Revision 3135258e

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libavcodec/qdm2.c
1
/*
2
 * QDM2 compatible decoder
3
 * Copyright (c) 2003 Ewald Snel
4
 * Copyright (c) 2005 Benjamin Larsson
5
 * Copyright (c) 2005 Alex Beregszaszi
6
 * Copyright (c) 2005 Roberto Togni
7
 *
8
 * This library is free software; you can redistribute it and/or
9
 * modify it under the terms of the GNU Lesser General Public
10
 * License as published by the Free Software Foundation; either
11
 * version 2 of the License, or (at your option) any later version.
12
 *
13
 * This library is distributed in the hope that it will be useful,
14
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
16
 * Lesser General Public License for more details.
17
 *
18
 * You should have received a copy of the GNU Lesser General Public
19
 * License along with this library; if not, write to the Free Software
20
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
21
 *
22
 */
23

  
24
/**
25
 * @file qdm2.c
26
 * QDM2 decoder
27
 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
28
 * The decoder is not perfect yet, there are still some distorions expecially
29
 * on files encoded with 16 or 8 subbands
30
 */
31

  
32
#include <math.h>
33
#include <stddef.h>
34
#include <stdio.h>
35

  
36
#define ALT_BITSTREAM_READER_LE
37
#include "avcodec.h"
38
#include "bitstream.h"
39
#include "dsputil.h"
40

  
41
#ifdef CONFIG_MPEGAUDIO_HP
42
#define USE_HIGHPRECISION
43
#endif
44

  
45
#include "mpegaudio.h"
46

  
47
#include "qdm2data.h"
48

  
49
#undef NDEBUG
50
#include <assert.h>
51

  
52

  
53
#define SOFTCLIP_THRESHOLD 27600
54
#define HARDCLIP_THRESHOLD 35716
55

  
56

  
57
#define QDM2_LIST_ADD(list, size, packet) \
58
do { \
59
      if (size > 0) { \
60
    list[size - 1].next = &list[size]; \
61
      } \
62
      list[size].packet = packet; \
63
      list[size].next = NULL; \
64
      size++; \
65
} while(0)
66

  
67
// Result is 8, 16 or 30
68
#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
69

  
70
#define FIX_NOISE_IDX(noise_idx) \
71
  if ((noise_idx) >= 3840) \
72
    (noise_idx) -= 3840; \
73

  
74
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
75

  
76
#define BITS_LEFT(length,gb) ((length) - get_bits_count ((gb)))
77

  
78
#define SAMPLES_NEEDED \
79
     av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
80

  
81
#define SAMPLES_NEEDED_2(why) \
82
     av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
83

  
84

  
85
typedef int8_t sb_int8_array[2][30][64];
86

  
87
/**
88
 * Subpacket
89
 */
90
typedef struct {
91
    int type;            ///< subpacket type
92
    unsigned int size;   ///< subpacket size
93
    const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
94
} QDM2SubPacket;
95

  
96
/**
97
 * A node in subpacket list
98
 */
99
typedef struct _QDM2SubPNode {
100
    QDM2SubPacket *packet;      ///< packet
101
    struct _QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
102
} QDM2SubPNode;
103

  
104
typedef struct {
105
    float level;
106
    float *samples_im;
107
    float *samples_re;
108
    float *table;
109
    int   phase;
110
    int   phase_shift;
111
    int   duration;
112
    short time_index;
113
    short cutoff;
114
} FFTTone;
115

  
116
typedef struct {
117
    int16_t sub_packet;
118
    uint8_t channel;
119
    int16_t offset;
120
    int16_t exp;
121
    uint8_t phase;
122
} FFTCoefficient;
123

  
124
typedef struct {
125
    float re;
126
    float im;
127
} QDM2Complex;
128

  
129
typedef struct {
130
    QDM2Complex complex[256 + 1] __attribute__((aligned(16)));
131
    float       samples_im[MPA_MAX_CHANNELS][256];
132
    float       samples_re[MPA_MAX_CHANNELS][256];
133
} QDM2FFT;
134

  
135
/**
136
 * QDM2 decoder context
137
 */
138
typedef struct {
139
    /// Parameters from codec header, do not change during playback
140
    int nb_channels;         ///< number of channels
141
    int channels;            ///< number of channels
142
    int group_size;          ///< size of frame group (16 frames per group)
143
    int fft_size;            ///< size of FFT, in complex numbers
144
    int checksum_size;       ///< size of data block, used also for checksum
145

  
146
    /// Parameters built from header parameters, do not change during playback
147
    int group_order;         ///< order of frame group
148
    int fft_order;           ///< order of FFT (actually fftorder+1)
149
    int fft_frame_size;      ///< size of fft frame, in components (1 comples = re + im)
150
    int frame_size;          ///< size of data frame
151
    int frequency_range;
152
    int sub_sampling;        ///< subsampling: 0=25%, 1=50%, 2=100% */
153
    int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
154
    int cm_table_select;     ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
155

  
156
    /// Packets and packet lists
157
    QDM2SubPacket sub_packets[16];      ///< the packets themselves
158
    QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
159
    QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
160
    int sub_packets_B;                  ///< number of packets on 'B' list
161
    QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
162
    QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
163

  
164
    /// FFT and tones
165
    FFTTone fft_tones[1000];
166
    int fft_tone_start;
167
    int fft_tone_end;
168
    FFTCoefficient fft_coefs[1000];
169
    int fft_coefs_index;
170
    int fft_coefs_min_index[5];
171
    int fft_coefs_max_index[5];
172
    int fft_level_exp[6];
173
    FFTContext fft_ctx;
174
    FFTComplex exptab[128];
175
    QDM2FFT fft;
176

  
177
    /// I/O data
178
    uint8_t *compressed_data;
179
    int compressed_size;
180
    float output_buffer[1024];
181

  
182
    /// Synthesis filter
183
    MPA_INT synth_buf[MPA_MAX_CHANNELS][512*2] __attribute__((aligned(16)));
184
    int synth_buf_offset[MPA_MAX_CHANNELS];
185
    int32_t sb_samples[MPA_MAX_CHANNELS][128][SBLIMIT] __attribute__((aligned(16)));
186

  
187
    /// Mixed temporary data used in decoding
188
    float tone_level[MPA_MAX_CHANNELS][30][64];
189
    int8_t coding_method[MPA_MAX_CHANNELS][30][64];
190
    int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
191
    int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
192
    int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
193
    int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
194
    int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
195
    int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
196
    int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
197

  
198
    // Flags
199
    int has_errors;         ///< packet have errors
200
    int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
201
    int do_synth_filter;    ///< used to perform or skip synthesis filter
202

