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QDM2 compatible decoder
author rtognimp
date Tue, 18 Oct 2005 20:31:12 +0000
parents
children 767156b9aa0d
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2913:cc55bc1f8d92 2914:4a52affac0e0
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 library is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2 of the License, or (at your option) any later version.
12 *
13 * This library is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with this library; if not, write to the Free Software
20 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
21 *
22 */
23
24 /**
25 * @file qdm2.c
26 * QDM2 decoder
27 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
28 * The decoder is not perfect yet, there are still some distorions expecially
29 * on files encoded with 16 or 8 subbands
30 */
31
32 #include <math.h>
33 #include <stddef.h>
34 #include <stdio.h>
35
36 #define ALT_BITSTREAM_READER_LE
37 #include "avcodec.h"
38 #include "bitstream.h"
39 #include "dsputil.h"
40
41 #ifdef CONFIG_MPEGAUDIO_HP
42 #define USE_HIGHPRECISION
43 #endif
44
45 #include "mpegaudio.h"
46
47 #include "qdm2data.h"
48
49 #undef NDEBUG
50 #include <assert.h>
51
52
53 #define SOFTCLIP_THRESHOLD 27600
54 #define HARDCLIP_THRESHOLD 35716
55
56
57 #define QDM2_LIST_ADD(list, size, packet) \
58 do { \
59 if (size > 0) { \
60 list[size - 1].next = &list[size]; \
61 } \
62 list[size].packet = packet; \
63 list[size].next = NULL; \
64 size++; \
65 } while(0)
66
67 // Result is 8, 16 or 30
68 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
69
70 #define FIX_NOISE_IDX(noise_idx) \
71 if ((noise_idx) >= 3840) \
72 (noise_idx) -= 3840; \
73
74 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
75
76 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
77
78 #define SAMPLES_NEEDED \
79 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
80
81 #define SAMPLES_NEEDED_2(why) \
82 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
83
84
85 typedef int8_t sb_int8_array[2][30][64];
86
87 /**
88 * Subpacket
89 */
90 typedef struct {
91 int type; ///< subpacket type
92 unsigned int size; ///< subpacket size
93 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
94 } QDM2SubPacket;
95
96 /**
97 * A node in subpacket list
98 */
99 typedef struct _QDM2SubPNode {
100 QDM2SubPacket *packet; ///< packet
101 struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
102 } QDM2SubPNode;
103
104 typedef struct {
105 float level;
106 float *samples_im;
107 float *samples_re;
108 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 float re;
126 float im;
127 } QDM2Complex;
128
129 typedef struct {
130 QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
131 float samples_im[MPA_MAX_CHANNELS][256];
132 float samples_re[MPA_MAX_CHANNELS][256];
133 } QDM2FFT;
134
135 /**
136 * QDM2 decoder context
137 */
138 typedef struct {
139 /// Parameters from codec header, do not change during playback
140 int nb_channels; ///< number of channels
141 int channels; ///< number of channels
142 int group_size; ///< size of frame group (16 frames per group)
143 int fft_size; ///< size of FFT, in complex numbers
144 int checksum_size; ///< size of data block, used also for checksum
145
146 /// Parameters built from header parameters, do not change during playback
147 int group_order; ///< order of frame group
148 int fft_order; ///< order of FFT (actually fftorder+1)
149 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
150 int frame_size; ///< size of data frame
151 int frequency_range;
152 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
153 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
154 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
155
156 /// Packets and packet lists
157 QDM2SubPacket sub_packets[16]; ///< the packets themselves
158 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
159 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
160 int sub_packets_B; ///< number of packets on 'B' list
161 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
162 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
163
164 /// FFT and tones
165 FFTTone fft_tones[1000];
166 int fft_tone_start;
167 int fft_tone_end;
168 FFTCoefficient fft_coefs[1000];
169 int fft_coefs_index;
170 int fft_coefs_min_index[5];
171 int fft_coefs_max_index[5];
172 int fft_level_exp[6];
173 FFTContext fft_ctx;
174 FFTComplex exptab[128];
175 QDM2FFT fft;
176
177 /// I/O data
178 uint8_t *compressed_data;
179 int compressed_size;
180 float output_buffer[1024];
181
182 /// Synthesis filter
183 MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
184 int synth_buf_offset[MPA_MAX_CHANNELS];
185 int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
186
187 /// Mixed temporary data used in decoding
188 float tone_level[MPA_MAX_CHANNELS][30][64];
189 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
190 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
191 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
192 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
193 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
194 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
195 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
196 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
197
198 // Flags
199 int has_errors; ///< packet have errors
200 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
201 int do_synth_filter; ///< used to perform or skip synthesis filter
202
203 int sub_packet;
204 int noise_idx; ///< Index for dithering noise table
205 } QDM2Context;
206
207
208 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
209
210 static VLC vlc_tab_level;
211 static VLC vlc_tab_diff;
212 static VLC vlc_tab_run;
213 static VLC fft_level_exp_alt_vlc;
214 static VLC fft_level_exp_vlc;
215 static VLC fft_stereo_exp_vlc;
216 static VLC fft_stereo_phase_vlc;
217 static VLC vlc_tab_tone_level_idx_hi1;
218 static VLC vlc_tab_tone_level_idx_mid;
219 static VLC vlc_tab_tone_level_idx_hi2;
220 static VLC vlc_tab_type30;
221 static VLC vlc_tab_type34;
222 static VLC vlc_tab_fft_tone_offset[5];
223
224 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
225 static float noise_table[4096];
226 static uint8_t random_dequant_index[256][5];
227 static uint8_t random_dequant_type24[128][3];
228 static float noise_samples[128];
229
230 static MPA_INT mpa_window[512] __attribute__((aligned(16)));
231
232
233 static void softclip_table_init() {
234 int i;
235 double dfl = SOFTCLIP_THRESHOLD - 32767;
236 float delta = 1.0 / -dfl;
237 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
238 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
239 }
240
241
242 // random generated table
243 static void rnd_table_init() {
244 int i,j;
245 uint32_t ldw,hdw;
246 uint64_t tmp64_1;
247 uint64_t random_seed = 0;
248 float delta = 1.0 / 16384.0;
249 for(i = 0; i < 4096 ;i++) {
250 random_seed = random_seed * 214013 + 2531011;
251 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
252 }
253
254 for (i = 0; i < 256 ;i++) {
255 random_seed = 81;
256 ldw = i;
257 for (j = 0; j < 5 ;j++) {
258 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
259 ldw = (uint32_t)ldw % (uint32_t)random_seed;
