GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/qdm2.c Lines: 473 852 55.5 %
Date: 2020-10-23 17:01:47 Branches: 280 673 41.6 %

Line Branch Exec Source
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
27
 * QDM2 decoder
28
 * @author Ewald Snel, Benjamin Larsson, Alex Beregszaszi, Roberto Togni
29
 *
30
 * The decoder is not perfect yet, there are still some distortions
31
 * especially on files encoded with 16 or 8 subbands.
32
 */
33
34
#include <math.h>
35
#include <stddef.h>
36
#include <stdio.h>
37
38
#include "libavutil/channel_layout.h"
39
40
#define BITSTREAM_READER_LE
41
#include "avcodec.h"
42
#include "get_bits.h"
43
#include "bytestream.h"
44
#include "internal.h"
45
#include "mpegaudio.h"
46
#include "mpegaudiodsp.h"
47
#include "rdft.h"
48
49
#include "qdm2_tablegen.h"
50
51
#define QDM2_LIST_ADD(list, size, packet) \
52
do { \
53
      if (size > 0) { \
54
    list[size - 1].next = &list[size]; \
55
      } \
56
      list[size].packet = packet; \
57
      list[size].next = NULL; \
58
      size++; \
59
} while(0)
60
61
// Result is 8, 16 or 30
62
#define QDM2_SB_USED(sub_sampling) (((sub_sampling) >= 2) ? 30 : 8 << (sub_sampling))
63
64
#define FIX_NOISE_IDX(noise_idx) \
65
  if ((noise_idx) >= 3840) \
66
    (noise_idx) -= 3840; \
67
68
#define SB_DITHERING_NOISE(sb,noise_idx) (noise_table[(noise_idx)++] * sb_noise_attenuation[(sb)])
69
70
#define SAMPLES_NEEDED \
71
     av_log (NULL,AV_LOG_INFO,"This file triggers some untested code. Please contact the developers.\n");
72
73
#define SAMPLES_NEEDED_2(why) \
74
     av_log (NULL,AV_LOG_INFO,"This file triggers some missing code. Please contact the developers.\nPosition: %s\n",why);
75
76
#define QDM2_MAX_FRAME_SIZE 512
77
78
typedef int8_t sb_int8_array[2][30][64];
79
80
/**
81
 * Subpacket
82
 */
83
typedef struct QDM2SubPacket {
84
    int type;            ///< subpacket type
85
    unsigned int size;   ///< subpacket size
86
    const uint8_t *data; ///< pointer to subpacket data (points to input data buffer, it's not a private copy)
87
} QDM2SubPacket;
88
89
/**
90
 * A node in the subpacket list
91
 */
92
typedef struct QDM2SubPNode {
93
    QDM2SubPacket *packet;      ///< packet
94
    struct QDM2SubPNode *next; ///< pointer to next packet in the list, NULL if leaf node
95
} QDM2SubPNode;
96
97
typedef struct QDM2Complex {
98
    float re;
99
    float im;
100
} QDM2Complex;
101
102
typedef struct FFTTone {
103
    float level;
104
    QDM2Complex *complex;
105
    const float *table;
106
    int   phase;
107
    int   phase_shift;
108
    int   duration;
109
    short time_index;
110
    short cutoff;
111
} FFTTone;
112
113
typedef struct FFTCoefficient {
114
    int16_t sub_packet;
115
    uint8_t channel;
116
    int16_t offset;
117
    int16_t exp;
118
    uint8_t phase;
119
} FFTCoefficient;
120
121
typedef struct QDM2FFT {
122
    DECLARE_ALIGNED(32, QDM2Complex, complex)[MPA_MAX_CHANNELS][256];
123
} QDM2FFT;
124
125
/**
126
 * QDM2 decoder context
127
 */
128
typedef struct QDM2Context {
129
    /// Parameters from codec header, do not change during playback
130
    int nb_channels;         ///< number of channels
131
    int channels;            ///< number of channels
132
    int group_size;          ///< size of frame group (16 frames per group)
133
    int fft_size;            ///< size of FFT, in complex numbers
134
    int checksum_size;       ///< size of data block, used also for checksum
135
136
    /// Parameters built from header parameters, do not change during playback
137
    int group_order;         ///< order of frame group
138
    int fft_order;           ///< order of FFT (actually fftorder+1)
139
    int frame_size;          ///< size of data frame
140
    int frequency_range;
141
    int sub_sampling;        ///< subsampling: 0=25%, 1=50%, 2=100% */
142
    int coeff_per_sb_select; ///< selector for "num. of coeffs. per subband" tables. Can be 0, 1, 2
143
    int cm_table_select;     ///< selector for "coding method" tables. Can be 0, 1 (from init: 0-4)
144
145
    /// Packets and packet lists
146
    QDM2SubPacket sub_packets[16];      ///< the packets themselves
147
    QDM2SubPNode sub_packet_list_A[16]; ///< list of all packets
148
    QDM2SubPNode sub_packet_list_B[16]; ///< FFT packets B are on list
149
    int sub_packets_B;                  ///< number of packets on 'B' list
150
    QDM2SubPNode sub_packet_list_C[16]; ///< packets with errors?
151
    QDM2SubPNode sub_packet_list_D[16]; ///< DCT packets
152
153
    /// FFT and tones
154
    FFTTone fft_tones[1000];
155
    int fft_tone_start;
156
    int fft_tone_end;
157
    FFTCoefficient fft_coefs[1000];
158
    int fft_coefs_index;
159
    int fft_coefs_min_index[5];
160
    int fft_coefs_max_index[5];
161
    int fft_level_exp[6];
162
    RDFTContext rdft_ctx;
163
    QDM2FFT fft;
164
165
    /// I/O data
166
    const uint8_t *compressed_data;
167
    int compressed_size;
168
    float output_buffer[QDM2_MAX_FRAME_SIZE * MPA_MAX_CHANNELS * 2];
169
170
    /// Synthesis filter
171
    MPADSPContext mpadsp;
172
    DECLARE_ALIGNED(32, float, synth_buf)[MPA_MAX_CHANNELS][512*2];
173
    int synth_buf_offset[MPA_MAX_CHANNELS];
174
    DECLARE_ALIGNED(32, float, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT];
175
    DECLARE_ALIGNED(32, float, samples)[MPA_MAX_CHANNELS * MPA_FRAME_SIZE];
176
177
    /// Mixed temporary data used in decoding
178
    float tone_level[MPA_MAX_CHANNELS][30][64];
179
    int8_t coding_method[MPA_MAX_CHANNELS][30][64];
180
    int8_t quantized_coeffs[MPA_MAX_CHANNELS][10][8];
181
    int8_t tone_level_idx_base[MPA_MAX_CHANNELS][30][8];
182
    int8_t tone_level_idx_hi1[MPA_MAX_CHANNELS][3][8][8];
183
    int8_t tone_level_idx_mid[MPA_MAX_CHANNELS][26][8];
184
    int8_t tone_level_idx_hi2[MPA_MAX_CHANNELS][26];
185
    int8_t tone_level_idx[MPA_MAX_CHANNELS][30][64];
186
    int8_t tone_level_idx_temp[MPA_MAX_CHANNELS][30][64];
187
188
    // Flags
189
    int has_errors;         ///< packet has errors
190
    int superblocktype_2_3; ///< select fft tables and some algorithm based on superblock type
191
    int do_synth_filter;    ///< used to perform or skip synthesis filter
192
193
    int sub_packet;
194
    int noise_idx; ///< index for dithering noise table
195
} QDM2Context;
196
197
static const int switchtable[23] = {
198
    0, 5, 1, 5, 5, 5, 5, 5, 2, 5, 5, 5, 5, 5, 5, 5, 3, 5, 5, 5, 5, 5, 4
199
};
200
201
66888
static int qdm2_get_vlc(GetBitContext *gb, const VLC *vlc, int flag, int depth)
202
{
203
    int value;
204
205
66888
    value = get_vlc2(gb, vlc->table, vlc->bits, depth);
206
207
    /* stage-2, 3 bits exponent escape sequence */
208
66888
    if (value-- == 0)
209
555
        value = get_bits(gb, get_bits(gb, 3) + 1);
210
211
    /* stage-3, optional */
212
66888
    if (flag) {
213
        int tmp;
214
215
18134
        if (value >= 60) {
216
            av_log(NULL, AV_LOG_ERROR, "value %d in qdm2_get_vlc too large\n", value);
217
            return 0;
218
        }
219
220
18134
        tmp= vlc_stage3_values[value];
221
222
18134
        if ((value & ~3) > 0)
223
14090
            tmp += get_bits(gb, (value >> 2));
224
18134
        value = tmp;
225
    }
226
227
66888
    return value;
228
}
229
230
10896
static int qdm2_get_se_vlc(const VLC *vlc, GetBitContext *gb, int depth)
231
{
232
10896
    int value = qdm2_get_vlc(gb, vlc, 0, depth);
233
234
10896
    return (value & 1) ? ((value + 1) >> 1) : -(value >> 1);
235
}
236
237
/**
238
 * QDM2 checksum
239
 *
240
 * @param data      pointer to data to be checksummed
241
 * @param length    data length
242
 * @param value     checksum value
243
 *
244
 * @return          0 if checksum is OK
245
 */
246
138
static uint16_t qdm2_packet_checksum(const uint8_t *data, int length, int value)
247
{
248
    int i;
249
250
51198
    for (i = 0; i < length; i++)
251
51060
        value -= data[i];
252
253
138
    return (uint16_t)(value & 0xffff);
254
}
255
256
/**
257
 * Fill a QDM2SubPacket structure with packet type, size, and data pointer.
258
 *
259
 * @param gb            bitreader context
260
 * @param sub_packet    packet under analysis
261
 */
262
1104
static void qdm2_decode_sub_packet_header(GetBitContext *gb,
263
                                          QDM2SubPacket *sub_packet)
264
{
265
1104
    sub_packet->type = get_bits(gb, 8);
266
267
1104
    if (sub_packet->type == 0) {
268
138
        sub_packet->size = 0;
269
138
        sub_packet->data = NULL;
270
    } else {
271
966
        sub_packet->size = get_bits(gb, 8);
272
273
966
        if (sub_packet->type & 0x80) {
274
138
            sub_packet->size <<= 8;
275
138
            sub_packet->size  |= get_bits(gb, 8);
276
138
            sub_packet->type  &= 0x7f;
277
        }
278
279
966
        if (sub_packet->type == 0x7f)
280
            sub_packet->type |= (get_bits(gb, 8) << 8);
281
282
        // FIXME: this depends on bitreader-internal data
283
966
        sub_packet->data = &gb->buffer[get_bits_count(gb) / 8];
284
    }
285
286
1104
    av_log(NULL, AV_LOG_DEBUG, "Subpacket: type=%d size=%d start_offs=%x\n",
287
1104
           sub_packet->type, sub_packet->size, get_bits_count(gb) / 8);
288
1104
}
289
290
/**
291
 * Return node pointer to first packet of requested type in list.
292
 *
293
 * @param list    list of subpackets to be scanned
294
 * @param type    type of searched subpacket
295
 * @return        node pointer for subpacket if found, else NULL
296
 */
297
552
static QDM2SubPNode *qdm2_search_subpacket_type_in_list(QDM2SubPNode *list,
298
                                                        int type)
299
{
300

