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flacdec: change frame bps validation to return an error value if bps
[FFMpeg-mirror/lagarith.git] / libavcodec / qdm2.c
bloba3373a16d963b98550f666ec1597bfcbe2f0892f
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 libavcodec/qdm2.c
27 * QDM2 decoder
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29 * The decoder is not perfect yet, there are still some distortions
30 * especially on files encoded with 16 or 8 subbands.
31 */
32
33 #include <math.h>
34 #include <stddef.h>
35 #include <stdio.h>
36
37 #define ALT_BITSTREAM_READER_LE
38 #include "avcodec.h"
39 #include "bitstream.h"
40 #include "dsputil.h"
41 #include "mpegaudio.h"
42
43 #include "qdm2data.h"
44
45 #undef NDEBUG
46 #include <assert.h>
47
48
49 #define SOFTCLIP_THRESHOLD 27600
50 #define HARDCLIP_THRESHOLD 35716
51
52
53 #define QDM2_LIST_ADD(list, size, packet) \
54 do { \
55 if (size > 0) { \
56 list[size - 1].next = &list[size]; \
57 } \
58 list[size].packet = packet; \
59 list[size].next = NULL; \
60 size++; \
61 } while(0)
62
63 // Result is 8, 16 or 30
64 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
65
66 #define FIX_NOISE_IDX(noise_idx) \
67 if ((noise_idx) >= 3840) \
68 (noise_idx) -= 3840; \
69
70 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
71
72 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
73
74 #define SAMPLES_NEEDED \
75 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
76
77 #define SAMPLES_NEEDED_2(why) \
78 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
79
80
81 typedef int8_t sb_int8_array[2][30][64];
82
83 /**
84 * Subpacket
85 */
86 typedef struct {
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 {
101 float re;
102 float im;
103 } QDM2Complex;
104
105 typedef struct {
106 float level;
107 QDM2Complex *complex;
108 const float *table;
109 int phase;
110 int phase_shift;
111 int duration;
112 short time_index;
113 short cutoff;
114 } FFTTone;
115
116 typedef struct {
117 int16_t sub_packet;
118 uint8_t channel;
119 int16_t offset;
120 int16_t exp;
121 uint8_t phase;
122 } FFTCoefficient;
123
124 typedef struct {
125 DECLARE_ALIGNED_16(QDM2Complex, complex[MPA_MAX_CHANNELS][256]);
126 } QDM2FFT;
127
128 /**
129 * QDM2 decoder context
130 */
131 typedef struct {
132 /// Parameters from codec header, do not change during playback
133 int nb_channels; ///< number of channels
134 int channels; ///< number of channels
135 int group_size; ///< size of frame group (16 frames per group)
136 int fft_size; ///< size of FFT, in complex numbers
137 int checksum_size; ///< size of data block, used also for checksum
138
139 /// Parameters built from header parameters, do not change during playback
140 int group_order; ///< order of frame group
141 int fft_order; ///< order of FFT (actually fftorder+1)
142 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
143 int frame_size; ///< size of data frame
144 int frequency_range;
145 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
146 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
147 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
148
149 /// Packets and packet lists
150 QDM2SubPacket sub_packets[16]; ///< the packets themselves
151 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
152 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
153 int sub_packets_B; ///< number of packets on 'B' list
154 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
155 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
156
157 /// FFT and tones
158 FFTTone fft_tones[1000];
159 int fft_tone_start;
160 int fft_tone_end;
161 FFTCoefficient fft_coefs[1000];
162 int fft_coefs_index;
163 int fft_coefs_min_index[5];
164 int fft_coefs_max_index[5];
165 int fft_level_exp[6];
166 RDFTContext rdft_ctx;
167 QDM2FFT fft;
168
169 /// I/O data
170 const uint8_t *compressed_data;
171 int compressed_size;
172 float output_buffer[1024];
173
174 /// Synthesis filter
175 DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
176 int synth_buf_offset[MPA_MAX_CHANNELS];
177 DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);
178
179 /// Mixed temporary data used in decoding
180 float tone_level[MPA_MAX_CHANNELS][30][64];
181 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
182 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
183 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
184 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
185 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
186 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
187 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
188 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
189
190 // Flags
191 int has_errors; ///< packet has errors
192 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
193 int do_synth_filter; ///< used to perform or skip synthesis filter
194
195 int sub_packet;
196 int noise_idx; ///< index for dithering noise table
197 } QDM2Context;
198
199
200 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
201
202 static VLC vlc_tab_level;
203 static VLC vlc_tab_diff;
204 static VLC vlc_tab_run;
205 static VLC fft_level_exp_alt_vlc;
206 static VLC fft_level_exp_vlc;
207 static VLC fft_stereo_exp_vlc;
208 static VLC fft_stereo_phase_vlc;
209 static VLC vlc_tab_tone_level_idx_hi1;
210 static VLC vlc_tab_tone_level_idx_mid;
211 static VLC vlc_tab_tone_level_idx_hi2;
212 static VLC vlc_tab_type30;
213 static VLC vlc_tab_type34;
214 static VLC vlc_tab_fft_tone_offset[5];
215
216 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
217 static float noise_table[4096];
218 static uint8_t random_dequant_index[256][5];
219 static uint8_t random_dequant_type24[128][3];
220 static float noise_samples[128];
221
222 static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
223
224
225 static av_cold void softclip_table_init(void) {
226 int i;
227 double dfl = SOFTCLIP_THRESHOLD - 32767;
228 float delta = 1.0 / -dfl;
229 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
230 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
231 }
232
233
234 // random generated table
235 static av_cold void rnd_table_init(void) {
236 int i,j;
237 uint32_t ldw,hdw;
238 uint64_t tmp64_1;
239 uint64_t random_seed = 0;
240 float delta = 1.0 / 16384.0;
241 for(i = 0; i < 4096 ;i++) {
242 random_seed = random_seed * 214013 + 2531011;
243 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
244 }
245
246 for (i = 0; i < 256 ;i++) {
247 random_seed = 81;
248 ldw = i;
249 for (j = 0; j < 5 ;j++) {
250 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
251 ldw = (uint32_t)ldw % (uint32_t)random_seed;
252 tmp64_1 = (random_seed * 0x55555556);
253 hdw = (uint32_t)(tmp64_1 >> 32);
254 random_seed = (uint64_t)(hdw + (ldw >> 31));
255 }
256 }
257 for (i = 0; i < 128 ;i++) {
258 random_seed = 25;
259 ldw = i;
260 for (j = 0; j < 3 ;j++) {
261 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
262 ldw = (uint32_t)ldw % (uint32_t)random_seed;
