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cosmetics: line wrap and vertical alignment
[FFMpeg-mirror/lagarith.git] / libavcodec / qdm2.c
blobe3e4c783674b285bc5b081db925c757c22c81b7e
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 "get_bits.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
711 multres = 0x66666667 * (acc * 10);
712 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
713 for (ch = 0; ch < nb_channels; ch++)
714 for (sb = 0; sb < 30; sb++)
715 for (j = 0; j < 64; j++) {
716 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
717 if (comp < 0)
718 comp += 0xff;
719 comp /= 256; // signed shift
720 switch(sb) {
721 case 0:
722 if (comp < 30)
723 comp = 30;
724 comp += 15;
725 break;
726 case 1:
727 if (comp < 24)
728 comp = 24;
729 comp += 10;
730 break;
731 case 2:
732 case 3:
733 case 4:
734 if (comp < 16)
735 comp = 16;
736 }
737 if (comp <= 5)
738 tmp = 0;
739 else if (comp <= 10)
740 tmp = 10;
741 else if (comp <= 16)
742 tmp = 16;
743 else if (comp <= 24)
744 tmp = -1;
745 else
746 tmp = 0;
747 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
748 }
749 for (sb = 0; sb < 30; sb++)
750 fix_coding_method_array(sb, nb_channels, coding_method);
751 for (ch = 0; ch < nb_channels; ch++)
752 for (sb = 0; sb < 30; sb++)
753 for (j = 0; j < 64; j++)
754 if (sb >= 10) {
755 if (coding_method[ch][sb][j] < 10)
756 coding_method[ch][sb][j] = 10;
757 } else {
758 if (sb >= 2) {
759 if (coding_method[ch][sb][j] < 16)
760 coding_method[ch][sb][j] = 16;
761 } else {
762 if (coding_method[ch][sb][j] < 30)
763 coding_method[ch][sb][j] = 30;
764 }
765 }
766 } else { // superblocktype_2_3 != 0
767 for (ch = 0; ch < nb_channels; ch++)
768 for (sb = 0; sb < 30; sb++)
769 for (j = 0; j < 64; j++)
770 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
771 }
772
773 return;
774 }
775
776
777 /**
778 *
779 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
780 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
781 *
782 * @param q context
783 * @param gb bitreader context
784 * @param length packet length in bits
785 * @param sb_min lower subband processed (sb_min included)
786 * @param sb_max higher subband processed (sb_max excluded)
787 */
788 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
789 {
790 int sb, j, k, n, ch, run, channels;
791 int joined_stereo, zero_encoding, chs;
792 int type34_first;
793 float type34_div = 0;
794 float type34_predictor;
795 float samples[10], sign_bits[16];
796
797 if (length == 0) {
798 // If no data use noise
799 for (sb=sb_min; sb < sb_max; sb++)
800 build_sb_samples_from_noise (q, sb);
801
802 return;
803 }
804
805 for (sb = sb_min; sb < sb_max; sb++) {
806 FIX_NOISE_IDX(q->noise_idx);
807
808 channels = q->nb_channels;
809
810 if (q->nb_channels <= 1 || sb < 12)
811 joined_stereo = 0;
812 else if (sb >= 24)
813 joined_stereo = 1;
814 else
815 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
816
817 if (joined_stereo) {
818 if (BITS_LEFT(length,gb) >= 16)
819 for (j = 0; j < 16; j++)
820 sign_bits[j] = get_bits1 (gb);
821
822 for (j = 0; j < 64; j++)
823 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
824 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
825
826 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
827 channels = 1;
828 }
829
830 for (ch = 0; ch < channels; ch++) {
831 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
832 type34_predictor = 0.0;
833 type34_first = 1;
834
835 for (j = 0; j < 128; ) {
836 switch (q->coding_method[ch][sb][j / 2]) {
837 case 8:
838 if (BITS_LEFT(length,gb) >= 10) {
839 if (zero_encoding) {
840 for (k = 0; k < 5; k++) {
841 if ((j + 2 * k) >= 128)
842 break;
843 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
844 }
845 } else {
846 n = get_bits(gb, 8);
847 for (k = 0; k < 5; k++)
848 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
849 }
850 for (k = 0; k < 5; k++)
851 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
852 } else {
853 for (k = 0; k < 10; k++)
854 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
855 }
856 run = 10;
857 break;
858
859 case 10:
860 if (BITS_LEFT(length,gb) >= 1) {
861 float f = 0.81;
862
863 if (get_bits1(gb))
864 f = -f;
865 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
866 samples[0] = f;
867 } else {
