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Move #defines that are mostly used in h264.c out of h264data.h and into h264.h.
[FFMpeg-mirror/ffmpeg-vdpau.git] / libavcodec / qdm2.c
blobe1b67d0c19176945409acafe068d498623aa30ba
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 qdm2.c
27 * QDM2 decoder
28 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29 * The decoder is not perfect yet, there are still some distortions
30 * especially on files encoded with 16 or 8 subbands.
31 */
32
33 #include <math.h>
34 #include <stddef.h>
35 #include <stdio.h>
36
37 #define ALT_BITSTREAM_READER_LE
38 #include "avcodec.h"
39 #include "bitstream.h"
40 #include "dsputil.h"
41
42 #ifdef CONFIG_MPEGAUDIO_HP
43 #define USE_HIGHPRECISION
44 #endif
45
46 #include "mpegaudio.h"
47
48 #include "qdm2data.h"
49
50 #undef NDEBUG
51 #include <assert.h>
52
53
54 #define SOFTCLIP_THRESHOLD 27600
55 #define HARDCLIP_THRESHOLD 35716
56
57
58 #define QDM2_LIST_ADD(list, size, packet) \
59 do { \
60 if (size > 0) { \
61 list[size - 1].next = &list[size]; \
62 } \
63 list[size].packet = packet; \
64 list[size].next = NULL; \
65 size++; \
66 } while(0)
67
68 // Result is 8, 16 or 30
69 #define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
70
71 #define FIX_NOISE_IDX(noise_idx) \
72 if ((noise_idx) >= 3840) \
73 (noise_idx) -= 3840; \
74
75 #define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
76
77 #define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
78
79 #define SAMPLES_NEEDED \
80 av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
81
82 #define SAMPLES_NEEDED_2(why) \
83 av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
84
85
86 typedef int8_t sb_int8_array[2][30][64];
87
88 /**
89 * Subpacket
90 */
91 typedef struct {
92 int type; ///< subpacket type
93 unsigned int size; ///< subpacket size
94 const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
95 } QDM2SubPacket;
96
97 /**
98 * A node in the subpacket list
99 */
100 typedef struct QDM2SubPNode {
101 QDM2SubPacket *packet; ///< packet
102 struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
103 } QDM2SubPNode;
104
105 typedef struct {
106 float level;
107 float *samples_im;
108 float *samples_re;
109 const float *table;
110 int phase;
111 int phase_shift;
112 int duration;
113 short time_index;
114 short cutoff;
115 } FFTTone;
116
117 typedef struct {
118 int16_t sub_packet;
119 uint8_t channel;
120 int16_t offset;
121 int16_t exp;
122 uint8_t phase;
123 } FFTCoefficient;
124
125 typedef struct {
126 float re;
127 float im;
128 } QDM2Complex;
129
130 typedef struct {
131 DECLARE_ALIGNED_16(QDM2Complex, complex[256 + 1]);
132 float samples_im[MPA_MAX_CHANNELS][256];
133 float samples_re[MPA_MAX_CHANNELS][256];
134 } QDM2FFT;
135
136 /**
137 * QDM2 decoder context
138 */
139 typedef struct {
140 /// Parameters from codec header, do not change during playback
141 int nb_channels; ///< number of channels
142 int channels; ///< number of channels
143 int group_size; ///< size of frame group (16 frames per group)
144 int fft_size; ///< size of FFT, in complex numbers
145 int checksum_size; ///< size of data block, used also for checksum
146
147 /// Parameters built from header parameters, do not change during playback
148 int group_order; ///< order of frame group
149 int fft_order; ///< order of FFT (actually fftorder+1)
150 int fft_frame_size; ///< size of fft frame, in components (1 comples = re + im)
151 int frame_size; ///< size of data frame
152 int frequency_range;
153 int sub_sampling; ///< subsampling: 0=25%, 1=50%, 2=100% */
154 int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
155 int cm_table_select; ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
156
157 /// Packets and packet lists
158 QDM2SubPacket sub_packets[16]; ///< the packets themselves
159 QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
160 QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
161 int sub_packets_B; ///< number of packets on 'B' list
162 QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
163 QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
164
165 /// FFT and tones
166 FFTTone fft_tones[1000];
167 int fft_tone_start;
168 int fft_tone_end;
169 FFTCoefficient fft_coefs[1000];
170 int fft_coefs_index;
171 int fft_coefs_min_index[5];
172 int fft_coefs_max_index[5];
173 int fft_level_exp[6];
174 FFTContext fft_ctx;
175 FFTComplex exptab[128];
176 QDM2FFT fft;
177
178 /// I/O data
179 const uint8_t *compressed_data;
180 int compressed_size;
181 float output_buffer[1024];
182
183 /// Synthesis filter
184 DECLARE_ALIGNED_16(MPA_INT, synth_buf[MPA_MAX_CHANNELS][512*2]);
185 int synth_buf_offset[MPA_MAX_CHANNELS];
186 DECLARE_ALIGNED_16(int32_t, sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT]);
187
188 /// Mixed temporary data used in decoding
189 float tone_level[MPA_MAX_CHANNELS][30][64];
190 int8_t coding_method[MPA_MAX_CHANNELS][30][64];
191 int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
192 int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
193 int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
194 int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
195 int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
196 int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
197 int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
198
199 // Flags
200 int has_errors; ///< packet has errors
201 int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
202 int do_synth_filter; ///< used to perform or skip synthesis filter
203
204 int sub_packet;
205 int noise_idx; ///< index for dithering noise table
206 } QDM2Context;
207
208
209 static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
210
211 static VLC vlc_tab_level;
212 static VLC vlc_tab_diff;
213 static VLC vlc_tab_run;
214 static VLC fft_level_exp_alt_vlc;
215 static VLC fft_level_exp_vlc;
216 static VLC fft_stereo_exp_vlc;
217 static VLC fft_stereo_phase_vlc;
218 static VLC vlc_tab_tone_level_idx_hi1;
219 static VLC vlc_tab_tone_level_idx_mid;
220 static VLC vlc_tab_tone_level_idx_hi2;
221 static VLC vlc_tab_type30;
222 static VLC vlc_tab_type34;
223 static VLC vlc_tab_fft_tone_offset[5];
224
225 static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
226 static float noise_table[4096];
227 static uint8_t random_dequant_index[256][5];
228 static uint8_t random_dequant_type24[128][3];
229 static float noise_samples[128];
230
231 static DECLARE_ALIGNED_16(MPA_INT, mpa_window[512]);
232
233
234 static void softclip_table_init(void) {
235 int i;
236 double dfl = SOFTCLIP_THRESHOLD - 32767;
237 float delta = 1.0 / -dfl;
238 for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
239 softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
240 }
241
242
243 // random generated table
244 static void rnd_table_init(void) {
245 int i,j;
246 uint32_t ldw,hdw;
247 uint64_t tmp64_1;
248 uint64_t random_seed = 0;
249 float delta = 1.0 / 16384.0;
250 for(i = 0; i < 4096 ;i++) {
251 random_seed = random_seed * 214013 + 2531011;
252 noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
253 }
254
255 for (i = 0; i < 256 ;i++) {
256 random_seed = 81;
257 ldw = i;
258 for (j = 0; j < 5 ;j++) {
259 random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
260 ldw = (uint32_t)ldw % (uint32_t)random_seed;
261 tmp64_1 = (random_seed * 0x55555556);
262 hdw = (uint32_t)(tmp64_1 >> 32);