966
    while (list && list->packet) {
301
552
        if (list->packet->type == type)
302
138
            return list;
303
414
        list = list->next;
304
    }
305
414
    return NULL;
306
}
307
308
/**
309
 * Replace 8 elements with their average value.
310
 * Called by qdm2_decode_superblock before starting subblock decoding.
311
 *
312
 * @param q       context
313
 */
314
138
static void average_quantized_coeffs(QDM2Context *q)
315
{
316
    int i, j, n, ch, sum;
317
318
138
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
319
320
414
    for (ch = 0; ch < q->nb_channels; ch++)
321
3036
        for (i = 0; i < n; i++) {
322
2760
            sum = 0;
323
324
24840
            for (j = 0; j < 8; j++)
325
22080
                sum += q->quantized_coeffs[ch][i][j];
326
327
2760
            sum /= 8;
328
2760
            if (sum > 0)
329
2466
                sum--;
330
331
24840
            for (j = 0; j < 8; j++)
332
22080
                q->quantized_coeffs[ch][i][j] = sum;
333
        }
334
138
}
335
336
/**
337
 * Build subband samples with noise weighted by q->tone_level.
338
 * Called by synthfilt_build_sb_samples.
339
 *
340
 * @param q     context
341
 * @param sb    subband index
342
 */
343
4140
static void build_sb_samples_from_noise(QDM2Context *q, int sb)
344
{
345
    int ch, j;
346
347
4140
    FIX_NOISE_IDX(q->noise_idx);
348
349
4140
    if (!q->nb_channels)
350
        return;
351
352
12420
    for (ch = 0; ch < q->nb_channels; ch++) {
353
538200
        for (j = 0; j < 64; j++) {
354
529920
            q->sb_samples[ch][j * 2][sb] =
355
529920
                SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
356
529920
            q->sb_samples[ch][j * 2 + 1][sb] =
357
529920
                SB_DITHERING_NOISE(sb, q->noise_idx) * q->tone_level[ch][sb][j];
358
        }
359
    }
360
}
361
362
/**
363
 * Called while processing data from subpackets 11 and 12.
364
 * Used after making changes to coding_method array.
365
 *
366
 * @param sb               subband index
367
 * @param channels         number of channels
368
 * @param coding_method    q->coding_method[0][0][0]
369
 */
370
static int fix_coding_method_array(int sb, int channels,
371
                                   sb_int8_array coding_method)
372
{
373
    int j, k;
374
    int ch;
375
    int run, case_val;
376
377
    for (ch = 0; ch < channels; ch++) {
378
        for (j = 0; j < 64; ) {
379
            if (coding_method[ch][sb][j] < 8)
380
                return -1;
381
            if ((coding_method[ch][sb][j] - 8) > 22) {
382
                run      = 1;
383
                case_val = 8;
384
            } else {
385
                switch (switchtable[coding_method[ch][sb][j] - 8]) {
386
                case 0: run  = 10;
387
                    case_val = 10;
388
                    break;
389
                case 1: run  = 1;
390
                    case_val = 16;
391
                    break;
392
                case 2: run  = 5;
393
                    case_val = 24;
394
                    break;
395
                case 3: run  = 3;
396
                    case_val = 30;
397
                    break;
398
                case 4: run  = 1;
399
                    case_val = 30;
400
                    break;
401
                case 5: run  = 1;
402
                    case_val = 8;
403
                    break;
404
                default: run = 1;
405
                    case_val = 8;
406
                    break;
407
                }
408
            }
409
            for (k = 0; k < run; k++) {
410
                if (j + k < 128) {
411
                    int sbjk = sb + (j + k) / 64;
412
                    if (sbjk > 29) {
413
                        SAMPLES_NEEDED
414
                        continue;
415
                    }
416
                    if (coding_method[ch][sbjk][(j + k) % 64] > coding_method[ch][sb][j]) {
417
                        if (k > 0) {
418
                            SAMPLES_NEEDED
419
                            //not debugged, almost never used
420
                            memset(&coding_method[ch][sb][j + k], case_val,
421
                                   k *sizeof(int8_t));
422
                            memset(&coding_method[ch][sb][j + k], case_val,
423
                                   3 * sizeof(int8_t));
424
                        }
425
                    }
426
                }
427
            }
428
            j += run;
429
        }
430
    }
431
    return 0;
432
}
433
434
/**
435
 * Related to synthesis filter
436
 * Called by process_subpacket_10
437
 *
438
 * @param q       context
439
 * @param flag    1 if called after getting data from subpacket 10, 0 if no subpacket 10
440
 */
441
138
static void fill_tone_level_array(QDM2Context *q, int flag)
442
{
443
    int i, sb, ch, sb_used;
444
    int tmp, tab;
445
446
414
    for (ch = 0; ch < q->nb_channels; ch++)
447
8556
        for (sb = 0; sb < 30; sb++)
448
74520
            for (i = 0; i < 8; i++) {
449
66240
                if ((tab=coeff_per_sb_for_dequant[q->coeff_per_sb_select][sb]) < (last_coeff[q->coeff_per_sb_select] - 1))
450
52992
                    tmp = q->quantized_coeffs[ch][tab + 1][i] * dequant_table[q->coeff_per_sb_select][tab + 1][sb]+
451
52992
                          q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
452
                else
453
13248
                    tmp = q->quantized_coeffs[ch][tab][i] * dequant_table[q->coeff_per_sb_select][tab][sb];
454
66240
                if(tmp < 0)
455
                    tmp += 0xff;
456
66240
                q->tone_level_idx_base[ch][sb][i] = (tmp / 256) & 0xff;
457
            }
458
459
138
    sb_used = QDM2_SB_USED(q->sub_sampling);
460
461