260 tmp64_1 = (random_seed * 0x55555556);
261 hdw = (uint32_t)(tmp64_1 >> 32);
262 random_seed = (uint64_t)(hdw + (ldw >> 31));
263 }
264 }
265 for (i = 0; i < 128 ;i++) {
266 random_seed = 25;
267 ldw = i;
268 for (j = 0; j < 3 ;j++) {
269 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
270 ldw = (uint32_t)ldw % (uint32_t)random_seed;
271 tmp64_1 = (random_seed * 0x66666667);
272 hdw = (uint32_t)(tmp64_1 >> 33);
273 random_seed = hdw + (ldw >> 31);
274 }
275 }
276 }
277
278
279 static void init_noise_samples() {
280 int i;
281 int random_seed = 0;
282 float delta = 1.0 / 16384.0;
283 for (i = 0; i < 128;i++) {
284 random_seed = random_seed * 214013 + 2531011;
285 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
286 }
287 }
288
289
290 static void qdm2_init_vlc()
291 {
292 init_vlc (&vlc_tab_level, 8, 24,
293 vlc_tab_level_huffbits, 1, 1,
294 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
295
296 init_vlc (&vlc_tab_diff, 8, 37,
297 vlc_tab_diff_huffbits, 1, 1,
298 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
299
300 init_vlc (&vlc_tab_run, 5, 6,
301 vlc_tab_run_huffbits, 1, 1,
302 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
303
304 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
305 fft_level_exp_alt_huffbits, 1, 1,
306 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
307
308 init_vlc (&fft_level_exp_vlc, 8, 20,
309 fft_level_exp_huffbits, 1, 1,
310 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
311
312 init_vlc (&fft_stereo_exp_vlc, 6, 7,
313 fft_stereo_exp_huffbits, 1, 1,
314 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
315
316 init_vlc (&fft_stereo_phase_vlc, 6, 9,
317 fft_stereo_phase_huffbits, 1, 1,
318 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
319
320 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
321 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
322 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
323
324 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
325 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
326 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
327
328 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
329 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
330 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
331
332 init_vlc (&vlc_tab_type30, 6, 9,
333 vlc_tab_type30_huffbits, 1, 1,
334 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
335
336 init_vlc (&vlc_tab_type34, 5, 10,
337 vlc_tab_type34_huffbits, 1, 1,
338 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
339
340 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
341 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
342 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
343
344 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
345 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
346 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
347
348 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
349 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
350 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
351
352 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
353 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
354 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
355
356 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
357 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
358 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
359 }
360
361
362 /* for floating point to fixed point conversion */
363 static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
364
365
366 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
367 {
368 int value;
369
370 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
371
372 /* stage-2, 3 bits exponent escape sequence */
373 if (value-- == 0)
374 value = get_bits (gb, get_bits (gb, 3) + 1);
375
376 /* stage-3, optional */
377 if (flag) {
378 int tmp = vlc_stage3_values[value];
379
380 if ((value & ~3) > 0)
381 tmp += get_bits (gb, (value >> 2));
382 value = tmp;
383 }
384
385 return value;
386 }
387
388
389 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
390 {
391 int value = qdm2_get_vlc (gb, vlc, 0, depth);
392
393 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
394 }
395
396
397 /**
398 * QDM2 checksum
399 *
400 * @param data pointer to data to be checksum'ed
401 * @param length data length
402 * @param value checksum value
403 *
404 * @return 0 if checksum is ok
405 */
406 static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
407 int i;
408
409 for (i=0; i < length; i++)
410 value -= data[i];
411
412 return (uint16_t)(value & 0xffff);
413 }
414
415
416 /**
417 * Fills a QDM2SubPacket structure with packet type, size, and data pointer
418 *
419 * @param gb bitreader context
420 * @param sub_packet packet under analysis
421 */
422 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
423 {
424 sub_packet->type = get_bits (gb, 8);
425
426 if (sub_packet->type == 0) {
427 sub_packet->size = 0;
428 sub_packet->data = NULL;
429 } else {
430 sub_packet->size = get_bits (gb, 8);
431
432 if (sub_packet->type & 0x80) {
433 sub_packet->size <<= 8;
434 sub_packet->size |= get_bits (gb, 8);
435 sub_packet->type &= 0x7f;
436 }
437
438 if (sub_packet->type == 0x7f)
439 sub_packet->type |= (get_bits (gb, 8) << 8);
440
441 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
442 }
443
444 av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n",
445 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
446 }
447
448
449 /**
450 * Return node pointer to first packet of requested type in list
451 *
452 * @param list list of subpacket to be scanned
453 * @param type type of searched subpacket
454 * @return node pointer for subpacket if found, else NULL
455 */
456 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
457 {
458 while (list != NULL && list->packet != NULL) {
459 if (list->packet->type == type)
460 return list;
461 list = list->next;
462 }
463 return NULL;
464 }
465
466
467 /**
468 * Replaces 8 elements with their average value
469 * Called by qdm2_decode_superblock before starting subblocks decoding
470 *
471 * @param q context
472 */
473 static void average_quantized_coeffs (QDM2Context *q)
474 {
475 int i, j, n, ch, sum;
476
477 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
478
479 for (ch = 0; ch < q->nb_channels; ch++)
480 for (i = 0; i < n; i++) {
481 sum = 0;
482
483 for (j = 0; j < 8; j++)
484 sum += q->quantized_coeffs[ch][i][j];
485
486 sum /= 8;
487 if (sum > 0)
488 sum--;
489
490 for (j=0; j < 8; j++)
491 q->quantized_coeffs[ch][i][j] = sum;
492 }
493 }
494
495
496 /**
497 * Build subband samples with noise weighted by q->tone_level
498 * Called by synthfilt_build_sb_samples
499 *
500 * @param q context
501 * @param sb subband index
502 */
503 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
504 {
505 int ch, j;
506
507 FIX_NOISE_IDX(q->noise_idx);
508
509 if (!q->nb_channels)
510 return;
511
512 for (ch = 0; ch < q->nb_channels; ch++)
513 for (j = 0; j < 64; j++) {
514 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);
515 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);
516 }
517 }
518
519
520 /**
521 * Called while processing data from subpackets 11 and 12
522 * Used after making changes to coding_method array