  
203
    int sub_packet;
204
    int noise_idx; ///< Index for dithering noise table
205
} QDM2Context;
206

  
207

  
208
static uint8_t empty_buffer[FF_INPUT_BUFFER_PADDING_SIZE];
209

  
210
static VLC vlc_tab_level;
211
static VLC vlc_tab_diff;
212
static VLC vlc_tab_run;
213
static VLC fft_level_exp_alt_vlc;
214
static VLC fft_level_exp_vlc;
215
static VLC fft_stereo_exp_vlc;
216
static VLC fft_stereo_phase_vlc;
217
static VLC vlc_tab_tone_level_idx_hi1;
218
static VLC vlc_tab_tone_level_idx_mid;
219
static VLC vlc_tab_tone_level_idx_hi2;
220
static VLC vlc_tab_type30;
221
static VLC vlc_tab_type34;
222
static VLC vlc_tab_fft_tone_offset[5];
223

  
224
static uint16_t softclip_table[HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1];
225
static float noise_table[4096];
226
static uint8_t random_dequant_index[256][5];
227
static uint8_t random_dequant_type24[128][3];
228
static float noise_samples[128];
229

  
230
static MPA_INT mpa_window[512] __attribute__((aligned(16)));
231

  
232

  
233
static void softclip_table_init() {
234
    int i;
235
    double dfl = SOFTCLIP_THRESHOLD - 32767;
236
    float delta = 1.0 / -dfl;
237
    for (i = 0; i < HARDCLIP_THRESHOLD - SOFTCLIP_THRESHOLD + 1; i++)
238
        softclip_table[i] = SOFTCLIP_THRESHOLD - ((int)(sin((float)i * delta) * dfl) & 0x0000FFFF);
239
}
240

  
241

  
242
// random generated table
243
static void rnd_table_init() {
244
    int i,j;
245
    uint32_t ldw,hdw;
246
    uint64_t tmp64_1;
247
    uint64_t random_seed = 0;
248
    float delta = 1.0 / 16384.0;
249
    for(i = 0; i < 4096 ;i++) {
250
        random_seed = random_seed * 214013 + 2531011;
251
        noise_table[i] = (delta * (float)(((int32_t)random_seed >> 16) & 0x00007FFF)- 1.0) * 1.3;
252
    }
253

  
254
    for (i = 0; i < 256 ;i++) {
255
        random_seed = 81;
256
        ldw = i;
257
        for (j = 0; j < 5 ;j++) {
258
            random_dequant_index[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
259
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
260
            tmp64_1 = (random_seed * 0x55555556);
261
            hdw = (uint32_t)(tmp64_1 >> 32);
262
            random_seed = (uint64_t)(hdw + (ldw >> 31));
263
        }
264
    }
265
    for (i = 0; i < 128 ;i++) {
266
        random_seed = 25;
267
        ldw = i;
268
        for (j = 0; j < 3 ;j++) {
269
            random_dequant_type24[i][j] = (uint8_t)((ldw / random_seed) & 0xFF);
270
            ldw = (uint32_t)ldw % (uint32_t)random_seed;
271
            tmp64_1 = (random_seed * 0x66666667);
272
            hdw = (uint32_t)(tmp64_1 >> 33);
273
            random_seed = hdw + (ldw >> 31);
274
        }
275
    }
276
}
277

  
278

  
279
static void init_noise_samples() {
280
    int i;
281
    int random_seed = 0;
282
    float delta = 1.0 / 16384.0;
283
    for (i = 0; i < 128;i++) {
284
        random_seed = random_seed * 214013 + 2531011;
285
        noise_samples[i] = (delta * (float)((random_seed >> 16) & 0x00007fff) - 1.0);
286
    }
287
}
288

  
289

  
290
static void qdm2_init_vlc()
291
{
292
    init_vlc (&vlc_tab_level, 8, 24,
293
        vlc_tab_level_huffbits, 1, 1,
294
        vlc_tab_level_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
295

  
296
    init_vlc (&vlc_tab_diff, 8, 37,
297
        vlc_tab_diff_huffbits, 1, 1,
298
        vlc_tab_diff_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
299

  
300
    init_vlc (&vlc_tab_run, 5, 6,
301
        vlc_tab_run_huffbits, 1, 1,
302
        vlc_tab_run_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
303

  
304
    init_vlc (&fft_level_exp_alt_vlc, 8, 28,
305
        fft_level_exp_alt_huffbits, 1, 1,
306
        fft_level_exp_alt_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
307

  
308
    init_vlc (&fft_level_exp_vlc, 8, 20,
309
        fft_level_exp_huffbits, 1, 1,
310
        fft_level_exp_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
311

  
312
    init_vlc (&fft_stereo_exp_vlc, 6, 7,
313
        fft_stereo_exp_huffbits, 1, 1,
314
        fft_stereo_exp_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
315

  
316
    init_vlc (&fft_stereo_phase_vlc, 6, 9,
317
        fft_stereo_phase_huffbits, 1, 1,
318
        fft_stereo_phase_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
319

  
320
    init_vlc (&vlc_tab_tone_level_idx_hi1, 8, 20,
321
        vlc_tab_tone_level_idx_hi1_huffbits, 1, 1,
322
        vlc_tab_tone_level_idx_hi1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
323

  
324
    init_vlc (&vlc_tab_tone_level_idx_mid, 8, 24,
325
        vlc_tab_tone_level_idx_mid_huffbits, 1, 1,
326
        vlc_tab_tone_level_idx_mid_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
327

  
328
    init_vlc (&vlc_tab_tone_level_idx_hi2, 8, 24,
329
        vlc_tab_tone_level_idx_hi2_huffbits, 1, 1,
330
        vlc_tab_tone_level_idx_hi2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
331

  
332
    init_vlc (&vlc_tab_type30, 6, 9,
333
        vlc_tab_type30_huffbits, 1, 1,
334
        vlc_tab_type30_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
335

  
336
    init_vlc (&vlc_tab_type34, 5, 10,
337
        vlc_tab_type34_huffbits, 1, 1,
338
        vlc_tab_type34_huffcodes, 1, 1, INIT_VLC_USE_STATIC | INIT_VLC_LE);
339

  
340
    init_vlc (&vlc_tab_fft_tone_offset[0], 8, 23,
341
        vlc_tab_fft_tone_offset_0_huffbits, 1, 1,
342
        vlc_tab_fft_tone_offset_0_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
343

  
344
    init_vlc (&vlc_tab_fft_tone_offset[1], 8, 28,
345
        vlc_tab_fft_tone_offset_1_huffbits, 1, 1,
346
        vlc_tab_fft_tone_offset_1_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
347