263 tmp64_1 = (random_seed * 0x66666667);
264 hdw = (uint32_t)(tmp64_1 >> 33);
265 random_seed = hdw + (ldw >> 31);
266 }
267 }
268 }
269
270
271 static av_cold void init_noise_samples(void) {
272 int i;
273 int random_seed = 0;
274 float delta = 1.0 / 16384.0;
275 for (i = 0; i < 128;i++) {
276 random_seed = random_seed * 214013 + 2531011;
277 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
278 }
279 }
280
281
282 static av_cold void qdm2_init_vlc(void)
283 {
284 init_vlc (&vlc_tab_level, 8, 24,
285 vlc_tab_level_huffbits, 1, 1,
286 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
287
288 init_vlc (&vlc_tab_diff, 8, 37,
289 vlc_tab_diff_huffbits, 1, 1,
290 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
291
292 init_vlc (&vlc_tab_run, 5, 6,
293 vlc_tab_run_huffbits, 1, 1,
294 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
295
296 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
297 fft_level_exp_alt_huffbits, 1, 1,
298 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
299
300 init_vlc (&fft_level_exp_vlc, 8, 20,
301 fft_level_exp_huffbits, 1, 1,
302 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
303
304 init_vlc (&fft_stereo_exp_vlc, 6, 7,
305 fft_stereo_exp_huffbits, 1, 1,
306 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
307
308 init_vlc (&fft_stereo_phase_vlc, 6, 9,
309 fft_stereo_phase_huffbits, 1, 1,
310 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
311
312 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
313 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
314 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
315
316 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
317 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
318 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
319
320 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
321 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
322 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
323
324 init_vlc (&vlc_tab_type30, 6, 9,
325 vlc_tab_type30_huffbits, 1, 1,
326 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
327
328 init_vlc (&vlc_tab_type34, 5, 10,
329 vlc_tab_type34_huffbits, 1, 1,
330 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
331
332 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
333 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
334 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
335
336 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
337 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
338 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
339
340 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
341 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
342 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
343
344 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
345 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
346 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
347
348 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
349 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
350 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
351 }
352
353
354 /* for floating point to fixed point conversion */
355 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
356
357
358 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
359 {
360 int value;
361
362 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
363
364 /* stage-2, 3 bits exponent escape sequence */
365 if (value-- == 0)
366 value = get_bits (gb, get_bits (gb, 3) + 1);
367
368 /* stage-3, optional */
369 if (flag) {
370 int tmp = vlc_stage3_values[value];
371
372 if ((value & ~3) > 0)
373 tmp += get_bits (gb, (value >> 2));
374 value = tmp;
375 }
376
377 return value;
378 }
379
380
381 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
382 {
383 int value = qdm2_get_vlc (gb, vlc, 0, depth);
384
385 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
386 }
387
388
389 /**
390 * QDM2 checksum
391 *
392 * @param data pointer to data to be checksum'ed
393 * @param length data length
394 * @param value checksum value
395 *
396 * @return 0 if checksum is OK
397 */
398 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
399 int i;
400
401 for (i=0; i < length; i++)
402 value -= data[i];
403
404 return (uint16_t)(value & 0xffff);
405 }
406
407
408 /**
409 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
410 *
411 * @param gb bitreader context
412 * @param sub_packet packet under analysis
413 */
414 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
415 {
416 sub_packet->type = get_bits (gb, 8);
417
418 if (sub_packet->type == 0) {
419 sub_packet->size = 0;
420 sub_packet->data = NULL;
421 } else {
422 sub_packet->size = get_bits (gb, 8);
423
424 if (sub_packet->type & 0x80) {
425 sub_packet->size <<= 8;
426 sub_packet->size |= get_bits (gb, 8);
427 sub_packet->type &= 0x7f;
428 }
429
430 if (sub_packet->type == 0x7f)
431 sub_packet->type |= (get_bits (gb, 8) << 8);
432
433 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
434 }
435
436 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
437 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
438 }
439
440
441 /**
442 * Return node pointer to first packet of requested type in list.
443 *
444 * @param list list of subpackets to be scanned
445 * @param type type of searched subpacket
446 * @return node pointer for subpacket if found, else NULL
447 */
448 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
449 {
450 while (list != NULL && list->packet != NULL) {
451 if (list->packet->type == type)
452 return list;
453 list = list->next;
454 }
455 return NULL;
456 }
457
458
459 /**
460 * Replaces 8 elements with their average value.
461 * Called by qdm2_decode_superblock before starting subblock decoding.
462 *
463 * @param q context
464 */
465 static void average_quantized_coeffs (QDM2Context *q)
466 {
467 int i, j, n, ch, sum;
468
469 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
470
471 for (ch = 0; ch < q->nb_channels; ch++)
472 for (i = 0; i < n; i++) {
473 sum = 0;
474
475 for (j = 0; j < 8; j++)
476 sum += q->quantized_coeffs[ch][i][j];
477
478 sum /= 8;
479 if (sum > 0)
480 sum--;
481
482 for (j=0; j < 8; j++)
483 q->quantized_coeffs[ch][i][j] = sum;
484 }
485 }
486
487
488 /**
489 * Build subband samples with noise weighted by q->tone_level.
490 * Called by synthfilt_build_sb_samples.
491 *
492 * @param q context
493 * @param sb subband index
494 */
495 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
496 {
497 int ch, j;
498
499 FIX_NOISE_IDX(q->noise_idx);
500
501 if (!q->nb_channels)
502 return;
503
504 for (ch = 0; ch < q->nb_channels; ch++)
505 for (j = 0; j < 64; j++) {
506 q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
507 q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
508 }
509 }
510
511
512 /**
513 * Called while processing data from subpackets 11 and 12.
514 * Used after making changes to coding_method array.