138
    if ((q->superblocktype_2_3 != 0) && !flag) {
462
4278
        for (sb = 0; sb < sb_used; sb++)
463
12420
            for (ch = 0; ch < q->nb_channels; ch++)
464
538200
                for (i = 0; i < 64; i++) {
465
529920
                    q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
466
529920
                    if (q->tone_level_idx[ch][sb][i] < 0)
467
                        q->tone_level[ch][sb][i] = 0;
468
                    else
469
529920
                        q->tone_level[ch][sb][i] = fft_tone_level_table[0][q->tone_level_idx[ch][sb][i] & 0x3f];
470
                }
471
    } else {
472
        tab = q->superblocktype_2_3 ? 0 : 1;
473
        for (sb = 0; sb < sb_used; sb++) {
474
            if ((sb >= 4) && (sb <= 23)) {
475
                for (ch = 0; ch < q->nb_channels; ch++)
476
                    for (i = 0; i < 64; i++) {
477
                        tmp = q->tone_level_idx_base[ch][sb][i / 8] -
478
                              q->tone_level_idx_hi1[ch][sb / 8][i / 8][i % 8] -
479
                              q->tone_level_idx_mid[ch][sb - 4][i / 8] -
480
                              q->tone_level_idx_hi2[ch][sb - 4];
481
                        q->tone_level_idx[ch][sb][i] = tmp & 0xff;
482
                        if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
483
                            q->tone_level[ch][sb][i] = 0;
484
                        else
485
                            q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
486
                }
487
            } else {
488
                if (sb > 4) {
489
                    for (ch = 0; ch < q->nb_channels; ch++)
490
                        for (i = 0; i < 64; i++) {
491
                            tmp = q->tone_level_idx_base[ch][sb][i / 8] -
492
                                  q->tone_level_idx_hi1[ch][2][i / 8][i % 8] -
493
                                  q->tone_level_idx_hi2[ch][sb - 4];
494
                            q->tone_level_idx[ch][sb][i] = tmp & 0xff;
495
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
496
                                q->tone_level[ch][sb][i] = 0;
497
                            else
498
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
499
                    }
500
                } else {
501
                    for (ch = 0; ch < q->nb_channels; ch++)
502
                        for (i = 0; i < 64; i++) {
503
                            tmp = q->tone_level_idx[ch][sb][i] = q->tone_level_idx_base[ch][sb][i / 8];
504
                            if ((tmp < 0) || (!q->superblocktype_2_3 && !tmp))
505
                                q->tone_level[ch][sb][i] = 0;
506
                            else
507
                                q->tone_level[ch][sb][i] = fft_tone_level_table[tab][tmp & 0x3f];
508
                        }
509
                }
510
            }
511
        }
512
    }
513
138
}
514
515
/**
516
 * Related to synthesis filter
517
 * Called by process_subpacket_11
518
 * c is built with data from subpacket 11
519
 * Most of this function is used only if superblock_type_2_3 == 0,
520
 * never seen it in samples.
521
 *
522
 * @param tone_level_idx
523
 * @param tone_level_idx_temp
524
 * @param coding_method        q->coding_method[0][0][0]
525
 * @param nb_channels          number of channels
526
 * @param c                    coming from subpacket 11, passed as 8*c
527
 * @param superblocktype_2_3   flag based on superblock packet type
528
 * @param cm_table_select      q->cm_table_select
529
 */
530
static void fill_coding_method_array(sb_int8_array tone_level_idx,
531
                                     sb_int8_array tone_level_idx_temp,
532
                                     sb_int8_array coding_method,
533
                                     int nb_channels,
534
                                     int c, int superblocktype_2_3,
535
                                     int cm_table_select)
536
{
537
    int ch, sb, j;
538
    int tmp, acc, esp_40, comp;
539
    int add1, add2, add3, add4;
540
    int64_t multres;
541
542
    if (!superblocktype_2_3) {
543
        /* This case is untested, no samples available */
544
        avpriv_request_sample(NULL, "!superblocktype_2_3");
545
        return;
546
        for (ch = 0; ch < nb_channels; ch++) {
547
            for (sb = 0; sb < 30; sb++) {
548
                for (j = 1; j < 63; j++) {  // The loop only iterates to 63 so the code doesn't overflow the buffer
549
                    add1 = tone_level_idx[ch][sb][j] - 10;
550
                    if (add1 < 0)
551
                        add1 = 0;
552
                    add2 = add3 = add4 = 0;
553
                    if (sb > 1) {
554
                        add2 = tone_level_idx[ch][sb - 2][j] + tone_level_idx_offset_table[sb][0] - 6;
555
                        if (add2 < 0)
556
                            add2 = 0;
557
                    }
558
                    if (sb > 0) {
559
                        add3 = tone_level_idx[ch][sb - 1][j] + tone_level_idx_offset_table[sb][1] - 6;
560
                        if (add3 < 0)
561
                            add3 = 0;
562
                    }
563
                    if (sb < 29) {
564
                        add4 = tone_level_idx[ch][sb + 1][j] + tone_level_idx_offset_table[sb][3] - 6;
565
                        if (add4 < 0)
566
                            add4 = 0;
567
                    }
568
                    tmp = tone_level_idx[ch][sb][j + 1] * 2 - add4 - add3 - add2 - add1;
569
                    if (tmp < 0)
570
                        tmp = 0;
571
                    tone_level_idx_temp[ch][sb][j + 1] = tmp & 0xff;
572
                }
573
                tone_level_idx_temp[ch][sb][0] = tone_level_idx_temp[ch][sb][1];
574
            }
575
        }
576
        acc = 0;
577
        for (ch = 0; ch < nb_channels; ch++)
578
            for (sb = 0; sb < 30; sb++)
579
                for (j = 0; j < 64; j++)
580
                    acc += tone_level_idx_temp[ch][sb][j];
581
582
        multres = 0x66666667LL * (acc * 10);
583
        esp_40 = (multres >> 32) / 8 + ((multres & 0xffffffff) >> 31);
584
        for (ch = 0;  ch < nb_channels; ch++)
585
            for (sb = 0; sb < 30; sb++)
586
                for (j = 0; j < 64; j++) {
587
                    comp = tone_level_idx_temp[ch][sb][j]* esp_40 * 10;
588
                    if (comp < 0)
589
                        comp += 0xff;
590
                    comp /= 256; // signed shift
591
                    switch(sb) {
592
                        case 0:
593
                            if (comp < 30)
594
                                comp = 30;
595
                            comp += 15;
596
                            break;
597
                        case 1:
598
                            if (comp < 24)
599
                                comp = 24;
600
                            comp += 10;
601
                            break;
602
                        case 2:
603
                        case 3:
604
                        case 4:
605
                            if (comp < 16)
606
                                comp = 16;
607
                    }
608
                    if (comp <= 5)
609
                        tmp = 0;
610
                    else if (comp <= 10)
611
                        tmp = 10;
612
                    else if (comp <= 16)
613
                        tmp = 16;
614
                    else if (comp <= 24)
615
                        tmp = -1;
616
                    else
617
                        tmp = 0;
618
                    coding_method[ch][sb][j] = ((tmp & 0xfffa) + 30 )& 0xff;
619
                }
620
        for (sb = 0; sb < 30; sb++)
621
            fix_coding_method_array(sb, nb_channels, coding_method);
622
        for (ch = 0; ch < nb_channels; ch++)
623
            for (sb = 0; sb < 30; sb++)
624
                for (j = 0; j < 64; j++)
625
                    if (sb >= 10) {
626
                        if (coding_method[ch][sb][j] < 10)
627
                            coding_method[ch][sb][j] = 10;
628
                    } else {
629
                        if (sb >= 2) {
630
                            if (coding_method[ch][sb][j] < 16)
631
                                coding_method[ch][sb][j] = 16;
632
                        } else {
633
                            if (coding_method[ch][sb][j] < 30)
634
                                coding_method[ch][sb][j] = 30;
635
                        }
636
                    }
637
    } else { // superblocktype_2_3 != 0
638
        for (ch = 0; ch < nb_channels; ch++)
639
            for (sb = 0; sb < 30; sb++)
640
                for (j = 0; j < 64; j++)
641
                    coding_method[ch][sb][j] = coding_method_table[cm_table_select][sb];
642
    }
643
}
644
645
/**
646
 * Called by process_subpacket_11 to process more data from subpacket 11
647
 * with sb 0-8.
648
 * Called by process_subpacket_12 to process data from subpacket 12 with
649
 * sb 8-sb_used.
650
 *
651
 * @param q         context
652
 * @param gb        bitreader context
653
 * @param length    packet length in bits
654
 * @param sb_min    lower subband processed (sb_min included)
655
 * @param sb_max    higher subband processed (sb_max excluded)
656
 */
657
276
static int synthfilt_build_sb_samples(QDM2Context *q, GetBitContext *gb,
658
                                       int length, int sb_min, int sb_max)
659
{
660
    int sb, j, k, n, ch, run, channels;
661
    int joined_stereo, zero_encoding;
662
    int type34_first;
663
276
    float type34_div = 0;
664
    float type34_predictor;
665
    float samples[10];
666
276
    int sign_bits[16] = {0};
667
668
276
    if (length == 0) {
669
        // If no data use noise
670
4416
        for (sb=sb_min; sb < sb_max; sb++)
671
4140
            build_sb_samples_from_noise(q, sb);
672
673
276
        return 0;
674
    }
675
676
    for (sb = sb_min; sb < sb_max; sb++) {
677
        channels = q->nb_channels;
678
679
        if (q->nb_channels <= 1 || sb < 12)
680
            joined_stereo = 0;
681
        else if (sb >= 24)
682
            joined_stereo = 1;
683
        else
684
            joined_stereo = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
685
686
        if (joined_stereo) {
687
            if (get_bits_left(gb) >= 16)
688
                for (j = 0; j < 16; j++)
689
                    sign_bits[j] = get_bits1(gb);
690
691
            for (j = 0; j < 64; j++)
692
                if (q->coding_method[1][sb][j] > q->coding_method[0][sb][j])
693
                    q->coding_method[0][sb][j] = q->coding_method[1][sb][j];
694
695
            if (fix_coding_method_array(sb, q->nb_channels,
696
                                            q->coding_method)) {
697
                av_log(NULL, AV_LOG_ERROR, "coding method invalid\n");
698
                build_sb_samples_from_noise(q, sb);
699
                continue;
700
            }
701
            channels = 1;
702
        }
703
704
        for (ch = 0; ch < channels; ch++) {
705
            FIX_NOISE_IDX(q->noise_idx);
706
            zero_encoding = (get_bits_left(gb) >= 1) ? get_bits1(gb) : 0;
707
            type34_predictor = 0.0;
708
            type34_first = 1;
709
710
            for (j = 0; j < 128; ) {
711
                switch (q->coding_method[ch][sb][j / 2]) {
712
                    case 8:
713
                        if (get_bits_left(gb) >= 10) {
714
                            if (zero_encoding) {
715
                                for (k = 0; k < 5; k++) {
716
                                    if ((j + 2 * k) >= 128)
717
                                        break;
718
                                    samples[2 * k] = get_bits1(gb) ? dequant_1bit[joined_stereo][2 * get_bits1(gb)] : 0;
719
                                }
720
                            } else {
721
                                n = get_bits(gb, 8);
722
                                if (n >= 243) {
723
                                    av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
724
                                    return AVERROR_INVALIDDATA;
725
                                }
726
727
                                for (k = 0; k < 5; k++)
728
                                    samples[2 * k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
729
                            }
730
                            for (k = 0; k < 5; k++)
731
                                samples[2 * k + 1] = SB_DITHERING_NOISE(sb,q->noise_idx);
732
                        } else {
733
                            for (k = 0; k < 10; k++)
734
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
735
                        }
736
                        run = 10;
737
                        break;
738
739
                    case 10:
740
                        if (get_bits_left(gb) >= 1) {
741
                            float f = 0.81;
742
743
                            if (get_bits1(gb))
744
                                f = -f;
745
                            f -= noise_samples[((sb + 1) * (j +5 * ch + 1)) & 127] * 9.0 / 40.0;
746
                            samples[0] = f;
747
                        } else {
748
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
749
                        }
750
                        run = 1;
751
                        break;
752
753
                    case 16:
754