523 *
524 * @param sb subband index
525 * @param channels number of channels
526 * @param coding_method q->coding_method[0][0][0]
527 */
528 void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
529 {
530 int j,k;
531 int ch;
532 int run, case_val;
533 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};
534
535 for (ch = 0; ch < channels; ch++) {
536 for (j = 0; j < 64; ) {
537 if((coding_method[ch][sb][j] - 8) > 22) {
538 run = 1;
539 case_val = 8;
540 } else {
541 switch (switchtable[coding_method[ch][sb][j]]) {
542 case 0: run = 10; case_val = 10; break;
543 case 1: run = 1; case_val = 16; break;
544 case 2: run = 5; case_val = 24; break;
545 case 3: run = 3; case_val = 30; break;
546 case 4: run = 1; case_val = 30; break;
547 case 5: run = 1; case_val = 8; break;
548 default: run = 1; case_val = 8; break;
549 }
550 }
551 for (k = 0; k < run; k++)
552 if (j + k < 128)
553 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
554 if (k > 0) {
555 SAMPLES_NEEDED
556 //not debugged, almost never used
557 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
558 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
559 }
560 j += run;
561 }
562 }
563 }
564
565
566 /**
567 * Related to synthesis filter
568 * Called by process_subpacket_10
569 *
570 * @param q context
571 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
572 */
573 static void fill_tone_level_array (QDM2Context *q, int flag)
574 {
575 int i, sb, ch, sb_used;
576 int tmp, tab;
577
578 // This should never happen
579 if (q->nb_channels <= 0)
580 return;
581
582 for (ch = 0; ch < q->nb_channels; ch++)
583 for (sb = 0; sb < 30; sb++)
584 for (i = 0; i < 8; i++) {
585 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
586 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
587 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
588 else
589 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
590 if(tmp < 0)
591 tmp += 0xff;
592 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
593 }
594
595 sb_used = QDM2_SB_USED(q->sub_sampling);
596
597 if ((q->superblocktype_2_3 != 0) && !flag) {
598 for (sb = 0; sb < sb_used; sb++)
599 for (ch = 0; ch < q->nb_channels; ch++)
600 for (i = 0; i < 64; i++) {
601 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
602 if (q->tone_level_idx[ch][sb][i] < 0)
603 q->tone_level[ch][sb][i] = 0;
604 else
605 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
606 }
607 } else {
608 tab = q->superblocktype_2_3 ? 0 : 1;
609 for (sb = 0; sb < sb_used; sb++) {
610 if ((sb >= 4) && (sb <= 23)) {
611 for (ch = 0; ch < q->nb_channels; ch++)
612 for (i = 0; i < 64; i++) {
613 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
614 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
615 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
616 q->tone_level_idx_hi2[ch][sb - 4];
617 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
618 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
619 q->tone_level[ch][sb][i] = 0;
620 else
621 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
622 }
623 } else {
624 if (sb > 4) {
625 for (ch = 0; ch < q->nb_channels; ch++)
626 for (i = 0; i < 64; i++) {
627 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
628 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
629 q->tone_level_idx_hi2[ch][sb - 4];
630 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
631 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
632 q->tone_level[ch][sb][i] = 0;
633 else
634 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
635 }
636 } else {
637 for (ch = 0; ch < q->nb_channels; ch++)
638 for (i = 0; i < 64; i++) {
639 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
640 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
641 q->tone_level[ch][sb][i] = 0;
642 else
643 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
644 }
645 }
646 }
647 }
648 }
649
650 return;
651 }
652
653
654 /**
655 * Related to synthesis filter
656 * Called by process_subpacket_11
657 * c is built with data from subpacket 11
658 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
659 *
660 * @param tone_level_idx
661 * @param tone_level_idx_temp
662 * @param coding_method q->coding_method[0][0][0]
663 * @param nb_channels number of channels
664 * @param c coming from subpacket 11, passed as 8*c
665 * @param superblocktype_2_3 flag based on superblock packet type
666 * @param cm_table_select q->cm_table_select
667 */
668 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
669 sb_int8_array coding_method, int nb_channels,
670 int c, int superblocktype_2_3, int cm_table_select)
671 {
672 int ch, sb, j;
673 int tmp, acc, esp_40, comp;
674 int add1, add2, add3, add4;
675 int64_t multres;
676
677 // This should never happen
678 if (nb_channels <= 0)
679 return;
680
681 if (!superblocktype_2_3) {
682 /* This case is untested, no samples available */
683 SAMPLES_NEEDED
684 for (ch = 0; ch < nb_channels; ch++)
685 for (sb = 0; sb < 30; sb++) {
686 for (j = 1; j < 64; j++) {
687 add1 = tone_level_idx[ch][sb][j] - 10;
688 if (add1 < 0)
689 add1 = 0;
690 add2 = add3 = add4 = 0;
691 if (sb > 1) {
692 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
693 if (add2 < 0)
694 add2 = 0;
695 }
696 if (sb > 0) {
697 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
698 if (add3 < 0)
699 add3 = 0;
700 }
701 if (sb < 29) {
702 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
703 if (add4 < 0)
704 add4 = 0;
705 }
706 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
707 if (tmp < 0)
708 tmp = 0;
709 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
710 }
711 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
712 }
713 acc = 0;
714 for (ch = 0; ch < nb_channels; ch++)
715 for (sb = 0; sb < 30; sb++)
716 for (j = 0; j < 64; j++)
717 acc += tone_level_idx_temp[ch][sb][j];
718 if (acc)
719 tmp = c * 256 / (acc & 0xffff);
720 multres = 0x66666667 * (acc * 10);
721 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
722 for (ch = 0; ch < nb_channels; ch++)
723 for (sb = 0; sb < 30; sb++)
724 for (j = 0; j < 64; j++) {
725 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
726 if (comp < 0)
727 comp += 0xff;
728 comp /= 256; // signed shift
729 switch(sb) {
730 case 0:
731 if (comp < 30)
732 comp = 30;
733 comp += 15;
734 break;
735 case 1:
736 if (comp < 24)
737 comp = 24;
738 comp += 10;
739 break;
740 case 2:
741 case 3:
742 case 4:
743 if (comp < 16)
744 comp = 16;
745 }
746 if (comp <= 5)
747 tmp = 0;
748 else if (comp <= 10)
749 tmp = 10;
750 else if (comp <= 16)
751 tmp = 16;
752 else if (comp <= 24)
753 tmp = -1;
754 else
755 tmp = 0;
756 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
757 }
758 for (sb = 0; sb < 30; sb++)
759 fix_coding_method_array(sb, nb_channels, coding_method);
760 for (ch = 0; ch < nb_channels; ch++)
761 for (sb = 0; sb < 30; sb++)
762 for (j = 0; j < 64; j++)
763 if (sb >= 10) {
764 if (coding_method[ch][sb][j] < 10)
765 coding_method[ch][sb][j] = 10;
766 } else {
767 if (sb >= 2) {
768 if (coding_method[ch][sb][j] < 16)
769 coding_method[ch][sb][j] = 16;
770 } else {
771 if (coding_method[ch][sb][j] < 30)
772 coding_method[ch][sb][j] = 30;
773 }
774 }
775 } else { // superblocktype_2_3 != 0
776 for (ch = 0; ch < nb_channels; ch++)
