32(&gb);
1710 if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
1711 av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
1712 return AVERROR_INVALIDDATA;
1713 }
1714
1715 s->fft_order = av_log2(s->fft_size) + 1;
1716
1717 // Fail on unknown fft order
1718 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1719 avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1720 return AVERROR_PATCHWELCOME;
1721 }
1722
1723 // something like max decodable tones
1724 s->group_order = av_log2(s->group_size) + 1;
1725 s->frame_size = s->group_size / 16; // 16 iterations per super block
1726
1727 if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1728 return AVERROR_INVALIDDATA;
1729
1730 s->sub_sampling = s->fft_order - 7;
1731 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1732
1733 if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
1734 avpriv_request_sample(avctx, "large frames");
1735 return AVERROR_PATCHWELCOME;
1736 }
1737
1738 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1739 case 0: tmp = 40; break;
1740 case 1: tmp = 48; break;
1741 case 2: tmp = 56; break;
1742 case 3: tmp = 72; break;
1743 case 4: tmp = 80; break;
1744 case 5: tmp = 100;break;
1745 default: tmp=s->sub_sampling; break;
1746 }
1747 tmp_val = 0;
1748 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1749 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1750 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1751 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1752 s->cm_table_select = tmp_val;
1753
1754 if (avctx->bit_rate <= 8000)
1755 s->coeff_per_sb_select = 0;
1756 else if (avctx->bit_rate < 16000)
1757 s->coeff_per_sb_select = 1;
1758 else
1759 s->coeff_per_sb_select = 2;
1760
1761 if (s->fft_size != (1 << (s->fft_order - 1))) {
1762 av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1763 return AVERROR_INVALIDDATA;
1764 }
1765
1766 ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1767 ff_mpadsp_init(&s->mpadsp);
1768
1769 avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1770
1771 return 0;
1772 }
1773
qdm2_decode_close(AVCodecContext * avctx)1774 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1775 {
1776 QDM2Context *s = avctx->priv_data;
1777
1778 ff_rdft_end(&s->rdft_ctx);
1779
1780 return 0;
1781 }
1782
qdm2_decode(QDM2Context * q,const uint8_t * in,int16_t * out)1783 static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1784 {
1785 int ch, i;
1786 const int frame_size = (q->frame_size * q->channels);
1787
1788 if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1789 return -1;
1790
1791 /* select input buffer */
1792 q->compressed_data = in;
1793 q->compressed_size = q->checksum_size;
1794
1795 /* copy old block, clear new block of output samples */
1796 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1797 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1798
1799 /* decode block of QDM2 compressed data */
1800 if (q->sub_packet == 0) {
1801 q->has_errors = 0; // zero it for a new super block
1802 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1803 qdm2_decode_super_block(q);
1804 }
1805
1806 /* parse subpackets */
1807 if (!q->has_errors) {
1808 if (q->sub_packet == 2)
1809 qdm2_decode_fft_packets(q);
1810
1811 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1812 }
1813
1814 /* sound synthesis stage 1 (FFT) */
1815 for (ch = 0; ch < q->channels; ch++) {
1816 qdm2_calculate_fft(q, ch, q->sub_packet);
1817
1818 if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1819 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1820 return -1;
1821 }
1822 }
1823
1824 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1825 if (!q->has_errors && q->do_synth_filter)
1826 qdm2_synthesis_filter(q, q->sub_packet);
1827
1828 q->sub_packet = (q->sub_packet + 1) % 16;
1829
1830 /* clip and convert output float[] to 16-bit signed samples */
1831 for (i = 0; i < frame_size; i++) {
1832 int value = (int)q->output_buffer[i];
1833
1834 if (value > SOFTCLIP_THRESHOLD)
1835 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1836 else if (value < -SOFTCLIP_THRESHOLD)
1837 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1838
1839 out[i] = value;
1840 }
1841
1842 return 0;
1843 }
1844
qdm2_decode_frame(AVCodecContext * avctx,void * data,int * got_frame_ptr,AVPacket * avpkt)1845 static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1846 int *got_frame_ptr, AVPacket *avpkt)
1847 {
1848 AVFrame *frame = data;
1849 const uint8_t *buf = avpkt->data;
1850 int buf_size = avpkt->size;
1851 QDM2Context *s = avctx->priv_data;
1852 int16_t *out;
1853 int i, ret;
1854
1855 if(!buf)
1856 return 0;
1857 if(buf_size < s->checksum_size)
1858 return -1;
1859
1860 /* get output buffer */
1861 frame->nb_samples = 16 * s->frame_size;
1862 if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1863 return ret;
1864 out = (int16_t *)frame->data[0];
1865
1866 for (i = 0; i < 16; i++) {
1867 if ((ret = qdm2_decode(s, buf, out)) < 0)
1868 return ret;
1869 out += s->channels * s->frame_size;
1870 }
1871
1872 *got_frame_ptr = 1;
1873
1874 return s->checksum_size;
1875 }
1876
1877 AVCodec ff_qdm2_decoder = {
1878 .name = "qdm2",
1879 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1880 .type = AVMEDIA_TYPE_AUDIO,
1881 .id = AV_CODEC_ID_QDM2,
1882 .priv_data_size = sizeof(QDM2Context),
1883 .init = qdm2_decode_init,
1884 .close = qdm2_decode_close,
1885 .decode = qdm2_decode_frame,
1886 .capabilities = AV_CODEC_CAP_DR1,
1887 };
1888