                        if (get_bits_left(gb) >= 10) {
755
                            if (zero_encoding) {
756
                                for (k = 0; k < 5; k++) {
757
                                    if ((j + k) >= 128)
758
                                        break;
759
                                    samples[k] = (get_bits1(gb) == 0) ? 0 : dequant_1bit[joined_stereo][2 * get_bits1(gb)];
760
                                }
761
                            } else {
762
                                n = get_bits (gb, 8);
763
                                if (n >= 243) {
764
                                    av_log(NULL, AV_LOG_ERROR, "Invalid 8bit codeword\n");
765
                                    return AVERROR_INVALIDDATA;
766
                                }
767
768
                                for (k = 0; k < 5; k++)
769
                                    samples[k] = dequant_1bit[joined_stereo][random_dequant_index[n][k]];
770
                            }
771
                        } else {
772
                            for (k = 0; k < 5; k++)
773
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
774
                        }
775
                        run = 5;
776
                        break;
777
778
                    case 24:
779
                        if (get_bits_left(gb) >= 7) {
780
                            n = get_bits(gb, 7);
781
                            if (n >= 125) {
782
                                av_log(NULL, AV_LOG_ERROR, "Invalid 7bit codeword\n");
783
                                return AVERROR_INVALIDDATA;
784
                            }
785
786
                            for (k = 0; k < 3; k++)
787
                                samples[k] = (random_dequant_type24[n][k] - 2.0) * 0.5;
788
                        } else {
789
                            for (k = 0; k < 3; k++)
790
                                samples[k] = SB_DITHERING_NOISE(sb,q->noise_idx);
791
                        }
792
                        run = 3;
793
                        break;
794
795
                    case 30:
796
                        if (get_bits_left(gb) >= 4) {
797
                            unsigned index = qdm2_get_vlc(gb, &vlc_tab_type30, 0, 1);
798
                            if (index >= FF_ARRAY_ELEMS(type30_dequant)) {
799
                                av_log(NULL, AV_LOG_ERROR, "index %d out of type30_dequant array\n", index);
800
                                return AVERROR_INVALIDDATA;
801
                            }
802
                            samples[0] = type30_dequant[index];
803
                        } else
804
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
805
806
                        run = 1;
807
                        break;
808
809
                    case 34:
810
                        if (get_bits_left(gb) >= 7) {
811
                            if (type34_first) {
812
                                type34_div = (float)(1 << get_bits(gb, 2));
813
                                samples[0] = ((float)get_bits(gb, 5) - 16.0) / 15.0;
814
                                type34_predictor = samples[0];
815
                                type34_first = 0;
816
                            } else {
817
                                unsigned index = qdm2_get_vlc(gb, &vlc_tab_type34, 0, 1);
818
                                if (index >= FF_ARRAY_ELEMS(type34_delta)) {
819
                                    av_log(NULL, AV_LOG_ERROR, "index %d out of type34_delta array\n", index);
820
                                    return AVERROR_INVALIDDATA;
821
                                }
822
                                samples[0] = type34_delta[index] / type34_div + type34_predictor;
823
                                type34_predictor = samples[0];
824
                            }
825
                        } else {
826
                            samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
827
                        }
828
                        run = 1;
829
                        break;
830
831
                    default:
832
                        samples[0] = SB_DITHERING_NOISE(sb,q->noise_idx);
833
                        run = 1;
834
                        break;
835
                }
836
837
                if (joined_stereo) {
838
                    for (k = 0; k < run && j + k < 128; k++) {
839
                        q->sb_samples[0][j + k][sb] =
840
                            q->tone_level[0][sb][(j + k) / 2] * samples[k];
841
                        if (q->nb_channels == 2) {
842
                            if (sign_bits[(j + k) / 8])
843
                                q->sb_samples[1][j + k][sb] =
844
                                    q->tone_level[1][sb][(j + k) / 2] * -samples[k];
845
                            else
846
                                q->sb_samples[1][j + k][sb] =
847
                                    q->tone_level[1][sb][(j + k) / 2] * samples[k];
848
                        }
849
                    }
850
                } else {
851
                    for (k = 0; k < run; k++)
852
                        if ((j + k) < 128)
853
                            q->sb_samples[ch][j + k][sb] = q->tone_level[ch][sb][(j + k)/2] * samples[k];
854
                }
855
856
                j += run;
857
            } // j loop
858
        } // channel loop
859
    } // subband loop
860
    return 0;
861
}
862
863
/**
864
 * Init the first element of a channel in quantized_coeffs with data
865
 * from packet 10 (quantized_coeffs[ch][0]).
866
 * This is similar to process_subpacket_9, but for a single channel
867
 * and for element [0]
868
 * same VLC tables as process_subpacket_9 are used.
869
 *
870
 * @param quantized_coeffs    pointer to quantized_coeffs[ch][0]
871
 * @param gb        bitreader context
872
 */
873
static int init_quantized_coeffs_elem0(int8_t *quantized_coeffs,
874
                                        GetBitContext *gb)
875
{
876
    int i, k, run, level, diff;
877
878
    if (get_bits_left(gb) < 16)
879
        return -1;
880
    level = qdm2_get_vlc(gb, &vlc_tab_level, 0, 2);
881
882
    quantized_coeffs[0] = level;
883
884
    for (i = 0; i < 7; ) {
885
        if (get_bits_left(gb) < 16)
886
            return -1;
887
        run = qdm2_get_vlc(gb, &vlc_tab_run, 0, 1) + 1;
888
889
        if (i + run >= 8)
890
            return -1;
891
892
        if (get_bits_left(gb) < 16)
893
            return -1;
894
        diff = qdm2_get_se_vlc(&vlc_tab_diff, gb, 2);
895
896
        for (k = 1; k <= run; k++)
897
            quantized_coeffs[i + k] = (level + ((k * diff) / run));
898
899
        level += diff;
900
        i += run;
901
    }
902
    return 0;
903
}
904
905
/**
906
 * Related to synthesis filter, process data from packet 10
907
 * Init part of quantized_coeffs via function init_quantized_coeffs_elem0
908
 * Init tone_level_idx_hi1, tone_level_idx_hi2, tone_level_idx_mid with
909
 * data from packet 10
910
 *
911
 * @param q         context
912
 * @param gb        bitreader context
913
 */
914
static void init_tone_level_dequantization(QDM2Context *q, GetBitContext *gb)
915
{
916
    int sb, j, k, n, ch;
917
918
    for (ch = 0; ch < q->nb_channels; ch++) {
919
        init_quantized_coeffs_elem0(q->quantized_coeffs[ch][0], gb);
920
921
        if (get_bits_left(gb) < 16) {
922
            memset(q->quantized_coeffs[ch][0], 0, 8);
923
            break;
924
        }
925
    }
926
927
    n = q->sub_sampling + 1;
928
929
    for (sb = 0; sb < n; sb++)
930
        for (ch = 0; ch < q->nb_channels; ch++)
931
            for (j = 0; j < 8; j++) {
932
                if (get_bits_left(gb) < 1)
933
                    break;
934
                if (get_bits1(gb)) {
935
                    for (k=0; k < 8; k++) {
936
                        if (get_bits_left(gb) < 16)
937
                            break;
938
                        q->tone_level_idx_hi1[ch][sb][j][k] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi1, 0, 2);
939
                    }
940
                } else {
941
                    for (k=0; k < 8; k++)
942
                        q->tone_level_idx_hi1[ch][sb][j][k] = 0;
943
                }
944
            }
945
946
    n = QDM2_SB_USED(q->sub_sampling) - 4;
947
948
    for (sb = 0; sb < n; sb++)
949
        for (ch = 0; ch < q->nb_channels; ch++) {
950
            if (get_bits_left(gb) < 16)
951
                break;
952
            q->tone_level_idx_hi2[ch][sb] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_hi2, 0, 2);
953
            if (sb > 19)
954
                q->tone_level_idx_hi2[ch][sb] -= 16;
955
            else
956
                for (j = 0; j < 8; j++)
957
                    q->tone_level_idx_mid[ch][sb][j] = -16;
958
        }
959
960
    n = QDM2_SB_USED(q->sub_sampling) - 5;
961
962
    for (sb = 0; sb < n; sb++)
963
        for (ch = 0; ch < q->nb_channels; ch++)
964
            for (j = 0; j < 8; j++) {
965
                if (get_bits_left(gb) < 16)
966
                    break;
967
                q->tone_level_idx_mid[ch][sb][j] = qdm2_get_vlc(gb, &vlc_tab_tone_level_idx_mid, 0, 2) - 32;
968
            }
969
}
970
971
/**
972
 * Process subpacket 9, init quantized_coeffs with data from it
973
 *
974
 * @param q       context
975
 * @param node    pointer to node with packet
976
 */
977
138
static int process_subpacket_9(QDM2Context *q, QDM2SubPNode *node)
978
{
979
    GetBitContext gb;
980
    int i, j, k, n, ch, run, level, diff;
981
982
138
    init_get_bits(&gb, node->packet->data, node->packet->size * 8);
983
984
138
    n = coeff_per_sb_for_avg[q->coeff_per_sb_select][QDM2_SB_USED(q->sub_sampling) - 1] + 1;
985
986
1380
    for (i = 1; i < n; i++)
987
3726
        for (ch = 0; ch < q->nb_channels; ch++) {
988
2484
            level = qdm2_get_vlc(&gb, &vlc_tab_level, 0, 2);
989
2484
            q->quantized_coeffs[ch][i][0] = level;
990
991
13380
            for (j = 0; j < (8 - 1); ) {
992
10896
                run  = qdm2_get_vlc(&gb, &vlc_tab_run, 0, 1) + 1;
993
10896
                diff = qdm2_get_se_vlc(&vlc_tab_diff, &gb, 2);
994
995
10896
                if (j + run >= 8)
996
                    return -1;
997
998
28284
                for (k = 1; k <= run; k++)
999
17388
                    q->quantized_coeffs[ch][i][j + k] = (level + ((k * diff) / run));
1000
1001
10896
                level += diff;
1002
10896
                j     += run;
1003
            }
1004
        }
1005
1006
414
    for (ch = 0; ch < q->nb_channels; ch++)
1007
2484
        for (i = 0; i < 8; i++)
1008
2208
            q->quantized_coeffs[ch][0][i] = 0;
1009
1010
138
    return 0;
1011
}
1012
1013
/**
1014
 * Process subpacket 10 if not null, else
1015
 *
1016
 * @param q         context
1017
 * @param node      pointer to node with packet
1018
 */
1019
138
static void process_subpacket_10(QDM2Context *q, QDM2SubPNode *node)
1020
{
1021
    GetBitContext gb;
1022
1023
138
    if (node) {
1024
        init_get_bits(&gb, node->packet->data, node->packet->size * 8);
1025
        init_tone_level_dequantization(q, &gb);
1026
        fill_tone_level_array(q, 1);
1027
    } else {
1028
138
        fill_tone_level_array(q, 0);
1029
    }
1030
138
}
1031
1032
/**
1033
 * Process subpacket 11
1034
 *
1035
 * @param q         context
1036
 * @param node      pointer to node with packet
1037
 */
1038
138
static void process_subpacket_11(QDM2Context *q, QDM2SubPNode *node)
1039
{
1040
    GetBitContext gb;
1041
138
    int length = 0;
1042
1043
138
    if (node) {
1044
        length = node->packet->size * 8;
1045
        init_get_bits(&gb, node->packet->data, length);
1046
    }
1047
1048
138
    if (length >= 32) {
1049
        int c = get_bits(&gb, 13);
1050
1051
        if (c > 3)
1052
            fill_coding_method_array(q->tone_level_idx,
1053
                                     q->tone_level_idx_temp, q->coding_method,
1054
                                     q->nb_channels, 8 * c,
1055
                                     q->superblocktype_2_3, q->cm_table_select);
1056
    }
1057
1058
138
    synthfilt_build_sb_samples(q, &gb, length, 0, 8);
1059
138
}
1060
1061
/**
1062
 * Process subpacket 12
1063
 *
1064
 * @param q         context
1065
 * @param node      pointer to node with packet
1066
 */
1067
138
static void process_subpacket_12(QDM2Context *q, QDM2SubPNode *node)
1068
{
1069
    GetBitContext gb;
1070
138
    int length = 0;
1071
1072
138
    if (node) {
1073
        length = node->packet->size * 8;
1074
        init_get_bits(&gb, node->packet->data, length);
1075
    }
1076
1077
138
    synthfilt_build_sb_samples(q, &gb, length, 8, QDM2_SB_USED(q->sub_sampling));
1078
138
}
1079
1080
/**
1081
 * Process new subpackets for synthesis filter
1082
 *
1083
 * @param q       context
1084
 * @param list    list with synthesis filter packets (list D)
1085
 */
1086
138
static void process_synthesis_subpackets(QDM2Context *q, QDM2SubPNode *list)
1087
{
1088
    QDM2SubPNode *nodes[4];
1089
1090
138
    nodes[0] = qdm2_search_subpacket_type_in_list(list, 9);
1091
138
    if (nodes[0])
1092
138
        process_subpacket_9(q, nodes[0]);
1093
1094
138
    nodes[1] = qdm2_search_subpacket_type_in_list(list, 10);
1095
138
    if (nodes[1])
1096
        process_subpacket_10(q, nodes[1]);
1097
    else
1098
138
        process_subpacket_10(q, NULL);
1099
1100
138
    nodes[2] = qdm2_search_subpacket_type_in_list(list, 11);
1101