777 for (sb = 0; sb < 30; sb++)
778 for (j = 0; j < 64; j++)
779 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
780 }
781
782 return;
783 }
784
785
786 /**
787 *
788 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
789 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
790 *
791 * @param q context
792 * @param gb bitreader context
793 * @param length packet length in bit
794 * @param sb_min lower subband processed (sb_min included)
795 * @param sb_max higher subband processed (sb_max excluded)
796 */
797 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
798 {
799 int sb, j, k, n, ch, run, channels;
800 int joined_stereo, zero_encoding, chs;
801 int type34_first;
802 float type34_div = 0;
803 float type34_predictor;
804 float samples[10], sign_bits[16];
805
806 if (length == 0) {
807 // If no data use noise
808 for (sb=sb_min; sb < sb_max; sb++)
809 build_sb_samples_from_noise (q, sb);
810
811 return;
812 }
813
814 for (sb = sb_min; sb < sb_max; sb++) {
815 FIX_NOISE_IDX(q->noise_idx);
816
817 channels = q->nb_channels;
818
819 if (q->nb_channels <= 1 || sb < 12)
820 joined_stereo = 0;
821 else if (sb >= 24)
822 joined_stereo = 1;
823 else
824 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
825
826 if (joined_stereo) {
827 if (BITS_LEFT(length,gb) >= 16)
828 for (j = 0; j < 16; j++)
829 sign_bits[j] = get_bits1 (gb);
830
831 for (j = 0; j < 64; j++)
832 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
833 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
834
835 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
836 channels = 1;
837 }
838
839 for (ch = 0; ch < channels; ch++) {
840 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
841 type34_predictor = 0.0;
842 type34_first = 1;
843
844 for (j = 0; j < 128; ) {
845 switch (q->coding_method[ch][sb][j / 2]) {
846 case 8:
847 if (BITS_LEFT(length,gb) >= 10) {
848 if (zero_encoding) {
849 for (k = 0; k < 5; k++) {
850 if ((j + 2 * k) >= 128)
851 break;
852 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
853 }
854 } else {
855 n = get_bits(gb, 8);
856 for (k = 0; k < 5; k++)
857 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
858 }
859 for (k = 0; k < 5; k++)
860 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
861 } else {
862 for (k = 0; k < 10; k++)
863 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
864 }
865 run = 10;
866 break;
867
868 case 10:
869 if (BITS_LEFT(length,gb) >= 1) {
870 float f = 0.81;
871
872 if (get_bits1(gb))
873 f = -f;
874 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
875 samples[0] = f;
876 } else {
877 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
878 }
879 run = 1;
880 break;
881
882 case 16:
883 if (BITS_LEFT(length,gb) >= 10) {
884 if (zero_encoding) {
885 for (k = 0; k < 5; k++) {
886 if ((j + k) >= 128)
887 break;
888 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
889 }
890 } else {
891 n = get_bits (gb, 8);
892 for (k = 0; k < 5; k++)
893 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
894 }
895 } else {
896 for (k = 0; k < 5; k++)
897 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
898 }
899 run = 5;
900 break;
901
902 case 24:
903 if (BITS_LEFT(length,gb) >= 7) {
904 n = get_bits(gb, 7);
905 for (k = 0; k < 3; k++)
906 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
907 } else {
908 for (k = 0; k < 3; k++)
909 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
910 }
911 run = 3;
912 break;
913
914 case 30:
915 if (BITS_LEFT(length,gb) >= 4)
916 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
917 else
918 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
919
920 run = 1;
921 break;
922
923 case 34:
924 if (BITS_LEFT(length,gb) >= 7) {
925 if (type34_first) {
926 type34_div = (float)(1 << get_bits(gb, 2));
927 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
928 type34_predictor = samples[0];
929 type34_first = 0;
930 } else {
931 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
932 type34_predictor = samples[0];
933 }
934 } else {
935 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
936 }
937 run = 1;
938 break;
939
940 default:
941 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
942 run = 1;
943 break;
944 }
945
946 if (joined_stereo) {
947 float tmp[10][MPA_MAX_CHANNELS];
948
949 for (k = 0; k < run; k++) {
950 tmp[k][0] = samples[k];
951 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
952 }
953 for (chs = 0; chs < q->nb_channels; chs++)
954 for (k = 0; k < run; k++)
955 if ((j + k) < 128)
956 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
957 } else {
958 for (k = 0; k < run; k++)
959 if ((j + k) < 128)
960 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
961 }
962
963 j += run;
964 } // j loop
965 } // channel loop
966 } // subband loop
967 }
968
969
970 /**
971 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0])
972 * This is similar to process_subpacket_9, but for a single channel and for element [0]
973 * same VLC tables as process_subpacket_9 are used
974 *
975 * @param q context
976 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
977 * @param gb bitreader context
978 * @param length packet length in bit
979 */
980 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
981 {
982 int i, k, run, level, diff;
983
984 if (BITS_LEFT(length,gb) < 16)
985 return;
986 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
987
988 quantized_coeffs[0] = level;
989
990 for (i = 0; i < 7; ) {
991 if (BITS_LEFT(length,gb) < 16)
992 break;
993 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
994
995 if (BITS_LEFT(length,gb) < 16)
996 break;
997 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
998
999 for (k = 1; k <= run; k++)
1000 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1001
1002 level += diff;
1003 i += run;
1004 }
1005 }
1006
1007
1008 /**
1009 * Related to synthesis filter, process data from packet 10
1010 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1011 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1012 *
1013 * @param q context
1014 * @param gb bitreader context
1015 * @param length packet length in bit
1016 */
1017 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1018 {
1019 int sb, j, k, n, ch;
1020
1021 for (ch = 0; ch < q->nb_channels; ch++) {
1022 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1023
1024 if (BITS_LEFT(length,gb) < 16) {
1025 memset(q->quantized_coeffs[ch][0], 0, 8);
1026 break;
1027 }
1028 }
1029
1030 n = q->sub_sampling + 1;
1031
1032 for (sb = 0; sb < n; sb++)
1033 for (ch = 0; ch < q->nb_channels; ch++)
1034 for (j = 0; j < 8; j++) {
1035 if (BITS_LEFT(length,gb) < 1)
1036 break;
1037 if (get_bits1(gb)) {
1038 for (k=0; k < 8; k++) {
1039 if (BITS_LEFT(length,gb) < 16)
1040 break;
1041 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1042 }
1043 } else {
1044 for (k=0; k < 8; k++)
1045 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1046 }
1047 }
1048
1049 n = QDM2_SB_USED(q->sub_sampling) - 4;
1050
1051 for (sb = 0; sb < n; sb++)
1052 for (ch = 0; ch < q->nb_channels; ch++) {
1053 if (BITS_LEFT(length,gb) < 16)
1054 break;
1055 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1056 if (sb > 19)
1057 q->tone_level_idx_hi2[ch][sb] -= 16;
1058 else
1059 for (j = 0; j < 8; j++)
