GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/cook.c Lines: 440 538 81.8 %
Date: 2021-04-20 15:25:36 Branches: 196 273 71.8 %

Line Branch Exec Source
1
/*
2
 * COOK compatible decoder
3
 * Copyright (c) 2003 Sascha Sommer
4
 * Copyright (c) 2005 Benjamin Larsson
5
 *
6
 * This file is part of FFmpeg.
7
 *
8
 * FFmpeg 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 2.1 of the License, or (at your option) any later version.
12
 *
13
 * FFmpeg 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 FFmpeg; if not, write to the Free Software
20
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21
 */
22
23
/**
24
 * @file
25
 * Cook compatible decoder. Bastardization of the G.722.1 standard.
26
 * This decoder handles RealNetworks, RealAudio G2 data.
27
 * Cook is identified by the codec name cook in RM files.
28
 *
29
 * To use this decoder, a calling application must supply the extradata
30
 * bytes provided from the RM container; 8+ bytes for mono streams and
31
 * 16+ for stereo streams (maybe more).
32
 *
33
 * Codec technicalities (all this assume a buffer length of 1024):
34
 * Cook works with several different techniques to achieve its compression.
35
 * In the timedomain the buffer is divided into 8 pieces and quantized. If
36
 * two neighboring pieces have different quantization index a smooth
37
 * quantization curve is used to get a smooth overlap between the different
38
 * pieces.
39
 * To get to the transformdomain Cook uses a modulated lapped transform.
40
 * The transform domain has 50 subbands with 20 elements each. This
41
 * means only a maximum of 50*20=1000 coefficients are used out of the 1024
42
 * available.
43
 */
44
45
#include "libavutil/channel_layout.h"
46
#include "libavutil/lfg.h"
47
#include "libavutil/mem_internal.h"
48
49
#include "audiodsp.h"
50
#include "avcodec.h"
51
#include "get_bits.h"
52
#include "bytestream.h"
53
#include "fft.h"
54
#include "internal.h"
55
#include "sinewin.h"
56
#include "unary.h"
57
58
#include "cookdata.h"
59
60
/* the different Cook versions */
61
#define MONO            0x1000001
62
#define STEREO          0x1000002
63
#define JOINT_STEREO    0x1000003
64
#define MC_COOK         0x2000000
65
66
#define SUBBAND_SIZE    20
67
#define MAX_SUBPACKETS   5
68
69
#define QUANT_VLC_BITS    9
70
#define COUPLING_VLC_BITS 6
71
72
typedef struct cook_gains {
73
    int *now;
74
    int *previous;
75
} cook_gains;
76
77
typedef struct COOKSubpacket {
78
    int                 ch_idx;
79
    int                 size;
80
    int                 num_channels;
81
    int                 cookversion;
82
    int                 subbands;
83
    int                 js_subband_start;
84
    int                 js_vlc_bits;
85
    int                 samples_per_channel;
86
    int                 log2_numvector_size;
87
    unsigned int        channel_mask;
88
    VLC                 channel_coupling;
89
    int                 joint_stereo;
90
    int                 bits_per_subpacket;
91
    int                 bits_per_subpdiv;
92
    int                 total_subbands;
93
    int                 numvector_size;       // 1 << log2_numvector_size;
94
95
    float               mono_previous_buffer1[1024];
96
    float               mono_previous_buffer2[1024];
97
98
    cook_gains          gains1;
99
    cook_gains          gains2;
100
    int                 gain_1[9];
101
    int                 gain_2[9];
102
    int                 gain_3[9];
103
    int                 gain_4[9];
104
} COOKSubpacket;
105
106
typedef struct cook {
107
    /*
108
     * The following 5 functions provide the lowlevel arithmetic on
109
     * the internal audio buffers.
110
     */
111
    void (*scalar_dequant)(struct cook *q, int index, int quant_index,
112
                           int *subband_coef_index, int *subband_coef_sign,
113
                           float *mlt_p);
114
115
    void (*decouple)(struct cook *q,
116
                     COOKSubpacket *p,
117
                     int subband,
118
                     float f1, float f2,
119
                     float *decode_buffer,
120
                     float *mlt_buffer1, float *mlt_buffer2);
121
122
    void (*imlt_window)(struct cook *q, float *buffer1,
123
                        cook_gains *gains_ptr, float *previous_buffer);
124
125
    void (*interpolate)(struct cook *q, float *buffer,
126
                        int gain_index, int gain_index_next);
127
128
    void (*saturate_output)(struct cook *q, float *out);
129
130
    AVCodecContext*     avctx;
131
    AudioDSPContext     adsp;
132
    GetBitContext       gb;
133
    /* stream data */
134
    int                 num_vectors;
135
    int                 samples_per_channel;
136
    /* states */
137
    AVLFG               random_state;
138
    int                 discarded_packets;
139
140
    /* transform data */
141
    FFTContext          mdct_ctx;
142
    float*              mlt_window;
143
144
    /* VLC data */
145
    VLC                 envelope_quant_index[13];
146
    VLC                 sqvh[7];          // scalar quantization
147
148
    /* generate tables and related variables */
149
    int                 gain_size_factor;
150
    float               gain_table[31];
151
152
    /* data buffers */
153
154
    uint8_t*            decoded_bytes_buffer;
155
    DECLARE_ALIGNED(32, float, mono_mdct_output)[2048];