138
    if (nodes[0] && nodes[1] && nodes[2])
1102
        process_subpacket_11(q, nodes[2]);
1103
    else
1104
138
        process_subpacket_11(q, NULL);
1105
1106
138
    nodes[3] = qdm2_search_subpacket_type_in_list(list, 12);
1107

138
    if (nodes[0] && nodes[1] && nodes[3])
1108
        process_subpacket_12(q, nodes[3]);
1109
    else
1110
138
        process_subpacket_12(q, NULL);
1111
138
}
1112
1113
/**
1114
 * Decode superblock, fill packet lists.
1115
 *
1116
 * @param q    context
1117
 */
1118
138
static void qdm2_decode_super_block(QDM2Context *q)
1119
{
1120
    GetBitContext gb;
1121
    QDM2SubPacket header, *packet;
1122
    int i, packet_bytes, sub_packet_size, sub_packets_D;
1123
138
    unsigned int next_index = 0;
1124
1125
138
    memset(q->tone_level_idx_hi1, 0, sizeof(q->tone_level_idx_hi1));
1126
138
    memset(q->tone_level_idx_mid, 0, sizeof(q->tone_level_idx_mid));
1127
138
    memset(q->tone_level_idx_hi2, 0, sizeof(q->tone_level_idx_hi2));
1128
1129
138
    q->sub_packets_B = 0;
1130
138
    sub_packets_D    = 0;
1131
1132
138
    average_quantized_coeffs(q); // average elements in quantized_coeffs[max_ch][10][8]
1133
1134
138
    init_get_bits(&gb, q->compressed_data, q->compressed_size * 8);
1135
138
    qdm2_decode_sub_packet_header(&gb, &header);
1136
1137

138
    if (header.type < 2 || header.type >= 8) {
1138
        q->has_errors = 1;
1139
        av_log(NULL, AV_LOG_ERROR, "bad superblock type\n");
1140
        return;
1141
    }
1142
1143

138
    q->superblocktype_2_3 = (header.type == 2 || header.type == 3);
1144
138
    packet_bytes          = (q->compressed_size - get_bits_count(&gb) / 8);
1145
1146
138
    init_get_bits(&gb, header.data, header.size * 8);
1147
1148

138
    if (header.type == 2 || header.type == 4 || header.type == 5) {
1149
138
        int csum = 257 * get_bits(&gb, 8);
1150
138
        csum += 2 * get_bits(&gb, 8);
1151
1152
138
        csum = qdm2_packet_checksum(q->compressed_data, q->checksum_size, csum);
1153
1154
138
        if (csum != 0) {
1155
            q->has_errors = 1;
1156
            av_log(NULL, AV_LOG_ERROR, "bad packet checksum\n");
1157
            return;
1158
        }
1159
    }
1160
1161
138
    q->sub_packet_list_B[0].packet = NULL;
1162
138
    q->sub_packet_list_D[0].packet = NULL;
1163
1164
966
    for (i = 0; i < 6; i++)
1165
828
        if (--q->fft_level_exp[i] < 0)
1166
828
            q->fft_level_exp[i] = 0;
1167
1168
966
    for (i = 0; packet_bytes > 0; i++) {
1169
        int j;
1170
1171
966
        if (i >= FF_ARRAY_ELEMS(q->sub_packet_list_A)) {
1172
            SAMPLES_NEEDED_2("too many packet bytes");
1173
            return;
1174
        }
1175
1176
966
        q->sub_packet_list_A[i].next = NULL;
1177
1178
966
        if (i > 0) {
1179
828
            q->sub_packet_list_A[i - 1].next = &q->sub_packet_list_A[i];
1180
1181
            /* seek to next block */
1182
828
            init_get_bits(&gb, header.data, header.size * 8);
1183
828
            skip_bits(&gb, next_index * 8);
1184
1185
828
            if (next_index >= header.size)
1186
                break;
1187
        }
1188
1189
        /* decode subpacket */
1190
966
        packet = &q->sub_packets[i];
1191
966
        qdm2_decode_sub_packet_header(&gb, packet);
1192
966
        next_index      = packet->size + get_bits_count(&gb) / 8;
1193
966
        sub_packet_size = ((packet->size > 0xff) ? 1 : 0) + packet->size + 2;
1194
1195
966
        if (packet->type == 0)
1196
138
            break;
1197
1198
828
        if (sub_packet_size > packet_bytes) {
1199
            if (packet->type != 10 && packet->type != 11 && packet->type != 12)
1200
                break;
1201
            packet->size += packet_bytes - sub_packet_size;
1202
        }
1203
1204
828
        packet_bytes -= sub_packet_size;
1205
1206
        /* add subpacket to 'all subpackets' list */
1207
828
        q->sub_packet_list_A[i].packet = packet;
1208
1209
        /* add subpacket to related list */
1210
828
        if (packet->type == 8) {
1211
            SAMPLES_NEEDED_2("packet type 8");
1212
            return;
1213

828
        } else if (packet->type >= 9 && packet->type <= 12) {
1214
            /* packets for MPEG Audio like Synthesis Filter */
1215
138
            QDM2_LIST_ADD(q->sub_packet_list_D, sub_packets_D, packet);
1216
690
        } else if (packet->type == 13) {
1217
            for (j = 0; j < 6; j++)
1218
                q->fft_level_exp[j] = get_bits(&gb, 6);
1219
690
        } else if (packet->type == 14) {
1220
            for (j = 0; j < 6; j++)
1221
                q->fft_level_exp[j] = qdm2_get_vlc(&gb, &fft_level_exp_vlc, 0, 2);
1222
690
        } else if (packet->type == 15) {
1223
            SAMPLES_NEEDED_2("packet type 15")
1224
            return;
1225