1060 q->tone_level_idx_mid[ch][sb][j] = -16;
1061 }
1062
1063 n = QDM2_SB_USED(q->sub_sampling) - 5;
1064
1065 for (sb = 0; sb < n; sb++)
1066 for (ch = 0; ch < q->nb_channels; ch++)
1067 for (j = 0; j < 8; j++) {
1068 if (BITS_LEFT(length,gb) < 16)
1069 break;
1070 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1071 }
1072 }
1073
1074 /**
1075 * Process subpacket 9, init quantized_coeffs with data from it
1076 *
1077 * @param q context
1078 * @param node pointer to node with packet
1079 */
1080 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1081 {
1082 GetBitContext gb;
1083 int i, j, k, n, ch, run, level, diff;
1084
1085 init_get_bits(&gb, node->packet->data, node->packet->size);
1086
1087 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1088
1089 for (i = 1; i < n; i++)
1090 for (ch=0; ch < q->nb_channels; ch++) {
1091 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1092 q->quantized_coeffs[ch][i][0] = level;
1093
1094 for (j = 0; j < (8 - 1); ) {
1095 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1096 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1097
1098 for (k = 1; k <= run; k++)
1099 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1100
1101 level += diff;
1102 j += run;
1103 }
1104 }
1105
1106 for (ch = 0; ch < q->nb_channels; ch++)
1107 for (i = 0; i < 8; i++)
1108 q->quantized_coeffs[ch][0][i] = 0;
1109 }
1110
1111
1112 /**
1113 * Process subpacket 10 if not null, else
1114 *
1115 * @param q context
1116 * @param node pointer to node with packet
1117 * @param length packet length in bit
1118 */
1119 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1120 {
1121 GetBitContext gb;
1122
1123 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
1124
1125 if (length != 0) {
1126 init_tone_level_dequantization(q, &gb, length);
1127 fill_tone_level_array(q, 1);
1128 } else {
1129 fill_tone_level_array(q, 0);
1130 }
1131 }
1132
1133
1134 /**
1135 * Process subpacket 11
1136 *
1137 * @param q context
1138 * @param node pointer to node with packet
1139 * @param length packet length in bit
1140 */
1141 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1142 {
1143 GetBitContext gb;
1144
1145 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
1146 if (length >= 32) {
1147 int c = get_bits (&gb, 13);
1148
1149 if (c > 3)
1150 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1151 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1152 }
1153
1154 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1155 }
1156
1157
1158 /**
1159 * Process subpacket 12
1160 *
1161 * @param q context
1162 * @param node pointer to node with packet
1163 * @param length packet length in bit
1164 */
1165 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1166 {
1167 GetBitContext gb;
1168
1169 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
1170 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1171 }
1172
1173 /*
1174 * Process new subpackets for synthesis filter
1175 *
1176 * @param q context
1177 * @param list list with synthesis filter packets (list D)
1178 */
1179 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1180 {
1181 QDM2SubPNode *nodes[4];
1182
1183 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1184 if (nodes[0] != NULL)
1185 process_subpacket_9(q, nodes[0]);
1186
1187 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1188 if (nodes[1] != NULL)
1189 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1190 else
1191 process_subpacket_10(q, NULL, 0);
1192
1193 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1194 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1195 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1196 else
1197 process_subpacket_11(q, NULL, 0);
1198
1199 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1200 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1201 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1202 else
1203 process_subpacket_12(q, NULL, 0);
1204 }
1205
1206
1207 /*
1208 * Decode superblock, fill packet lists
1209 *
1210 * @param q context
1211 */
1212 static void qdm2_decode_super_block (QDM2Context *q)
1213 {
1214 GetBitContext gb;
1215 QDM2SubPacket header, *packet;
1216 int i, packet_bytes, sub_packet_size, sub_packets_D;
1217 unsigned int next_index = 0;
1218
1219 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1220 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1221 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1222
1223 q->sub_packets_B = 0;
1224 sub_packets_D = 0;
1225
1226 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1227
1228 init_get_bits(&gb, q->compressed_data, q->compressed_size);
1229 qdm2_decode_sub_packet_header(&gb, &header);
1230
1231 if (header.type < 2 || header.type >= 8) {
1232 q->has_errors = 1;
1233 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1234 return;
1235 }
1236
1237 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1238 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1239
1240 init_get_bits(&gb, header.data, header.size);
1241
1242 if (header.type == 2 || header.type == 4 || header.type == 5) {
1243 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1244
1245 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1246
1247 if (csum != 0) {
1248 q->has_errors = 1;
1249 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1250 return;
1251 }
1252 }
1253
1254 q->sub_packet_list_B[0].packet = NULL;
1255 q->sub_packet_list_D[0].packet = NULL;
1256
1257 for (i = 0; i < 6; i++)
1258 if (--q->fft_level_exp[i] < 0)
1259 q->fft_level_exp[i] = 0;
1260
1261 for (i = 0; packet_bytes > 0; i++) {
1262 int j;
1263
1264 q->sub_packet_list_A[i].next = NULL;
1265
1266 if (i > 0) {
1267 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1268
1269 /* seek to next block */
1270 init_get_bits(&gb, header.data, header.size);
1271 skip_bits(&gb, next_index*8);
1272
1273 if (next_index >= header.size)
1274 break;
1275 }
1276
1277 /* decode sub packet */
1278 packet = &q->sub_packets[i];
1279 qdm2_decode_sub_packet_header(&gb, packet);
1280 next_index = packet->size + get_bits_count(&gb) / 8;
1281 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1282
1283 if (packet->type == 0)
1284 break;
1285
1286 if (sub_packet_size > packet_bytes) {
1287 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1288 break;
1289 packet->size += packet_bytes - sub_packet_size;
1290 }
1291
1292 packet_bytes -= sub_packet_size;
1293
1294 /* add sub packet to 'all sub packets' list */
1295 q->sub_packet_list_A[i].packet = packet;
1296
1297 /* add sub packet to related list */
1298 if (packet->type == 8) {
1299 SAMPLES_NEEDED_2("packet type 8");
1300 return;
1301 } else if (packet->type >= 9 && packet->type <= 12) {
1302 /* packets for MPEG Audio like Synthesis Filter */
1303 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1304 } else if (packet->type == 13) {
1305 for (j = 0; j < 6; j++)
1306 q->fft_level_exp[j] = get_bits(&gb, 6);
1307 } else if (packet->type == 14) {
1308 for (j = 0; j < 6; j++)
1309 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1310 } else if (packet->type == 15) {
1311 SAMPLES_NEEDED_2("packet type 15")
1312 return;
1313 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1314 /* packets for FFT */
1315 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1316 }