  
348
    init_vlc (&vlc_tab_fft_tone_offset[2], 8, 32,
349
        vlc_tab_fft_tone_offset_2_huffbits, 1, 1,
350
        vlc_tab_fft_tone_offset_2_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
351

  
352
    init_vlc (&vlc_tab_fft_tone_offset[3], 8, 35,
353
        vlc_tab_fft_tone_offset_3_huffbits, 1, 1,
354
        vlc_tab_fft_tone_offset_3_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
355

  
356
    init_vlc (&vlc_tab_fft_tone_offset[4], 8, 38,
357
        vlc_tab_fft_tone_offset_4_huffbits, 1, 1,
358
        vlc_tab_fft_tone_offset_4_huffcodes, 2, 2, INIT_VLC_USE_STATIC | INIT_VLC_LE);
359
}
360

  
361

  
362
/* for floating point to fixed point conversion */
363
static float f2i_scale = (float) (1 << (FRAC_BITS - 15));
364

  
365

  
366
static int qdm2_get_vlc (GetBitContext *gb, VLC *vlc, int flag, int depth)
367
{
368
    int value;
369

  
370
    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
371

  
372
    /* stage-2, 3 bits exponent escape sequence */
373
    if (value-- == 0)
374
        value = get_bits (gb, get_bits (gb, 3) + 1);
375

  
376
    /* stage-3, optional */
377
    if (flag) {
378
        int tmp = vlc_stage3_values[value];
379

  
380
        if ((value & ~3) > 0)
381
            tmp += get_bits (gb, (value >> 2));
382
        value = tmp;
383
    }
384

  
385
    return value;
386
}
387

  
388

  
389
static int qdm2_get_se_vlc (VLC *vlc, GetBitContext *gb, int depth)
390
{
391
    int value = qdm2_get_vlc (gb, vlc, 0, depth);
392

  
393
    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
394
}
395

  
396

  
397
/**
398
 * QDM2 checksum
399
 *
400
 * @param data      pointer to data to be checksum'ed
401
 * @param length    data length
402
 * @param value     checksum value
403
 *
404
 * @return          0 if checksum is ok
405
 */
406
static uint16_t qdm2_packet_checksum (uint8_t *data, int length, int value) {
407
    int i;
408

  
409
    for (i=0; i < length; i++)
410
        value -= data[i];
411

  
412
    return (uint16_t)(value & 0xffff);
413
}
414

  
415

  
416
/**
417
 * Fills a QDM2SubPacket structure with packet type, size, and data pointer
418
 *
419
 * @param gb            bitreader context
420
 * @param sub_packet    packet under analysis
421
 */
422
static void qdm2_decode_sub_packet_header (GetBitContext *gb, QDM2SubPacket *sub_packet)
423
{
424
    sub_packet->type = get_bits (gb, 8);
425

  
426
    if (sub_packet->type == 0) {
427
        sub_packet->size = 0;
428
        sub_packet->data = NULL;
429
    } else {
430
        sub_packet->size = get_bits (gb, 8);
431

  
432
      if (sub_packet->type & 0x80) {
433
          sub_packet->size <<= 8;
434
          sub_packet->size  |= get_bits (gb, 8);
435
          sub_packet->type  &= 0x7f;
436
      }
437

  
438
      if (sub_packet->type == 0x7f)
439
          sub_packet->type |= (get_bits (gb, 8) << 8);
440

  
441
      sub_packet->data = &gb->buffer[get_bits_count(gb) / 8]; // FIXME: this depends on bitreader internal data
442
    }
443

  
444
    av_log(NULL,AV_LOG_DEBUG,"Sub packet: type=%d size=%d start_offs=%x\n",
445
        sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
446
}
447

  
448

  
449
/**
450
 * Return node pointer to first packet of requested type in list
451
 *
452
 * @param list    list of subpacket to be scanned
453
 * @param type    type of searched subpacket
454
 * @return        node pointer for subpacket if found, else NULL
455
 */
456
static QDM2SubPNode* qdm2_search_subpacket_type_in_list (QDM2SubPNode *list, int type)
457
{
458
    while (list != NULL && list->packet != NULL) {
459
        if (list->packet->type == type)
460
            return list;
461
        list = list->next;
462
    }
463
    return NULL;
464
}
465

  
466

  
467
/**
468
 * Replaces 8 elements with their average value
469
 * Called by qdm2_decode_superblock before starting subblocks decoding
470
 *
471
 * @param q       context
472
 */
473
static void average_quantized_coeffs (QDM2Context *q)
474
{
475
    int i, j, n, ch, sum;
476

  
477
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
478

  
479
    for (ch = 0; ch < q->nb_channels; ch++)
480
        for (i = 0; i < n; i++) {
481
            sum = 0;
482

  
483
            for (j = 0; j < 8; j++)
484
                sum += q->quantized_coeffs[ch][i][j];
485

  
486
            sum /= 8;
487
            if (sum > 0)
488
                sum--;
489

  
490
            for (j=0; j < 8; j++)
491
                q->quantized_coeffs[ch][i][j] = sum;
492
        }
493
}
494

  
495

  
496
/**
497
 * Build subband samples with noise weighted by q->tone_level
498
 * Called by synthfilt_build_sb_samples
499
 *
500
 * @param q     context
501
 * @param sb    subband index
502
 */
503
static void build_sb_samples_from_noise (QDM2Context *q, int sb)
504
{
505
    int ch, j;
506

  
507
    FIX_NOISE_IDX(q->noise_idx);
508

  
509
    if (!q->nb_channels)
510
        return;
511

  
512
    for (ch = 0; ch < q->nb_channels; ch++)
513
        for (j = 0; j < 64; j++) {
514
            q->sb_samples[ch][j * 2][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
515
            q->sb_samples[ch][j * 2 + 1][sb] = (int32_t)(f2i_scale * SB_DITHERING_NOISE(sb,q->noise_idx) * q->tone_level[ch][sb][j] + .5);
516
        }
517
}
518

  
519

  
520
/**
521
 * Called while processing data from subpackets 11 and 12
522
 * Used after making changes to coding_method array
523
 *
524
 * @param sb               subband index
525
 * @param channels         number of channels
526
 * @param coding_method    q->coding_method[0][0][0]
527
 */
528
 void fix_coding_method_array (int sb, int channels, sb_int8_array coding_method)
529
{
530
    int j,k;
531
    int ch;
532
    int run, case_val;
533
    int switchtable[23] = {0,5,1,5,5,5,5,5,2,5,5,5,5,5,5,5,3,5,5,5,5,5,4};
534