156
    float               decode_buffer_1[1024];
157
    float               decode_buffer_2[1024];
158
    float               decode_buffer_0[1060]; /* static allocation for joint decode */
159
160
    const float         *cplscales[5];
161
    int                 num_subpackets;
162
    COOKSubpacket       subpacket[MAX_SUBPACKETS];
163
} COOKContext;
164
165
static float     pow2tab[127];
166
static float rootpow2tab[127];
167
168
/*************** init functions ***************/
169
170
/* table generator */
171
6
static av_cold void init_pow2table(void)
172
{
173
    /* fast way of computing 2^i and 2^(0.5*i) for -63 <= i < 64 */
174
    int i;
175
    static const float exp2_tab[2] = {1, M_SQRT2};
176
6
    float exp2_val = powf(2, -63);
177
6
    float root_val = powf(2, -32);
178
768
    for (i = -63; i < 64; i++) {
179
762
        if (!(i & 1))
180
378
            root_val *= 2;
181
762
        pow2tab[63 + i] = exp2_val;
182
762
        rootpow2tab[63 + i] = root_val * exp2_tab[i & 1];
183
762
        exp2_val *= 2;
184
    }
185
6
}
186
187
/* table generator */
188
6
static av_cold void init_gain_table(COOKContext *q)
189
{
190
    int i;
191
6
    q->gain_size_factor = q->samples_per_channel / 8;
192
192
    for (i = 0; i < 31; i++)
193
186
        q->gain_table[i] = pow(pow2tab[i + 48],
194
186
                               (1.0 / (double) q->gain_size_factor));
195
6
}
196
197
124
static av_cold int build_vlc(VLC *vlc, int nb_bits, const uint8_t counts[16],
198
                             const void *syms, int symbol_size, int offset,
199
                             void *logctx)
200
{
201
    uint8_t lens[MAX_COOK_VLC_ENTRIES];
202
124
    unsigned num = 0;
203
204
2108
    for (int i = 0; i < 16; i++)
205
11636
        for (unsigned count = num + counts[i]; num < count; num++)
206
9652
            lens[num] = i + 1;
207
208
124
    return ff_init_vlc_from_lengths(vlc, nb_bits, num, lens, 1,
209
                                    syms, symbol_size, symbol_size,
210
                                    offset, 0, logctx);
211
}
212
213
6
static av_cold int init_cook_vlc_tables(COOKContext *q)
214
{
215
    int i, result;
216
217
6
    result = 0;
218
84
    for (i = 0; i < 13; i++) {
219
78
        result |= build_vlc(&q->envelope_quant_index[i], QUANT_VLC_BITS,
220
78
                            envelope_quant_index_huffcounts[i],
221
78
                            envelope_quant_index_huffsyms[i], 1, -12, q->avctx);
222
    }
223
6
    av_log(q->avctx, AV_LOG_DEBUG, "sqvh VLC init\n");
224
48
    for (i = 0; i < 7; i++) {
225
42
        int sym_size = 1 + (i == 3);
226
42
        result |= build_vlc(&q->sqvh[i], vhvlcsize_tab[i],
227
42
                            cvh_huffcounts[i],
228
42
                            cvh_huffsyms[i], sym_size, 0, q->avctx);
229
    }
230
231
12
    for (i = 0; i < q->num_subpackets; i++) {
232
6
        if (q->subpacket[i].joint_stereo == 1) {
233
8
            result |= build_vlc(&q->subpacket[i].channel_coupling, COUPLING_VLC_BITS,
234
4
                                ccpl_huffcounts[q->subpacket[i].js_vlc_bits - 2],
235
4
                                ccpl_huffsyms[q->subpacket[i].js_vlc_bits - 2], 1,
236
4
                                0, q->avctx);
237
4
            av_log(q->avctx, AV_LOG_DEBUG, "subpacket %i Joint-stereo VLC used.\n", i);
238
        }
239
    }
240
241
6
    av_log(q->avctx, AV_LOG_DEBUG, "VLC tables initialized.\n");
242
6
    return result;
243
}
244
245
6
static av_cold int init_cook_mlt(COOKContext *q)
246
{
247
    int j, ret;
248
6
    int mlt_size = q->samples_per_channel;
249
250
6
    if ((q->mlt_window = av_malloc_array(mlt_size, sizeof(*q->mlt_window))) == 0)
251
        return AVERROR(ENOMEM);
252
253
    /* Initialize the MLT window: simple sine window. */
254
6
    ff_sine_window_init(q->mlt_window, mlt_size);
255
6150
    for (j = 0; j < mlt_size; j++)
256
6144
        q->mlt_window[j] *= sqrt(2.0 / q->samples_per_channel);
257
258
    /* Initialize the MDCT. */
259
6
    if ((ret = ff_mdct_init(&q->mdct_ctx, av_log2(mlt_size) + 1, 1, 1.0 / 32768.0))) {
260
        av_freep(&q->mlt_window);
261
        return ret;
262
    }
263
6
    av_log(q->avctx, AV_LOG_DEBUG, "MDCT initialized, order = %d.\n",
264
6
           av_log2(mlt_size) + 1);
265
266
6
    return 0;
267
}
268
269
6
static av_cold void init_cplscales_table(COOKContext *q)
270
{
271
    int i;
272
36
    for (i = 0; i < 5; i++)
273
30
        q->cplscales[i] = cplscales[i];
274
6
}
275
276
/*************** init functions end ***********/
277
278
#define DECODE_BYTES_PAD1(bytes) (3 - ((bytes) + 3) % 4)
279
#define DECODE_BYTES_PAD2(bytes) ((bytes) % 4 + DECODE_BYTES_PAD1(2 * (bytes)))
280
281
/**
282
 * Cook indata decoding, every 32 bits are XORed with 0x37c511f2.
283
 * Why? No idea, some checksum/error detection method maybe.
284
 *
285
 * Out buffer size: extra bytes are needed to cope with
286
 * padding/misalignment.
287
 * Subpackets passed to the decoder can contain two, consecutive
288
 * half-subpackets, of identical but arbitrary size.
289
 *          1234 1234 1234 1234  extraA extraB
290
 * Case 1:  AAAA BBBB              0      0
291
 * Case 2:  AAAA ABBB BB--         3      3
292
 * Case 3:  AAAA AABB BBBB         2      2
293
 * Case 4:  AAAA AAAB BBBB BB--    1      5
294
 *
295
 * Nice way to waste CPU cycles.