1317 } // Packet bytes loop
1318
1319 /* **************************************************************** */
1320 if (q->sub_packet_list_D[0].packet != NULL) {
1321 process_synthesis_subpackets(q, q->sub_packet_list_D);
1322 q->do_synth_filter = 1;
1323 } else if (q->do_synth_filter) {
1324 process_subpacket_10(q, NULL, 0);
1325 process_subpacket_11(q, NULL, 0);
1326 process_subpacket_12(q, NULL, 0);
1327 }
1328 /* **************************************************************** */
1329 }
1330
1331
1332 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1333 int offset, int duration, int channel,
1334 int exp, int phase)
1335 {
1336 if (q->fft_coefs_min_index[duration] < 0)
1337 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1338
1339 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1340 q->fft_coefs[q->fft_coefs_index].channel = channel;
1341 q->fft_coefs[q->fft_coefs_index].offset = offset;
1342 q->fft_coefs[q->fft_coefs_index].exp = exp;
1343 q->fft_coefs[q->fft_coefs_index].phase = phase;
1344 q->fft_coefs_index++;
1345 }
1346
1347
1348 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1349 {
1350 int channel, stereo, phase, exp;
1351 int local_int_4, local_int_8, stereo_phase, local_int_10;
1352 int local_int_14, stereo_exp, local_int_20, local_int_28;
1353 int n, offset;
1354
1355 local_int_4 = 0;
1356 local_int_28 = 0;
1357 local_int_20 = 2;
1358 local_int_8 = (4 - duration);
1359 local_int_10 = 1 << (q->group_order - duration - 1);
1360 offset = 1;
1361
1362 while (1) {
1363 if (q->superblocktype_2_3) {
1364 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1365 offset = 1;
1366 if (n == 0) {
1367 local_int_4 += local_int_10;
1368 local_int_28 += (1 << local_int_8);
1369 } else {
1370 local_int_4 += 8*local_int_10;
1371 local_int_28 += (8 << local_int_8);
1372 }
1373 }
1374 offset += (n - 2);
1375 } else {
1376 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1377 while (offset >= (local_int_10 - 1)) {
1378 offset += (1 - (local_int_10 - 1));
1379 local_int_4 += local_int_10;
1380 local_int_28 += (1 << local_int_8);
1381 }
1382 }
1383
1384 if (local_int_4 >= q->group_size)
1385 return;
1386
1387 local_int_14 = (offset >> local_int_8);
1388
1389 if (q->nb_channels > 1) {
1390 channel = get_bits1(gb);
1391 stereo = get_bits1(gb);
1392 } else {
1393 channel = 0;
1394 stereo = 0;
1395 }
1396
1397 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1398 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1399 exp = (exp < 0) ? 0 : exp;
1400
1401 phase = get_bits(gb, 3);
1402 stereo_exp = 0;
1403 stereo_phase = 0;
1404
1405 if (stereo) {
1406 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1407 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1408 if (stereo_phase < 0)
1409 stereo_phase += 8;
1410 }
1411
1412 if (q->frequency_range > (local_int_14 + 1)) {
1413 int sub_packet = (local_int_20 + local_int_28);
1414
1415 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1416 if (stereo)
1417 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1418 }
1419
1420 offset++;
1421 }
1422 }
1423
1424
1425 static void qdm2_decode_fft_packets (QDM2Context *q)
1426 {
1427 int i, j, min, max, value, type, unknown_flag;
1428 GetBitContext gb;
1429
1430 if (q->sub_packet_list_B[0].packet == NULL)
1431 return;
1432
1433 /* reset minimum indices for FFT coefficients */
1434 q->fft_coefs_index = 0;
1435 for (i=0; i < 5; i++)
1436 q->fft_coefs_min_index[i] = -1;
1437
1438 /* process sub packets ordered by type, largest type first */
1439 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1440 QDM2SubPacket *packet;
1441
1442 /* find sub packet with largest type less than max */
1443 for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1444 value = q->sub_packet_list_B[j].packet->type;
1445 if (value > min && value < max) {
1446 min = value;
1447 packet = q->sub_packet_list_B[j].packet;
1448 }
1449 }
1450
1451 max = min;
1452
1453 /* check for errors (?) */
1454 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1455 return;
1456
1457 /* decode FFT tones */
1458 init_get_bits (&gb, packet->data, packet->size);
1459
1460 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1461 unknown_flag = 1;
1462 else
1463 unknown_flag = 0;
1464
1465 type = packet->type;
1466
1467 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1468 int duration = q->sub_sampling + 5 - (type & 15);
1469
1470 if (duration >= 0 && duration < 4)
1471 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1472 } else if (type == 31) {
1473 for (i=0; i < 4; i++)
1474 qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
1475 } else if (type == 46) {
1476 for (i=0; i < 6; i++)
1477 q->fft_level_exp[i] = get_bits(&gb, 6);
1478 for (i=0; i < 4; i++)
1479 qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
1480 }
1481 } // Loop on B packets
1482
1483 /* calculate maximum indices for FFT coefficients */
1484 for (i = 0, j = -1; i < 5; i++)
1485 if (q->fft_coefs_min_index[i] >= 0) {
1486 if (j >= 0)
1487 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1488 j = i;
1489 }
1490 if (j >= 0)
1491 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1492 }
1493
1494
1495 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1496 {
1497 float level, f[6];
1498 int i;
1499 QDM2Complex c;
1500 const double iscale = 2.0*M_PI / 512.0;
1501
1502 tone->phase += tone->phase_shift;
1503
1504 /* calculate current level (maximum amplitude) of tone */
1505 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1506 c.im = level * sin(tone->phase*iscale);
1507 c.re = level * cos(tone->phase*iscale);
1508
1509 /* generate FFT coefficients for tone */
1510 if (tone->duration >= 3 || tone->cutoff >= 3) {
1511 tone->samples_im[0] += c.im;
1512 tone->samples_re[0] += c.re;
1513 tone->samples_im[1] -= c.im;
1514 tone->samples_re[1] -= c.re;
1515 } else {
1516 f[1] = -tone->table[4];
1517 f[0] = tone->table[3] - tone->table[0];
1518 f[2] = 1.0 - tone->table[2] - tone->table[3];
1519 f[3] = tone->table[1] + tone->table[4] - 1.0;
1520 f[4] = tone->table[0] - tone->table[1];
1521 f[5] = tone->table[2];
1522 for (i = 0; i < 2; i++) {
1523 tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1524 tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1525 }
1526 for (i = 0; i < 4; i++) {
1527 tone->samples_re[i] += c.re * f[i+2];
1528 tone->samples_im[i] += c.im * f[i+2];
1529 }
1530 }
1531
1532 /* copy the tone if it has not yet died out */
1533 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1534 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1535 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1536 }
1537 }
1538
1539
1540 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1541 {
1542 int i, j, ch;
1543 const double iscale = 0.25 * M_PI;
1544
1545 for (ch = 0; ch < q->channels; ch++) {
1546 memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1547 memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1548 }
1549
1550
1551 /* apply FFT tones with duration 4 (1 FFT period) */
1552 if (q->fft_coefs_min_index[4] >= 0)
1553 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1554 float level;
1555 QDM2Complex c;
1556
1557 if (q->fft_coefs[i].sub_packet != sub_packet)
1558 break;
1559