690
        } else if (packet->type >= 16 && packet->type < 48 &&
1226
690
                   !fft_subpackets[packet->type - 16]) {
1227
            /* packets for FFT */
1228
690
            QDM2_LIST_ADD(q->sub_packet_list_B, q->sub_packets_B, packet);
1229
        }
1230
    } // Packet bytes loop
1231
1232
138
    if (q->sub_packet_list_D[0].packet) {
1233
138
        process_synthesis_subpackets(q, q->sub_packet_list_D);
1234
138
        q->do_synth_filter = 1;
1235
    } else if (q->do_synth_filter) {
1236
        process_subpacket_10(q, NULL);
1237
        process_subpacket_11(q, NULL);
1238
        process_subpacket_12(q, NULL);
1239
    }
1240
}
1241
1242
20037
static void qdm2_fft_init_coefficient(QDM2Context *q, int sub_packet,
1243
                                      int offset, int duration, int channel,
1244
                                      int exp, int phase)
1245
{
1246
20037
    if (q->fft_coefs_min_index[duration] < 0)
1247
542
        q->fft_coefs_min_index[duration] = q->fft_coefs_index;
1248
1249
20037
    q->fft_coefs[q->fft_coefs_index].sub_packet =
1250
284
        ((sub_packet >= 16) ? (sub_packet - 16) : sub_packet);
1251
20037
    q->fft_coefs[q->fft_coefs_index].channel = channel;
1252
20037
    q->fft_coefs[q->fft_coefs_index].offset  = offset;
1253
20037
    q->fft_coefs[q->fft_coefs_index].exp     = exp;
1254
20037
    q->fft_coefs[q->fft_coefs_index].phase   = phase;
1255
20037
    q->fft_coefs_index++;
1256
20037
}
1257
1258
552
static void qdm2_fft_decode_tones(QDM2Context *q, int duration,
1259
                                  GetBitContext *gb, int b)
1260
{
1261
    int channel, stereo, phase, exp;
1262
    int local_int_4, local_int_8, stereo_phase, local_int_10;
1263
    int local_int_14, stereo_exp, local_int_20, local_int_28;
1264
    int n, offset;
1265
1266
552
    local_int_4  = 0;
1267
552
    local_int_28 = 0;
1268
552
    local_int_20 = 2;
1269
552
    local_int_8  = (4 - duration);
1270
552
    local_int_10 = 1 << (q->group_order - duration - 1);
1271
552
    offset       = 1;
1272
1273
16148
    while (get_bits_left(gb)>0) {
1274
16148
        if (q->superblocktype_2_3) {
1275
18134
            while ((n = qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2)) < 2) {
1276
2053
                if (get_bits_left(gb)<0) {
1277
67
                    if(local_int_4 < q->group_size)
1278
                        av_log(NULL, AV_LOG_ERROR, "overread in qdm2_fft_decode_tones()\n");
1279
67
                    return;
1280
                }
1281
1986
                offset = 1;
1282
1986
                if (n == 0) {
1283
1974
                    local_int_4  += local_int_10;
1284
1974
                    local_int_28 += (1 << local_int_8);
1285
                } else {
1286
12
                    local_int_4  += 8 * local_int_10;
1287
12
                    local_int_28 += (8 << local_int_8);
1288
                }
1289
            }
1290
16081
            offset += (n - 2);
1291
        } else {
1292
            if (local_int_10 <= 2) {
1293
                av_log(NULL, AV_LOG_ERROR, "qdm2_fft_decode_tones() stuck\n");
1294
                return;
1295
            }
1296
            offset += qdm2_get_vlc(gb, &vlc_tab_fft_tone_offset[local_int_8], 1, 2);
1297
            while (offset >= (local_int_10 - 1)) {
1298
                offset       += (1 - (local_int_10 - 1));
1299
                local_int_4  += local_int_10;
1300
                local_int_28 += (1 << local_int_8);
1301
            }
1302
        }
1303
1304
16081
        if (local_int_4 >= q->group_size)
1305
485
            return;
1306
1307
15596
        local_int_14 = (offset >> local_int_8);
1308
15596
        if (local_int_14 >= FF_ARRAY_ELEMS(fft_level_index_table))
1309
            return;
1310
1311
15596
        if (q->nb_channels > 1) {
1312
15596
            channel = get_bits1(gb);
1313
15596
            stereo  = get_bits1(gb);
1314
        } else {
1315
            channel = 0;
1316
            stereo  = 0;
1317
        }
1318
1319
15596
        exp  = qdm2_get_vlc(gb, (b ? &fft_level_exp_vlc : &fft_level_exp_alt_vlc), 0, 2);
1320
15596
        exp += q->fft_level_exp[fft_level_index_table[local_int_14]];
1321
15596
        exp  = (exp < 0) ? 0 : exp;
1322
1323
15596
        phase        = get_bits(gb, 3);
1324
15596
        stereo_exp   = 0;
1325
15596
        stereo_phase = 0;
1326
1327
15596
        if (stereo) {
1328
4441
            stereo_exp   = (exp - qdm2_get_vlc(gb, &fft_stereo_exp_vlc, 0, 1));
1329
4441
            stereo_phase = (phase - qdm2_get_vlc(gb, &fft_stereo_phase_vlc, 0, 1));
1330
4441
            if (stereo_phase < 0)
1331
497
                stereo_phase += 8;
1332
        }
1333
1334
15596
        if (q->frequency_range > (local_int_14 + 1)) {
1335
15596
            int sub_packet = (local_int_20 + local_int_28);
1336
1337
15596
            if (q->fft_coefs_index + stereo >= FF_ARRAY_ELEMS(q->fft_coefs))
1338
                return;
1339
1340
15596
            qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1341
                                      channel, exp, phase);
1342
15596
            if (stereo)
1343
4441
                qdm2_fft_init_coefficient(q, sub_packet, offset, duration,
1344
                                          1 - channel,
1345
                                          stereo_exp, stereo_phase);
1346
        }
1347
15596
        offset++;
1348
    }
1349
}
1350
1351
138
static void qdm2_decode_fft_packets(QDM2Context *q)
1352
{
1353
    int i, j, min, max, value, type, unknown_flag;
1354
    GetBitContext gb;
1355
1356
138
    if (!q->sub_packet_list_B[0].packet)
1357
        return;
1358
1359
    /* reset minimum indexes for FFT coefficients */
1360
138
    q->fft_coefs_index = 0;
1361
828
    for (i = 0; i < 5; i++)
1362
690
        q->fft_coefs_min_index[i] = -1;
1363
1364
    /* process subpackets ordered by type, largest type first */
1365
828
    for (i = 0, max = 256; i < q->sub_packets_B; i++) {
1366
690
        QDM2SubPacket *packet = NULL;
1367
1368
        /* find subpacket with largest type less than max */
1369
4140
        for (j = 0, min = 0; j < q->sub_packets_B; j++) {
1370
3450
            value = q->sub_packet_list_B[j].packet->type;
1371

3450
            if (value > min && value < max) {
1372
690
                min    = value;
1373
690
                packet = q->sub_packet_list_B[j].packet;
1374
            }
1375
        }
1376
1377
690
        max = min;
1378
1379
        /* check for errors (?) */
1380
690
        if (!packet)
1381
            return;
1382
1383
690
        if (i == 0 &&
1384

138
            (packet->type < 16 || packet->type >= 48 ||
1385
138
             fft_subpackets[packet->type - 16]))
1386
            return;
1387
1388
        /* decode FFT tones */
1389
690
        init_get_bits(&gb, packet->data, packet->size * 8);
1390
1391

690
        if (packet->type >= 32 && packet->type < 48 && !fft_subpackets[packet->type - 16])
1392
            unknown_flag = 1;
1393
        else
1394
690
            unknown_flag = 0;
1395
1396
690
        type = packet->type;
1397
1398


1380
        if ((type >= 17 && type < 24) || (type >= 33 && type < 40)) {
1399
690
            int duration = q->sub_sampling + 5 - (type & 15);
1400
1401

690
            if (duration >= 0 && duration < 4)
1402
552
                qdm2_fft_decode_tones(q, duration, &gb, unknown_flag);
1403
        } else if (type == 31) {
1404
            for (j = 0; j < 4; j++)
1405
                qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1406
        } else if (type == 46) {
1407
            for (j = 0; j < 6; j++)
1408
                q->fft_level_exp[j] = get_bits(&gb, 6);
1409
            for (j = 0; j < 4; j++)
1410
                qdm2_fft_decode_tones(q, j, &gb, unknown_flag);
1411
        }
1412
    } // Loop on B packets
1413
1414
    /* calculate maximum indexes for FFT coefficients */
1415
828
    for (i = 0, j = -1; i < 5; i++)
1416
690
        if (q->fft_coefs_min_index[i] >= 0) {
1417
542
            if (j >= 0)
1418
404
                q->fft_coefs_max_index[j] = q->fft_coefs_min_index[i];
1419
542
            j = i;
1420
        }
1421
138
    if (j >= 0)
1422
138
        q->fft_coefs_max_index[j] = q->fft_coefs_index;
1423
}
1424
1425
340988
static void qdm2_fft_generate_tone(QDM2Context *q, FFTTone *tone)
1426
{
1427
    float level, f[6];
1428
    int i;
1429
    QDM2Complex c;
1430
340988
    const double iscale = 2.0 * M_PI / 512.0;
1431
1432
340988
    tone->phase += tone->phase_shift;
1433
1434
    /* calculate current level (maximum amplitude) of tone */
1435
340988
    level = fft_tone_envelope_table[tone->duration][tone->time_index] * tone->level;
1436
340988
    c.im  = level * sin(tone->phase * iscale);
1437
340988
    c.re  = level * cos(tone->phase * iscale);
1438
1439
    /* generate FFT coefficients for tone */
1440