296
 *
297
 * @param inbuffer  pointer to byte array of indata
298
 * @param out       pointer to byte array of outdata
299
 * @param bytes     number of bytes
300
 */
301
240
static inline int decode_bytes(const uint8_t *inbuffer, uint8_t *out, int bytes)
302
{
303
    static const uint32_t tab[4] = {
304
        AV_BE2NE32C(0x37c511f2u), AV_BE2NE32C(0xf237c511u),
305
        AV_BE2NE32C(0x11f237c5u), AV_BE2NE32C(0xc511f237u),
306
    };
307
    int i, off;
308
    uint32_t c;
309
    const uint32_t *buf;
310
240
    uint32_t *obuf = (uint32_t *) out;
311
    /* FIXME: 64 bit platforms would be able to do 64 bits at a time.
312
     * I'm too lazy though, should be something like
313
     * for (i = 0; i < bitamount / 64; i++)
314
     *     (int64_t) out[i] = 0x37c511f237c511f2 ^ av_be2ne64(int64_t) in[i]);
315
     * Buffer alignment needs to be checked. */
316
317
240
    off = (intptr_t) inbuffer & 3;
318
240
    buf = (const uint32_t *) (inbuffer - off);
319
240
    c = tab[off];
320
240
    bytes += 3 + off;
321
11520
    for (i = 0; i < bytes / 4; i++)
322
11280
        obuf[i] = c ^ buf[i];
323
324
240
    return off;
325
}
326
327
6
static av_cold int cook_decode_close(AVCodecContext *avctx)
328
{
329
    int i;
330
6
    COOKContext *q = avctx->priv_data;
331
6
    av_log(avctx, AV_LOG_DEBUG, "Deallocating memory.\n");
332
333
    /* Free allocated memory buffers. */
334
6
    av_freep(&q->mlt_window);
335
6
    av_freep(&q->decoded_bytes_buffer);
336
337
    /* Free the transform. */
338
6
    ff_mdct_end(&q->mdct_ctx);
339
340
    /* Free the VLC tables. */
341
84
    for (i = 0; i < 13; i++)
342
78
        ff_free_vlc(&q->envelope_quant_index[i]);
343
48
    for (i = 0; i < 7; i++)
344
42
        ff_free_vlc(&q->sqvh[i]);
345
12
    for (i = 0; i < q->num_subpackets; i++)
346
6
        ff_free_vlc(&q->subpacket[i].channel_coupling);
347
348
6
    av_log(avctx, AV_LOG_DEBUG, "Memory deallocated.\n");
349
350
6
    return 0;
351
}
352
353
/**
354
 * Fill the gain array for the timedomain quantization.
355
 *
356
 * @param gb          pointer to the GetBitContext
357
 * @param gaininfo    array[9] of gain indexes
358
 */
359
240
static void decode_gain_info(GetBitContext *gb, int *gaininfo)
360
{
361
    int i, n;
362
363
240
    n = get_unary(gb, 0, get_bits_left(gb));     // amount of elements*2 to update
364
365
240
    i = 0;
366
241
    while (n--) {
367
1
        int index = get_bits(gb, 3);
368
1
        int gain = get_bits1(gb) ? get_bits(gb, 4) - 7 : -1;
369
370
8
        while (i <= index)
371
7
            gaininfo[i++] = gain;
372
    }
373
2393
    while (i <= 8)
374
2153
        gaininfo[i++] = 0;
375
240
}
376
377
/**
378
 * Create the quant index table needed for the envelope.
379
 *
380
 * @param q                 pointer to the COOKContext
381
 * @param quant_index_table pointer to the array
382
 */
383
240
static int decode_envelope(COOKContext *q, COOKSubpacket *p,
384
                           int *quant_index_table)
385
{
386
    int i, j, vlc_index;
387
388
240
    quant_index_table[0] = get_bits(&q->gb, 6) - 6; // This is used later in categorize
389
390
10320
    for (i = 1; i < p->total_subbands; i++) {
391
10080
        vlc_index = i;
392
10080
        if (i >= p->js_subband_start * 2) {
393
7440
            vlc_index -= p->js_subband_start;
394
        } else {
395
2640
            vlc_index /= 2;
396
2640
            if (vlc_index < 1)
397
240
                vlc_index = 1;
398
        }
399
10080
        if (vlc_index > 13)
400
5520
            vlc_index = 13; // the VLC tables >13 are identical to No. 13
401
402
10080
        j = get_vlc2(&q->gb, q->envelope_quant_index[vlc_index - 1].table,
403
                     QUANT_VLC_BITS, 2);
404
10080
        quant_index_table[i] = quant_index_table[i - 1] + j; // differential encoding
405

10080
        if (quant_index_table[i] > 63 || quant_index_table[i] < -63) {
406
            av_log(q->avctx, AV_LOG_ERROR,
407
                   "Invalid quantizer %d at position %d, outside [-63, 63] range\n",
408
                   quant_index_table[i], i);
409
            return AVERROR_INVALIDDATA;
410
        }
411
    }
412
413
240
    return 0;
414
}
415
416
/**
417
 * Calculate the category and category_index vector.