  
535
    for (ch = 0; ch < channels; ch++) {
536
        for (j = 0; j < 64; ) {
537
            if((coding_method[ch][sb][j] - 8) > 22) {
538
                run = 1;
539
                case_val = 8;
540
            } else {
541
                switch (switchtable[coding_method[ch][sb][j]]) {
542
                    case 0: run = 10; case_val = 10; break;
543
                    case 1: run = 1; case_val = 16; break;
544
                    case 2: run = 5; case_val = 24; break;
545
                    case 3: run = 3; case_val = 30; break;
546
                    case 4: run = 1; case_val = 30; break;
547
                    case 5: run = 1; case_val = 8; break;
548
                    default: run = 1; case_val = 8; break;
549
                }
550
            }
551
            for (k = 0; k < run; k++)
552
                if (j + k < 128)
553
                    if (coding_method[ch][sb + (j + k) / 64][(j + k) % 64] > coding_method[ch][sb][j])
554
                        if (k > 0) {
555
                           SAMPLES_NEEDED
556
                            //not debugged, almost never used
557
                            memset(&coding_method[ch][sb][j + k], case_val, k * sizeof(int8_t));
558
                            memset(&coding_method[ch][sb][j + k], case_val, 3 * sizeof(int8_t));
559
                        }
560
            j += run;
561
        }
562
    }
563
}
564

  
565

  
566
/**
567
 * Related to synthesis filter
568
 * Called by process_subpacket_10
569
 *
570
 * @param q       context
571
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
572
 */
573
static void fill_tone_level_array (QDM2Context *q, int flag)
574
{
575
    int i, sb, ch, sb_used;
576
    int tmp, tab;
577

  
578
    // This should never happen
579
    if (q->nb_channels <= 0)
580
        return;
581

  
582
    for (ch = 0; ch < q->nb_channels; ch++)
583
        for (sb = 0; sb < 30; sb++)
584
            for (i = 0; i < 8; i++) {
585
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
586
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
587
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
588
                else
589
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
590
                if(tmp < 0)
591
                    tmp += 0xff;
592
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
593
            }
594

  
595
    sb_used = QDM2_SB_USED(q->sub_sampling);
596

  
597
    if ((q->superblocktype_2_3 != 0) && !flag) {
598
        for (sb = 0; sb < sb_used; sb++)
599
            for (ch = 0; ch < q->nb_channels; ch++)
600
                for (i = 0; i < 64; i++) {
601
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
602
                    if (q->tone_level_idx[ch][sb][i] < 0)
603
                        q->tone_level[ch][sb][i] = 0;
604
                    else
605
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
606
                }
607
    } else {
608
        tab = q->superblocktype_2_3 ? 0 : 1;
609
        for (sb = 0; sb < sb_used; sb++) {
610
            if ((sb >= 4) && (sb <= 23)) {
611
                for (ch = 0; ch < q->nb_channels; ch++)
612
                    for (i = 0; i < 64; i++) {
613
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
614
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
615
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
616
                              q->tone_level_idx_hi2[ch][sb - 4];
617
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
618
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
619
                            q->tone_level[ch][sb][i] = 0;
620
                        else
621
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
622
                }
623
            } else {
624
                if (sb > 4) {
625
                    for (ch = 0; ch < q->nb_channels; ch++)
626
                        for (i = 0; i < 64; i++) {
627
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
628
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
629
                                  q->tone_level_idx_hi2[ch][sb - 4];
630
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
631
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
632
                                q->tone_level[ch][sb][i] = 0;
633
                            else
634
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
635
                    }
636
                } else {
637
                    for (ch = 0; ch < q->nb_channels; ch++)
638
                        for (i = 0; i < 64; i++) {
639
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
640
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
641
                                q->tone_level[ch][sb][i] = 0;
642
                            else
643
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
644
                        }
645
                }
646
            }
647
        }
648
    }
649

  
650
    return;
651
}
652

  
653

  
654
/**
655
 * Related to synthesis filter
656
 * Called by process_subpacket_11
657
 * c is built with data from subpacket 11
658
 * Most of this function is used only if superblock_type_2_3 == 0, never seen it in samples
659
 *
660
 * @param tone_level_idx           
661
 * @param tone_level_idx_temp
662
 * @param coding_method        q->coding_method[0][0][0]
663
 * @param nb_channels          number of channels
664
 * @param c                    coming from subpacket 11, passed as 8*c
665
 * @param superblocktype_2_3   flag based on superblock packet type
666
 * @param cm_table_select      q->cm_table_select
667
 */
668
static void fill_coding_method_array (sb_int8_array tone_level_idx, sb_int8_array tone_level_idx_temp,
669
                sb_int8_array coding_method, int nb_channels,
670
                int c, int superblocktype_2_3, int cm_table_select)
671
{
672
    int ch, sb, j;
673
    int tmp, acc, esp_40, comp;
674
    int add1, add2, add3, add4;
675
    int64_t multres;
676

  
677
    // This should never happen
678
    if (nb_channels <= 0)
679
        return;
680