263 random_seed = (uint64_t)(hdw + (ldw >> 31));
264 }
265 }
266 for (i = 0; i < 128 ;i++) {
267 random_seed = 25;
268 ldw = i;
269 for (j = 0; j < 3 ;j++) {
270 random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
271 ldw = (uint32_t)ldw % (uint32_t)random_seed;
272 tmp64_1 = (random_seed * 0x66666667);
273 hdw = (uint32_t)(tmp64_1 >> 33);
274 random_seed = hdw + (ldw >> 31);
275 }
276 }
277 }
278
279
280 static void init_noise_samples(void) {
281 int i;
282 int random_seed = 0;
283 float delta = 1.0 / 16384.0;
284 for (i = 0; i < 128;i++) {
285 random_seed = random_seed * 214013 + 2531011;
286 noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
287 }
288 }
289
290
291 static void qdm2_init_vlc(void)
292 {
293 init_vlc (&vlc_tab_level, 8, 24,
294 vlc_tab_level_huffbits, 1, 1,
295 vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
296
297 init_vlc (&vlc_tab_diff, 8, 37,
298 vlc_tab_diff_huffbits, 1, 1,
299 vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
300
301 init_vlc (&vlc_tab_run, 5, 6,
302 vlc_tab_run_huffbits, 1, 1,
303 vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
304
305 init_vlc (&fft_level_exp_alt_vlc, 8, 28,
306 fft_level_exp_alt_huffbits, 1, 1,
307 fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
308
309 init_vlc (&fft_level_exp_vlc, 8, 20,
310 fft_level_exp_huffbits, 1, 1,
311 fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
312
313 init_vlc (&fft_stereo_exp_vlc, 6, 7,
314 fft_stereo_exp_huffbits, 1, 1,
315 fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
316
317 init_vlc (&fft_stereo_phase_vlc, 6, 9,
318 fft_stereo_phase_huffbits, 1, 1,
319 fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
320
321 init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
322 vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
323 vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
324
325 init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
326 vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
327 vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
328
329 init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
330 vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
331 vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
332
333 init_vlc (&vlc_tab_type30, 6, 9,
334 vlc_tab_type30_huffbits, 1, 1,
335 vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
336
337 init_vlc (&vlc_tab_type34, 5, 10,
338 vlc_tab_type34_huffbits, 1, 1,
339 vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
340
341 init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
342 vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
343 vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
344
345 init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
346 vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
347 vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
348
349 init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
350 vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
351 vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
352
353 init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
354 vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
355 vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
356
357 init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
358 vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
359 vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
360 }
361
362
363 /* for floating point to fixed point conversion */
364 static const float f2i_scale = (float) (1 << (FRAC_BITS - 15));
365
366
367 static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
368 {
369 int value;
370
371 value = get_vlc2(gb, vlc->table, vlc->bits, depth);
372
373 /* stage-2, 3 bits exponent escape sequence */
374 if (value-- == 0)
375 value = get_bits (gb, get_bits (gb, 3) + 1);
376
377 /* stage-3, optional */
378 if (flag) {
379 int tmp = vlc_stage3_values[value];
380
381 if ((value & ~3) > 0)
382 tmp += get_bits (gb, (value >> 2));
383 value = tmp;
384 }
385
386 return value;
387 }
388
389
390 static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
391 {
392 int value = qdm2_get_vlc (gb, vlc, 0, depth);
393
394 return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
395 }
396
397
398 /**
399 * QDM2 checksum
400 *
401 * @param data pointer to data to be checksum'ed
402 * @param length data length
403 * @param value checksum value
404 *
405 * @return 0 if checksum is OK
406 */
407 static uint16_t qdm2_packet_checksum (const uint8_t *data, int length, int value) {
408 int i;
409
410 for (i=0; i < length; i++)
411 value -= data[i];
412
413 return (uint16_t)(value & 0xffff);
414 }
415
416
417 /**
418 * Fills a QDM2SubPacket structure with packet type, size, and data pointer.
419 *
420 * @param gb bitreader context
421 * @param sub_packet packet under analysis
422 */
423 static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
424 {
425 sub_packet->type = get_bits (gb, 8);
426
427 if (sub_packet->type == 0) {
428 sub_packet->size = 0;
429 sub_packet->data = NULL;
430 } else {
431 sub_packet->size = get_bits (gb, 8);
432
433 if (sub_packet->type & 0x80) {
434 sub_packet->size <<= 8;
435 sub_packet->size |= get_bits (gb, 8);
436 sub_packet->type &= 0x7f;
437 }
438
439 if (sub_packet->type == 0x7f)
440 sub_packet->type |= (get_bits (gb, 8) << 8);
441
442 sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
443 }
444
445 av_log(NULL,AV_LOG_DEBUG,"Subpacket: type=%d size=%d start_offs=%x\n",
446 sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
447 }
448
449
450 /**
451 * Return node pointer to first packet of requested type in list.
452 *
453 * @param list list of subpackets to be scanned
454 * @param type type of searched subpacket
455 * @return node pointer for subpacket if found, else NULL
456 */
457 static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
458 {
459 while (list != NULL && list->packet != NULL) {
460 if (list->packet->type == type)
461 return list;
462 list = list->next;
463 }
464 return NULL;
465 }
466
467
468 /**
469 * Replaces 8 elements with their average value.
470 * Called by qdm2_decode_superblock before starting subblock decoding.
471 *
472 * @param q context
473 */
474 static void average_quantized_coeffs (QDM2Context *q)
475 {
476 int i, j, n, ch, sum;
477
478 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
479
480 for (ch = 0; ch < q->nb_channels; ch++)
481 for (i = 0; i < n; i++) {
482 sum = 0;
483
484 for (j = 0; j < 8; j++)
485 sum += q->quantized_coeffs[ch][i][j];
486
487 sum /= 8;
488 if (sum > 0)
489 sum--;
490
491 for (j=0; j < 8; j++)
492 q->quantized_coeffs[ch][i][j] = sum;
493 }
494 }
495
496
497 /**
498 * Build subband samples with noise weighted by q->tone_level.
499 * Called by synthfilt_build_sb_samples.
500 *
501 * @param q context
502 * @param sb subband index
503 */
504 static void build_sb_samples_from_noise (QDM2Context *q, int sb)
505 {
506 int ch, j;
507
508 FIX_NOISE_IDX(q->noise_idx);
509
510 if (!q->nb_channels)
511 return;
512
513 for (ch = 0; ch < q->nb_channels; ch++)
514 for (j = 0; j < 64; j++) {
515 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);
516 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);
517 }
518 }
519
520
521 /**
522 * Called while processing data from subpackets 11 and 12.
523 * Used after making changes to coding_method array.