418
 *
419
 * @param q                     pointer to the COOKContext
420
 * @param quant_index_table     pointer to the array
421
 * @param category              pointer to the category array
422
 * @param category_index        pointer to the category_index array
423
 */
424
240
static void categorize(COOKContext *q, COOKSubpacket *p, const int *quant_index_table,
425
                       int *category, int *category_index)
426
{
427
    int exp_idx, bias, tmpbias1, tmpbias2, bits_left, num_bits, index, v, i, j;
428
240
    int exp_index2[102] = { 0 };
429
240
    int exp_index1[102] = { 0 };
430
431
240
    int tmp_categorize_array[128 * 2] = { 0 };
432
240
    int tmp_categorize_array1_idx = p->numvector_size;
433
240
    int tmp_categorize_array2_idx = p->numvector_size;
434
435
240
    bits_left = p->bits_per_subpacket - get_bits_count(&q->gb);
436
437
240
    if (bits_left > q->samples_per_channel)
438
240
        bits_left = q->samples_per_channel +
439
240
                    ((bits_left - q->samples_per_channel) * 5) / 8;
440
441
240
    bias = -32;
442
443
    /* Estimate bias. */
444
1680
    for (i = 32; i > 0; i = i / 2) {
445
1440
        num_bits = 0;
446
1440
        index    = 0;
447
63360
        for (j = p->total_subbands; j > 0; j--) {
448
61920
            exp_idx = av_clip_uintp2((i - quant_index_table[index] + bias) / 2, 3);
449
61920
            index++;
450
61920
            num_bits += expbits_tab[exp_idx];
451
        }
452
1440
        if (num_bits >= bits_left - 32)
453
1333
            bias += i;
454
    }
455
456
    /* Calculate total number of bits. */
457
240
    num_bits = 0;
458
10560
    for (i = 0; i < p->total_subbands; i++) {
459
10320
        exp_idx = av_clip_uintp2((bias - quant_index_table[i]) / 2, 3);
460
10320
        num_bits += expbits_tab[exp_idx];
461
10320
        exp_index1[i] = exp_idx;
462
10320
        exp_index2[i] = exp_idx;
463
    }
464
240
    tmpbias1 = tmpbias2 = num_bits;
465
466
30720
    for (j = 1; j < p->numvector_size; j++) {
467
30480
        if (tmpbias1 + tmpbias2 > 2 * bits_left) {  /* ---> */
468
16408
            int max = -999999;
469
16408
            index = -1;
470
721952
            for (i = 0; i < p->total_subbands; i++) {
471
705544
                if (exp_index1[i] < 7) {
472
543313
                    v = (-2 * exp_index1[i]) - quant_index_table[i] + bias;
473
543313
                    if (v >= max) {
474
186262
                        max   = v;
475
186262
                        index = i;
476
                    }
477
                }
478
            }
479
16408
            if (index == -1)
480
                break;
481
16408
            tmp_categorize_array[tmp_categorize_array1_idx++] = index;
482
16408
            tmpbias1 -= expbits_tab[exp_index1[index]] -
483
16408
                        expbits_tab[exp_index1[index] + 1];
484
16408
            ++exp_index1[index];
485
        } else {  /* <--- */
486
14072
            int min = 999999;
487
14072
            index = -1;
488
619168
            for (i = 0; i < p->total_subbands; i++) {
489
605096
                if (exp_index2[i] > 0) {
490
527917
                    v = (-2 * exp_index2[i]) - quant_index_table[i] + bias;
491
527917
                    if (v < min) {
492
31097
                        min   = v;
493
31097
                        index = i;
494
                    }
495
                }
496
            }
497
14072
            if (index == -1)
498
                break;
499
14072
            tmp_categorize_array[--tmp_categorize_array2_idx] = index;
500
14072
            tmpbias2 -= expbits_tab[exp_index2[index]] -
501
14072
                        expbits_tab[exp_index2[index] - 1];
502
14072
            --exp_index2[index];
503
        }
504
    }
505
506
10560
    for (i = 0; i < p->total_subbands; i++)
507
10320
        category[i] = exp_index2[i];
508
509
30720
    for (i = 0; i < p->numvector_size - 1; i++)
510
30480
        category_index[i] = tmp_categorize_array[tmp_categorize_array2_idx++];
511
240
}
512
513
514
/**
515
 * Expand the category vector.
516
 *
517
 * @param q                     pointer to the COOKContext
518
 * @param category              pointer to the category array
519
 * @param category_index        pointer to the category_index array
520
 */
521
240
static inline void expand_category(COOKContext *q, int *category,
522
                                   int *category_index)
523
{
524
    int i;
525
15141
    for (i = 0; i < q->num_vectors; i++)
526
    {
527
14901
        int idx = category_index[i];
528
14901
        if (++category[idx] >= FF_ARRAY_ELEMS(dither_tab))
529
            --category[idx];
530
    }
531
240
}
532
533
/**
534
 * The real requantization of the mltcoefs
535
 *
536
 * @param q                     pointer to the COOKContext
537
 * @param index                 index
538
 * @param quant_index           quantisation index
539
 * @param subband_coef_index    array of indexes to quant_centroid_tab
540
 * @param subband_coef_sign     signs of coefficients
541
 * @param mlt_p                 pointer into the mlt buffer
542
 */
543
10320
static void scalar_dequant_float(COOKContext *q, int index, int quant_index,
544
                                 int *subband_coef_index, int *subband_coef_sign,
545
                                 float *mlt_p)
546
{
547
    int i;
548
    float f1;
549
550
216720
    for (i = 0; i < SUBBAND_SIZE; i++) {
551
206400
        if (subband_coef_index[i]) {
552
67248
            f1 = quant_centroid_tab[index][subband_coef_index[i]];
553
67248
            if (subband_coef_sign[i])
554
33713
                f1 = -f1;
555
        } else {
556
            /* noise coding if subband_coef_index[i] == 0 */
557
139152
            f1 = dither_tab[index];
558
139152
            if (av_lfg_get(&q->random_state) < 0x80000000)
559
69511
                f1 = -f1;
560
        }
561
206400
        mlt_p[i] = f1 * rootpow2tab[quant_index + 63];
562
    }
563
10320
}
564
/**
565
 * Unpack the subband_coef_index and subband_coef_sign vectors.
566
 *
567
 * @param q                     pointer to the COOKContext
568
 * @param category              pointer to the category array
569
 * @param subband_coef_index    array of indexes to quant_centroid_tab
570
 * @param subband_coef_sign     signs of coefficients
571
 */
572
8969
static int unpack_SQVH(COOKContext *q, COOKSubpacket *p, int category,
573
                       int *subband_coef_index, int *subband_coef_sign)
574
{
575
    int i, j;
576
    int vlc, vd, tmp, result;
577
578
8969
    vd = vd_tab[category];
579
8969
    result = 0;
580
65444
    for (i = 0; i < vpr_tab[category]; i++) {
581
56475
        vlc = get_vlc2(&q->gb, q->sqvh[category].table, q->sqvh[category].bits, 3);
582
56475
        if (p->bits_per_subpacket < get_bits_count(&q->gb)) {
583
            vlc = 0;
584
            result = 1;
585
        }
586
235855
        for (j = vd - 1; j >= 0; j--) {
587
179380
            tmp = (vlc * invradix_tab[category]) / 0x100000;
588
179380
            subband_coef_index[vd * i + j] = vlc - tmp * (kmax_tab[category] + 1);
589
179380
            vlc = tmp;
590
        }
591
235855
        for (j = 0; j < vd; j++) {
592
179380
            if (subband_coef_index[i * vd + j]) {
593
67248
                if (get_bits_count(&q->gb) < p->bits_per_subpacket) {
594
67248
                    subband_coef_sign[i * vd + j] = get_bits1(&q->gb);
595
                } else {
596
                    result = 1;
597
                    subband_coef_sign[i * vd + j] = 0;
598
                }
599
            } else {
600
112132
                subband_coef_sign[i * vd + j] = 0;
601
            }
602
        }
603
    }
604
8969
    return result;
605
}
606
607
608
/**
609
 * Fill the mlt_buffer with mlt coefficients.