868 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
869 }
870 run = 1;
871 break;
872
873 case 16:
874 if (BITS_LEFT(length,gb) >= 10) {
875 if (zero_encoding) {
876 for (k = 0; k < 5; k++) {
877 if ((j + k) >= 128)
878 break;
879 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
880 }
881 } else {
882 n = get_bits (gb, 8);
883 for (k = 0; k < 5; k++)
884 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
885 }
886 } else {
887 for (k = 0; k < 5; k++)
888 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
889 }
890 run = 5;
891 break;
892
893 case 24:
894 if (BITS_LEFT(length,gb) >= 7) {
895 n = get_bits(gb, 7);
896 for (k = 0; k < 3; k++)
897 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
898 } else {
899 for (k = 0; k < 3; k++)
900 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
901 }
902 run = 3;
903 break;
904
905 case 30:
906 if (BITS_LEFT(length,gb) >= 4)
907 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
908 else
909 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
910
911 run = 1;
912 break;
913
914 case 34:
915 if (BITS_LEFT(length,gb) >= 7) {
916 if (type34_first) {
917 type34_div = (float)(1 << get_bits(gb, 2));
918 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
919 type34_predictor = samples[0];
920 type34_first = 0;
921 } else {
922 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
923 type34_predictor = samples[0];
924 }
925 } else {
926 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
927 }
928 run = 1;
929 break;
930
931 default:
932 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
933 run = 1;
934 break;
935 }
936
937 if (joined_stereo) {
938 float tmp[10][MPA_MAX_CHANNELS];
939
940 for (k = 0; k < run; k++) {
941 tmp[k][0] = samples[k];
942 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
943 }
944 for (chs = 0; chs < q->nb_channels; chs++)
945 for (k = 0; k < run; k++)
946 if ((j + k) < 128)
947 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
948 } else {
949 for (k = 0; k < run; k++)
950 if ((j + k) < 128)
951 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
952 }
953
954 j += run;
955 } // j loop
956 } // channel loop
957 } // subband loop
958 }
959
960
961 /**
962 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
963 * This is similar to process_subpacket_9, but for a single channel and for element [0]
964 * same VLC tables as process_subpacket_9 are used.
965 *
966 * @param q context
967 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
968 * @param gb bitreader context
969 * @param length packet length in bits
970 */
971 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
972 {
973 int i, k, run, level, diff;
974
975 if (BITS_LEFT(length,gb) < 16)
976 return;
977 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
978
979 quantized_coeffs[0] = level;
980
981 for (i = 0; i < 7; ) {
982 if (BITS_LEFT(length,gb) < 16)
983 break;
984 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
985
986 if (BITS_LEFT(length,gb) < 16)
987 break;
988 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
989
990 for (k = 1; k <= run; k++)
991 quantized_coeffs[i + k] = (level + ((k * diff) / run));
992
993 level += diff;
994 i += run;
995 }
996 }
997
998
999 /**
1000 * Related to synthesis filter, process data from packet 10
1001 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1002 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1003 *
1004 * @param q context
1005 * @param gb bitreader context
1006 * @param length packet length in bits
1007 */
1008 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1009 {
1010 int sb, j, k, n, ch;
1011
1012 for (ch = 0; ch < q->nb_channels; ch++) {
1013 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1014
1015 if (BITS_LEFT(length,gb) < 16) {
1016 memset(q->quantized_coeffs[ch][0], 0, 8);
1017 break;
1018 }
1019 }
1020
1021 n = q->sub_sampling + 1;
1022
1023 for (sb = 0; sb < n; sb++)
1024 for (ch = 0; ch < q->nb_channels; ch++)
1025 for (j = 0; j < 8; j++) {
1026 if (BITS_LEFT(length,gb) < 1)
1027 break;
1028 if (get_bits1(gb)) {
1029 for (k=0; k < 8; k++) {
1030 if (BITS_LEFT(length,gb) < 16)
1031 break;
1032 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1033 }
1034 } else {
1035 for (k=0; k < 8; k++)
1036 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1037 }
1038 }
1039
1040 n = QDM2_SB_USED(q->sub_sampling) - 4;