524 *
525 * @param sb subband index
526 * @param channels number of channels
527 * @param coding_method q->coding_method[0][0][0]
528 */
529 static void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
530 {
531 int j,k;
532 int ch;
533 int run, case_val;
534 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};
535
536 for (ch = 0; ch < channels; ch++) {
537 for (j = 0; j < 64; ) {
538 if((coding_method[ch][sb][j] - 8) > 22) {
539 run = 1;
540 case_val = 8;
541 } else {
542 switch (switchtable[coding_method[ch][sb][j]-8]) {
543 case 0: run = 10; case_val = 10; break;
544 case 1: run = 1; case_val = 16; break;
545 case 2: run = 5; case_val = 24; break;
546 case 3: run = 3; case_val = 30; break;
547 case 4: run = 1; case_val = 30; break;
548 case 5: run = 1; case_val = 8; break;
549 default: run = 1; case_val = 8; break;
550 }
551 }
552 for (k = 0; k < run; k++)
553 if (j + k < 128)
554 if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
555 if (k > 0) {
556 SAMPLES_NEEDED
557 //not debugged, almost never used
558 memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
559 memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
560 }
561 j += run;
562 }
563 }
564 }
565
566
567 /**
568 * Related to synthesis filter
569 * Called by process_subpacket_10
570 *
571 * @param q context
572 * @param flag 1 if called after getting data from subpacket 10, 0 if no subpacket 10
573 */
574 static void fill_tone_level_array (QDM2Context *q, int flag)
575 {
576 int i, sb, ch, sb_used;
577 int tmp, tab;
578
579 // This should never happen
580 if (q->nb_channels <= 0)
581 return;
582
583 for (ch = 0; ch < q->nb_channels; ch++)
584 for (sb = 0; sb < 30; sb++)
585 for (i = 0; i < 8; i++) {
586 if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
587 tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
588 q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
589 else
590 tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
591 if(tmp < 0)
592 tmp += 0xff;
593 q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
594 }
595
596 sb_used = QDM2_SB_USED(q->sub_sampling);
597
598 if ((q->superblocktype_2_3 != 0) && !flag) {
599 for (sb = 0; sb < sb_used; sb++)
600 for (ch = 0; ch < q->nb_channels; ch++)
601 for (i = 0; i < 64; i++) {
602 q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
603 if (q->tone_level_idx[ch][sb][i] < 0)
604 q->tone_level[ch][sb][i] = 0;
605 else
606 q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
607 }
608 } else {
609 tab = q->superblocktype_2_3 ? 0 : 1;
610 for (sb = 0; sb < sb_used; sb++) {
611 if ((sb >= 4) && (sb <= 23)) {
612 for (ch = 0; ch < q->nb_channels; ch++)
613 for (i = 0; i < 64; i++) {
614 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
615 q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
616 q->tone_level_idx_mid[ch][sb - 4][i / 8] -
617 q->tone_level_idx_hi2[ch][sb - 4];
618 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
619 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
620 q->tone_level[ch][sb][i] = 0;
621 else
622 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
623 }
624 } else {
625 if (sb > 4) {
626 for (ch = 0; ch < q->nb_channels; ch++)
627 for (i = 0; i < 64; i++) {
628 tmp = q->tone_level_idx_base[ch][sb][i / 8] -
629 q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
630 q->tone_level_idx_hi2[ch][sb - 4];
631 q->tone_level_idx[ch][sb][i] = tmp & 0xff;
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 } else {
638 for (ch = 0; ch < q->nb_channels; ch++)
639 for (i = 0; i < 64; i++) {
640 tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
641 if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
642 q->tone_level[ch][sb][i] = 0;
643 else
644 q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
645 }
646 }
647 }
648 }
649 }
650
651 return;
652 }
653
654
655 /**
656 * Related to synthesis filter
657 * Called by process_subpacket_11
658 * c is built with data from subpacket 11
659 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
660 *
661 * @param tone_level_idx
662 * @param tone_level_idx_temp
663 * @param coding_method q->coding_method[0][0][0]
664 * @param nb_channels number of channels
665 * @param c coming from subpacket 11, passed as 8*c
666 * @param superblocktype_2_3 flag based on superblock packet type
667 * @param cm_table_select q->cm_table_select
668 */
669 static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
670 sb_int8_array coding_method, int nb_channels,
671 int c, int superblocktype_2_3, int cm_table_select)
672 {
673 int ch, sb, j;
674 int tmp, acc, esp_40, comp;
675 int add1, add2, add3, add4;
676 int64_t multres;
677
678 // This should never happen
679 if (nb_channels <= 0)
680 return;
681
682 if (!superblocktype_2_3) {
683 /* This case is untested, no samples available */
684 SAMPLES_NEEDED
685 for (ch = 0; ch < nb_channels; ch++)
686 for (sb = 0; sb < 30; sb++) {
687 for (j = 1; j < 63; j++) { // The loop only iterates to 63 so the code doesn't overflow the buffer
688 add1 = tone_level_idx[ch][sb][j] - 10;
689 if (add1 < 0)
690 add1 = 0;
691 add2 = add3 = add4 = 0;
692 if (sb > 1) {
693 add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
694 if (add2 < 0)
695 add2 = 0;
696 }
697 if (sb > 0) {
698 add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
699 if (add3 < 0)
700 add3 = 0;
701 }
702 if (sb < 29) {
703 add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
704 if (add4 < 0)
705 add4 = 0;
706 }
707 tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
708 if (tmp < 0)
709 tmp = 0;
710 tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
711 }
712 tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
713 }
714 acc = 0;
715 for (ch = 0; ch < nb_channels; ch++)
716 for (sb = 0; sb < 30; sb++)
717 for (j = 0; j < 64; j++)
718 acc += tone_level_idx_temp[ch][sb][j];
719 if (acc)
720 tmp = c * 256 / (acc & 0xffff);
721 multres = 0x66666667 * (acc * 10);
722 esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
723 for (ch = 0; ch < nb_channels; ch++)
724 for (sb = 0; sb < 30; sb++)
725 for (j = 0; j < 64; j++) {
726 comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
727 if (comp < 0)
728 comp += 0xff;
729 comp /= 256; // signed shift
730 switch(sb) {
731 case 0:
732 if (comp < 30)
733 comp = 30;
734 comp += 15;
735 break;
736 case 1:
737 if (comp < 24)
738 comp = 24;
739 comp += 10;
740 break;
741 case 2:
742 case 3:
743 case 4:
744 if (comp < 16)
745 comp = 16;
746 }
747 if (comp <= 5)
748 tmp = 0;
749 else if (comp <= 10)
750 tmp = 10;
751 else if (comp <= 16)
752 tmp = 16;
753 else if (comp <= 24)
754 tmp = -1;
755 else
756 tmp = 0;
757 coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
758 }
759 for (sb = 0; sb < 30; sb++)
760 fix_coding_method_array(sb, nb_channels, coding_method);
761 for (ch = 0; ch < nb_channels; ch++)
762 for (sb = 0; sb < 30; sb++)
763 for (j = 0; j < 64; j++)
764 if (sb >= 10) {
765 if (coding_method[ch][sb][j] < 10)
766 coding_method[ch][sb][j] = 10;
767 } else {
768 if (sb >= 2) {
769 if (coding_method[ch][sb][j] < 16)
770 coding_method[ch][sb][j] = 16;
771 } else {
772 if (coding_method[ch][sb][j] < 30)
773 coding_method[ch][sb][j] = 30;
774 }
775 }
776 } else { // superblocktype_2_3 != 0
777 for (ch = 0; ch < nb_channels; ch++)
778 for (sb = 0; sb < 30; sb++)
779 for (j = 0; j < 64; j++)
780 coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
781 }
782
783 return;
784 }
785
786
787 /**
788 *
789 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
790 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
791 *
792 * @param q context
793 * @param gb bitreader context
794 * @param length packet length in bits
795 * @param sb_min lower subband processed (sb_min included)
796 * @param sb_max higher subband processed (sb_max excluded)
797 */
798 static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
799 {
800 int sb, j, k, n, ch, run, channels;
801 int joined_stereo, zero_encoding, chs;
802 int type34_first;
803 float type34_div = 0;
804 float type34_predictor;
805 float samples[10], sign_bits[16];
806
807 if (length == 0) {
808 // If no data use noise
809 for (sb=sb_min; sb < sb_max; sb++)
810 build_sb_samples_from_noise (q, sb);
811
812 return;
813 }
814
815 for (sb = sb_min; sb < sb_max; sb++) {
816 FIX_NOISE_IDX(q->noise_idx);
817
818 channels = q->nb_channels;
819
820 if (q->nb_channels <= 1 || sb < 12)
821 joined_stereo = 0;
822 else if (sb >= 24)
823 joined_stereo = 1;
824 else
825 joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
826
827 if (joined_stereo) {
828 if (BITS_LEFT(length,gb) >= 16)
829 for (j = 0; j < 16; j++)
830 sign_bits[j] = get_bits1 (gb);
831
832 for (j = 0; j < 64; j++)
833 if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
834 q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
835
836 fix_coding_method_array(sb, q->nb_channels, q->coding_method);
837 channels = 1;
838 }
839
840 for (ch = 0; ch < channels; ch++) {
841 zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
842 type34_predictor = 0.0;
843 type34_first = 1;
844
845 for (j = 0; j < 128; ) {
846 switch (q->coding_method[ch][sb][j / 2]) {
847 case 8:
848 if (BITS_LEFT(length,gb) >= 10) {
849 if (zero_encoding) {
850 for (k = 0; k < 5; k++) {
851 if ((j + 2 * k) >= 128)
852 break;
853 samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
854 }
855 } else {
856 n = get_bits(gb, 8);