610
 *
611
 * @param q                 pointer to the COOKContext
612
 * @param category          pointer to the category array
613
 * @param quant_index_table pointer to the array
614
 * @param mlt_buffer        pointer to mlt coefficients
615
 */
616
240
static void decode_vectors(COOKContext *q, COOKSubpacket *p, int *category,
617
                           int *quant_index_table, float *mlt_buffer)
618
{
619
    /* A zero in this table means that the subband coefficient is
620
       random noise coded. */
621
    int subband_coef_index[SUBBAND_SIZE];
622
    /* A zero in this table means that the subband coefficient is a
623
       positive multiplicator. */
624
    int subband_coef_sign[SUBBAND_SIZE];
625
    int band, j;
626
240
    int index = 0;
627
628
10560
    for (band = 0; band < p->total_subbands; band++) {
629
10320
        index = category[band];
630
10320
        if (category[band] < 7) {
631
8969
            if (unpack_SQVH(q, p, category[band], subband_coef_index, subband_coef_sign)) {
632
                index = 7;
633
                for (j = 0; j < p->total_subbands; j++)
634
                    category[band + j] = 7;
635
            }
636
        }
637
10320
        if (index >= 7) {
638
1351
            memset(subband_coef_index, 0, sizeof(subband_coef_index));
639
1351
            memset(subband_coef_sign,  0, sizeof(subband_coef_sign));
640
        }
641
10320
        q->scalar_dequant(q, index, quant_index_table[band],
642
                          subband_coef_index, subband_coef_sign,
643
10320
                          &mlt_buffer[band * SUBBAND_SIZE]);
644
    }
645
646
    /* FIXME: should this be removed, or moved into loop above? */
647
240
    if (p->total_subbands * SUBBAND_SIZE >= q->samples_per_channel)
648
        return;
649
}
650
651
652
240
static int mono_decode(COOKContext *q, COOKSubpacket *p, float *mlt_buffer)
653
{
654
240
    int category_index[128] = { 0 };
655
240
    int category[128]       = { 0 };
656
    int quant_index_table[102];
657
    int res, i;
658
659
240
    if ((res = decode_envelope(q, p, quant_index_table)) < 0)
660
        return res;
661
240
    q->num_vectors = get_bits(&q->gb, p->log2_numvector_size);
662
240
    categorize(q, p, quant_index_table, category, category_index);
663
240
    expand_category(q, category, category_index);
664
10560
    for (i=0; i<p->total_subbands; i++) {
665
10320
        if (category[i] > 7)
666
            return AVERROR_INVALIDDATA;
667
    }
668
240
    decode_vectors(q, p, category, quant_index_table, mlt_buffer);
669
670
240
    return 0;
671
}
672
673
674
/**
675
 * the actual requantization of the timedomain samples
676
 *
677
 * @param q                 pointer to the COOKContext
678
 * @param buffer            pointer to the timedomain buffer
679
 * @param gain_index        index for the block multiplier
680
 * @param gain_index_next   index for the next block multiplier
681
 */
682
14
static void interpolate_float(COOKContext *q, float *buffer,
683
                              int gain_index, int gain_index_next)
684
{
685
    int i;
686
    float fc1, fc2;
687
14
    fc1 = pow2tab[gain_index + 63];
688
689
14
    if (gain_index == gain_index_next) {             // static gain
690
1548
        for (i = 0; i < q->gain_size_factor; i++)
691
1536
            buffer[i] *= fc1;
692
    } else {                                        // smooth gain
693
2
        fc2 = q->gain_table[15 + (gain_index_next - gain_index)];
694
258
        for (i = 0; i < q->gain_size_factor; i++) {
695
256
            buffer[i] *= fc1;
696
256
            fc1       *= fc2;
697
        }
698
    }
699
14
}
700
701
/**
702
 * Apply transform window, overlap buffers.
703
 *
704
 * @param q                 pointer to the COOKContext
705
 * @param inbuffer          pointer to the mltcoefficients
706
 * @param gains_ptr         current and previous gains
707
 * @param previous_buffer   pointer to the previous buffer to be used for overlapping
708
 */
709
480
static void imlt_window_float(COOKContext *q, float *inbuffer,
710
                              cook_gains *gains_ptr, float *previous_buffer)
711
{
712
480
    const float fc = pow2tab[gains_ptr->previous[0] + 63];
713
    int i;
714
    /* The weird thing here, is that the two halves of the time domain
715
     * buffer are swapped. Also, the newest data, that we save away for
716
     * next frame, has the wrong sign. Hence the subtraction below.
717
     * Almost sounds like a complex conjugate/reverse data/FFT effect.
718
     */
719
720
    /* Apply window and overlap */
721
492000
    for (i = 0; i < q->samples_per_channel; i++)
722
491520
        inbuffer[i] = inbuffer[i] * fc * q->mlt_window[i] -
723
491520
                      previous_buffer[i] * q->mlt_window[q->samples_per_channel - 1 - i];
724
480
}
725
726
/**
727
 * The modulated lapped transform, this takes transform coefficients
728
 * and transforms them into timedomain samples.
729
 * Apply transform window, overlap buffers, apply gain profile
730
 * and buffer management.