340988
    if (tone->duration >= 3 || tone->cutoff >= 3) {
1441
46448
        tone->complex[0].im += c.im;
1442
46448
        tone->complex[0].re += c.re;
1443
46448
        tone->complex[1].im -= c.im;
1444
46448
        tone->complex[1].re -= c.re;
1445
    } else {
1446
294540
        f[1] = -tone->table[4];
1447
294540
        f[0] = tone->table[3] - tone->table[0];
1448
294540
        f[2] = 1.0 - tone->table[2] - tone->table[3];
1449
294540
        f[3] = tone->table[1] + tone->table[4] - 1.0;
1450
294540
        f[4] = tone->table[0] - tone->table[1];
1451
294540
        f[5] = tone->table[2];
1452
883620
        for (i = 0; i < 2; i++) {
1453
589080
            tone->complex[fft_cutoff_index_table[tone->cutoff][i]].re +=
1454
589080
                c.re * f[i];
1455
1178160
            tone->complex[fft_cutoff_index_table[tone->cutoff][i]].im +=
1456
589080
                c.im * ((tone->cutoff <= i) ? -f[i] : f[i]);
1457
        }
1458
1472700
        for (i = 0; i < 4; i++) {
1459
1178160
            tone->complex[i].re += c.re * f[i + 2];
1460
1178160
            tone->complex[i].im += c.im * f[i + 2];
1461
        }
1462
    }
1463
1464
    /* copy the tone if it has not yet died out */
1465
340988
    if (++tone->time_index < ((1 << (5 - tone->duration)) - 1)) {
1466
321230
        memcpy(&q->fft_tones[q->fft_tone_end], tone, sizeof(FFTTone));
1467
321230
        q->fft_tone_end = (q->fft_tone_end + 1) % 1000;
1468
    }
1469
340988
}
1470
1471
2208
static void qdm2_fft_tone_synthesizer(QDM2Context *q, int sub_packet)
1472
{
1473
    int i, j, ch;
1474
2208
    const double iscale = 0.25 * M_PI;
1475
1476
6624
    for (ch = 0; ch < q->channels; ch++) {
1477
4416
        memset(q->fft.complex[ch], 0, q->fft_size * sizeof(QDM2Complex));
1478
    }
1479
1480
1481
    /* apply FFT tones with duration 4 (1 FFT period) */
1482
2208
    if (q->fft_coefs_min_index[4] >= 0)
1483
2
        for (i = q->fft_coefs_min_index[4]; i < q->fft_coefs_max_index[4]; i++) {
1484
            float level;
1485
            QDM2Complex c;
1486
1487
            if (q->fft_coefs[i].sub_packet != sub_packet)
1488
                break;
1489
1490
            ch = (q->channels == 1) ? 0 : q->fft_coefs[i].channel;
1491
            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];
1492
1493
            c.re = level * cos(q->fft_coefs[i].phase * iscale);
1494
            c.im = level * sin(q->fft_coefs[i].phase * iscale);
1495
            q->fft.complex[ch][q->fft_coefs[i].offset + 0].re += c.re;
1496
            q->fft.complex[ch][q->fft_coefs[i].offset + 0].im += c.im;
1497
            q->fft.complex[ch][q->fft_coefs[i].offset + 1].re -= c.re;
1498
            q->fft.complex[ch][q->fft_coefs[i].offset + 1].im -= c.im;
1499
        }
1500
1501
    /* generate existing FFT tones */
1502
323159
    for (i = q->fft_tone_end; i != q->fft_tone_start; ) {
1503
320951
        qdm2_fft_generate_tone(q, &q->fft_tones[q->fft_tone_start]);
1504
320951
        q->fft_tone_start = (q->fft_tone_start + 1) % 1000;
1505
    }
1506
1507
    /* create and generate new FFT tones with duration 0 (long) to 3 (short) */
1508
11040
    for (i = 0; i < 4; i++)
1509
8832
        if (q->fft_coefs_min_index[i] >= 0) {
1510
28711
            for (j = q->fft_coefs_min_index[i]; j < q->fft_coefs_max_index[i]; j++) {
1511
                int offset, four_i;
1512
                FFTTone tone;
1513
1514
24279
                if (q->fft_coefs[j].sub_packet != sub_packet)
1515
4242
                    break;
1516
1517
20037
                four_i = (4 - i);
1518
20037
                offset = q->fft_coefs[j].offset >> four_i;
1519
20037
                ch = (q->channels == 1) ? 0 : q->fft_coefs[j].channel;
1520
1521
20037
                if (offset < q->frequency_range) {
1522
20037
                    if (offset < 2)
1523
2780
                        tone.cutoff = offset;
1524
                    else
1525
17257
                        tone.cutoff = (offset >= 60) ? 3 : 2;
1526
1527
20037
                    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];
1528
20037
                    tone.complex = &q->fft.complex[ch][offset];
1529
20037
                    tone.table = fft_tone_sample_table[i][q->fft_coefs[j].offset - (offset << four_i)];
1530
20037
                    tone.phase = 64 * q->fft_coefs[j].phase - (offset << 8) - 128;
1531
20037
                    tone.phase_shift = (2 * q->fft_coefs[j].offset + 1) << (7 - four_i);
1532
20037
                    tone.duration = i;
1533
20037
                    tone.time_index = 0;
1534
1535
20037
                    qdm2_fft_generate_tone(q, &tone);
1536
                }
1537
            }
1538
8674
            q->fft_coefs_min_index[i] = j;
1539
        }
1540
2208
}
1541
1542
4416
static void qdm2_calculate_fft(QDM2Context *q, int channel, int sub_packet)
1543
{
1544

4416
    const float gain = (q->channels == 1 && q->nb_channels == 2) ? 0.5f : 1.0f;
1545
4416
    float *out       = q->output_buffer + channel;
1546
    int i;
1547
4416
    q->fft.complex[channel][0].re *= 2.0f;
1548
4416
    q->fft.complex[channel][0].im  = 0.0f;
1549
4416
    q->rdft_ctx.rdft_calc(&q->rdft_ctx, (FFTSample *)q->fft.complex[channel]);
1550
    /* add samples to output buffer */
1551
1134912
    for (i = 0; i < FFALIGN(q->fft_size, 8); i++) {
1552
1130496
        out[0]           += q->fft.complex[channel][i].re * gain;
1553
1130496
        out[q->channels] += q->fft.complex[channel][i].im * gain;
1554
1130496
        out              += 2 * q->channels;
1555
    }
1556
4416
}
1557
1558
/**
1559
 * @param q        context
1560
 * @param index    subpacket number
1561
 */
1562
2208
static void qdm2_synthesis_filter(QDM2Context *q, int index)
1563
{
1564
2208
    int i, k, ch, sb_used, sub_sampling, dither_state = 0;
1565
1566
    /* copy sb_samples */
1567
2208
    sb_used = QDM2_SB_USED(q->sub_sampling);
1568
1569
6624
    for (ch = 0; ch < q->channels; ch++)
1570
39744
        for (i = 0; i < 8; i++)
1571
105984
            for (k = sb_used; k < SBLIMIT; k++)
1572
70656
                q->sb_samples[ch][(8 * index) + i][k] = 0;
1573
1574
6624
    for (ch = 0; ch < q->nb_channels; ch++) {
1575
4416
        float *samples_ptr = q->samples + ch;
1576
1577
39744
        for (i = 0; i < 8; i++) {
1578
35328
            ff_mpa_synth_filter_float(&q->mpadsp,
1579
35328
                                      q->synth_buf[ch], &(q->synth_buf_offset[ch]),
1580
                                      ff_mpa_synth_window_float, &dither_state,
1581
35328
                                      samples_ptr, q->nb_channels,
1582
35328
                                      q->sb_samples[ch][(8 * index) + i]);
1583
35328
            samples_ptr += 32 * q->nb_channels;
1584
        }
1585
    }
1586
1587
    /* add samples to output buffer */
1588
2208
    sub_sampling = (4 >> q->sub_sampling);
1589
1590
6624
    for (ch = 0; ch < q->channels; ch++)
1591
1134912
        for (i = 0; i < q->frame_size; i++)
1592
1130496
            q->output_buffer[q->channels * i + ch] += (1 << 23) * q->samples[q->nb_channels * sub_sampling * i + ch];
1593
2208
}
1594
1595
/**
1596
 * Init static data (does not depend on specific file)
1597
 *
1598
 * @param q    context
1599
 */
1600
2
static av_cold void qdm2_init_static_data(void) {
1601
    static int done;
1602
1603
2
    if(done)
1604
1
        return;
1605
1606
1
    qdm2_init_vlc();
1607
1
    ff_mpa_synth_init_float(ff_mpa_synth_window_float);
1608
1
    softclip_table_init();
1609
1
    rnd_table_init();
1610
1
    init_noise_samples();
1611
1612
1
    done = 1;
1613
}
1614
1615
/**
1616
 * Init parameters from codec extradata
1617
 */
1618
2
static av_cold int qdm2_decode_init(AVCodecContext *avctx)
1619
{
1620
2
    QDM2Context *s = avctx->priv_data;
1621
    int tmp_val, tmp, size;
1622
    GetByteContext gb;
1623
1624
2
    qdm2_init_static_data();
1625
1626
    /* extradata parsing
1627
1628
    Structure:
1629
    wave {
1630
        frma (QDM2)
1631
        QDCA
1632
        QDCP
1633
    }
1634
1635
    32  size (including this field)
1636
    32  tag (=frma)
1637
    32  type (=QDM2 or QDMC)
1638
1639
    32  size (including this field, in bytes)
1640
    32  tag (=QDCA) // maybe mandatory parameters
1641
    32  unknown (=1)
1642
    32  channels (=2)
1643
    32  samplerate (=44100)
1644
    32  bitrate (=96000)
1645
    32  block size (=4096)
1646
    32  frame size (=256) (for one channel)
1647
    32  packet size (=1300)
1648
1649
    32  size (including this field, in bytes)
1650
    32  tag (=QDCP) // maybe some tuneable parameters
1651
    32  float1 (=1.0)
1652
    32  zero ?
1653
    32  float2 (=1.0)
1654
    32  float3 (=1.0)
1655
    32  unknown (27)
1656
    32  unknown (8)
1657
    32  zero ?
1658
    */
1659
1660