857 for (k = 0; k < 5; k++)
858 samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
859 }
860 for (k = 0; k < 5; k++)
861 samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
862 } else {
863 for (k = 0; k < 10; k++)
864 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
865 }
866 run = 10;
867 break;
868
869 case 10:
870 if (BITS_LEFT(length,gb) >= 1) {
871 float f = 0.81;
872
873 if (get_bits1(gb))
874 f = -f;
875 f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
876 samples[0] = f;
877 } else {
878 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
879 }
880 run = 1;
881 break;
882
883 case 16:
884 if (BITS_LEFT(length,gb) >= 10) {
885 if (zero_encoding) {
886 for (k = 0; k < 5; k++) {
887 if ((j + k) >= 128)
888 break;
889 samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
890 }
891 } else {
892 n = get_bits (gb, 8);
893 for (k = 0; k < 5; k++)
894 samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
895 }
896 } else {
897 for (k = 0; k < 5; k++)
898 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
899 }
900 run = 5;
901 break;
902
903 case 24:
904 if (BITS_LEFT(length,gb) >= 7) {
905 n = get_bits(gb, 7);
906 for (k = 0; k < 3; k++)
907 samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
908 } else {
909 for (k = 0; k < 3; k++)
910 samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
911 }
912 run = 3;
913 break;
914
915 case 30:
916 if (BITS_LEFT(length,gb) >= 4)
917 samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
918 else
919 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
920
921 run = 1;
922 break;
923
924 case 34:
925 if (BITS_LEFT(length,gb) >= 7) {
926 if (type34_first) {
927 type34_div = (float)(1 << get_bits(gb, 2));
928 samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
929 type34_predictor = samples[0];
930 type34_first = 0;
931 } else {
932 samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
933 type34_predictor = samples[0];
934 }
935 } else {
936 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
937 }
938 run = 1;
939 break;
940
941 default:
942 samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
943 run = 1;
944 break;
945 }
946
947 if (joined_stereo) {
948 float tmp[10][MPA_MAX_CHANNELS];
949
950 for (k = 0; k < run; k++) {
951 tmp[k][0] = samples[k];
952 tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
953 }
954 for (chs = 0; chs < q->nb_channels; chs++)
955 for (k = 0; k < run; k++)
956 if ((j + k) < 128)
957 q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
958 } else {
959 for (k = 0; k < run; k++)
960 if ((j + k) < 128)
961 q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
962 }
963
964 j += run;
965 } // j loop
966 } // channel loop
967 } // subband loop
968 }
969
970
971 /**
972 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0]).
973 * This is similar to process_subpacket_9, but for a single channel and for element [0]
974 * same VLC tables as process_subpacket_9 are used.
975 *
976 * @param q context
977 * @param quantized_coeffs pointer to quantized_coeffs[ch][0]
978 * @param gb bitreader context
979 * @param length packet length in bits
980 */
981 static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
982 {
983 int i, k, run, level, diff;
984
985 if (BITS_LEFT(length,gb) < 16)
986 return;
987 level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
988
989 quantized_coeffs[0] = level;
990
991 for (i = 0; i < 7; ) {
992 if (BITS_LEFT(length,gb) < 16)
993 break;
994 run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
995
996 if (BITS_LEFT(length,gb) < 16)
997 break;
998 diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
999
1000 for (k = 1; k <= run; k++)
1001 quantized_coeffs[i + k] = (level + ((k * diff) / run));
1002
1003 level += diff;
1004 i += run;
1005 }
1006 }
1007
1008
1009 /**
1010 * Related to synthesis filter, process data from packet 10
1011 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1012 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1013 *
1014 * @param q context
1015 * @param gb bitreader context
1016 * @param length packet length in bits
1017 */
1018 static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1019 {
1020 int sb, j, k, n, ch;
1021
1022 for (ch = 0; ch < q->nb_channels; ch++) {
1023 init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1024
1025 if (BITS_LEFT(length,gb) < 16) {
1026 memset(q->quantized_coeffs[ch][0], 0, 8);
1027 break;
1028 }
1029 }
1030
1031 n = q->sub_sampling + 1;
1032
1033 for (sb = 0; sb < n; sb++)
1034 for (ch = 0; ch < q->nb_channels; ch++)
1035 for (j = 0; j < 8; j++) {
1036 if (BITS_LEFT(length,gb) < 1)
1037 break;
1038 if (get_bits1(gb)) {
1039 for (k=0; k < 8; k++) {
1040 if (BITS_LEFT(length,gb) < 16)
1041 break;
1042 q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1043 }
1044 } else {
1045 for (k=0; k < 8; k++)
1046 q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1047 }
1048 }
1049
1050 n = QDM2_SB_USED(q->sub_sampling) - 4;
1051
1052 for (sb = 0; sb < n; sb++)
1053 for (ch = 0; ch < q->nb_channels; ch++) {
1054 if (BITS_LEFT(length,gb) < 16)
1055 break;
1056 q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1057 if (sb > 19)
1058 q->tone_level_idx_hi2[ch][sb] -= 16;
1059 else
1060 for (j = 0; j < 8; j++)
1061 q->tone_level_idx_mid[ch][sb][j] = -16;
1062 }
1063
1064 n = QDM2_SB_USED(q->sub_sampling) - 5;
1065
1066 for (sb = 0; sb < n; sb++)
1067 for (ch = 0; ch < q->nb_channels; ch++)
1068 for (j = 0; j < 8; j++) {
1069 if (BITS_LEFT(length,gb) < 16)
1070 break;
1071 q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1072 }
1073 }
1074
1075 /**
1076 * Process subpacket 9, init quantized_coeffs with data from it
1077 *
1078 * @param q context
1079 * @param node pointer to node with packet
1080 */
1081 static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1082 {
1083 GetBitContext gb;
1084 int i, j, k, n, ch, run, level, diff;
1085
1086 init_get_bits(&gb, node->packet->data, node->packet->size*8);
1087
1088 n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1089
1090 for (i = 1; i < n; i++)
1091 for (ch=0; ch < q->nb_channels; ch++) {
1092 level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1093 q->quantized_coeffs[ch][i][0] = level;
1094
1095 for (j = 0; j < (8 - 1); ) {
1096 run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1097 diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1098
1099 for (k = 1; k <= run; k++)
1100 q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1101
1102 level += diff;
1103 j += run;
1104 }
1105 }
1106
1107 for (ch = 0; ch < q->nb_channels; ch++)
1108 for (i = 0; i < 8; i++)
1109 q->quantized_coeffs[ch][0][i] = 0;
1110 }
1111
1112
1113 /**
1114 * Process subpacket 10 if not null, else
1115 *
1116 * @param q context
1117 * @param node pointer to node with packet
1118 * @param length packet length in bits
1119 */
1120 static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1121 {
1122 GetBitContext gb;
1123
1124 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1125
1126 if (length != 0) {
1127 init_tone_level_dequantization(q, &gb, length);
1128 fill_tone_level_array(q, 1);
1129 } else {
1130 fill_tone_level_array(q, 0);
1131 }
1132 }
1133
1134
1135 /**
1136 * Process subpacket 11
1137 *
1138 * @param q context
1139 * @param node pointer to node with packet
1140 * @param length packet length in bit
1141 */
1142 static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1143 {
1144 GetBitContext gb;
1145
1146 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1147 if (length >= 32) {
1148 int c = get_bits (&gb, 13);
1149
1150 if (c > 3)
1151 fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1152 q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1153 }
1154
1155 synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1156 }
1157
1158
1159 /**
1160 * Process subpacket 12
1161 *
1162 * @param q context
1163 * @param node pointer to node with packet
1164 * @param length packet length in bits
1165 */
1166 static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1167 {
1168 GetBitContext gb;
1169
1170 init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size*8));
1171 synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1172 }
1173
1174 /*
1175 * Process new subpackets for synthesis filter
1176 *
1177 * @param q context
1178 * @param list list with synthesis filter packets (list D)
1179 */
1180 static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1181 {
1182 QDM2SubPNode *nodes[4];
1183
1184 nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1185 if (nodes[0] != NULL)
1186 process_subpacket_9(q, nodes[0]);
1187
1188 nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1189 if (nodes[1] != NULL)
1190 process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1191 else
1192 process_subpacket_10(q, NULL, 0);
1193
1194 nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1195 if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1196 process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1197 else
1198 process_subpacket_11(q, NULL, 0);
1199
1200 nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1201 if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1202 process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1203 else
1204 process_subpacket_12(q, NULL, 0);
1205 }
1206
1207
1208 /*
1209 * Decode superblock, fill packet lists.