1041
1042 for (sb = 0; sb < n; sb++)
1043 for (ch = 0; ch < q->nb_channels; ch++) {
1044 if (BITS_LEFT(length,gb) < 16)
1045 break;
1046 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1047 if (sb > 19)
1048 q->tone_level_idx_hi2[ch][sb] -= 16;
1049 else
1050 for (j = 0; j < 8; j++)
1051 q->tone_level_idx_mid[ch][sb][j] = -16;
1052 }
1053
1054 n = QDM2_SB_USED(q->sub_sampling) - 5;
1055
1056 for (sb = 0; sb < n; sb++)
1057 for (ch = 0; ch < q->nb_channels; ch++)
1058 for (j = 0; j < 8; j++) {
1059 if (BITS_LEFT(length,gb) < 16)
1060 break;
1061 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1062 }
1063 }
1064
1065 /**
1066 * Process subpacket 9, init quantized_coeffs with data from it
1067 *
1068 * @param q context
1069 * @param node pointer to node with packet
1070 */
1071 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1072 {
1073 GetBitContext gb;
1074 int i, j, k, n, ch, run, level, diff;
1075
1076 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1077
1078 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1079
1080 for (i = 1; i < n; i++)
1081 for (ch=0; ch < q->nb_channels; ch++) {
1082 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1083 q->quantized_coeffs[ch][i][0] = level;
1084
1085 for (j = 0; j < (8 - 1); ) {
1086 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1087 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1088
1089 for (k = 1; k <= run; k++)
1090 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1091
1092 level += diff;
1093 j += run;
1094 }
1095 }
1096
1097 for (ch = 0; ch < q->nb_channels; ch++)
1098 for (i = 0; i < 8; i++)
1099 q->quantized_coeffs[ch][0][i] = 0;
1100 }
1101
1102
1103 /**
1104 * Process subpacket 10 if not null, else
1105 *
1106 * @param q context
1107 * @param node pointer to node with packet
1108 * @param length packet length in bits
1109 */
1110 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1111 {
1112 GetBitContext gb;
1113
1114 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1115
1116 if (length != 0) {
1117 init_tone_level_dequantization(q, &gb, length);
1118 fill_tone_level_array(q, 1);
1119 } else {
1120 fill_tone_level_array(q, 0);
1121 }
1122 }
1123
1124
1125 /**
1126 * Process subpacket 11
1127 *
1128 * @param q context
1129 * @param node pointer to node with packet
1130 * @param length packet length in bit
1131 */
1132 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1133 {
1134 GetBitContext gb;
1135
1136 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1137 if (length >= 32) {
1138 int c = get_bits (&gb, 13);
1139
1140 if (c > 3)
1141 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1142 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1143 }
1144
1145 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1146 }
1147
1148
1149 /**
1150 * Process subpacket 12
1151 *
1152 * @param q context
1153 * @param node pointer to node with packet
1154 * @param length packet length in bits
1155 */
1156 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1157 {
1158 GetBitContext gb;
1159
1160 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1161 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1162 }
1163
1164 /*
1165 * Process new subpackets for synthesis filter
1166 *
1167 * @param q context
1168 * @param list list with synthesis filter packets (list D)
1169 */
1170 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1171 {
1172 QDM2SubPNode *nodes[4];
1173
1174 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1175 if (nodes[0] != NULL)
1176 process_subpacket_9(q, nodes[0]);
1177
1178 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1179 if (nodes[1] != NULL)
1180 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1181 else
1182 process_subpacket_10(q, NULL, 0);
1183
1184 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1185 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1186 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1187 else
1188 process_subpacket_11(q, NULL, 0);
1189
1190 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1191 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1192 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1193 else
1194 process_subpacket_12(q, NULL, 0);
1195 }
1196
1197
1198 /*
1199 * Decode superblock, fill packet lists.