515 *
516 * @param sb subband index
517 * @param channels number of channels
518 * @param coding_method q->coding_method[0][0][0]
519 */
520 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
521 {
522 int j,k;
523 int ch;
524 int run, case_val;
525 int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
526
527 for (ch = 0; ch < channels; ch++) {
528 for (j = 0; j < 64; ) {
529 if((coding_method[ch][sb][j] - 8) > 22) {
530 run = 1;
531 case_val = 8;
532 } else {
533 switch (switchtable[coding_method[ch][sb][j]-8]) {
534 case 0: run = 10; case_val = 10; break;
535 case 1: run = 1; case_val = 16; break;
536 case 2: run = 5; case_val = 24; break;
537 case 3: run = 3; case_val = 30; break;
538 case 4: run = 1; case_val = 30; break;
539 case 5: run = 1; case_val = 8; break;
540 default: run = 1; case_val = 8; break;
541 }
542 }
543 for (k = 0; k < run; k++)
544 if (j + k < 128)
545 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
546 if (k > 0) {
547 SAMPLES_NEEDED
548 //not debugged, almost never used
549 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
550 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
551 }
552 j += run;
553 }
554 }
555 }
556
557
558 /**
559 * Related to synthesis filter
560 * Called by process_subpacket_10
561 *
562 * @param q context
563 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
564 */
565 static void fill_tone_level_array (QDM2Context *q, int flag)
566 {
567 int i, sb, ch, sb_used;
568 int tmp, tab;
569
570 // This should never happen
571 if (q->nb_channels <= 0)
572 return;
573
574 for (ch = 0; ch < q->nb_channels; ch++)
575 for (sb = 0; sb < 30; sb++)
576 for (i = 0; i < 8; i++) {
577 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
578 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
579 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
580 else
581 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
582 if(tmp < 0)
583 tmp += 0xff;
584 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
585 }
586
587 sb_used = QDM2_SB_USED(q->sub_sampling);
588
589 if ((q->superblocktype_2_3 != 0) && !flag) {
590 for (sb = 0; sb < sb_used; sb++)
591 for (ch = 0; ch < q->nb_channels; ch++)
592 for (i = 0; i < 64; i++) {
593 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
594 if (q->tone_level_idx[ch][sb][i] < 0)
595 q->tone_level[ch][sb][i] = 0;
596 else
597 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
598 }
599 } else {
600 tab = q->superblocktype_2_3 ? 0 : 1;
601 for (sb = 0; sb < sb_used; sb++) {
602 if ((sb >= 4) && (sb <= 23)) {
603 for (ch = 0; ch < q->nb_channels; ch++)
604 for (i = 0; i < 64; i++) {
605 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
606 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
607 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
608 q->tone_level_idx_hi2[ch][sb - 4];
609 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
610 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
611 q->tone_level[ch][sb][i] = 0;
612 else
613 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
614 }
615 } else {
616 if (sb > 4) {
617 for (ch = 0; ch < q->nb_channels; ch++)
618 for (i = 0; i < 64; i++) {
619 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
620 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
621 q->tone_level_idx_hi2[ch][sb - 4];
622 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
623 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
624 q->tone_level[ch][sb][i] = 0;
625 else
626 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
627 }
628 } else {
629 for (ch = 0; ch < q->nb_channels; ch++)
630 for (i = 0; i < 64; i++) {
631 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
632 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
633 q->tone_level[ch][sb][i] = 0;
634 else
635 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
636 }
637 }
638 }
639 }
640 }
641
642 return;
643 }
644
645
646 /**
647 * Related to synthesis filter
648 * Called by process_subpacket_11
649 * c is built with data from subpacket 11
650 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
651 *
652 * @param tone_level_idx
653 * @param tone_level_idx_temp
654 * @param coding_method q->coding_method[0][0][0]
655 * @param nb_channels number of channels
656 * @param c coming from subpacket 11, passed as 8*c
657 * @param superblocktype_2_3 flag based on superblock packet type
658 * @param cm_table_select q->cm_table_select
659 */
660 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
661 sb_int8_array coding_method, int nb_channels,
662 int c, int superblocktype_2_3, int cm_table_select)
663 {
664 int ch, sb, j;
665 int tmp, acc, esp_40, comp;
666 int add1, add2, add3, add4;
667 int64_t multres;
668
669 // This should never happen
670 if (nb_channels <= 0)
671 return;
672
673 if (!superblocktype_2_3) {
674 /* This case is untested, no samples available */
675 SAMPLES_NEEDED
676 for (ch = 0; ch < nb_channels; ch++)
677 for (sb = 0; sb < 30; sb++) {
678 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
679 add1 = tone_level_idx[ch][sb][j] - 10;
680 if (add1 < 0)
681 add1 = 0;
682 add2 = add3 = add4 = 0;
683 if (sb > 1) {
684 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
685 if (add2 < 0)
686 add2 = 0;
687 }
688 if (sb > 0) {
689 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
690 if (add3 < 0)
691 add3 = 0;
692 }
693 if (sb < 29) {
694 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
695 if (add4 < 0)
696 add4 = 0;
697 }
698 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
699 if (tmp < 0)
700 tmp = 0;
701 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
702 }
703 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
704 }
705 acc = 0;
706 for (ch = 0; ch < nb_channels; ch++)
707 for (sb = 0; sb < 30; sb++)
708 for (j = 0; j < 64; j++)
709 acc += tone_level_idx_temp[ch][sb][j];
710 if (acc)
711 tmp = c * 256 / (acc & 0xffff);
712 multres = 0x66666667 * (acc * 10);
713 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
714 for (ch = 0; ch < nb_channels; ch++)
715 for (sb = 0; sb < 30; sb++)
716 for (j = 0; j < 64; j++) {
717 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