731
 *
732
 * @param q                 pointer to the COOKContext
733
 * @param inbuffer          pointer to the mltcoefficients
734
 * @param gains_ptr         current and previous gains
735
 * @param previous_buffer   pointer to the previous buffer to be used for overlapping
736
 */
737
480
static void imlt_gain(COOKContext *q, float *inbuffer,
738
                      cook_gains *gains_ptr, float *previous_buffer)
739
{
740
480
    float *buffer0 = q->mono_mdct_output;
741
480
    float *buffer1 = q->mono_mdct_output + q->samples_per_channel;
742
    int i;
743
744
    /* Inverse modified discrete cosine transform */
745
480
    q->mdct_ctx.imdct_calc(&q->mdct_ctx, q->mono_mdct_output, inbuffer);
746
747
480
    q->imlt_window(q, buffer1, gains_ptr, previous_buffer);
748
749
    /* Apply gain profile */
750
4320
    for (i = 0; i < 8; i++)
751

3840
        if (gains_ptr->now[i] || gains_ptr->now[i + 1])
752
14
            q->interpolate(q, &buffer1[q->gain_size_factor * i],
753
14
                           gains_ptr->now[i], gains_ptr->now[i + 1]);
754
755
    /* Save away the current to be previous block. */
756
480
    memcpy(previous_buffer, buffer0,
757
480
           q->samples_per_channel * sizeof(*previous_buffer));
758
480
}
759
760
761
/**
762
 * function for getting the jointstereo coupling information
763
 *
764
 * @param q                 pointer to the COOKContext
765
 * @param decouple_tab      decoupling array
766
 */
767
240
static int decouple_info(COOKContext *q, COOKSubpacket *p, int *decouple_tab)
768
{
769
    int i;
770
240
    int vlc    = get_bits1(&q->gb);
771
240
    int start  = cplband[p->js_subband_start];
772
240
    int end    = cplband[p->subbands - 1];
773
240
    int length = end - start + 1;
774
775
240
    if (start > end)
776
        return 0;
777
778
240
    if (vlc)
779
1806
        for (i = 0; i < length; i++)
780
1677
            decouple_tab[start + i] = get_vlc2(&q->gb,
781
                                               p->channel_coupling.table,
782
                                               COUPLING_VLC_BITS, 3);
783
    else
784
1554
        for (i = 0; i < length; i++) {
785
1443
            int v = get_bits(&q->gb, p->js_vlc_bits);
786
1443
            if (v == (1<<p->js_vlc_bits)-1) {
787
                av_log(q->avctx, AV_LOG_ERROR, "decouple value too large\n");
788
                return AVERROR_INVALIDDATA;
789
            }
790
1443
            decouple_tab[start + i] = v;
791
        }
792
240
    return 0;
793
}
794
795
/**
796
 * function decouples a pair of signals from a single signal via multiplication.
797
 *
798
 * @param q                 pointer to the COOKContext
799
 * @param subband           index of the current subband
800
 * @param f1                multiplier for channel 1 extraction
801
 * @param f2                multiplier for channel 2 extraction
802
 * @param decode_buffer     input buffer
803
 * @param mlt_buffer1       pointer to left channel mlt coefficients
804
 * @param mlt_buffer2       pointer to right channel mlt coefficients
805
 */
806
7440
static void decouple_float(COOKContext *q,
807
                           COOKSubpacket *p,
808
                           int subband,
809
                           float f1, float f2,
810
                           float *decode_buffer,
811
                           float *mlt_buffer1, float *mlt_buffer2)
812
{
813
    int j, tmp_idx;
814
156240
    for (j = 0; j < SUBBAND_SIZE; j++) {
815
148800
        tmp_idx = ((p->js_subband_start + subband) * SUBBAND_SIZE) + j;
816
148800
        mlt_buffer1[SUBBAND_SIZE * subband + j] = f1 * decode_buffer[tmp_idx];
817
148800
        mlt_buffer2[SUBBAND_SIZE * subband + j] = f2 * decode_buffer[tmp_idx];
818
    }
819
7440
}
820
821
/**
822
 * function for decoding joint stereo data
823
 *
824
 * @param q                 pointer to the COOKContext
825
 * @param mlt_buffer1       pointer to left channel mlt coefficients
826
 * @param mlt_buffer2       pointer to right channel mlt coefficients
827
 */
828
240
static int joint_decode(COOKContext *q, COOKSubpacket *p,
829
                        float *mlt_buffer_left, float *mlt_buffer_right)
830
{
831
    int i, j, res;
832
240
    int decouple_tab[SUBBAND_SIZE] = { 0 };
833
240
    float *decode_buffer = q->decode_buffer_0;
834
    int idx, cpl_tmp;
835
    float f1, f2;
836
    const float *cplscale;
837
838
240
    memset(decode_buffer, 0, sizeof(q->decode_buffer_0));
839
840
    /* Make sure the buffers are zeroed out. */
841
240
    memset(mlt_buffer_left,  0, 1024 * sizeof(*mlt_buffer_left));
842
240
    memset(mlt_buffer_right, 0, 1024 * sizeof(*mlt_buffer_right));
843
240
    if ((res = decouple_info(q, p, decouple_tab)) < 0)
844
        return res;
845
240
    if ((res = mono_decode(q, p, decode_buffer)) < 0)
846
        return res;
847
    /* The two channels are stored interleaved in decode_buffer. */
848
1680
    for (i = 0; i < p->js_subband_start; i++) {
849
30240
        for (j = 0; j < SUBBAND_SIZE; j++) {
850
28800
            mlt_buffer_left[i  * 20 + j] = decode_buffer[i * 40 + j];
851
28800
            mlt_buffer_right[i * 20 + j] = decode_buffer[i * 40 + 20 + j];
852
        }
853
    }
854
855
    /* When we reach js_subband_start (the higher frequencies)
856
       the coefficients are stored in a coupling scheme. */
857
240
    idx = (1 << p->js_vlc_bits) - 1;
858
7680
    for (i = p->js_subband_start; i < p->subbands; i++) {
859
7440
        cpl_tmp = cplband[i];
860
7440
        idx -= decouple_tab[cpl_tmp];
861
7440
        cplscale = q->cplscales[p->js_vlc_bits - 2];  // choose decoupler table
862
7440
        f1 = cplscale[decouple_tab[cpl_tmp] + 1];
863
7440
        f2 = cplscale[idx];
864
7440
        q->decouple(q, p, i, f1, f2, decode_buffer,
865
                    mlt_buffer_left, mlt_buffer_right);
866
7440
        idx = (1 << p->js_vlc_bits) - 1;
867
    }
868
869
240
    return 0;
870
}
871
872
/**
873
 * First part of subpacket decoding:
874
 *  decode raw stream bytes and read gain info.