1210 *
1211 * @param q context
1212 */
1213 static void qdm2_decode_super_block (QDM2Context *q)
1214 {
1215 GetBitContext gb;
1216 QDM2SubPacket header, *packet;
1217 int i, packet_bytes, sub_packet_size, sub_packets_D;
1218 unsigned int next_index = 0;
1219
1220 memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1221 memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1222 memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1223
1224 q->sub_packets_B = 0;
1225 sub_packets_D = 0;
1226
1227 average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1228
1229 init_get_bits(&gb, q->compressed_data, q->compressed_size*8);
1230 qdm2_decode_sub_packet_header(&gb, &header);
1231
1232 if (header.type < 2 || header.type >= 8) {
1233 q->has_errors = 1;
1234 av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1235 return;
1236 }
1237
1238 q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1239 packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1240
1241 init_get_bits(&gb, header.data, header.size*8);
1242
1243 if (header.type == 2 || header.type == 4 || header.type == 5) {
1244 int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1245
1246 csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1247
1248 if (csum != 0) {
1249 q->has_errors = 1;
1250 av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1251 return;
1252 }
1253 }
1254
1255 q->sub_packet_list_B[0].packet = NULL;
1256 q->sub_packet_list_D[0].packet = NULL;
1257
1258 for (i = 0; i < 6; i++)
1259 if (--q->fft_level_exp[i] < 0)
1260 q->fft_level_exp[i] = 0;
1261
1262 for (i = 0; packet_bytes > 0; i++) {
1263 int j;
1264
1265 q->sub_packet_list_A[i].next = NULL;
1266
1267 if (i > 0) {
1268 q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1269
1270 /* seek to next block */
1271 init_get_bits(&gb, header.data, header.size*8);
1272 skip_bits(&gb, next_index*8);
1273
1274 if (next_index >= header.size)
1275 break;
1276 }
1277
1278 /* decode subpacket */
1279 packet = &q->sub_packets[i];
1280 qdm2_decode_sub_packet_header(&gb, packet);
1281 next_index = packet->size + get_bits_count(&gb) / 8;
1282 sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1283
1284 if (packet->type == 0)
1285 break;
1286
1287 if (sub_packet_size > packet_bytes) {
1288 if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1289 break;
1290 packet->size += packet_bytes - sub_packet_size;
1291 }
1292
1293 packet_bytes -= sub_packet_size;
1294
1295 /* add subpacket to 'all subpackets' list */
1296 q->sub_packet_list_A[i].packet = packet;
1297
1298 /* add subpacket to related list */
1299 if (packet->type == 8) {
1300 SAMPLES_NEEDED_2("packet type 8");
1301 return;
1302 } else if (packet->type >= 9 && packet->type <= 12) {
1303 /* packets for MPEG Audio like Synthesis Filter */
1304 QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1305 } else if (packet->type == 13) {
1306 for (j = 0; j < 6; j++)
1307 q->fft_level_exp[j] = get_bits(&gb, 6);
1308 } else if (packet->type == 14) {
1309 for (j = 0; j < 6; j++)
1310 q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1311 } else if (packet->type == 15) {
1312 SAMPLES_NEEDED_2("packet type 15")
1313 return;
1314 } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1315 /* packets for FFT */
1316 QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1317 }
1318 } // Packet bytes loop
1319
1320 /* **************************************************************** */
1321 if (q->sub_packet_list_D[0].packet != NULL) {
1322 process_synthesis_subpackets(q, q->sub_packet_list_D);
1323 q->do_synth_filter = 1;
1324 } else if (q->do_synth_filter) {
1325 process_subpacket_10(q, NULL, 0);
1326 process_subpacket_11(q, NULL, 0);
1327 process_subpacket_12(q, NULL, 0);
1328 }
1329 /* **************************************************************** */
1330 }
1331
1332
1333 static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1334 int offset, int duration, int channel,
1335 int exp, int phase)
1336 {
1337 if (q->fft_coefs_min_index[duration] < 0)
1338 q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1339
1340 q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1341 q->fft_coefs[q->fft_coefs_index].channel = channel;
1342 q->fft_coefs[q->fft_coefs_index].offset = offset;
1343 q->fft_coefs[q->fft_coefs_index].exp = exp;
1344 q->fft_coefs[q->fft_coefs_index].phase = phase;
1345 q->fft_coefs_index++;
1346 }
1347
1348
1349 static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1350 {
1351 int channel, stereo, phase, exp;
1352 int local_int_4, local_int_8, stereo_phase, local_int_10;
1353 int local_int_14, stereo_exp, local_int_20, local_int_28;
1354 int n, offset;
1355
1356 local_int_4 = 0;
1357 local_int_28 = 0;
1358 local_int_20 = 2;
1359 local_int_8 = (4 - duration);
1360 local_int_10 = 1 << (q->group_order - duration - 1);
1361 offset = 1;
1362
1363 while (1) {
1364 if (q->superblocktype_2_3) {
1365 while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1366 offset = 1;
1367 if (n == 0) {
1368 local_int_4 += local_int_10;
1369 local_int_28 += (1 << local_int_8);
1370 } else {
1371 local_int_4 += 8*local_int_10;
1372 local_int_28 += (8 << local_int_8);
1373 }
1374 }
1375 offset += (n - 2);
1376 } else {
1377 offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1378 while (offset >= (local_int_10 - 1)) {
1379 offset += (1 - (local_int_10 - 1));
1380 local_int_4 += local_int_10;
1381 local_int_28 += (1 << local_int_8);
1382 }
1383 }
1384
1385 if (local_int_4 >= q->group_size)
1386 return;
1387
1388 local_int_14 = (offset >> local_int_8);
1389
1390 if (q->nb_channels > 1) {
1391 channel = get_bits1(gb);
1392 stereo = get_bits1(gb);
1393 } else {
1394 channel = 0;
1395 stereo = 0;
1396 }
1397
1398 exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1399 exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1400 exp = (exp < 0) ? 0 : exp;
1401
1402 phase = get_bits(gb, 3);
1403 stereo_exp = 0;
1404 stereo_phase = 0;
1405
1406 if (stereo) {
1407 stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1408 stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1409 if (stereo_phase < 0)
1410 stereo_phase += 8;
1411 }
1412