  
681
    if (!superblocktype_2_3) {
682
        /* This case is untested, no samples available */
683
        SAMPLES_NEEDED
684
        for (ch = 0; ch < nb_channels; ch++)
685
            for (sb = 0; sb < 30; sb++) {
686
                for (j = 1; j < 64; j++) {
687
                    add1 = tone_level_idx[ch][sb][j] - 10;
688
                    if (add1 < 0)
689
                        add1 = 0;
690
                    add2 = add3 = add4 = 0;
691
                    if (sb > 1) {
692
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
693
                        if (add2 < 0)
694
                            add2 = 0;
695
                    }
696
                    if (sb > 0) {
697
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
698
                        if (add3 < 0)
699
                            add3 = 0;
700
                    }
701
                    if (sb < 29) {
702
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
703
                        if (add4 < 0)
704
                            add4 = 0;
705
                    }
706
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
707
                    if (tmp < 0)
708
                        tmp = 0;
709
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
710
                }
711
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
712
            }
713
            acc = 0;
714
            for (ch = 0; ch < nb_channels; ch++)
715
                for (sb = 0; sb < 30; sb++)
716
                    for (j = 0; j < 64; j++)
717
                        acc += tone_level_idx_temp[ch][sb][j];
718
            if (acc)
719
                tmp = c * 256 / (acc & 0xffff);
720
            multres = 0x66666667 * (acc * 10);
721
            esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
722
            for (ch = 0;  ch < nb_channels; ch++)
723
                for (sb = 0; sb < 30; sb++)
724
                    for (j = 0; j < 64; j++) {
725
                        comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
726
                        if (comp < 0)
727
                            comp += 0xff;
728
                        comp /= 256; // signed shift
729
                        switch(sb) {
730
                            case 0:
731
                                if (comp < 30)
732
                                    comp = 30;
733
                                comp += 15;
734
                                break;
735
                            case 1:
736
                                if (comp < 24)
737
                                    comp = 24;
738
                                comp += 10;
739
                                break;
740
                            case 2:
741
                            case 3:
742
                            case 4:
743
                                if (comp < 16)
744
                                    comp = 16;
745
                        }
746
                        if (comp <= 5)
747
                            tmp = 0;
748
                        else if (comp <= 10)
749
                            tmp = 10;
750
                        else if (comp <= 16)
751
                            tmp = 16;
752
                        else if (comp <= 24)
753
                            tmp = -1;
754
                        else
755
                            tmp = 0;
756
                        coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
757
                    }
758
            for (sb = 0; sb < 30; sb++)
759
                fix_coding_method_array(sb, nb_channels, coding_method);
760
            for (ch = 0; ch < nb_channels; ch++)
761
                for (sb = 0; sb < 30; sb++)
762
                    for (j = 0; j < 64; j++)
763
                        if (sb >= 10) {
764
                            if (coding_method[ch][sb][j] < 10)
765
                                coding_method[ch][sb][j] = 10;
766
                        } else {
767
                            if (sb >= 2) {
768
                                if (coding_method[ch][sb][j] < 16)
769
                                    coding_method[ch][sb][j] = 16;
770
                            } else {
771
                                if (coding_method[ch][sb][j] < 30)
772
                                    coding_method[ch][sb][j] = 30;
773
                            }
774
                        }
775
    } else { // superblocktype_2_3 != 0
776
        for (ch = 0; ch < nb_channels; ch++)
777
            for (sb = 0; sb < 30; sb++)
778
                for (j = 0; j < 64; j++)
779
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
780
    }
781

  
782
    return;
783
}
784

  
785

  
786
/**
787
 *
788
 * Called by process_subpacket_11 to process more data from subpacket 11 with sb 0-8
789
 * Called by process_subpacket_12 to process data from subpacket 12 with sb 8-sb_used
790
 *
791
 * @param q         context
792
 * @param gb        bitreader context
793
 * @param length    packet length in bit
794
 * @param sb_min    lower subband processed (sb_min included)
795
 * @param sb_max    higher subband processed (sb_max excluded)
796
 */
797
static void synthfilt_build_sb_samples (QDM2Context *q, GetBitContext *gb, int length, int sb_min, int sb_max)
798
{
799
    int sb, j, k, n, ch, run, channels;
800
    int joined_stereo, zero_encoding, chs;
801
    int type34_first;
802
    float type34_div = 0;
803
    float type34_predictor;
804
    float samples[10], sign_bits[16];
805

  
806
    if (length == 0) {
807
        // If no data use noise
808
        for (sb=sb_min; sb < sb_max; sb++)
809
            build_sb_samples_from_noise (q, sb);
810

  
811
        return;
812
    }
813

  
814
    for (sb = sb_min; sb < sb_max; sb++) {
815
        FIX_NOISE_IDX(q->noise_idx);
816

  
817
        channels = q->nb_channels;
818

  
819
        if (q->nb_channels <= 1 || sb < 12)
820
            joined_stereo = 0;
821
        else if (sb >= 24)
822
            joined_stereo = 1;
823
        else
824
            joined_stereo = (BITS_LEFT(length,gb) >= 1) ? get_bits1 (gb) : 0;
825

  
826
        if (joined_stereo) {
827
            if (BITS_LEFT(length,gb) >= 16)
828
                for (j = 0; j < 16; j++)
829
                    sign_bits[j] = get_bits1 (gb);
830

  
831
            for (j = 0; j < 64; j++)
832
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
833
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
834

  
835
            fix_coding_method_array(sb, q->nb_channels, q->coding_method);
836
            channels = 1;
837
        }
838

  
839
        for (ch = 0; ch < channels; ch++) {
840
            zero_encoding = (BITS_LEFT(length,gb) >= 1) ? get_bits1(gb) : 0;
841
            type34_predictor = 0.0;
842
            type34_first = 1;
843

  
844
            for (j = 0; j < 128; ) {
845
                switch (q->coding_method[ch][sb][j / 2]) {
846
                    case 8:
847
                        if (BITS_LEFT(length,gb) >= 10) {
848
                            if (zero_encoding) {
849
                                for (k = 0; k < 5; k++) {
850
                                    if ((j + 2 * k) >= 128)
851
                                        break;
852
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
853
                                }
854
                            } else {
855
                                n = get_bits(gb, 8);
856
                                for (k = 0; k < 5; k++)
857
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
858
                            }
859
                            for (k = 0; k < 5; k++)
860
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
861
                        } else {
862
                            for (k = 0; k < 10; k++)
863
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
864
                        }
865
                        run = 10;
866
                        break;
867

  
868
                    case 10:
869
                        if (BITS_LEFT(length,gb) >= 1) {
870
                            float f = 0.81;
871

  
872
                            if (get_bits1(gb))
873
                                f = -f;
874
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
875
                            samples[0] = f;
876
                        } else {
877
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
878
                        }
879
                        run = 1;
880
                        break;
881

  
882
                    case 16:
883
                        if (BITS_LEFT(length,gb) >= 10) {
884
                            if (zero_encoding) {
885
                                for (k = 0; k < 5; k++) {
886
                                    if ((j + k) >= 128)
887
                                        break;
888
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
889
                                }
890
                            } else {
891
                                n = get_bits (gb, 8);
892
                                for (k = 0; k < 5; k++)
893
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
894
                            }
895
                        } else {
896
                            for (k = 0; k < 5; k++)
897
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
898
                        }
899
                        run = 5;
900
                        break;
901

  
902
                    case 24:
903
                        if (BITS_LEFT(length,gb) >= 7) {
904
                            n = get_bits(gb, 7);
905
                            for (k = 0; k < 3; k++)
906
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
907
                        } else {
908
                            for (k = 0; k < 3; k++)
909
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
910
                        }
911
                        run = 3;
912
                        break;
913

  
914
                    case 30:
915
                        if (BITS_LEFT(length,gb) >= 4)
916
                            samples[0] = type30_dequant[qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1)];
917
                        else
918
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
919
                        
920
                        run = 1;
921
                        break;
922

  
923
                    case 34:
924
                        if (BITS_LEFT(length,gb) >= 7) {
925
                            if (type34_first) {
926
                                type34_div = (float)(1 << get_bits(gb, 2));
927
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
928
                                type34_predictor = samples[0];
929
                                type34_first = 0;
930
                            } else {
931
                                samples[0] = type34_delta[qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1)] / type34_div + type34_predictor;
932
                                type34_predictor = samples[0];
933
                            }
934
                        } else {
935
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
936
                        }
937
                        run = 1;
938
                        break;
939