718 if (comp < 0)
719 comp += 0xff;
720 comp /= 256; // signed shift
721 switch(sb) {
722 case 0:
723 if (comp < 30)
724 comp = 30;
725 comp += 15;
726 break;
727 case 1:
728 if (comp < 24)
729 comp = 24;
730 comp += 10;
731 break;
732 case 2:
733 case 3:
734 case 4:
735 if (comp < 16)
736 comp = 16;
737 }
738 if (comp <= 5)
739 tmp = 0;
740 else if (comp <= 10)
741 tmp = 10;
742 else if (comp <= 16)
743 tmp = 16;
744 else if (comp <= 24)
745 tmp = -1;
746 else
747 tmp = 0;
748 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
749 }
750 for (sb = 0; sb < 30; sb++)
751 fix_coding_method_array(sb, nb_channels, coding_method);
752 for (ch = 0; ch < nb_channels; ch++)
753 for (sb = 0; sb < 30; sb++)
754 for (j = 0; j < 64; j++)
755 if (sb >= 10) {
756 if (coding_method[ch][sb][j] < 10)
757 coding_method[ch][sb][j] = 10;
758 } else {
759 if (sb >= 2) {
760 if (coding_method[ch][sb][j] < 16)
761 coding_method[ch][sb][j] = 16;
762 } else {
763 if (coding_method[ch][sb][j] < 30)
764 coding_method[ch][sb][j] = 30;
765 }
766 }
767 } else { // superblocktype_2_3 != 0
768 for (ch = 0; ch < nb_channels; ch++)
769 for (sb = 0; sb < 30; sb++)
770 for (j = 0; j < 64; j++)
771 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
772 }
773
774 return;
775 }
776
777
778 /**
779 *
780 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
781 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
782 *
783 * @param q context
784 * @param gb bitreader context
785 * @param length packet length in bits
786 * @param sb_min lower subband processed (sb_min included)
787 * @param sb_max higher subband processed (sb_max excluded)
788 */
789 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
790 {
791 int sb, j, k, n, ch, run, channels;
792 int joined_stereo, zero_encoding, chs;
793 int type34_first;
794 float type34_div = 0;
795 float type34_predictor;
796 float samples[10], sign_bits[16];
797
798 if (length == 0) {
799 // If no data use noise
800 for (sb=sb_min; sb < sb_max; sb++)
801 build_sb_samples_from_noise (q, sb);
802
803 return;
804 }
805
806 for (sb = sb_min; sb < sb_max; sb++) {
807 FIX_NOISE_IDX(q->noise_idx);
808
809 channels = q->nb_channels;
810
811 if (q->nb_channels <= 1 || sb < 12)
812 joined_stereo = 0;
813 else if (sb >= 24)
814 joined_stereo = 1;
815 else
816 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
817
818 if (joined_stereo) {
819 if (BITS_LEFT(length,gb) >= 16)
820 for (j = 0; j < 16; j++)
821 sign_bits[j] = get_bits1 (gb);
822
823 for (j = 0; j < 64; j++)
824 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
825 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
826
827 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
828 channels = 1;
829 }
830
831 for (ch = 0; ch < channels; ch++) {
832 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
833 type34_predictor = 0.0;
834 type34_first = 1;
835
836 for (j = 0; j < 128; ) {
837 switch (q->coding_method[ch][sb][j / 2]) {
838 case 8:
839 if (BITS_LEFT(length,gb) >= 10) {
840 if (zero_encoding) {
841 for (k = 0; k < 5; k++) {
842 if ((j + 2 * k) >= 128)
843 break;
844 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
845 }
846 } else {
847 n = get_bits(gb, 8);
848 for (k = 0; k < 5; k++)
849 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
850 }
851 for (k = 0; k < 5; k++)
852 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
853 } else {
854 for (k = 0; k < 10; k++)
855 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
856 }
857 run = 10;
858 break;
859
860 case 10:
861 if (BITS_LEFT(length,gb) >= 1) {
862 float f = 0.81;
863
864 if (get_bits1(gb))
865 f = -f;
866 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
867 samples[0] = f;
868 } else {
869 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
870 }
871 run = 1;
872 break;
873
874 case 16:
875 if (BITS_LEFT(length,gb) >= 10) {
876 if (zero_encoding) {
877 for (k = 0; k < 5; k++) {
878 if ((j + k) >= 128)
879 break;
880 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
881 }
882 } else {
883 n = get_bits (gb, 8);
884 for (k = 0; k < 5; k++)
885 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
886 }
887 } else {
888 for (k = 0; k < 5; k++)
889 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
890 }
891 run = 5;
892 break;
893
894 case 24:
895 if (BITS_LEFT(length,gb) >= 7) {
896 n = get_bits(gb, 7);
897 for (k = 0; k < 3; k++)
898 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
899 } else {
900 for (k = 0; k < 3; k++)
901 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
902 }
903 run = 3;
904 break;
905
906 case 30:
907 if (BITS_LEFT(length,gb) >= 4)
908 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
909 else
910 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
911
912 run = 1;
913 break;
914
915 case 34:
916 if (BITS_LEFT(length,gb) >= 7) {
917 if (type34_first) {
918 type34_div = (float)(1 << get_bits(gb, 2));
919 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
920 type34_predictor = samples[0];
921 type34_first = 0;
922 } else {
923 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
924 type34_predictor = samples[0];
925 }
926 } else {
927 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
928 }
929 run = 1;
930 break;
931
932 default:
933 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
934 run = 1;
935 break;
936 }
937
938 if (joined_stereo) {
939 float tmp[10][MPA_MAX_CHANNELS];
940
941 for (k = 0; k < run; k++) {
942 tmp[k][0] = samples[k];
943 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
944 }
945 for (chs = 0; chs < q->nb_channels; chs++)
946 for (k = 0; k < run; k++)
947 if ((j + k) < 128)
948 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
949 } else {
950 for (k = 0; k < run; k++)
951 if ((j + k) < 128)
952 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
953 }
954
955 j += run;
956 } // j loop
957 } // channel loop
958 } // subband loop
959 }
960
961
962 /**
963 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
964 * This is similar to process_subpacket_9, but for a single channel and for element [0]
965 * same VLC tables as process_subpacket_9 are used.