1413 if (q->frequency_range > (local_int_14 + 1)) {
1414 int sub_packet = (local_int_20 + local_int_28);
1415
1416 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1417 if (stereo)
1418 qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1419 }
1420
1421 offset++;
1422 }
1423 }
1424
1425
1426 static void qdm2_decode_fft_packets (QDM2Context *q)
1427 {
1428 int i, j, min, max, value, type, unknown_flag;
1429 GetBitContext gb;
1430
1431 if (q->sub_packet_list_B[0].packet == NULL)
1432 return;
1433
1434 /* reset minimum indexes for FFT coefficients */
1435 q->fft_coefs_index = 0;
1436 for (i=0; i < 5; i++)
1437 q->fft_coefs_min_index[i] = -1;
1438
1439 /* process subpackets ordered by type, largest type first */
1440 for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1441 QDM2SubPacket *packet= NULL;
1442
1443 /* find subpacket with largest type less than max */
1444 for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1445 value = q->sub_packet_list_B[j].packet->type;
1446 if (value > min && value < max) {
1447 min = value;
1448 packet = q->sub_packet_list_B[j].packet;
1449 }
1450 }
1451
1452 max = min;
1453
1454 /* check for errors (?) */
1455 if (!packet)
1456 return;
1457
1458 if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1459 return;
1460
1461 /* decode FFT tones */
1462 init_get_bits (&gb, packet->data, packet->size*8);
1463
1464 if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1465 unknown_flag = 1;
1466 else
1467 unknown_flag = 0;
1468
1469 type = packet->type;
1470
1471 if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1472 int duration = q->sub_sampling + 5 - (type & 15);
1473
1474 if (duration >= 0 && duration < 4)
1475 qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1476 } else if (type == 31) {
1477 for (j=0; j < 4; j++)
1478 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1479 } else if (type == 46) {
1480 for (j=0; j < 6; j++)
1481 q->fft_level_exp[j] = get_bits(&gb, 6);
1482 for (j=0; j < 4; j++)
1483 qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1484 }
1485 } // Loop on B packets
1486
1487 /* calculate maximum indexes for FFT coefficients */
1488 for (i = 0, j = -1; i < 5; i++)
1489 if (q->fft_coefs_min_index[i] >= 0) {
1490 if (j >= 0)
1491 q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1492 j = i;
1493 }
1494 if (j >= 0)
1495 q->fft_coefs_max_index[j] = q->fft_coefs_index;
1496 }
1497
1498
1499 static void qdm2_fft_generate_tone (QDM2Context *q, FFTTone *tone)
1500 {
1501 float level, f[6];
1502 int i;
1503 QDM2Complex c;
1504 const double iscale = 2.0*M_PI / 512.0;
1505
1506 tone->phase += tone->phase_shift;
1507
1508 /* calculate current level (maximum amplitude) of tone */
1509 level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1510 c.im = level * sin(tone->phase*iscale);
1511 c.re = level * cos(tone->phase*iscale);
1512
1513 /* generate FFT coefficients for tone */
1514 if (tone->duration >= 3 || tone->cutoff >= 3) {
1515 tone->samples_im[0] += c.im;
1516 tone->samples_re[0] += c.re;
1517 tone->samples_im[1] -= c.im;
1518 tone->samples_re[1] -= c.re;
1519 } else {
1520 f[1] = -tone->table[4];
1521 f[0] = tone->table[3] - tone->table[0];
1522 f[2] = 1.0 - tone->table[2] - tone->table[3];
1523 f[3] = tone->table[1] + tone->table[4] - 1.0;
1524 f[4] = tone->table[0] - tone->table[1];
1525 f[5] = tone->table[2];
1526 for (i = 0; i < 2; i++) {
1527 tone->samples_re[fft_cutoff_index_table[tone->cutoff][i]] += c.re * f[i];
1528 tone->samples_im[fft_cutoff_index_table[tone->cutoff][i]] += c.im *((tone->cutoff <= i) ? -f[i] : f[i]);
1529 }
1530 for (i = 0; i < 4; i++) {
1531 tone->samples_re[i] += c.re * f[i+2];
1532 tone->samples_im[i] += c.im * f[i+2];
1533 }
1534 }
1535
1536 /* copy the tone if it has not yet died out */
1537 if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1538 memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1539 q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1540 }
1541 }
1542
1543
1544 static void qdm2_fft_tone_synthesizer (QDM2Context *q, int sub_packet)
1545 {
1546 int i, j, ch;
1547 const double iscale = 0.25 * M_PI;
1548
1549 for (ch = 0; ch < q->channels; ch++) {
1550 memset(q->fft.samples_im[ch], 0, q->fft_size * sizeof(float));
1551 memset(q->fft.samples_re[ch], 0, q->fft_size * sizeof(float));
1552 }
1553
1554
1555 /* apply FFT tones with duration 4 (1 FFT period) */
1556 if (q->fft_coefs_min_index[4] >= 0)
1557 for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1558 float level;
1559 QDM2Complex c;
1560
1561 if (q->fft_coefs[i].sub_packet != sub_packet)
1562 break;
1563
1564 ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1565 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];
1566
1567 c.re = level * cos(q->fft_coefs[i].phase * iscale);
1568 c.im = level * sin(q->fft_coefs[i].phase * iscale);
1569 q->fft.samples_re[ch][q->fft_coefs[i].offset + 0] += c.re;
1570 q->fft.samples_im[ch][q->fft_coefs[i].offset + 0] += c.im;
1571 q->fft.samples_re[ch][q->fft_coefs[i].offset + 1] -= c.re;
1572 q->fft.samples_im[ch][q->fft_coefs[i].offset + 1] -= c.im;
1573 }
1574
1575 /* generate existing FFT tones */
1576 for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1577 qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1578 q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1579 }
1580
1581 /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1582 for (i = 0; i < 4; i++)
1583 if (q->fft_coefs_min_index[i] >= 0) {
1584 for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1585 int offset, four_i;
1586 FFTTone tone;
1587
1588 if (q->fft_coefs[j].sub_packet != sub_packet)
1589 break;
1590
1591 four_i = (4 - i);
1592 offset = q->fft_coefs[j].offset >> four_i;
1593 ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1594
1595 if (offset < q->frequency_range) {
1596 if (offset < 2)
1597 tone.cutoff = offset;
1598 else
1599 tone.cutoff = (offset >= 60) ? 3 : 2;
1600
1601 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];
1602 tone.samples_im = &q->fft.samples_im[ch][offset];
1603 tone.samples_re = &q->fft.samples_re[ch][offset];