1200 *
1201 * @param q context
1202 */
1203 static void qdm2_decode_super_block (QDM2Context *q)
1204 {
1205 GetBitContext gb;
1206 QDM2SubPacket header, *packet;
1207 int i, packet_bytes, sub_packet_size, sub_packets_D;
1208 unsigned int next_index = 0;
1209
1210 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1211 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1212 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1213
1214 q->sub_packets_B = 0;
1215 sub_packets_D = 0;
1216
1217 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1218
1219 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1220 qdm2_decode_sub_packet_header(&gb, &header);
1221
1222 if (header.type < 2 || header.type >= 8) {
1223 q->has_errors = 1;
1224 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1225 return;
1226 }
1227
1228 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1229 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1230
1231 init_get_bits(&gb, header.data, header.size*8);
1232
1233 if (header.type == 2 || header.type == 4 || header.type == 5) {
1234 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1235
1236 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1237
1238 if (csum != 0) {
1239 q->has_errors = 1;
1240 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1241 return;
1242 }
1243 }
1244
1245 q->sub_packet_list_B[0].packet = NULL;
1246 q->sub_packet_list_D[0].packet = NULL;
1247
1248 for (i = 0; i < 6; i++)
1249 if (--q->fft_level_exp[i] < 0)
1250 q->fft_level_exp[i] = 0;
1251
1252 for (i = 0; packet_bytes > 0; i++) {
1253 int j;
1254
1255 q->sub_packet_list_A[i].next = NULL;
1256
1257 if (i > 0) {
1258 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1259
1260 /* seek to next block */
1261 init_get_bits(&gb, header.data, header.size*8);
1262 skip_bits(&gb, next_index*8);
1263
1264 if (next_index >= header.size)
1265 break;
1266 }
1267
1268 /* decode subpacket */
1269 packet = &q->sub_packets[i];
1270 qdm2_decode_sub_packet_header(&gb, packet);
1271 next_index = packet->size + get_bits_count(&gb) / 8;
1272 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1273
1274 if (packet->type == 0)
1275 break;
1276
1277 if (sub_packet_size > packet_bytes) {
1278 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1279 break;
1280 packet->size += packet_bytes - sub_packet_size;
1281 }
1282
1283 packet_bytes -= sub_packet_size;
1284
1285 /* add subpacket to 'all subpackets' list */
1286 q->sub_packet_list_A[i].packet = packet;
1287
1288 /* add subpacket to related list */
1289 if (packet->type == 8) {
1290 SAMPLES_NEEDED_2("packet type 8");
1291 return;
1292 } else if (packet->type >= 9 && packet->type <= 12) {
1293 /* packets for MPEG Audio like Synthesis Filter */
1294 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1295 } else if (packet->type == 13) {
1296 for (j = 0; j < 6; j++)
1297 q->fft_level_exp[j] = get_bits(&gb, 6);
1298 } else if (packet->type == 14) {
1299 for (j = 0; j < 6; j++)
1300 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1301 } else if (packet->type == 15) {
1302 SAMPLES_NEEDED_2("packet type 15")
1303 return;
1304 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1305 /* packets for FFT */
1306 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1307 }
1308 } // Packet bytes loop
1309
1310 /* **************************************************************** */
1311 if (q->sub_packet_list_D[0].packet != NULL) {
1312 process_synthesis_subpackets(q, q->sub_packet_list_D);
1313 q->do_synth_filter = 1;
1314 } else if (q->do_synth_filter) {
1315 process_subpacket_10(q, NULL, 0);
1316 process_subpacket_11(q, NULL, 0);
1317 process_subpacket_12(q, NULL, 0);
1318 }
1319 /* **************************************************************** */
1320 }
1321
1322
1323 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1324 int offset, int duration, int channel,
1325 int exp, int phase)
1326 {
1327 if (q->fft_coefs_min_index[duration] < 0)
1328 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1329
1330 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1331 q->fft_coefs[q->fft_coefs_index].channel = channel;
1332 q->fft_coefs[q->fft_coefs_index].offset = offset;
1333 q->fft_coefs[q->fft_coefs_index].exp = exp;