966 *
967 * @param q context
968 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
969 * @param gb bitreader context
970 * @param length packet length in bits
971 */
972 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
973 {
974 int i, k, run, level, diff;
975
976 if (BITS_LEFT(length,gb) < 16)
977 return;
978 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
979
980 quantized_coeffs[0] = level;
981
982 for (i = 0; i < 7; ) {
983 if (BITS_LEFT(length,gb) < 16)
984 break;
985 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
986
987 if (BITS_LEFT(length,gb) < 16)
988 break;
989 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
990
991 for (k = 1; k <= run; k++)
992 quantized_coeffs[i + k] = (level + ((k * diff) / run));
993
994 level += diff;
995 i += run;
996 }
997 }
998
999
1000 /**
1001 * Related to synthesis filter, process data from packet 10
1002 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1003 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1004 *
1005 * @param q context
1006 * @param gb bitreader context
1007 * @param length packet length in bits
1008 */
1009 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1010 {
1011 int sb, j, k, n, ch;
1012
1013 for (ch = 0; ch < q->nb_channels; ch++) {
1014 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1015
1016 if (BITS_LEFT(length,gb) < 16) {
1017 memset(q->quantized_coeffs[ch][0], 0, 8);
1018 break;
1019 }
1020 }
1021
1022 n = q->sub_sampling + 1;
1023
1024 for (sb = 0; sb < n; sb++)
1025 for (ch = 0; ch < q->nb_channels; ch++)
1026 for (j = 0; j < 8; j++) {
1027 if (BITS_LEFT(length,gb) < 1)
1028 break;
1029 if (get_bits1(gb)) {
1030 for (k=0; k < 8; k++) {
1031 if (BITS_LEFT(length,gb) < 16)
1032 break;
1033 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1034 }
1035 } else {
1036 for (k=0; k < 8; k++)
1037 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1038 }
1039 }
1040
1041 n = QDM2_SB_USED(q->sub_sampling) - 4;
1042
1043 for (sb = 0; sb < n; sb++)
1044 for (ch = 0; ch < q->nb_channels; ch++) {
1045 if (BITS_LEFT(length,gb) < 16)
1046 break;
1047 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1048 if (sb > 19)
1049 q->tone_level_idx_hi2[ch][sb] -= 16;
1050 else
1051 for (j = 0; j < 8; j++)
1052 q->tone_level_idx_mid[ch][sb][j] = -16;
1053 }
1054
1055 n = QDM2_SB_USED(q->sub_sampling) - 5;
1056
1057 for (sb = 0; sb < n; sb++)
1058 for (ch = 0; ch < q->nb_channels; ch++)
1059 for (j = 0; j < 8; j++) {
1060 if (BITS_LEFT(length,gb) < 16)
1061 break;
1062 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1063 }
1064 }
1065
1066 /**
1067 * Process subpacket 9, init quantized_coeffs with data from it
1068 *
1069 * @param q context
1070 * @param node pointer to node with packet
1071 */
1072 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1073 {
1074 GetBitContext gb;
1075 int i, j, k, n, ch, run, level, diff;
1076
1077 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1078
1079 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1080
1081 for (i = 1; i < n; i++)
1082 for (ch=0; ch < q->nb_channels; ch++) {
1083 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1084 q->quantized_coeffs[ch][i][0] = level;
1085
1086 for (j = 0; j < (8 - 1); ) {
1087 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1088 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1089
1090 for (k = 1; k <= run; k++)
1091 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1092
1093 level += diff;
1094 j += run;
1095 }
1096 }
1097
1098 for (ch = 0; ch < q->nb_channels; ch++)
1099 for (i = 0; i < 8; i++)
1100 q->quantized_coeffs[ch][0][i] = 0;
1101 }
1102
1103
1104 /**
1105 * Process subpacket 10 if not null, else
1106 *
1107 * @param q context
1108 * @param node pointer to node with packet
1109 * @param length packet length in bits
1110 */
1111 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1112 {
1113 GetBitContext gb;
1114
1115 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1116
1117 if (length != 0) {
1118 init_tone_level_dequantization(q, &gb, length);
1119 fill_tone_level_array(q, 1);
1120 } else {
1121 fill_tone_level_array(q, 0);
1122 }
1123 }
1124
1125
1126 /**
1127 * Process subpacket 11
1128 *
1129 * @param q context
1130 * @param node pointer to node with packet
1131 * @param length packet length in bit
1132 */
1133 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1134 {
1135 GetBitContext gb;
1136
1137 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1138 if (length >= 32) {
1139 int c = get_bits (&gb, 13);
1140
1141 if (c > 3)
1142 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1143 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1144 }
1145
1146 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1147 }
1148
1149
1150 /**
1151 * Process subpacket 12
1152 *
1153 * @param q context
1154 * @param node pointer to node with packet
1155 * @param length packet length in bits
1156 */
1157 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1158 {
1159 GetBitContext gb;
1160
1161 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1162 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1163 }
1164
1165 /*
1166 * Process new subpackets for synthesis filter
1167 *
1168 * @param q context
1169 * @param list list with synthesis filter packets (list D)
1170 */
1171 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1172 {
1173 QDM2SubPNode *nodes[4];
1174
1175 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1176 if (nodes[0] != NULL)
1177 process_subpacket_9(q, nodes[0]);
1178
1179 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1180 if (nodes[1] != NULL)
1181 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1182 else
1183 process_subpacket_10(q, NULL, 0);
1184
1185 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1186 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1187 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1188 else
1189 process_subpacket_11(q, NULL, 0);
1190
1191 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1192 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1193 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1194 else
1195 process_subpacket_12(q, NULL, 0);
1196 }
1197
1198
1199 /*
1200 * Decode superblock, fill packet lists.
1201 *
1202 * @param q context
1203 */
1204 static void qdm2_decode_super_block (QDM2Context *q)
1205 {
1206 GetBitContext gb;
1207 QDM2SubPacket header, *packet;
1208 int i, packet_bytes, sub_packet_size, sub_packets_D;
1209 unsigned int next_index = 0;
1210
1211 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1212 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1213 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1214
1215 q->sub_packets_B = 0;
1216 sub_packets_D = 0;
1217
1218 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1219
1220 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1221 qdm2_decode_sub_packet_header(&gb, &header);
1222
1223 if (header.type < 2 || header.type >= 8) {
1224 q->has_errors = 1;
1225 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1226 return;
1227 }
1228
1229 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1230 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1231
1232 init_get_bits(&gb, header.data, header.size*8);
1233
1234 if (header.type == 2 || header.type == 4 || header.type == 5) {
1235 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1236
1237 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1238
1239 if (csum != 0) {
1240 q->has_errors = 1;