1604 tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1605 tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1606 tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1607 tone.duration = i;
1608 tone.time_index = 0;
1609
1610 qdm2_fft_generate_tone(q, &tone);
1611 }
1612 }
1613 q->fft_coefs_min_index[i] = j;
1614 }
1615 }
1616
1617
1618 static void qdm2_calculate_fft (QDM2Context *q, int channel, int sub_packet)
1619 {
1620 const int n = 1 << (q->fft_order - 1);
1621 const int n2 = n >> 1;
1622 const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.25f : 0.50f;
1623 float c, s, f0, f1, f2, f3;
1624 int i, j;
1625
1626 /* prerotation (or something like that) */
1627 for (i=1; i < n2; i++) {
1628 j = (n - i);
1629 c = q->exptab[i].re;
1630 s = -q->exptab[i].im;
1631 f0 = (q->fft.samples_re[channel][i] - q->fft.samples_re[channel][j]) * gain;
1632 f1 = (q->fft.samples_im[channel][i] + q->fft.samples_im[channel][j]) * gain;
1633 f2 = (q->fft.samples_re[channel][i] + q->fft.samples_re[channel][j]) * gain;
1634 f3 = (q->fft.samples_im[channel][i] - q->fft.samples_im[channel][j]) * gain;
1635 q->fft.complex[i].re = s * f0 - c * f1 + f2;
1636 q->fft.complex[i].im = c * f0 + s * f1 + f3;
1637 q->fft.complex[j].re = -s * f0 + c * f1 + f2;
1638 q->fft.complex[j].im = c * f0 + s * f1 - f3;
1639 }
1640
1641 q->fft.complex[ 0].re = q->fft.samples_re[channel][ 0] * gain * 2.0;
1642 q->fft.complex[ 0].im = q->fft.samples_re[channel][ 0] * gain * 2.0;
1643 q->fft.complex[n2].re = q->fft.samples_re[channel][n2] * gain * 2.0;
1644 q->fft.complex[n2].im = -q->fft.samples_im[channel][n2] * gain * 2.0;
1645
1646 ff_fft_permute(&q->fft_ctx, (FFTComplex *) q->fft.complex);
1647 ff_fft_calc (&q->fft_ctx, (FFTComplex *) q->fft.complex);
1648 /* add samples to output buffer */
1649 for (i = 0; i < ((q->fft_frame_size + 15) & ~15); i++)
1650 q->output_buffer[q->channels * i + channel] += ((float *) q->fft.complex)[i];
1651 }
1652
1653
1654 /**
1655 * @param q context
1656 * @param index subpacket number
1657 */
1658 static void qdm2_synthesis_filter (QDM2Context *q, int index)
1659 {
1660 OUT_INT samples[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
1661 int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1662
1663 /* copy sb_samples */
1664 sb_used = QDM2_SB_USED(q->sub_sampling);
1665
1666 for (ch = 0; ch < q->channels; ch++)
1667 for (i = 0; i < 8; i++)
1668 for (k=sb_used; k < SBLIMIT; k++)
1669 q->sb_samples[ch][(8 * index) + i][k] = 0;
1670
1671 for (ch = 0; ch < q->nb_channels; ch++) {
1672 OUT_INT *samples_ptr = samples + ch;
1673
1674 for (i = 0; i < 8; i++) {
1675 ff_mpa_synth_filter(q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1676 mpa_window, &dither_state,
1677 samples_ptr, q->nb_channels,
1678 q->sb_samples[ch][(8 * index) + i]);
1679 samples_ptr += 32 * q->nb_channels;
1680 }
1681 }
1682
1683 /* add samples to output buffer */
1684 sub_sampling = (4 >> q->sub_sampling);
1685
1686 for (ch = 0; ch < q->channels; ch++)
1687 for (i = 0; i < q->frame_size; i++)
1688 q->output_buffer[q->channels * i + ch] += (float)(samples[q->nb_channels * sub_sampling * i + ch] >> (sizeof(OUT_INT)*8-16));
1689 }
1690
1691
1692 /**
1693 * Init static data (does not depend on specific file)
1694 *
1695 * @param q context
1696 */
1697 static void qdm2_init(QDM2Context *q) {
1698 static int initialized = 0;
1699
1700 if (initialized != 0)
1701 return;
1702 initialized = 1;
1703
1704 qdm2_init_vlc();
1705 ff_mpa_synth_init(mpa_window);
1706 softclip_table_init();
1707 rnd_table_init();
1708 init_noise_samples();
1709
1710 av_log(NULL, AV_LOG_DEBUG, "init done\n");
1711 }
1712
1713
1714 #if 0
1715 static void dump_context(QDM2Context *q)
1716 {
1717 int i;
1718 #define PRINT(a,b) av_log(NULL,AV_LOG_DEBUG," %s = %d\n", a, b);
1719 PRINT("compressed_data",q->compressed_data);
1720 PRINT("compressed_size",q->compressed_size);
1721 PRINT("frame_size",q->frame_size);
1722 PRINT("checksum_size",q->checksum_size);
1723 PRINT("channels",q->channels);
1724 PRINT("nb_channels",q->nb_channels);
1725 PRINT("fft_frame_size",q->fft_frame_size);
1726 PRINT("fft_size",q->fft_size);
1727 PRINT("sub_sampling",q->sub_sampling);
1728 PRINT("fft_order",q->fft_order);
1729 PRINT("group_order",q->group_order);
1730 PRINT("group_size",q->group_size);
1731 PRINT("sub_packet",q->sub_packet);
1732 PRINT("frequency_range",q->frequency_range);
1733 PRINT("has_errors",q->has_errors);
1734 PRINT("fft_tone_end",q->fft_tone_end);
1735 PRINT("fft_tone_start",q->fft_tone_start);
1736 PRINT("fft_coefs_index",q->fft_coefs_index);
1737 PRINT("coeff_per_sb_select",q->coeff_per_sb_select);
1738 PRINT("cm_table_select",q->cm_table_select);
1739 PRINT("noise_idx",q->noise_idx);
1740
1741 for (i = q->fft_tone_start; i < q->fft_tone_end; i++)
1742 {
1743 FFTTone *t = &q->fft_tones[i];
1744
1745 av_log(NULL,AV_LOG_DEBUG,"Tone (%d) dump:\n", i);
1746 av_log(NULL,AV_LOG_DEBUG," level = %f\n", t->level);
1747 // PRINT(" level", t->level);
1748 PRINT(" phase", t->phase);
1749 PRINT(" phase_shift", t->phase_shift);
1750 PRINT(" duration", t->duration);
1751 PRINT(" samples_im", t->samples_im);
1752 PRINT(" samples_re", t->samples_re);
1753 PRINT(" table", t->table);
1754 }
1755
1756 }
1757 #endif
1758
1759
1760 /**
1761 * Init parameters from codec extradata
1762 */
1763 static int qdm2_decode_init(AVCodecContext *avctx)
1764 {
1765 QDM2Context *s = avctx->priv_data;
1766 uint8_t *extradata;
1767 int extradata_size;
1768 int tmp_val, tmp, size;
1769 int i;
1770 float alpha;
1771
1772 /* extradata parsing
1773
1774 Structure:
1775 wave {
1776 frma (QDM2)
1777 QDCA
1778 QDCP
1779 }
1780
1781 32 size (including this field)
1782 32 tag (=frma)
1783 32 type (=QDM2 or QDMC)
1784
1785 32 size (including this field, in bytes)
1786 32 tag (=QDCA) // maybe mandatory parameters
1787 32 unknown (=1)
1788 32 channels (=2)
1789 32 samplerate (=44100)
1790 32 bitrate (=96000)
1791 32 block size (=4096)
1792 32 frame size (=256) (for one channel)
1793 32 packet size (=1300)
1794
1795 32 size (including this field, in bytes)
1796 32 tag (=QDCP) // maybe some tuneable parameters
1797 32 float1 (=1.0)
1798 32 zero ?
1799 32 float2 (=1.0)
1800 32 float3 (=1.0)
1801 32 unknown (27)
1802 32 unknown (8)
1803 32 zero ?