1334 q->fft_coefs[q->fft_coefs_index].phase = phase;
1335 q->fft_coefs_index++;
1336 }
1337
1338
1339 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1340 {
1341 int channel, stereo, phase, exp;
1342 int local_int_4, local_int_8, stereo_phase, local_int_10;
1343 int local_int_14, stereo_exp, local_int_20, local_int_28;
1344 int n, offset;
1345
1346 local_int_4 = 0;
1347 local_int_28 = 0;
1348 local_int_20 = 2;
1349 local_int_8 = (4 - duration);
1350 local_int_10 = 1 << (q->group_order - duration - 1);
1351 offset = 1;
1352
1353 while (1) {
1354 if (q->superblocktype_2_3) {
1355 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1356 offset = 1;
1357 if (n == 0) {
1358 local_int_4 += local_int_10;
1359 local_int_28 += (1 << local_int_8);
1360 } else {
1361 local_int_4 += 8*local_int_10;
1362 local_int_28 += (8 << local_int_8);
1363 }
1364 }
1365 offset += (n - 2);
1366 } else {
1367 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1368 while (offset >= (local_int_10 - 1)) {
1369 offset += (1 - (local_int_10 - 1));
1370 local_int_4 += local_int_10;
1371 local_int_28 += (1 << local_int_8);
1372 }
1373 }
1374
1375 if (local_int_4 >= q->group_size)
1376 return;
1377
1378 local_int_14 = (offset >> local_int_8);
1379
1380 if (q->nb_channels > 1) {
1381 channel = get_bits1(gb);
1382 stereo = get_bits1(gb);
1383 } else {
1384 channel = 0;
1385 stereo = 0;
1386 }
1387
1388 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1389 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1390 exp = (exp < 0) ? 0 : exp;
1391
1392 phase = get_bits(gb, 3);
1393 stereo_exp = 0;
1394 stereo_phase = 0;
1395
1396 if (stereo) {
1397 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1398 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1399 if (stereo_phase < 0)
1400 stereo_phase += 8;
1401 }
1402
1403 if (q->frequency_range > (local_int_14 + 1)) {
1404 int sub_packet = (local_int_20 + local_int_28);
1405
1406 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1407 if (stereo)
1408 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1409 }
1410
1411 offset++;
1412 }
1413 }
1414
1415
1416 static void qdm2_decode_fft_packets (QDM2Context *q)
1417 {
1418 int i, j, min, max, value, type, unknown_flag;
1419 GetBitContext gb;
1420
1421 if (q->sub_packet_list_B[0].packet == NULL)
1422 return;
1423
1424 /* reset minimum indexes for FFT coefficients */
1425 q->fft_coefs_index = 0;
1426 for (i=0; i < 5; i++)
1427 q->fft_coefs_min_index[i] = -1;
1428
1429 /* process subpackets ordered by type, largest type first */
1430 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1431 QDM2SubPacket *packet= NULL;
1432
1433 /* find subpacket with largest type less than max */
1434 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1435 value = q->sub_packet_list_B[j].packet->type;
1436 if (value > min && value < max) {
1437 min = value;
1438 packet = q->sub_packet_list_B[j].packet;
1439 }
1440 }
1441
1442 max = min;
1443
1444 /* check for errors (?) */
1445 if (!packet)
1446 return;
1447
1448 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1449 return;
1450
1451 /* decode FFT tones */
1452 init_get_bits (&gb, packet->data, packet->size*8);
1453
1454 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1455 unknown_flag = 1;
1456 else
1457 unknown_flag = 0;
1458
1459 type = packet->type;
1460
1461 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1462 int duration = q->sub_sampling + 5 - (type & 15);
1463
1464 if (duration >= 0 && duration < 4)
1465 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1466 } else if (type == 31) {
1467 for (j=0; j < 4; j++)
1468 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1469 } else if (type == 46) {
1470 for (j=0; j < 6; j++)
1471 q->fft_level_exp[j] = get_bits(&gb, 6);
1472 for (j=0; j < 4; j++)
1473 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1474 }
1475 } // Loop on B packets
1476
1477 /* calculate maximum indexes for FFT coefficients */
1478 for (i = 0, j = -1; i < 5; i++)
1479 if (q->fft_coefs_min_index[i] >= 0) {
1480 if (j >= 0)
1481 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1482 j = i;
1483 }
1484 if (j >= 0)
1485 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1486 }