1241 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1242 return;
1243 }
1244 }
1245
1246 q->sub_packet_list_B[0].packet = NULL;
1247 q->sub_packet_list_D[0].packet = NULL;
1248
1249 for (i = 0; i < 6; i++)
1250 if (--q->fft_level_exp[i] < 0)
1251 q->fft_level_exp[i] = 0;
1252
1253 for (i = 0; packet_bytes > 0; i++) {
1254 int j;
1255
1256 q->sub_packet_list_A[i].next = NULL;
1257
1258 if (i > 0) {
1259 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1260
1261 /* seek to next block */
1262 init_get_bits(&gb, header.data, header.size*8);
1263 skip_bits(&gb, next_index*8);
1264
1265 if (next_index >= header.size)
1266 break;
1267 }
1268
1269 /* decode subpacket */
1270 packet = &q->sub_packets[i];
1271 qdm2_decode_sub_packet_header(&gb, packet);
1272 next_index = packet->size + get_bits_count(&gb) / 8;
1273 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1274
1275 if (packet->type == 0)
1276 break;
1277
1278 if (sub_packet_size > packet_bytes) {
1279 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1280 break;
1281 packet->size += packet_bytes - sub_packet_size;
1282 }
1283
1284 packet_bytes -= sub_packet_size;
1285
1286 /* add subpacket to 'all subpackets' list */
1287 q->sub_packet_list_A[i].packet = packet;
1288
1289 /* add subpacket to related list */
1290 if (packet->type == 8) {
1291 SAMPLES_NEEDED_2("packet type 8");
1292 return;
1293 } else if (packet->type >= 9 && packet->type <= 12) {
1294 /* packets for MPEG Audio like Synthesis Filter */
1295 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1296 } else if (packet->type == 13) {
1297 for (j = 0; j < 6; j++)
1298 q->fft_level_exp[j] = get_bits(&gb, 6);
1299 } else if (packet->type == 14) {
1300 for (j = 0; j < 6; j++)
1301 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1302 } else if (packet->type == 15) {
1303 SAMPLES_NEEDED_2("packet type 15")
1304 return;
1305 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1306 /* packets for FFT */
1307 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1308 }
1309 } // Packet bytes loop
1310
1311 /* **************************************************************** */
1312 if (q->sub_packet_list_D[0].packet != NULL) {
1313 process_synthesis_subpackets(q, q->sub_packet_list_D);
1314 q->do_synth_filter = 1;
1315 } else if (q->do_synth_filter) {
1316 process_subpacket_10(q, NULL, 0);
1317 process_subpacket_11(q, NULL, 0);
1318 process_subpacket_12(q, NULL, 0);
1319 }
1320 /* **************************************************************** */
1321 }
1322
1323
1324 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1325 int offset, int duration, int channel,
1326 int exp, int phase)
1327 {
1328 if (q->fft_coefs_min_index[duration] < 0)
1329 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1330
1331 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1332 q->fft_coefs[q->fft_coefs_index].channel = channel;
1333 q->fft_coefs[q->fft_coefs_index].offset = offset;
1334 q->fft_coefs[q->fft_coefs_index].exp = exp;
1335 q->fft_coefs[q->fft_coefs_index].phase = phase;
1336 q->fft_coefs_index++;
1337 }
1338
1339
1340 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1341 {
1342 int channel, stereo, phase, exp;
1343 int local_int_4, local_int_8, stereo_phase, local_int_10;
1344 int local_int_14, stereo_exp, local_int_20, local_int_28;
1345 int n, offset;
1346
1347 local_int_4 = 0;
1348 local_int_28 = 0;
1349 local_int_20 = 2;
1350 local_int_8 = (4 - duration);
1351 local_int_10 = 1 << (q->group_order - duration - 1);
1352 offset = 1;
1353
1354 while (1) {
1355 if (q->superblocktype_2_3) {
1356 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1357 offset = 1;
1358 if (n == 0) {
1359 local_int_4 += local_int_10;
1360 local_int_28 += (1 << local_int_8);
1361 } else {
1362 local_int_4 += 8*local_int_10;
1363 local_int_28 += (8 << local_int_8);
1364 }
1365 }
1366 offset += (n - 2);
1367 } else {
1368 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1369 while (offset >= (local_int_10 - 1)) {
1370 offset += (1 - (local_int_10 - 1));
1371 local_int_4 += local_int_10;
1372 local_int_28 += (1 << local_int_8);
1373 }
1374 }
1375
1376 if (local_int_4 >= q->group_size)
1377 return;
1378
1379 local_int_14 = (offset >> local_int_8);
1380
1381 if (q->nb_channels > 1) {
1382 channel = get_bits1(gb);
1383 stereo = get_bits1(gb);
1384 } else {
1385 channel = 0;
1386 stereo = 0;
1387 }
1388
1389 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1390 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1391 exp = (exp < 0) ? 0 : exp;
1392
1393 phase = get_bits(gb, 3);
1394 stereo_exp = 0;
1395 stereo_phase = 0;
1396
1397 if (stereo) {
1398 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1399 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1400 if (stereo_phase < 0)
1401 stereo_phase += 8;
1402 }
1403
1404 if (q->frequency_range > (local_int_14 + 1)) {
1405 int sub_packet = (local_int_20 + local_int_28);
1406
1407 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1408 if (stereo)
1409 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1410 }
1411
1412 offset++;
1413 }
1414 }
1415
1416
1417 static void qdm2_decode_fft_packets (QDM2Context *q)
1418 {
1419 int i, j, min, max, value, type, unknown_flag;
1420 GetBitContext gb;
1421
1422 if (q->sub_packet_list_B[0].packet == NULL)
1423 return;
1424
1425 /* reset minimum indexes for FFT coefficients */
1426 q->fft_coefs_index = 0;
1427 for (i=0; i < 5; i++)
1428 q->fft_coefs_min_index[i] = -1;
1429
1430 /* process subpackets ordered by type, largest type first */
1431 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1432 QDM2SubPacket *packet= NULL;
1433
1434 /* find subpacket with largest type less than max */
1435 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1436 value = q->sub_packet_list_B[j].packet->type;
1437 if (value > min && value < max) {
1438 min = value;
1439 packet = q->sub_packet_list_B[j].packet;
1440 }
1441 }
1442
1443 max = min;
1444
1445 /* check for errors (?) */
1446 if (!packet)
1447 return;
1448
1449 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1450 return;
1451
1452 /* decode FFT tones */
1453 init_get_bits (&gb, packet->data, packet->size*8);
1454
1455 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1456 unknown_flag = 1;
1457 else
1458 unknown_flag = 0;
1459
1460 type = packet->type;
1461
1462 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1463 int duration = q->sub_sampling + 5 - (type & 15);
1464
1465 if (duration >= 0 && duration < 4)
1466 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 for (i = 0, j = -1; i < 5; i++)
1480 if (q->fft_coefs_min_index[i] >= 0) {
1481 if (j >= 0)
1482 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1483 j = i;
1484 }
1485 if (j >= 0)
1486 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1487 }
1488
1489
1490 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1491 {
1492 float level, f[6];
1493 int i;
1494 QDM2Complex c;
1495 const double iscale = 2.0*M_PI / 512.0;
1496
1497 tone->phase += tone->phase_shift;
1498
1499 /* calculate current level (maximum amplitude) of tone */
1500 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1501 c.im = level * sin(tone->phase*iscale);
1502 c.re = level * cos(tone->phase*iscale);
1503
1504 /* generate FFT coefficients for tone */
1505 if (tone->duration >= 3 || tone->cutoff >= 3) {
1506 tone->complex[0].im += c.im;
1507 tone->complex[0].re += c.re;
1508 tone->complex[1].im -= c.im;
1509 tone->complex[1].re -= c.re;
1510 } else {
1511 f[1] = -tone->table[4];
1512 f[0] = tone->table[3] - tone->table[0];