  
940
                    default:
941
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
942
                        run = 1;
943
                        break;
944
                }
945

  
946
                if (joined_stereo) {
947
                    float tmp[10][MPA_MAX_CHANNELS];
948

  
949
                    for (k = 0; k < run; k++) {
950
                        tmp[k][0] = samples[k];
951
                        tmp[k][1] = (sign_bits[(j + k) / 8]) ? -samples[k] : samples[k];
952
                    }
953
                    for (chs = 0; chs < q->nb_channels; chs++)
954
                        for (k = 0; k < run; k++)
955
                            if ((j + k) < 128)
956
                                q->sb_samples[chs][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[chs][sb][((j + k)/2)] * tmp[k][chs] + .5);
957
                } else {
958
                    for (k = 0; k < run; k++)
959
                        if ((j + k) < 128)
960
                            q->sb_samples[ch][j + k][sb] = (int32_t)(f2i_scale * q->tone_level[ch][sb][(j + k)/2] * samples[k] + .5);
961
                }
962

  
963
                j += run;
964
            } // j loop
965
        } // channel loop
966
    } // subband loop
967
}
968

  
969

  
970
/**
971
 * Init the first element of a channel in quantized_coeffs with data from packet 10 (quantized_coeffs[ch][0])
972
 * This is similar to process_subpacket_9, but for a single channel and for element [0]
973
 * same VLC tables as process_subpacket_9 are used
974
 *
975
 * @param q         context
976
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
977
 * @param gb        bitreader context
978
 * @param length    packet length in bit
979
 */
980
static void init_quantized_coeffs_elem0 (int8_t *quantized_coeffs, GetBitContext *gb, int length)
981
{
982
    int i, k, run, level, diff;
983

  
984
    if (BITS_LEFT(length,gb) < 16)
985
        return;
986
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
987

  
988
    quantized_coeffs[0] = level;
989

  
990
    for (i = 0; i < 7; ) {
991
        if (BITS_LEFT(length,gb) < 16)
992
            break;
993
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
994

  
995
        if (BITS_LEFT(length,gb) < 16)
996
            break;
997
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
998
    
999
        for (k = 1; k <= run; k++)
1000
            quantized_coeffs[i + k] = (level + ((k * diff) / run));
1001
    
1002
        level += diff;
1003
        i += run;
1004
    }
1005
}
1006

  
1007

  
1008
/**
1009
 * Related to synthesis filter, process data from packet 10
1010
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
1011
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with data from packet 10
1012
 *
1013
 * @param q         context
1014
 * @param gb        bitreader context
1015
 * @param length    packet length in bit
1016
 */
1017
static void init_tone_level_dequantization (QDM2Context *q, GetBitContext *gb, int length)
1018
{
1019
    int sb, j, k, n, ch;
1020

  
1021
    for (ch = 0; ch < q->nb_channels; ch++) {
1022
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb, length);
1023

  
1024
        if (BITS_LEFT(length,gb) < 16) {
1025
            memset(q->quantized_coeffs[ch][0], 0, 8);
1026
            break;
1027
        }
1028
    }
1029

  
1030
    n = q->sub_sampling + 1;
1031

  
1032
    for (sb = 0; sb < n; sb++)
1033
        for (ch = 0; ch < q->nb_channels; ch++)
1034
            for (j = 0; j < 8; j++) {
1035
                if (BITS_LEFT(length,gb) < 1)
1036
                    break;
1037
                if (get_bits1(gb)) {
1038
                    for (k=0; k < 8; k++) {
1039
                        if (BITS_LEFT(length,gb) < 16)
1040
                            break;
1041
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
1042
                    }
1043
                } else {
1044
                    for (k=0; k < 8; k++)
1045
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
1046
                }
1047
            }
1048

  
1049
    n = QDM2_SB_USED(q->sub_sampling) - 4;
1050

  
1051
    for (sb = 0; sb < n; sb++)
1052
        for (ch = 0; ch < q->nb_channels; ch++) {
1053
            if (BITS_LEFT(length,gb) < 16)
1054
                break;
1055
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
1056
            if (sb > 19)
1057
                q->tone_level_idx_hi2[ch][sb] -= 16;
1058
            else
1059
                for (j = 0; j < 8; j++)
1060
                    q->tone_level_idx_mid[ch][sb][j] = -16;
1061
        }
1062

  
1063
    n = QDM2_SB_USED(q->sub_sampling) - 5;
1064

  
1065
    for (sb = 0; sb < n; sb++)
1066
        for (ch = 0; ch < q->nb_channels; ch++)
1067
            for (j = 0; j < 8; j++) {
1068
                if (BITS_LEFT(length,gb) < 16)
1069
                    break;
1070
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
1071
            }
1072
}
1073

  
1074
/**
1075
 * Process subpacket 9, init quantized_coeffs with data from it
1076
 *
1077
 * @param q       context
1078
 * @param node    pointer to node with packet
1079
 */
1080
static void process_subpacket_9 (QDM2Context *q, QDM2SubPNode *node)
1081
{
1082
    GetBitContext gb;
1083
    int i, j, k, n, ch, run, level, diff;
1084

  
1085
    init_get_bits(&gb, node->packet->data, node->packet->size);
1086

  
1087
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1; // same as averagesomething function
1088

  
1089
    for (i = 1; i < n; i++)
1090
        for (ch=0; ch < q->nb_channels; ch++) {
1091
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
1092
            q->quantized_coeffs[ch][i][0] = level;
1093

  
1094
            for (j = 0; j < (8 - 1); ) {
1095
                run = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
1096
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
1097

  
1098
                for (k = 1; k <= run; k++)
1099
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k*diff) / run));
1100

  
1101
                level += diff;
1102
                j += run;
1103
            }
1104
        }
1105

  
1106
    for (ch = 0; ch < q->nb_channels; ch++)
1107
        for (i = 0; i < 8; i++)
1108
            q->quantized_coeffs[ch][0][i] = 0;
1109
}
1110

  
1111

  
1112
/**
1113
 * Process subpacket 10 if not null, else
1114
 *
1115
 * @param q         context
1116
 * @param node      pointer to node with packet
1117
 * @param length    packet length in bit
1118
 */
1119
static void process_subpacket_10 (QDM2Context *q, QDM2SubPNode *node, int length)
1120
{
1121
    GetBitContext gb;
1122