1487
1488
1489 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1490 {
1491 float level, f[6];
1492 int i;
1493 QDM2Complex c;
1494 const double iscale = 2.0*M_PI / 512.0;
1495
1496 tone->phase += tone->phase_shift;
1497
1498 /* calculate current level (maximum amplitude) of tone */
1499 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1500 c.im = level * sin(tone->phase*iscale);
1501 c.re = level * cos(tone->phase*iscale);
1502
1503 /* generate FFT coefficients for tone */
1504 if (tone->duration >= 3 || tone->cutoff >= 3) {
1505 tone->complex[0].im += c.im;
1506 tone->complex[0].re += c.re;
1507 tone->complex[1].im -= c.im;
1508 tone->complex[1].re -= c.re;
1509 } else {
1510 f[1] = -tone->table[4];
1511 f[0] = tone->table[3] - tone->table[0];
1512 f[2] = 1.0 - tone->table[2] - tone->table[3];
1513 f[3] = tone->table[1] + tone->table[4] - 1.0;
1514 f[4] = tone->table[0] - tone->table[1];
1515 f[5] = tone->table[2];
1516 for (i = 0; i < 2; i++) {
1517 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re += c.re * f[i];
1518 tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1519 }
1520 for (i = 0; i < 4; i++) {
1521 tone->complex[i].re += c.re * f[i+2];
1522 tone->complex[i].im += c.im * f[i+2];
1523 }
1524 }
1525
1526 /* copy the tone if it has not yet died out */
1527 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1528 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1529 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1530 }
1531 }
1532
1533
1534 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1535 {
1536 int i, j, ch;
1537 const double iscale = 0.25 * M_PI;
1538
1539 for (ch = 0; ch < q->channels; ch++) {
1540 memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1541 }
1542
1543
1544 /* apply FFT tones with duration 4 (1 FFT period) */
1545 if (q->fft_coefs_min_index[4] >= 0)
1546 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1547 float level;
1548 QDM2Complex c;
1549
1550 if (q->fft_coefs[i].sub_packet != sub_packet)
1551 break;
1552
1553 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1554 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];
1555
1556 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1557 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1558 q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1559 q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1560 q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1561 q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1562 }
1563
1564 /* generate existing FFT tones */
1565 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1566 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1567 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1568 }
1569
1570 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1571 for (i = 0; i < 4; i++)
1572 if (q->fft_coefs_min_index[i] >= 0) {
1573 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1574 int offset, four_i;
1575 FFTTone tone;
1576
1577 if (q->fft_coefs[j].sub_packet != sub_packet)
1578 break;
1579
1580 four_i = (4 - i);
1581 offset = q->fft_coefs[j].offset >> four_i;
1582 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1583
1584 if (offset < q->frequency_range) {
1585 if (offset < 2)
1586 tone.cutoff = offset;
1587 else
1588 tone.cutoff = (offset >= 60) ? 3 : 2;
1589
1590 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];
1591 tone.complex = &q->fft.complex[ch][offset];
1592 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1593 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1594 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1595 tone.duration = i;
1596 tone.time_index = 0;
1597
1598 qdm2_fft_generate_tone(q, &tone);
1599 }
1600 }
1601 q->fft_coefs_min_index[i] = j;
1602 }
1603 }
1604
1605
1606 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1607 {
1608 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1609 int i;
1610 q->fft.complex[channel][0].re *= 2.0f;
1611 q->fft.complex[channel][0].im = 0.0f;
1612 ff_rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1613 /* add samples to output buffer */
1614 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1615 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex[channel])[i] * gain;