1513 f[2] = 1.0 - tone->table[2] - tone->table[3];
1514 f[3] = tone->table[1] + tone->table[4] - 1.0;
1515 f[4] = tone->table[0] - tone->table[1];
1516 f[5] = tone->table[2];
1517 for (i = 0; i < 2; i++) {
1518 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1519 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1520 }
1521 for (i = 0; i < 4; i++) {
1522 tone->complex[i].re += c.re * f[i+2];
1523 tone->complex[i].im += c.im * f[i+2];
1524 }
1525 }
1526
1527 /* copy the tone if it has not yet died out */
1528 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1529 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1530 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1531 }
1532 }
1533
1534
1535 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1536 {
1537 int i, j, ch;
1538 const double iscale = 0.25 * M_PI;
1539
1540 for (ch = 0; ch < q->channels; ch++) {
1541 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1542 }
1543
1544
1545 /* apply FFT tones with duration 4 (1 FFT period) */
1546 if (q->fft_coefs_min_index[4] >= 0)
1547 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1548 float level;
1549 QDM2Complex c;
1550
1551 if (q->fft_coefs[i].sub_packet != sub_packet)
1552 break;
1553
1554 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1555 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];
1556
1557 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1558 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1559 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1560 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1561 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1562 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1563 }
1564
1565 /* generate existing FFT tones */
1566 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1567 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1568 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1569 }
1570
1571 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1572 for (i = 0; i < 4; i++)
1573 if (q->fft_coefs_min_index[i] >= 0) {
1574 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1575 int offset, four_i;
1576 FFTTone tone;
1577
1578 if (q->fft_coefs[j].sub_packet != sub_packet)
1579 break;
1580
1581 four_i = (4 - i);
1582 offset = q->fft_coefs[j].offset >> four_i;
1583 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1584
1585 if (offset < q->frequency_range) {
1586 if (offset < 2)
1587 tone.cutoff = offset;
1588 else
1589 tone.cutoff = (offset >= 60) ? 3 : 2;
1590
1591 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];
1592 tone.complex = &q->fft.complex[ch][offset];
1593 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1594 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1595 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1596 tone.duration = i;
1597 tone.time_index = 0;
1598
1599 qdm2_fft_generate_tone(q, &tone);
1600 }
1601 }
1602 q->fft_coefs_min_index[i] = j;
1603 }
1604 }
1605
1606
1607 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1608 {
1609 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1610 int i;
1611 q->fft.complex[channel][0].re *= 2.0f;
1612 q->fft.complex[channel][0].im = 0.0f;
1613 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1614 /* add samples to output buffer */
1615 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1616 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1617 }
1618
1619
1620 /**
1621 * @param q context
1622 * @param index subpacket number
1623 */
1624 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1625 {
1626 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1627 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1628
1629 /* copy sb_samples */
1630 sb_used = QDM2_SB_USED(q->sub_sampling);
1631
1632 for (ch = 0; ch < q->channels; ch++)
1633 for (i = 0; i < 8; i++)
1634 for (k=sb_used; k < SBLIMIT; k++)
1635 q->sb_samples[ch][(8 * index) + i][k] = 0;
1636
1637 for (ch = 0; ch < q->nb_channels; ch++) {
1638 OUT_INT *samples_ptr = samples + ch;
1639
1640 for (i = 0; i < 8; i++) {
1641 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1642 mpa_window, &dither_state,
1643 samples_ptr, q->nb_channels,
1644 q->sb_samples[ch][(8 * index) + i]);
1645 samples_ptr += 32 * q->nb_channels;
1646 }
1647 }
1648
1649 /* add samples to output buffer */
1650 sub_sampling = (4 >> q->sub_sampling);
1651
1652 for (ch = 0; ch < q->channels; ch++)
1653 for (i = 0; i < q->frame_size; i++)
1654 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1655 }
1656
1657
1658 /**
1659 * Init static data (does not depend on specific file)
1660 *
1661 * @param q context
1662 */
1663 static av_cold void qdm2_init(QDM2Context *q) {
1664 static int initialized = 0;
1665
1666 if (initialized != 0)
1667 return;
1668 initialized = 1;
1669
1670 qdm2_init_vlc();
1671 ff_mpa_synth_init(mpa_window);
1672 softclip_table_init();
1673 rnd_table_init();
1674 init_noise_samples();
1675
1676 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1677 }
1678
1679
1680 #if 0
1681 static void dump_context(QDM2Context *q)
1682 {
1683 int i;
1684 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1685 PRINT("compressed_data",q->compressed_data);
1686 PRINT("compressed_size",q->compressed_size);
1687 PRINT("frame_size",q->frame_size);
1688 PRINT("checksum_size",q->checksum_size);
1689 PRINT("channels",q->channels);
1690 PRINT("nb_channels",q->nb_channels);
1691 PRINT("fft_frame_size",q->fft_frame_size);
1692 PRINT("fft_size",q->fft_size);
1693 PRINT("sub_sampling",q->sub_sampling);
1694 PRINT("fft_order",q->fft_order);
1695 PRINT("group_order",q->group_order);
1696 PRINT("group_size",q->group_size);
1697 PRINT("sub_packet",q->sub_packet);
1698 PRINT("frequency_range",q->frequency_range);
1699 PRINT("has_errors",q->has_errors);
1700 PRINT("fft_tone_end",q->fft_tone_end);
1701 PRINT("fft_tone_start",q->fft_tone_start);
1702 PRINT("fft_coefs_index",q->fft_coefs_index);
1703 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1704 PRINT("cm_table_select",q->cm_table_select);
1705 PRINT("noise_idx",q->noise_idx);
1706
1707 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1708 {
1709 FFTTone *t = &q->fft_tones[i];
1710
1711 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1712 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1713 // PRINT(" level", t->level);
1714 PRINT(" phase", t->phase);
1715 PRINT(" phase_shift", t->phase_shift);
1716 PRINT(" duration", t->duration);
1717 PRINT(" samples_im", t->samples_im);
1718 PRINT(" samples_re", t->samples_re);
1719 PRINT(" table", t->table);
1720 }
1721
1722 }
1723 #endif
1724
1725
1726 /**
1727 * Init parameters from codec extradata
1728 */
1729 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1730 {
1731 QDM2Context *s = avctx->priv_data;
1732 uint8_t *extradata;
1733 int extradata_size;
1734 int tmp_val, tmp, size;
1735
1736 /* extradata parsing
1737
1738 Structure:
1739 wave {
1740 frma (QDM2)
1741 QDCA
1742 QDCP
1743 }
1744
1745 32 size (including this field)
1746 32 tag (=frma)
1747 32 type (=QDM2 or QDMC)
1748
1749 32 size (including this field, in bytes)
1750 32 tag (=QDCA) // maybe mandatory parameters
1751 32 unknown (=1)
1752 32 channels (=2)
1753 32 samplerate (=44100)
1754 32 bitrate (=96000)
1755 32 block size (=4096)
1756 32 frame size (=256) (for one channel)
1757 32 packet size (=1300)
1758
1759 32 size (including this field, in bytes)
1760 32 tag (=QDCP) // maybe some tuneable parameters
1761 32 float1 (=1.0)
1762 32 zero ?