875
 *
876
 * @param q                 pointer to the COOKContext
877
 * @param inbuffer          pointer to raw stream data
878
 * @param gains_ptr         array of current/prev gain pointers
879
 */
880
240
static inline void decode_bytes_and_gain(COOKContext *q, COOKSubpacket *p,
881
                                         const uint8_t *inbuffer,
882
                                         cook_gains *gains_ptr)
883
{
884
    int offset;
885
886
240
    offset = decode_bytes(inbuffer, q->decoded_bytes_buffer,
887
240
                          p->bits_per_subpacket / 8);
888
240
    init_get_bits(&q->gb, q->decoded_bytes_buffer + offset,
889
                  p->bits_per_subpacket);
890
240
    decode_gain_info(&q->gb, gains_ptr->now);
891
892
    /* Swap current and previous gains */
893
240
    FFSWAP(int *, gains_ptr->now, gains_ptr->previous);
894
240
}
895
896
/**
897
 * Saturate the output signal and interleave.
898
 *
899
 * @param q                 pointer to the COOKContext
900
 * @param out               pointer to the output vector
901
 */
902
476
static void saturate_output_float(COOKContext *q, float *out)
903
{
904
476
    q->adsp.vector_clipf(out, q->mono_mdct_output + q->samples_per_channel,
905
476
                         FFALIGN(q->samples_per_channel, 8), -1.0f, 1.0f);
906
476
}
907
908
909
/**
910
 * Final part of subpacket decoding:
911
 *  Apply modulated lapped transform, gain compensation,
912
 *  clip and convert to integer.
913
 *
914
 * @param q                 pointer to the COOKContext
915
 * @param decode_buffer     pointer to the mlt coefficients
916
 * @param gains_ptr         array of current/prev gain pointers
917
 * @param previous_buffer   pointer to the previous buffer to be used for overlapping
918
 * @param out               pointer to the output buffer
919
 */
920
480
static inline void mlt_compensate_output(COOKContext *q, float *decode_buffer,
921
                                         cook_gains *gains_ptr, float *previous_buffer,
922
                                         float *out)
923
{
924
480
    imlt_gain(q, decode_buffer, gains_ptr, previous_buffer);
925
480
    if (out)
926
476
        q->saturate_output(q, out);
927
480
}
928
929
930
/**
931
 * Cook subpacket decoding. This function returns one decoded subpacket,
932
 * usually 1024 samples per channel.
933
 *
934
 * @param q                 pointer to the COOKContext
935
 * @param inbuffer          pointer to the inbuffer
936
 * @param outbuffer         pointer to the outbuffer
937
 */
938
240
static int decode_subpacket(COOKContext *q, COOKSubpacket *p,
939
                            const uint8_t *inbuffer, float **outbuffer)
940
{
941
240
    int sub_packet_size = p->size;
942
    int res;
943
944
240
    memset(q->decode_buffer_1, 0, sizeof(q->decode_buffer_1));
945
240
    decode_bytes_and_gain(q, p, inbuffer, &p->gains1);
946
947
240
    if (p->joint_stereo) {
948
240
        if ((res = joint_decode(q, p, q->decode_buffer_1, q->decode_buffer_2)) < 0)
949
            return res;
950
    } else {
951
        if ((res = mono_decode(q, p, q->decode_buffer_1)) < 0)
952
            return res;
953
954
        if (p->num_channels == 2) {
955
            decode_bytes_and_gain(q, p, inbuffer + sub_packet_size / 2, &p->gains2);
956
            if ((res = mono_decode(q, p, q->decode_buffer_2)) < 0)
957
                return res;
958
        }
959
    }
960
961
240
    mlt_compensate_output(q, q->decode_buffer_1, &p->gains1,
962
240
                          p->mono_previous_buffer1,
963
238
                          outbuffer ? outbuffer[p->ch_idx] : NULL);
964
965
240
    if (p->num_channels == 2) {
966
240
        if (p->joint_stereo)
967
240
            mlt_compensate_output(q, q->decode_buffer_2, &p->gains1,
968
240
                                  p->mono_previous_buffer2,
969
238
                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
970
        else
971
            mlt_compensate_output(q, q->decode_buffer_2, &p->gains2,
972
                                  p->mono_previous_buffer2,
973
                                  outbuffer ? outbuffer[p->ch_idx + 1] : NULL);
974
    }
975
976
240
    return 0;
977
}
978
979
980
240
static int cook_decode_frame(AVCodecContext *avctx, void *data,
981
                             int *got_frame_ptr, AVPacket *avpkt)
982
{
983
240
    AVFrame *frame     = data;
984
240
    const uint8_t *buf = avpkt->data;
985
240
    int buf_size = avpkt->size;
986
240
    COOKContext *q = avctx->priv_data;
987
240
    float **samples = NULL;
988
    int i, ret;
989
240
    int offset = 0;
990
240
    int chidx = 0;
991
992
240
    if (buf_size < avctx->block_align)
993
        return buf_size;
994
995
    /* get output buffer */
996
240
    if (q->discarded_packets >= 2) {
997
238
        frame->nb_samples = q->samples_per_channel;
998
238
        if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
999
            return ret;
1000
238
        samples = (float **)frame->extended_data;
1001
    }
1002
1003
    /* estimate subpacket sizes */
1004
240
    q->subpacket[0].size = avctx->block_align;
1005
1006
240
    for (i = 1; i < q->num_subpackets; i++) {