1616 }
1617
1618
1619 /**
1620 * @param q context
1621 * @param index subpacket number
1622 */
1623 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1624 {
1625 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1626 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1627
1628 /* copy sb_samples */
1629 sb_used = QDM2_SB_USED(q->sub_sampling);
1630
1631 for (ch = 0; ch < q->channels; ch++)
1632 for (i = 0; i < 8; i++)
1633 for (k=sb_used; k < SBLIMIT; k++)
1634 q->sb_samples[ch][(8 * index) + i][k] = 0;
1635
1636 for (ch = 0; ch < q->nb_channels; ch++) {
1637 OUT_INT *samples_ptr = samples + ch;
1638
1639 for (i = 0; i < 8; i++) {
1640 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1641 mpa_window, &dither_state,
1642 samples_ptr, q->nb_channels,
1643 q->sb_samples[ch][(8 * index) + i]);
1644 samples_ptr += 32 * q->nb_channels;
1645 }
1646 }
1647
1648 /* add samples to output buffer */
1649 sub_sampling = (4 >> q->sub_sampling);
1650
1651 for (ch = 0; ch < q->channels; ch++)
1652 for (i = 0; i < q->frame_size; i++)
1653 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1654 }
1655
1656
1657 /**
1658 * Init static data (does not depend on specific file)
1659 *
1660 * @param q context
1661 */
1662 static av_cold void qdm2_init(QDM2Context *q) {
1663 static int initialized = 0;
1664
1665 if (initialized != 0)
1666 return;
1667 initialized = 1;
1668
1669 qdm2_init_vlc();
1670 ff_mpa_synth_init(mpa_window);
1671 softclip_table_init();
1672 rnd_table_init();
1673 init_noise_samples();
1674
1675 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1676 }
1677
1678
1679 #if 0
1680 static void dump_context(QDM2Context *q)
1681 {
1682 int i;
1683 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1684 PRINT("compressed_data",q->compressed_data);
1685 PRINT("compressed_size",q->compressed_size);
1686 PRINT("frame_size",q->frame_size);
1687 PRINT("checksum_size",q->checksum_size);
1688 PRINT("channels",q->channels);
1689 PRINT("nb_channels",q->nb_channels);
1690 PRINT("fft_frame_size",q->fft_frame_size);
1691 PRINT("fft_size",q->fft_size);
1692 PRINT("sub_sampling",q->sub_sampling);
1693 PRINT("fft_order",q->fft_order);
1694 PRINT("group_order",q->group_order);
1695 PRINT("group_size",q->group_size);
1696 PRINT("sub_packet",q->sub_packet);
1697 PRINT("frequency_range",q->frequency_range);
1698 PRINT("has_errors",q->has_errors);
1699 PRINT("fft_tone_end",q->fft_tone_end);
1700 PRINT("fft_tone_start",q->fft_tone_start);
1701 PRINT("fft_coefs_index",q->fft_coefs_index);
1702 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1703 PRINT("cm_table_select",q->cm_table_select);
1704 PRINT("noise_idx",q->noise_idx);
1705
1706 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1707 {
1708 FFTTone *t = &q->fft_tones[i];
1709
1710 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1711 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1712 // PRINT(" level", t->level);
1713 PRINT(" phase", t->phase);
1714 PRINT(" phase_shift", t->phase_shift);
1715 PRINT(" duration", t->duration);
1716 PRINT(" samples_im", t->samples_im);
1717 PRINT(" samples_re", t->samples_re);
1718 PRINT(" table", t->table);
1719 }
1720
1721 }
1722 #endif
1723
1724
1725 /**
1726 * Init parameters from codec extradata
1727 */
1728 static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1729 {
1730 QDM2Context *s = avctx->priv_data;
1731 uint8_t *extradata;
1732 int extradata_size;
1733 int tmp_val, tmp, size;
1734
1735 /* extradata parsing
1736
1737 Structure:
1738 wave {
1739 frma (QDM2)
1740 QDCA
1741 QDCP
1742 }
1743
1744 32 size (including this field)
1745 32 tag (=frma)
1746 32 type (=QDM2 or QDMC)
1747
1748 32 size (including this field, in bytes)
1749 32 tag (=QDCA) // maybe mandatory parameters
1750 32 unknown (=1)
1751 32 channels (=2)
1752 32 samplerate (=44100)
1753 32 bitrate (=96000)
1754 32 block size (=4096)
1755 32 frame size (=256) (for one channel)
1756 32 packet size (=1300)
1757
1758 32 size (including this field, in bytes)
1759 32 tag (=QDCP) // maybe some tuneable parameters
1760 32 float1 (=1.0)
1761 32 zero ?
1762 32 float2 (=1.0)
1763 32 float3 (=1.0)
1764 32 unknown (27)
1765 32 unknown (8)
1766 32 zero ?