1804 */
1805
1806 if (!avctx->extradata || (avctx->extradata_size < 48)) {
1807 av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1808 return -1;
1809 }
1810
1811 extradata = avctx->extradata;
1812 extradata_size = avctx->extradata_size;
1813
1814 while (extradata_size > 7) {
1815 if (!memcmp(extradata, "frmaQDM", 7))
1816 break;
1817 extradata++;
1818 extradata_size--;
1819 }
1820
1821 if (extradata_size < 12) {
1822 av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1823 extradata_size);
1824 return -1;
1825 }
1826
1827 if (memcmp(extradata, "frmaQDM", 7)) {
1828 av_log(avctx, AV_LOG_ERROR, "invalid headers, QDM? not found\n");
1829 return -1;
1830 }
1831
1832 if (extradata[7] == 'C') {
1833 // s->is_qdmc = 1;
1834 av_log(avctx, AV_LOG_ERROR, "stream is QDMC version 1, which is not supported\n");
1835 return -1;
1836 }
1837
1838 extradata += 8;
1839 extradata_size -= 8;
1840
1841 size = AV_RB32(extradata);
1842
1843 if(size > extradata_size){
1844 av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1845 extradata_size, size);
1846 return -1;
1847 }
1848
1849 extradata += 4;
1850 av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1851 if (AV_RB32(extradata) != MKBETAG('Q','D','C','A')) {
1852 av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1853 return -1;
1854 }
1855
1856 extradata += 8;
1857
1858 avctx->channels = s->nb_channels = s->channels = AV_RB32(extradata);
1859 extradata += 4;
1860
1861 avctx->sample_rate = AV_RB32(extradata);
1862 extradata += 4;
1863
1864 avctx->bit_rate = AV_RB32(extradata);
1865 extradata += 4;
1866
1867 s->group_size = AV_RB32(extradata);
1868 extradata += 4;
1869
1870 s->fft_size = AV_RB32(extradata);
1871 extradata += 4;
1872
1873 s->checksum_size = AV_RB32(extradata);
1874 extradata += 4;
1875
1876 s->fft_order = av_log2(s->fft_size) + 1;
1877 s->fft_frame_size = 2 * s->fft_size; // complex has two floats
1878
1879 // something like max decodable tones
1880 s->group_order = av_log2(s->group_size) + 1;
1881 s->frame_size = s->group_size / 16; // 16 iterations per super block
1882
1883 s->sub_sampling = s->fft_order - 7;
1884 s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1885
1886 switch ((s->sub_sampling * 2 + s->channels - 1)) {
1887 case 0: tmp = 40; break;
1888 case 1: tmp = 48; break;
1889 case 2: tmp = 56; break;
1890 case 3: tmp = 72; break;
1891 case 4: tmp = 80; break;
1892 case 5: tmp = 100;break;
1893 default: tmp=s->sub_sampling; break;
1894 }
1895 tmp_val = 0;
1896 if ((tmp * 1000) < avctx->bit_rate) tmp_val = 1;
1897 if ((tmp * 1440) < avctx->bit_rate) tmp_val = 2;
1898 if ((tmp * 1760) < avctx->bit_rate) tmp_val = 3;
1899 if ((tmp * 2240) < avctx->bit_rate) tmp_val = 4;
1900 s->cm_table_select = tmp_val;
1901
1902 if (s->sub_sampling == 0)
1903 tmp = 7999;
1904 else
1905 tmp = ((-(s->sub_sampling -1)) & 8000) + 20000;
1906 /*
1907 0: 7999 -> 0
1908 1: 20000 -> 2
1909 2: 28000 -> 2
1910 */
1911 if (tmp < 8000)
1912 s->coeff_per_sb_select = 0;
1913 else if (tmp <= 16000)
1914 s->coeff_per_sb_select = 1;
1915 else
1916 s->coeff_per_sb_select = 2;
1917
1918 // Fail on unknown fft order, if it's > 9 it can overflow s->exptab[]
1919 if ((s->fft_order < 7) || (s->fft_order > 9)) {
1920 av_log(avctx, AV_LOG_ERROR, "Unknown FFT order (%d), contact the developers!\n", s->fft_order);
1921 return -1;
1922 }
1923
1924 ff_fft_init(&s->fft_ctx, s->fft_order - 1, 1);
1925
1926 for (i = 1; i < (1 << (s->fft_order - 2)); i++) {
1927 alpha = 2 * M_PI * (float)i / (float)(1 << (s->fft_order - 1));
1928 s->exptab[i].re = cos(alpha);
1929 s->exptab[i].im = sin(alpha);
1930 }
1931
1932 qdm2_init(s);
1933
1934 avctx->sample_fmt = SAMPLE_FMT_S16;
1935
1936 // dump_context(s);
1937 return 0;
1938 }
1939
1940
1941 static int qdm2_decode_close(AVCodecContext *avctx)
1942 {
1943 QDM2Context *s = avctx->priv_data;
1944
1945 ff_fft_end(&s->fft_ctx);
1946
1947 return 0;
1948 }
1949
1950
1951 static void qdm2_decode (QDM2Context *q, const uint8_t *in, int16_t *out)
1952 {
1953 int ch, i;
1954 const int frame_size = (q->frame_size * q->channels);
1955
1956 /* select input buffer */
1957 q->compressed_data = in;
1958 q->compressed_size = q->checksum_size;
1959
1960 // dump_context(q);
1961
1962 /* copy old block, clear new block of output samples */
1963 memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1964 memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1965
1966 /* decode block of QDM2 compressed data */
1967 if (q->sub_packet == 0) {
1968 q->has_errors = 0; // zero it for a new super block
1969 av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1970 qdm2_decode_super_block(q);
1971 }
1972
1973 /* parse subpackets */
1974 if (!q->has_errors) {
1975 if (q->sub_packet == 2)
1976 qdm2_decode_fft_packets(q);
1977
1978 qdm2_fft_tone_synthesizer(q, q->sub_packet);
1979 }
1980
1981 /* sound synthesis stage 1 (FFT) */
1982 for (ch = 0; ch < q->channels; ch++) {
1983 qdm2_calculate_fft(q, ch, q->sub_packet);
1984
1985 if (!q->has_errors && q->sub_packet_list_C[0].packet != NULL) {
1986 SAMPLES_NEEDED_2("has errors, and C list is not empty")
1987 return;
1988 }
1989 }
1990
1991 /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1992 if (!q->has_errors && q->do_synth_filter)
1993 qdm2_synthesis_filter(q, q->sub_packet);
1994
1995 q->sub_packet = (q->sub_packet + 1) % 16;
1996
1997 /* clip and convert output float[] to 16bit signed samples */
1998 for (i = 0; i < frame_size; i++) {
1999 int value = (int)q->output_buffer[i];
2000
2001 if (value > SOFTCLIP_THRESHOLD)
2002 value = (value > HARDCLIP_THRESHOLD) ? 32767 : softclip_table[ value - SOFTCLIP_THRESHOLD];
2003 else if (value < -SOFTCLIP_THRESHOLD)
2004 value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
2005
2006 out[i] = value;
2007 }
2008 }
2009
2010
2011 static int qdm2_decode_frame(AVCodecContext *avctx,
2012 void *data, int *data_size,
2013 const uint8_t *buf, int buf_size)
2014 {
2015 QDM2Context *s = avctx->priv_data;
2016
2017 if(!buf)
2018 return 0;
2019 if(buf_size < s->checksum_size)
2020 return -1;
2021
2022 *data_size = s->channels * s->frame_size * sizeof(int16_t);
2023
2024 av_log(avctx, AV_LOG_DEBUG, "decode(%d): %p[%d] -> %p[%d]\n",
2025 buf_size, buf, s->checksum_size, data, *data_size);
2026
2027 qdm2_decode(s, buf, data);
2028
2029 // reading only when next superblock found
2030 if (s->sub_packet == 0) {
2031 return s->checksum_size;
2032 }
2033
2034 return 0;
2035 }
2036
2037 AVCodec qdm2_decoder =
2038 {
2039 .name = "qdm2",
2040 .type = CODEC_TYPE_AUDIO,
2041 .id = CODEC_ID_QDM2,
2042 .priv_data_size = sizeof(QDM2Context),
2043 .init = qdm2_decode_init,
2044 .close = qdm2_decode_close,
2045 .decode = qdm2_decode_frame,
2046 .long_name = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
2047 };
VDPAU for FFmpeg