  
1123
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
1124

  
1125
    if (length != 0) {
1126
        init_tone_level_dequantization(q, &gb, length);
1127
        fill_tone_level_array(q, 1);
1128
    } else {
1129
        fill_tone_level_array(q, 0);
1130
    }
1131
}
1132

  
1133

  
1134
/**
1135
 * Process subpacket 11
1136
 *
1137
 * @param q         context
1138
 * @param node      pointer to node with packet
1139
 * @param length    packet length in bit
1140
 */
1141
static void process_subpacket_11 (QDM2Context *q, QDM2SubPNode *node, int length)
1142
{
1143
    GetBitContext gb;
1144

  
1145
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
1146
    if (length >= 32) {
1147
        int c = get_bits (&gb, 13);
1148

  
1149
        if (c > 3)
1150
            fill_coding_method_array (q->tone_level_idx, q->tone_level_idx_temp, q->coding_method,
1151
                                      q->nb_channels, 8*c, q->superblocktype_2_3, q->cm_table_select);
1152
    }
1153

  
1154
    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1155
}
1156

  
1157

  
1158
/**
1159
 * Process subpacket 12
1160
 *
1161
 * @param q         context
1162
 * @param node      pointer to node with packet
1163
 * @param length    packet length in bit
1164
 */
1165
static void process_subpacket_12 (QDM2Context *q, QDM2SubPNode *node, int length)
1166
{
1167
    GetBitContext gb;
1168

  
1169
    init_get_bits(&gb, ((node == NULL) ? empty_buffer : node->packet->data), ((node == NULL) ? 0 : node->packet->size));
1170
    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1171
}
1172

  
1173
/*
1174
 * Process new subpackets for synthesis filter
1175
 *
1176
 * @param q       context
1177
 * @param list    list with synthesis filter packets (list D)
1178
 */
1179
static void process_synthesis_subpackets (QDM2Context *q, QDM2SubPNode *list)
1180
{
1181
    QDM2SubPNode *nodes[4];
1182

  
1183
    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1184
    if (nodes[0] != NULL)
1185
        process_subpacket_9(q, nodes[0]);
1186

  
1187
    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1188
    if (nodes[1] != NULL)
1189
        process_subpacket_10(q, nodes[1], nodes[1]->packet->size << 3);
1190
    else
1191
        process_subpacket_10(q, NULL, 0);
1192

  
1193
    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1194
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[2] != NULL)
1195
        process_subpacket_11(q, nodes[2], (nodes[2]->packet->size << 3));
1196
    else
1197
        process_subpacket_11(q, NULL, 0);
1198

  
1199
    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1200
    if (nodes[0] != NULL && nodes[1] != NULL && nodes[3] != NULL)
1201
        process_subpacket_12(q, nodes[3], (nodes[3]->packet->size << 3));
1202
    else
1203
        process_subpacket_12(q, NULL, 0);
1204
}
1205

  
1206

  
1207
/*
1208
 * Decode superblock, fill packet lists
1209
 *
1210
 * @param q    context
1211
 */
1212
static void qdm2_decode_super_block (QDM2Context *q)
1213
{
1214
    GetBitContext gb;
1215
    QDM2SubPacket header, *packet;
1216
    int i, packet_bytes, sub_packet_size, sub_packets_D;
1217
    unsigned int next_index = 0;
1218

  
1219
    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1220
    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1221
    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1222

  
1223
    q->sub_packets_B = 0;
1224
    sub_packets_D = 0;
1225

  
1226
    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1227

  
1228
    init_get_bits(&gb, q->compressed_data, q->compressed_size);
1229
    qdm2_decode_sub_packet_header(&gb, &header);
1230

  
1231
    if (header.type < 2 || header.type >= 8) {
1232
        q->has_errors = 1;
1233
        av_log(NULL,AV_LOG_ERROR,"bad superblock type\n");
1234
        return;
1235
    }
1236

  
1237
    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1238
    packet_bytes = (q->compressed_size - get_bits_count(&gb) / 8);
1239

  
1240
    init_get_bits(&gb, header.data, header.size);
1241

  
1242
    if (header.type == 2 || header.type == 4 || header.type == 5) {
1243
        int csum = 257 * get_bits(&gb, 8) + 2 * get_bits(&gb, 8);
1244

  
1245
        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1246

  
1247
        if (csum != 0) {
1248
            q->has_errors = 1;
1249
            av_log(NULL,AV_LOG_ERROR,"bad packet checksum\n");
1250
            return;
1251
        }
1252
    }
1253

  
1254
    q->sub_packet_list_B[0].packet = NULL;
1255
    q->sub_packet_list_D[0].packet = NULL;
1256

  
1257
    for (i = 0; i < 6; i++)
1258
        if (--q->fft_level_exp[i] < 0)
1259
            q->fft_level_exp[i] = 0;
1260

  
1261
    for (i = 0; packet_bytes > 0; i++) {
1262
        int j;
1263

  
1264
        q->sub_packet_list_A[i].next = NULL;
1265

  
1266
        if (i > 0) {
1267
            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1268

  
1269
            /* seek to next block */
1270
            init_get_bits(&gb, header.data, header.size);
1271
            skip_bits(&gb, next_index*8);
1272

  
1273
            if (next_index >= header.size)
1274
                break;
1275
        }
1276

  
1277
        /* decode sub packet */
1278
        packet = &q->sub_packets[i];
1279
        qdm2_decode_sub_packet_header(&gb, packet);
1280
        next_index = packet->size + get_bits_count(&gb) / 8;
1281
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1282

  
1283
        if (packet->type == 0)
1284
            break;
1285

  
1286
        if (sub_packet_size > packet_bytes) {
1287
            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1288
                break;
1289
            packet->size += packet_bytes - sub_packet_size;
1290
        }
1291

  
1292
        packet_bytes -= sub_packet_size;
1293

  
1294
        /* add sub packet to 'all sub packets' list */
1295
        q->sub_packet_list_A[i].packet = packet;
1296

  
1297
        /* add sub packet to related list */
1298
        if (packet->type == 8) {
1299
            SAMPLES_NEEDED_2("packet type 8");
1300
            return;
1301
        } else if (packet->type >= 9 && packet->type <= 12) {
1302
            /* packets for MPEG Audio like Synthesis Filter */
1303
            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1304
        } else if (packet->type == 13) {
1305
            for (j = 0; j < 6; j++)
1306
                q->fft_level_exp[j] = get_bits(&gb, 6);
1307
        } else if (packet->type == 14) {
1308
            for (j = 0; j < 6; j++)
1309
                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1310
        } else if (packet->type == 15) {
1311
            SAMPLES_NEEDED_2("packet type 15")
1312
            return;
1313
        } else if (packet->type >= 16 && packet->type < 48 && !fft_subpackets[packet->type - 16]) {
1314
            /* packets for FFT */
1315
            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1316
        }
1317
    } // Packet bytes loop
1318