1560 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1561 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];
1562
1563 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1564 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1565 q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1566 q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1567 q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1568 q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1569 }
1570
1571 /* generate existing FFT tones */
1572 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1573 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1574 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1575 }
1576
1577 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1578 for (i = 0; i < 4; i++)
1579 if (q->fft_coefs_min_index[i] >= 0) {
1580 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1581 int offset, four_i;
1582 FFTTone tone;
1583
1584 if (q->fft_coefs[j].sub_packet != sub_packet)
1585 break;
1586
1587 four_i = (4 - i);
1588 offset = q->fft_coefs[j].offset >> four_i;
1589 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1590
1591 if (offset < q->frequency_range) {
1592 if (offset < 2)
1593 tone.cutoff = offset;
1594 else
1595 tone.cutoff = (offset >= 60) ? 3 : 2;
1596
1597 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];
1598 tone.samples_im = &q->fft.samples_im[ch][offset];
1599 tone.samples_re = &q->fft.samples_re[ch][offset];
1600 tone.table = (float*)fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1601 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1602 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1603 tone.duration = i;
1604 tone.time_index = 0;
1605
1606 qdm2_fft_generate_tone(q, &tone);
1607 }
1608 }
1609 q->fft_coefs_min_index[i] = j;
1610 }
1611 }
1612
1613
1614 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1615 {
1616 const int n = 1 << (q->fft_order - 1);
1617 const int n2 = n >> 1;
1618 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1619 float c, s, f0, f1, f2, f3;
1620 int i, j;
1621
1622 /* pre rotation (or something like that) */
1623 for (i=1; i < n2; i++) {
1624 j = (n - i);
1625 c = q->exptab[i].re;
1626 s = -q->exptab[i].im;
1627 f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1628 f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1629 f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1630 f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1631 q->fft.complex[i].re = s * f0 - c * f1 + f2;
1632 q->fft.complex[i].im = c * f0 + s * f1 + f3;
1633 q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1634 q->fft.complex[j].im = c * f0 + s * f1 - f3;
1635 }
1636
1637 q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0;
1638 q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0;
1639 q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0;
1640 q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1641
1642 ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1643 ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1644 /* add samples to output buffer */
1645 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1646 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1647 }
1648
1649
1650 /**
1651 * @param q context
1652 * @param index subpacket number
1653 */
1654 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1655 {
1656 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1657 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1658
1659 /* copy sb_samples */
1660 sb_used = QDM2_SB_USED(q->sub_sampling);
1661
1662 for (ch = 0; ch < q->channels; ch++)
1663 for (i = 0; i < 8; i++)
1664 for (k=sb_used; k < SBLIMIT; k++)
1665 q->sb_samples[ch][(8 * index) + i][k] = 0;
1666
1667 for (ch = 0; ch < q->nb_channels; ch++) {
1668 OUT_INT *samples_ptr = samples + ch;
1669
1670 for (i = 0; i < 8; i++) {
1671 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1672 mpa_window, &dither_state,
1673 samples_ptr, q->nb_channels,
1674 q->sb_samples[ch][(8 * index) + i]);
1675 samples_ptr += 32 * q->nb_channels;
1676 }
1677 }
1678
1679 /* add samples to output buffer */
1680 sub_sampling = (4 >> q->sub_sampling);
1681
1682 for (ch = 0; ch < q->channels; ch++)
1683 for (i = 0; i < q->frame_size; i++)
1684 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1685 }
1686
1687
1688 /**
1689 * Init static data (does not depend on specific file)
1690 *
1691 * @param q context
1692 */
1693 void qdm2_init(QDM2Context *q) {
1694 static int inited = 0;
1695
1696 if (inited != 0)
1697 return;
1698 inited = 1;
1699
1700 qdm2_init_vlc();
1701 ff_mpa_synth_init(mpa_window);
1702 softclip_table_init();
1703 rnd_table_init();
1704 init_noise_samples();
1705
1706 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1707 }
1708
1709
1710 #if 0
1711 static void dump_context(QDM2Context *q)
1712 {
1713 int i;
1714 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1715 PRINT("compressed_data",q->compressed_data);
1716 PRINT("compressed_size",q->compressed_size);
1717 PRINT("frame_size",q->frame_size);
1718 PRINT("checksum_size",q->checksum_size);
1719 PRINT("channels",q->channels);
1720 PRINT("nb_channels",q->nb_channels);
1721 PRINT("fft_frame_size",q->fft_frame_size);
1722 PRINT("fft_size",q->fft_size);
1723 PRINT("sub_sampling",q->sub_sampling);
1724 PRINT("fft_order",q->fft_order);
1725 PRINT("group_order",q->group_order);
1726 PRINT("group_size",q->group_size);
1727 PRINT("sub_packet",q->sub_packet);
1728 PRINT("frequency_range",q->frequency_range);
1729 PRINT("has_errors",q->has_errors);
1730 PRINT("fft_tone_end",q->fft_tone_end);
1731 PRINT("fft_tone_start",q->fft_tone_start);
1732 PRINT("fft_coefs_index",q->fft_coefs_index);
1733 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1734 PRINT("cm_table_select",q->cm_table_select);
1735 PRINT("noise_idx",q->noise_idx);
1736
1737 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1738 {
1739 FFTTone *t = &q->fft_tones[i];
1740
1741 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1742 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1743 // PRINT(" level", t->level);
1744 PRINT(" phase", t->phase);
1745 PRINT(" phase_shift", t->phase_shift);
1746 PRINT(" duration", t->duration);
1747 PRINT(" samples_im", t->samples_im);
1748 PRINT(" samples_re", t->samples_re);
1749 PRINT(" table", t->table);
1750 }
1751
1752 }
1753 #endif
1754
1755
1756 /**
1757 * Init parameters from codec extradata
1758 */
1759 static int qdm2_decode_init(AVCodecContext *avctx)
1760 {
1761 QDM2Context *s = avctx->priv_data;
1762 uint8_t *extradata;
1763 int extradata_size;
1764 int tmp_val, tmp, size;
1765 int i;
1766 float alpha;
1767
1768 /* extradata parsing
1769
1770 Structure:
1771 wave {
1772 frma (QDM2)
1773 QDCA
1774 QDCP
1775 }
1776
1777 32 size (including this field)
1778 32 tag (=frma)
1779 32 type (=QDM2 or QDMC)
1780
1781 32 size (including this field, in bytes)
1782 32 tag (=QDCA) // maybe mandatory parameters
1783 32 unknown (=1)
1784 32 channels (=2)
1785 32 samplerate (=44100)
1786 32 bitrate (=96000)
1787 32 block size (=4096)
1788 32 frame size (=256) (for one channel)
1789 32 packet size (=1300)
1790
1791 32 size (including this field, in bytes)
1792 32 tag (=QDCP) // maybe some tuneable parameters
1793 32 float1 (=1.0)
1794 32 zero ?