2
    if (!avctx->extradata || (avctx->extradata_size < 48)) {
1661
        av_log(avctx, AV_LOG_ERROR, "extradata missing or truncated\n");
1662
        return AVERROR_INVALIDDATA;
1663
    }
1664
1665
2
    bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1666
1667
10
    while (bytestream2_get_bytes_left(&gb) > 8) {
1668
10
        if (bytestream2_peek_be64(&gb) == (((uint64_t)MKBETAG('f','r','m','a') << 32) |
1669
                                            (uint64_t)MKBETAG('Q','D','M','2')))
1670
2
            break;
1671
8
        bytestream2_skip(&gb, 1);
1672
    }
1673
1674
2
    if (bytestream2_get_bytes_left(&gb) < 12) {
1675
        av_log(avctx, AV_LOG_ERROR, "not enough extradata (%i)\n",
1676
               bytestream2_get_bytes_left(&gb));
1677
        return AVERROR_INVALIDDATA;
1678
    }
1679
1680
2
    bytestream2_skip(&gb, 8);
1681
2
    size = bytestream2_get_be32(&gb);
1682
1683
2
    if (size > bytestream2_get_bytes_left(&gb)) {
1684
        av_log(avctx, AV_LOG_ERROR, "extradata size too small, %i < %i\n",
1685
               bytestream2_get_bytes_left(&gb), size);
1686
        return AVERROR_INVALIDDATA;
1687
    }
1688
1689
2
    av_log(avctx, AV_LOG_DEBUG, "size: %d\n", size);
1690
2
    if (bytestream2_get_be32(&gb) != MKBETAG('Q','D','C','A')) {
1691
        av_log(avctx, AV_LOG_ERROR, "invalid extradata, expecting QDCA\n");
1692
        return AVERROR_INVALIDDATA;
1693
    }
1694
1695
2
    bytestream2_skip(&gb, 4);
1696
1697
2
    avctx->channels = s->nb_channels = s->channels = bytestream2_get_be32(&gb);
1698

2
    if (s->channels <= 0 || s->channels > MPA_MAX_CHANNELS) {
1699
        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1700
        return AVERROR_INVALIDDATA;
1701
    }
1702
2
    avctx->channel_layout = avctx->channels == 2 ? AV_CH_LAYOUT_STEREO :
1703
                                                   AV_CH_LAYOUT_MONO;
1704
1705
2
    avctx->sample_rate = bytestream2_get_be32(&gb);
1706
2
    avctx->bit_rate = bytestream2_get_be32(&gb);
1707
2
    s->group_size = bytestream2_get_be32(&gb);
1708
2
    s->fft_size = bytestream2_get_be32(&gb);
1709
2
    s->checksum_size = bytestream2_get_be32(&gb);
1710

2
    if (s->checksum_size >= 1U << 28 || s->checksum_size <= 1) {
1711
        av_log(avctx, AV_LOG_ERROR, "data block size invalid (%u)\n", s->checksum_size);
1712
        return AVERROR_INVALIDDATA;
1713
    }
1714
1715
2
    s->fft_order = av_log2(s->fft_size) + 1;
1716
1717
    // Fail on unknown fft order
1718

2
    if ((s->fft_order < 7) || (s->fft_order > 9)) {
1719
        avpriv_request_sample(avctx, "Unknown FFT order %d", s->fft_order);
1720
        return AVERROR_PATCHWELCOME;
1721
    }
1722
1723
    // something like max decodable tones
1724
2
    s->group_order = av_log2(s->group_size) + 1;
1725
2
    s->frame_size = s->group_size / 16; // 16 iterations per super block
1726
1727
2
    if (s->frame_size > QDM2_MAX_FRAME_SIZE)
1728
        return AVERROR_INVALIDDATA;
1729
1730
2
    s->sub_sampling = s->fft_order - 7;
1731
2
    s->frequency_range = 255 / (1 << (2 - s->sub_sampling));
1732
1733
2
    if (s->frame_size * 4 >> s->sub_sampling > MPA_FRAME_SIZE) {
1734
        avpriv_request_sample(avctx, "large frames");
1735
        return AVERROR_PATCHWELCOME;
1736
    }
1737
1738

2
    switch ((s->sub_sampling * 2 + s->channels - 1)) {
1739
        case 0: tmp = 40; break;
1740
        case 1: tmp = 48; break;
1741
        case 2: tmp = 56; break;
1742
        case 3: tmp = 72; break;
1743
        case 4: tmp = 80; break;
1744
2
        case 5: tmp = 100;break;
1745
        default: tmp=s->sub_sampling; break;
1746
    }
1747
2
    tmp_val = 0;
1748
2
    if ((tmp * 1000) < avctx->bit_rate)  tmp_val = 1;
1749
2
    if ((tmp * 1440) < avctx->bit_rate)  tmp_val = 2;
1750
2
    if ((tmp * 1760) < avctx->bit_rate)  tmp_val = 3;
1751
2
    if ((tmp * 2240) < avctx->bit_rate)  tmp_val = 4;
1752
2
    s->cm_table_select = tmp_val;
1753
1754
2
    if (avctx->bit_rate <= 8000)
1755
        s->coeff_per_sb_select = 0;
1756
2
    else if (avctx->bit_rate < 16000)
1757
        s->coeff_per_sb_select = 1;
1758
    else
1759
2
        s->coeff_per_sb_select = 2;
1760
1761
2
    if (s->fft_size != (1 << (s->fft_order - 1))) {
1762
        av_log(avctx, AV_LOG_ERROR, "FFT size %d not power of 2.\n", s->fft_size);
1763
        return AVERROR_INVALIDDATA;
1764
    }
1765
1766
2
    ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R);
1767
2
    ff_mpadsp_init(&s->mpadsp);
1768
1769
2
    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
1770
1771
2
    return 0;
1772
}
1773
1774
2
static av_cold int qdm2_decode_close(AVCodecContext *avctx)
1775
{
1776
2
    QDM2Context *s = avctx->priv_data;
1777
1778
2
    ff_rdft_end(&s->rdft_ctx);
1779
1780
2
    return 0;
1781
}
1782
1783
2208
static int qdm2_decode(QDM2Context *q, const uint8_t *in, int16_t *out)
1784
{
1785
    int ch, i;
1786
2208
    const int frame_size = (q->frame_size * q->channels);
1787
1788
2208
    if((unsigned)frame_size > FF_ARRAY_ELEMS(q->output_buffer)/2)
1789
        return -1;
1790
1791
    /* select input buffer */
1792
2208
    q->compressed_data = in;
1793
2208
    q->compressed_size = q->checksum_size;
1794
1795
    /* copy old block, clear new block of output samples */
1796
2208
    memmove(q->output_buffer, &q->output_buffer[frame_size], frame_size * sizeof(float));
1797
2208
    memset(&q->output_buffer[frame_size], 0, frame_size * sizeof(float));
1798
1799
    /* decode block of QDM2 compressed data */
1800
2208
    if (q->sub_packet == 0) {
1801
138
        q->has_errors = 0; // zero it for a new super block
1802
138
        av_log(NULL,AV_LOG_DEBUG,"Superblock follows\n");
1803
138
        qdm2_decode_super_block(q);
1804
    }
1805
1806
    /* parse subpackets */
1807
2208
    if (!q->has_errors) {
1808
2208
        if (q->sub_packet == 2)
1809
138
            qdm2_decode_fft_packets(q);
1810
1811
2208
        qdm2_fft_tone_synthesizer(q, q->sub_packet);
1812
    }
1813
1814
    /* sound synthesis stage 1 (FFT) */
1815
6624
    for (ch = 0; ch < q->channels; ch++) {
1816
4416
        qdm2_calculate_fft(q, ch, q->sub_packet);
1817
1818

4416
        if (!q->has_errors && q->sub_packet_list_C[0].packet) {
1819
            SAMPLES_NEEDED_2("has errors, and C list is not empty")
1820
            return -1;
1821
        }
1822
    }
1823
1824
    /* sound synthesis stage 2 (MPEG audio like synthesis filter) */
1825

2208
    if (!q->has_errors && q->do_synth_filter)
1826
2208
        qdm2_synthesis_filter(q, q->sub_packet);
1827
1828
2208
    q->sub_packet = (q->sub_packet + 1) % 16;
1829
1830
    /* clip and convert output float[] to 16-bit signed samples */
1831
1132704
    for (i = 0; i < frame_size; i++) {
1832
1130496
        int value = (int)q->output_buffer[i];
1833
1834
1130496
        if (value > SOFTCLIP_THRESHOLD)
1835
245
            value = (value >  HARDCLIP_THRESHOLD) ?  32767 :  softclip_table[ value - SOFTCLIP_THRESHOLD];
1836
1130251
        else if (value < -SOFTCLIP_THRESHOLD)
1837
839
            value = (value < -HARDCLIP_THRESHOLD) ? -32767 : -softclip_table[-value - SOFTCLIP_THRESHOLD];
1838
1839
1130496
        out[i] = value;
1840
    }
1841
1842
2208
    return 0;
1843
}
1844
1845
138
static int qdm2_decode_frame(AVCodecContext *avctx, void *data,
1846
                             int *got_frame_ptr, AVPacket *avpkt)
1847
{
1848
138
    AVFrame *frame     = data;
1849
138
    const uint8_t *buf = avpkt->data;
1850
138
    int buf_size = avpkt->size;
1851
138
    QDM2Context *s = avctx->priv_data;
1852
    int16_t *out;
1853
    int i, ret;
1854
1855
138
    if(!buf)
1856
        return 0;
1857
138
    if(buf_size < s->checksum_size)
1858
        return -1;
1859
1860
    /* get output buffer */
1861
138
    frame->nb_samples = 16 * s->frame_size;
1862
138
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
1863
        return ret;
1864
138
    out = (int16_t *)frame->data[0];
1865
1866
2346
    for (i = 0; i < 16; i++) {
1867
2208
        if ((ret = qdm2_decode(s, buf, out)) < 0)
1868
            return ret;
1869
2208
        out += s->channels * s->frame_size;
1870
    }
1871
1872
138
    *got_frame_ptr = 1;
1873
1874
138
    return s->checksum_size;
1875
}
1876
1877
AVCodec ff_qdm2_decoder = {
1878
    .name             = "qdm2",
1879
    .long_name        = NULL_IF_CONFIG_SMALL("QDesign Music Codec 2"),
1880
    .type             = AVMEDIA_TYPE_AUDIO,
1881
    .id               = AV_CODEC_ID_QDM2,
1882
    .priv_data_size   = sizeof(QDM2Context),
1883
    .init             = qdm2_decode_init,
1884
    .close            = qdm2_decode_close,
1885
    .decode           = qdm2_decode_frame,
1886
    .capabilities     = AV_CODEC_CAP_DR1,
1887
};