  
1319
/* **************************************************************** */
1320
    if (q->sub_packet_list_D[0].packet != NULL) {
1321
        process_synthesis_subpackets(q, q->sub_packet_list_D);
1322
        q->do_synth_filter = 1;
1323
    } else if (q->do_synth_filter) {
1324
        process_subpacket_10(q, NULL, 0);
1325
        process_subpacket_11(q, NULL, 0);
1326
        process_subpacket_12(q, NULL, 0);
1327
    }
1328
/* **************************************************************** */
1329
}
1330

  
1331

  
1332
static void qdm2_fft_init_coefficient (QDM2Context *q, int sub_packet,
1333
                       int offset, int duration, int channel,
1334
                       int exp, int phase)
1335
{
1336
    if (q->fft_coefs_min_index[duration] < 0)
1337
        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1338

  
1339
    q->fft_coefs[q->fft_coefs_index].sub_packet = ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1340
    q->fft_coefs[q->fft_coefs_index].channel = channel;
1341
    q->fft_coefs[q->fft_coefs_index].offset = offset;
1342
    q->fft_coefs[q->fft_coefs_index].exp = exp;
1343
    q->fft_coefs[q->fft_coefs_index].phase = phase;
1344
    q->fft_coefs_index++;
1345
}
1346

  
1347

  
1348
static void qdm2_fft_decode_tones (QDM2Context *q, int duration, GetBitContext *gb, int b)
1349
{
1350
    int channel, stereo, phase, exp;
1351
    int local_int_4,  local_int_8,  stereo_phase,  local_int_10;
1352
    int local_int_14, stereo_exp, local_int_20, local_int_28;
1353
    int n, offset;
1354

  
1355
    local_int_4 = 0;
1356
    local_int_28 = 0;
1357
    local_int_20 = 2;
1358
    local_int_8 = (4 - duration);
1359
    local_int_10 = 1 << (q->group_order - duration - 1);
1360
    offset = 1;
1361

  
1362
    while (1) {
1363
        if (q->superblocktype_2_3) {
1364
            while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1365
                offset = 1;
1366
                if (n == 0) {
1367
                    local_int_4 += local_int_10;
1368
                    local_int_28 += (1 << local_int_8);
1369
                } else {
1370
                    local_int_4 += 8*local_int_10;
1371
                    local_int_28 += (8 << local_int_8);
1372
                }
1373
            }
1374
            offset += (n - 2);
1375
        } else {
1376
            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1377
            while (offset >= (local_int_10 - 1)) {
1378
                offset += (1 - (local_int_10 - 1));
1379
                local_int_4  += local_int_10;
1380
                local_int_28 += (1 << local_int_8);
1381
            }
1382
        }
1383

  
1384
        if (local_int_4 >= q->group_size)
1385
            return;
1386

  
1387
        local_int_14 = (offset >> local_int_8);
1388

  
1389
        if (q->nb_channels > 1) {
1390
            channel = get_bits1(gb);
1391
            stereo = get_bits1(gb);
1392
        } else {
1393
            channel = 0;
1394
            stereo = 0;
1395
        }
1396

  
1397
        exp = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1398
        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1399
        exp = (exp < 0) ? 0 : exp;
1400

  
1401
        phase = get_bits(gb, 3);
1402
        stereo_exp = 0;
1403
        stereo_phase = 0;
1404

  
1405
        if (stereo) {
1406
            stereo_exp = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1407
            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1408
            if (stereo_phase < 0)
1409
                stereo_phase += 8;
1410
        }
1411

  
1412
        if (q->frequency_range > (local_int_14 + 1)) {
1413
            int sub_packet = (local_int_20 + local_int_28);
1414

  
1415
            qdm2_fft_init_coefficient(q, sub_packet, offset, duration, channel, exp, phase);
1416
            if (stereo)
1417
                qdm2_fft_init_coefficient(q, sub_packet, offset, duration, (1 - channel), stereo_exp, stereo_phase);
1418
        }
1419

  
1420
        offset++;
1421
    }
1422
}
1423

  
1424

  
1425
static void qdm2_decode_fft_packets (QDM2Context *q)
1426
{
1427
    int i, j, min, max, value, type, unknown_flag;
1428
    GetBitContext gb;
1429

  
1430
    if (q->sub_packet_list_B[0].packet == NULL)
1431
        return;
1432

  
1433
    /* reset minimum indices for FFT coefficients */
1434
    q->fft_coefs_index = 0;
1435
    for (i=0; i < 5; i++)
1436
        q->fft_coefs_min_index[i] = -1;
1437

  
1438
    /* process sub packets ordered by type, largest type first */
1439
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1440
        QDM2SubPacket *packet;
1441

  
1442
        /* find sub packet with largest type less than max */
1443
        for (j = 0, min = 0, packet = NULL; j < q->sub_packets_B; j++) {
1444
            value = q->sub_packet_list_B[j].packet->type;
1445
            if (value > min && value < max) {
1446
                min = value;
1447
                packet = q->sub_packet_list_B[j].packet;
1448
            }
1449
        }
1450

  
1451
        max = min;
1452

  
1453
        /* check for errors (?) */
1454
        if (i == 0 && (packet->type < 16 || packet->type >= 48 || fft_subpackets[packet->type - 16]))
1455
            return;
1456

  
1457
        /* decode FFT tones */
1458
        init_get_bits (&gb, packet->data, packet->size);
1459

  
1460
        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1461
            unknown_flag = 1;
1462
        else
1463
            unknown_flag = 0;
1464

  
1465
        type = packet->type;
1466

  
1467
        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1468
            int duration = q->sub_sampling + 5 - (type & 15);
1469

  
1470
            if (duration >= 0 && duration < 4)
1471
                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1472
        } else if (type == 31) {
1473
            for (i=0; i < 4; i++)
1474
                qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
1475
        } else if (type == 46) {
1476
            for (i=0; i < 6; i++)
1477
                q->fft_level_exp[i] = get_bits(&gb, 6);
1478
            for (i=0; i < 4; i++)
1479
            qdm2_fft_decode_tones(q, i, &gb, unknown_flag);
1480
        }
1481
    } // Loop on B packets
1482

  
1483
    /* calculate maximum indices for FFT coefficients */
1484
    for (i = 0, j = -1; i < 5; i++)
1485
        if (q->fft_coefs_min_index[i] >= 0) {
1486
            if (j >= 0)
1487
                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
... This diff was truncated because it exceeds the maximum size that can be displayed.

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