1795 32 float2 (=1.0)
1796 32 float3 (=1.0)
1797 32 unknown (27)
1798 32 unknown (8)
1799 32 zero ?
1800 */
1801
1802 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1803 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1804 return -1;
1805 }
1806
1807 extradata = avctx->extradata;
1808 extradata_size = avctx->extradata_size;
1809
1810 while (extradata_size > 7) {
1811 if (!memcmp(extradata, "frmaQDM", 7))
1812 break;
1813 extradata++;
1814 extradata_size--;
1815 }
1816
1817 if (extradata_size < 12) {
1818 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1819 extradata_size);
1820 return -1;
1821 }
1822
1823 if (memcmp(extradata, "frmaQDM", 7)) {
1824 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1825 return -1;
1826 }
1827
1828 if (extradata[7] == 'C') {
1829 // s->is_qdmc = 1;
1830 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1831 return -1;
1832 }
1833
1834 extradata += 8;
1835 extradata_size -= 8;
1836
1837 size = BE_32(extradata);
1838
1839 if(size > extradata_size){
1840 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1841 extradata_size, size);
1842 return -1;
1843 }
1844
1845 extradata += 4;
1846 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1847 if (BE_32(extradata) != MKBETAG('Q','D','C','A')) {
1848 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1849 return -1;
1850 }
1851
1852 extradata += 8;
1853
1854 avctx->channels = s->nb_channels = s->channels = BE_32(extradata);
1855 extradata += 4;
1856
1857 avctx->sample_rate = BE_32(extradata);
1858 extradata += 4;
1859
1860 avctx->bit_rate = BE_32(extradata);
1861 extradata += 4;
1862
1863 s->group_size = BE_32(extradata);
1864 extradata += 4;
1865
1866 s->fft_size = BE_32(extradata);
1867 extradata += 4;
1868
1869 s->checksum_size = BE_32(extradata);
1870 extradata += 4;
1871
1872 s->fft_order = av_log2(s->fft_size) + 1;
1873 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1874
1875 // something like max decodable tones
1876 s->group_order = av_log2(s->group_size) + 1;
1877 s->frame_size = s->group_size / 16; // 16 iterations per super block
1878
1879 if (s->fft_order == 8)
1880 s->sub_sampling = 1;
1881 else
1882 s->sub_sampling = 2;
1883 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1884
1885 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1886 case 0: tmp = 40; break;
1887 case 1: tmp = 48; break;
1888 case 2: tmp = 56; break;
1889 case 3: tmp = 72; break;
1890 case 4: tmp = 80; break;
1891 case 5: tmp = 100;break;
1892 default: tmp=s->sub_sampling; break;
1893 }
1894 tmp_val = 0;
1895 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1896 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1897 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1898 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1899 s->cm_table_select = tmp_val;
1900
1901 if (s->sub_sampling == 0)
1902 tmp = 16000;
1903 else
1904 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1905 /*
1906 0: 16000 -> 1
1907 1: 20000 -> 2
1908 2: 28000 -> 2
1909 */
1910 if (tmp < 8000)
1911 s->coeff_per_sb_select = 0;
1912 else if (tmp <= 16000)
1913 s->coeff_per_sb_select = 1;
1914 else
1915 s->coeff_per_sb_select = 2;
1916
1917 if (s->fft_order != 8 && s->fft_order != 9)
1918 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1919
1920 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1921
1922 for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1923 alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1924 s->exptab[i].re = cos(alpha);
1925 s->exptab[i].im = sin(alpha);
1926 }
1927
1928 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1929 qdm2_init(s);
1930
1931 // dump_context(s);
1932 return 0;
1933 }
1934
1935
1936 static int qdm2_decode_close(AVCodecContext *avctx)
1937 {
1938 QDM2Context *s = avctx->priv_data;
1939
1940 ff_fft_end(&s->fft_ctx);
1941
1942 return 0;
1943 }
1944
1945
1946 void qdm2_decode (QDM2Context *q, uint8_t *in, int16_t *out)
1947 {
1948 int ch, i;
1949 const int frame_size = (q->frame_size * q->channels);
1950
1951 /* select input buffer */
1952 q->compressed_data = in;
1953 q->compressed_size = q->checksum_size;
1954
1955 // dump_context(q);
1956
1957 /* copy old block, clear new block of output samples */
1958 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1959 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1960
1961 /* decode block of QDM2 compressed data */
1962 if (q->sub_packet == 0) {
1963 q->has_errors = 0; // zero it for a new super block
1964 av_log(NULL,AV_LOG_DEBUG,"Super block follows\n");
1965 qdm2_decode_super_block(q);
1966 }
1967
1968 /* parse sub packets */
1969 if (!q->has_errors) {
1970 if (q->sub_packet == 2)
1971 qdm2_decode_fft_packets(q);
1972
1973 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1974 }
1975
1976 /* sound synthesis stage 1 (FFT) */
1977 for (ch = 0; ch < q->channels; ch++) {
1978 qdm2_calculate_fft(q, ch, q->sub_packet);
1979
1980 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1981 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1982 return;
1983 }
1984 }
1985
1986 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1987 if (!q->has_errors && q->do_synth_filter)
1988 qdm2_synthesis_filter(q, q->sub_packet);
1989
1990 q->sub_packet = (q->sub_packet + 1) % 16;
1991
1992 /* clip and convert output float[] to 16bit signed samples */
1993 for (i = 0; i < frame_size; i++) {
1994 int value = (int)q->output_buffer[i];
1995
1996 if (value > SOFTCLIP_THRESHOLD)
1997 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1998 else if (value < -SOFTCLIP_THRESHOLD)
1999 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2000
2001 out[i] = value;
2002 }
2003 }
2004
2005
2006 static int qdm2_decode_frame(AVCodecContext *avctx,
2007 void *data, int *data_size,
2008 uint8_t *buf, int buf_size)
2009 {
2010 QDM2Context *s = avctx->priv_data;
2011
2012 if((buf == NULL) || (buf_size < s->checksum_size))
2013 return 0;
2014
2015 *data_size = s->channels * s->frame_size * sizeof(int16_t);
2016
2017 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2018 buf_size, buf, s->checksum_size, data, *data_size);
2019
2020 qdm2_decode(s, buf, data);
2021
2022 // reading only when next superblock found
2023 if (s->sub_packet == 0) {
2024 return s->checksum_size;
2025 }
2026
2027 return 0;
2028 }
2029
2030 AVCodec qdm2_decoder =
2031 {
2032 .name = "qdm2",
2033 .type = CODEC_TYPE_AUDIO,
2034 .id = CODEC_ID_QDM2,
2035 .priv_data_size = sizeof(QDM2Context),
2036 .init = qdm2_decode_init,
2037 .close = qdm2_decode_close,
2038 .decode = qdm2_decode_frame,
2039 };