1767 */
1768
1769 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1770 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1771 return -1;
1772 }
1773
1774 extradata = avctx->extradata;
1775 extradata_size = avctx->extradata_size;
1776
1777 while (extradata_size > 7) {
1778 if (!memcmp(extradata, "frmaQDM", 7))
1779 break;
1780 extradata++;
1781 extradata_size--;
1782 }
1783
1784 if (extradata_size < 12) {
1785 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1786 extradata_size);
1787 return -1;
1788 }
1789
1790 if (memcmp(extradata, "frmaQDM", 7)) {
1791 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1792 return -1;
1793 }
1794
1795 if (extradata[7] == 'C') {
1796 // s->is_qdmc = 1;
1797 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1798 return -1;
1799 }
1800
1801 extradata += 8;
1802 extradata_size -= 8;
1803
1804 size = AV_RB32(extradata);
1805
1806 if(size > extradata_size){
1807 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1808 extradata_size, size);
1809 return -1;
1810 }
1811
1812 extradata += 4;
1813 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1814 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1815 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1816 return -1;
1817 }
1818
1819 extradata += 8;
1820
1821 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1822 extradata += 4;
1823
1824 avctx->sample_rate = AV_RB32(extradata);
1825 extradata += 4;
1826
1827 avctx->bit_rate = AV_RB32(extradata);
1828 extradata += 4;
1829
1830 s->group_size = AV_RB32(extradata);
1831 extradata += 4;
1832
1833 s->fft_size = AV_RB32(extradata);
1834 extradata += 4;
1835
1836 s->checksum_size = AV_RB32(extradata);
1837
1838 s->fft_order = av_log2(s->fft_size) + 1;
1839 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1840
1841 // something like max decodable tones
1842 s->group_order = av_log2(s->group_size) + 1;
1843 s->frame_size = s->group_size / 16; // 16 iterations per super block
1844
1845 s->sub_sampling = s->fft_order - 7;
1846 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1847
1848 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1849 case 0: tmp = 40; break;
1850 case 1: tmp = 48; break;
1851 case 2: tmp = 56; break;
1852 case 3: tmp = 72; break;
1853 case 4: tmp = 80; break;
1854 case 5: tmp = 100;break;
1855 default: tmp=s->sub_sampling; break;
1856 }
1857 tmp_val = 0;
1858 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1859 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1860 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1861 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1862 s->cm_table_select = tmp_val;
1863
1864 if (s->sub_sampling == 0)
1865 tmp = 7999;
1866 else
1867 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1868 /*
1869 0: 7999 -> 0
1870 1: 20000 -> 2
1871 2: 28000 -> 2
1872 */
1873 if (tmp < 8000)
1874 s->coeff_per_sb_select = 0;
1875 else if (tmp <= 16000)
1876 s->coeff_per_sb_select = 1;
1877 else
1878 s->coeff_per_sb_select = 2;
1879
1880 // Fail on unknown fft order
1881 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1882 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1883 return -1;
1884 }
1885
1886 ff_rdft_init(&s->rdft_ctx, s->fft_order, IRDFT);
1887
1888 qdm2_init(s);
1889
1890 avctx->sample_fmt = SAMPLE_FMT_S16;
1891
1892 // dump_context(s);
1893 return 0;
1894 }
1895
1896
1897 static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1898 {
1899 QDM2Context *s = avctx->priv_data;
1900
1901 ff_rdft_end(&s->rdft_ctx);
1902
1903 return 0;
1904 }
1905
1906
1907 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1908 {
1909 int ch, i;
1910 const int frame_size = (q->frame_size * q->channels);
1911
1912 /* select input buffer */
1913 q->compressed_data = in;
1914 q->compressed_size = q->checksum_size;
1915
1916 // dump_context(q);
1917
1918 /* copy old block, clear new block of output samples */
1919 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1920 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1921
1922 /* decode block of QDM2 compressed data */
1923 if (q->sub_packet == 0) {
1924 q->has_errors = 0; // zero it for a new super block
1925 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1926 qdm2_decode_super_block(q);
1927 }
1928
1929 /* parse subpackets */
1930 if (!q->has_errors) {
1931 if (q->sub_packet == 2)
1932 qdm2_decode_fft_packets(q);
1933
1934 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1935 }
1936
1937 /* sound synthesis stage 1 (FFT) */
1938 for (ch = 0; ch < q->channels; ch++) {
1939 qdm2_calculate_fft(q, ch, q->sub_packet);
1940
1941 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1942 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1943 return;
1944 }
1945 }
1946
1947 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1948 if (!q->has_errors && q->do_synth_filter)
1949 qdm2_synthesis_filter(q, q->sub_packet);
1950
1951 q->sub_packet = (q->sub_packet + 1) % 16;
1952
1953 /* clip and convert output float[] to 16bit signed samples */
1954 for (i = 0; i < frame_size; i++) {
1955 int value = (int)q->output_buffer[i];
1956
1957 if (value > SOFTCLIP_THRESHOLD)
1958 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
1959 else if (value < -SOFTCLIP_THRESHOLD)
1960 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1961
1962 out[i] = value;
1963 }
1964 }
1965
1966
1967 static int qdm2_decode_frame(AVCodecContext *avctx,
1968 void *data, int *data_size,
1969 AVPacket *avpkt)
1970 {
1971 const uint8_t *buf = avpkt->data;
1972 int buf_size = avpkt->size;
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 };
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