1763 32 float2 (=1.0)
1764 32 float3 (=1.0)
1765 32 unknown (27)
1766 32 unknown (8)
1767 32 zero ?
1768 */
1769
1770 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1771 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1772 return -1;
1773 }
1774
1775 extradata = avctx->extradata;
1776 extradata_size = avctx->extradata_size;
1777
1778 while (extradata_size > 7) {
1779 if (!memcmp(extradata, "frmaQDM", 7))
1780 break;
1781 extradata++;
1782 extradata_size--;
1783 }
1784
1785 if (extradata_size < 12) {
1786 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1787 extradata_size);
1788 return -1;
1789 }
1790
1791 if (memcmp(extradata, "frmaQDM", 7)) {
1792 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1793 return -1;
1794 }
1795
1796 if (extradata[7] == 'C') {
1797 // s->is_qdmc = 1;
1798 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1799 return -1;
1800 }
1801
1802 extradata += 8;
1803 extradata_size -= 8;
1804
1805 size = AV_RB32(extradata);
1806
1807 if(size > extradata_size){
1808 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1809 extradata_size, size);
1810 return -1;
1811 }
1812
1813 extradata += 4;
1814 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1815 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1816 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1817 return -1;
1818 }
1819
1820 extradata += 8;
1821
1822 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1823 extradata += 4;
1824
1825 avctx->sample_rate = AV_RB32(extradata);
1826 extradata += 4;
1827
1828 avctx->bit_rate = AV_RB32(extradata);
1829 extradata += 4;
1830
1831 s->group_size = AV_RB32(extradata);
1832 extradata += 4;
1833
1834 s->fft_size = AV_RB32(extradata);
1835 extradata += 4;
1836
1837 s->checksum_size = AV_RB32(extradata);
1838 extradata += 4;
1839
1840 s->fft_order = av_log2(s->fft_size) + 1;
1841 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1842
1843 // something like max decodable tones
1844 s->group_order = av_log2(s->group_size) + 1;
1845 s->frame_size = s->group_size / 16; // 16 iterations per super block
1846
1847 s->sub_sampling = s->fft_order - 7;
1848 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1849
1850 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1851 case 0: tmp = 40; break;
1852 case 1: tmp = 48; break;
1853 case 2: tmp = 56; break;
1854 case 3: tmp = 72; break;
1855 case 4: tmp = 80; break;
1856 case 5: tmp = 100;break;
1857 default: tmp=s->sub_sampling; break;
1858 }
1859 tmp_val = 0;
1860 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1861 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1862 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1863 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1864 s->cm_table_select = tmp_val;
1865
1866 if (s->sub_sampling == 0)
1867 tmp = 7999;
1868 else
1869 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1870 /*
1871 0: 7999 -> 0
1872 1: 20000 -> 2
1873 2: 28000 -> 2
1874 */
1875 if (tmp < 8000)
1876 s->coeff_per_sb_select = 0;
1877 else if (tmp <= 16000)
1878 s->coeff_per_sb_select = 1;
1879 else
1880 s->coeff_per_sb_select = 2;
1881
1882 // Fail on unknown fft order
1883 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1884 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1885 return -1;
1886 }
1887
1888 ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
1889
1890 qdm2_init(s);
1891
1892 avctx->sample_fmt = SAMPLE_FMT_S16;
1893
1894 // dump_context(s);
1895 return 0;
1896 }
1897
1898
1899 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1900 {
1901 QDM2Context *s = avctx->priv_data;
1902
1903 ff_rdft_end(&s->rdft_ctx);
1904
1905 return 0;
1906 }
1907
1908
1909 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1910 {
1911 int ch, i;
1912 const int frame_size = (q->frame_size * q->channels);
1913
1914 /* select input buffer */
1915 q->compressed_data = in;
1916 q->compressed_size = q->checksum_size;
1917
1918 // dump_context(q);
1919
1920 /* copy old block, clear new block of output samples */
1921 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1922 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1923
1924 /* decode block of QDM2 compressed data */
1925 if (q->sub_packet == 0) {
1926 q->has_errors = 0; // zero it for a new super block
1927 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1928 qdm2_decode_super_block(q);
1929 }
1930
1931 /* parse subpackets */
1932 if (!q->has_errors) {
1933 if (q->sub_packet == 2)
1934 qdm2_decode_fft_packets(q);
1935
1936 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1937 }
1938
1939 /* sound synthesis stage 1 (FFT) */
1940 for (ch = 0; ch < q->channels; ch++) {
1941 qdm2_calculate_fft(q, ch, q->sub_packet);
1942
1943 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1944 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1945 return;
1946 }
1947 }
1948
1949 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1950 if (!q->has_errors && q->do_synth_filter)
1951 qdm2_synthesis_filter(q, q->sub_packet);
1952
1953 q->sub_packet = (q->sub_packet + 1) % 16;
1954
1955 /* clip and convert output float[] to 16bit signed samples */
1956 for (i = 0; i < frame_size; i++) {
1957 int value = (int)q->output_buffer[i];
1958
1959 if (value > SOFTCLIP_THRESHOLD)
1960 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1961 else if (value < -SOFTCLIP_THRESHOLD)
1962 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1963
1964 out[i] = value;
1965 }
1966 }
1967
1968
1969 static int qdm2_decode_frame(AVCodecContext *avctx,
1970 void *data, int *data_size,
1971 const uint8_t *buf, int buf_size)
1972 {
1973 QDM2Context *s = avctx->priv_data;
1974
1975 if(!buf)
1976 return 0;
1977 if(buf_size < s->checksum_size)
1978 return -1;
1979
1980 *data_size = s->channels * s->frame_size * sizeof(int16_t);
1981
1982 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
1983 buf_size, buf, s->checksum_size, data, *data_size);
1984
1985 qdm2_decode(s, buf, data);
1986
1987 // reading only when next superblock found
1988 if (s->sub_packet == 0) {
1989 return s->checksum_size;
1990 }
1991
1992 return 0;
1993 }
1994
1995 AVCodec qdm2_decoder =
1996 {
1997 .name = "qdm2",
1998 .type = CODEC_TYPE_AUDIO,
1999 .id = CODEC_ID_QDM2,
2000 .priv_data_size = sizeof(QDM2Context),
2001 .init = qdm2_decode_init,
2002 .close = qdm2_decode_close,
2003 .decode = qdm2_decode_frame,
2004 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2005 };
ffmpeg Lagarith lossless decoder