1007
        q->subpacket[i].size = 2 * buf[avctx->block_align - q->num_subpackets + i];
1008
        q->subpacket[0].size -= q->subpacket[i].size + 1;
1009
        if (q->subpacket[0].size < 0) {
1010
            av_log(avctx, AV_LOG_DEBUG,
1011
                   "frame subpacket size total > avctx->block_align!\n");
1012
            return AVERROR_INVALIDDATA;
1013
        }
1014
    }
1015
1016
    /* decode supbackets */
1017
480
    for (i = 0; i < q->num_subpackets; i++) {
1018
240
        q->subpacket[i].bits_per_subpacket = (q->subpacket[i].size * 8) >>
1019
240
                                              q->subpacket[i].bits_per_subpdiv;
1020
240
        q->subpacket[i].ch_idx = chidx;
1021
240
        av_log(avctx, AV_LOG_DEBUG,
1022
               "subpacket[%i] size %i js %i %i block_align %i\n",
1023
               i, q->subpacket[i].size, q->subpacket[i].joint_stereo, offset,
1024
               avctx->block_align);
1025
1026
240
        if ((ret = decode_subpacket(q, &q->subpacket[i], buf + offset, samples)) < 0)
1027
            return ret;
1028
240
        offset += q->subpacket[i].size;
1029
240
        chidx += q->subpacket[i].num_channels;
1030
480
        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i] %i %i\n",
1031
240
               i, q->subpacket[i].size * 8, get_bits_count(&q->gb));
1032
    }
1033
1034
    /* Discard the first two frames: no valid audio. */
1035
240
    if (q->discarded_packets < 2) {
1036
2
        q->discarded_packets++;
1037
2
        *got_frame_ptr = 0;
1038
2
        return avctx->block_align;
1039
    }
1040
1041
238
    *got_frame_ptr = 1;
1042
1043
238
    return avctx->block_align;
1044
}
1045
1046
6
static void dump_cook_context(COOKContext *q)
1047
{
1048
    //int i=0;
1049
#define PRINT(a, b) ff_dlog(q->avctx, " %s = %d\n", a, b);
1050
    ff_dlog(q->avctx, "COOKextradata\n");
1051
    ff_dlog(q->avctx, "cookversion=%x\n", q->subpacket[0].cookversion);
1052
6
    if (q->subpacket[0].cookversion > STEREO) {
1053
        PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1054
        PRINT("js_vlc_bits", q->subpacket[0].js_vlc_bits);
1055
    }
1056
    ff_dlog(q->avctx, "COOKContext\n");
1057
    PRINT("nb_channels", q->avctx->channels);
1058
    PRINT("bit_rate", (int)q->avctx->bit_rate);
1059
    PRINT("sample_rate", q->avctx->sample_rate);
1060
    PRINT("samples_per_channel", q->subpacket[0].samples_per_channel);
1061
    PRINT("subbands", q->subpacket[0].subbands);
1062
    PRINT("js_subband_start", q->subpacket[0].js_subband_start);
1063
    PRINT("log2_numvector_size", q->subpacket[0].log2_numvector_size);
1064
    PRINT("numvector_size", q->subpacket[0].numvector_size);
1065
    PRINT("total_subbands", q->subpacket[0].total_subbands);
1066
6
}
1067
1068
/**
1069
 * Cook initialization
1070
 *
1071
 * @param avctx     pointer to the AVCodecContext
1072
 */
1073
6
static av_cold int cook_decode_init(AVCodecContext *avctx)
1074
{
1075
6
    COOKContext *q = avctx->priv_data;
1076
    GetByteContext gb;
1077
6
    int s = 0;
1078
6
    unsigned int channel_mask = 0;
1079
6
    int samples_per_frame = 0;
1080
    int ret;
1081
6
    q->avctx = avctx;
1082
1083
    /* Take care of the codec specific extradata. */
1084
6
    if (avctx->extradata_size < 8) {
1085
        av_log(avctx, AV_LOG_ERROR, "Necessary extradata missing!\n");
1086
        return AVERROR_INVALIDDATA;
1087
    }
1088
6
    av_log(avctx, AV_LOG_DEBUG, "codecdata_length=%d\n", avctx->extradata_size);
1089
1090
6
    bytestream2_init(&gb, avctx->extradata, avctx->extradata_size);
1091
1092
    /* Take data from the AVCodecContext (RM container). */
1093
6
    if (!avctx->channels) {
1094
        av_log(avctx, AV_LOG_ERROR, "Invalid number of channels\n");
1095
        return AVERROR_INVALIDDATA;
1096
    }
1097
1098
6
    if (avctx->block_align >= INT_MAX / 8)
1099
        return AVERROR(EINVAL);
1100
1101
    /* Initialize RNG. */
1102
6
    av_lfg_init(&q->random_state, 0);
1103
1104
6
    ff_audiodsp_init(&q->adsp);
1105
1106
12
    while (bytestream2_get_bytes_left(&gb)) {
1107
6
        if (s >= FFMIN(MAX_SUBPACKETS, avctx->block_align)) {
1108
            avpriv_request_sample(avctx, "subpackets > %d", FFMIN(MAX_SUBPACKETS, avctx->block_align));
1109
            return AVERROR_PATCHWELCOME;
1110
        }
1111
        /* 8 for mono, 16 for stereo, ? for multichannel
1112
           Swap to right endianness so we don't need to care later on. */
1113
6
        q->subpacket[s].cookversion      = bytestream2_get_be32(&gb);
1114
6
        samples_per_frame                = bytestream2_get_be16(&gb);
1115
6
        q->subpacket[s].subbands         = bytestream2_get_be16(&gb);
1116
6
        bytestream2_get_be32(&gb);    // Unknown unused
1117
6
        q->subpacket[s].js_subband_start = bytestream2_get_be16(&gb);
1118
6
        if (q->subpacket[s].js_subband_start >= 51) {
1119
            av_log(avctx, AV_LOG_ERROR, "js_subband_start %d is too large\n", q->subpacket[s].js_subband_start);
1120
            return AVERROR_INVALIDDATA;
1121
        }
1122
6
        q->subpacket[s].js_vlc_bits      = bytestream2_get_be16(&gb);
1123
1124
        /* Initialize extradata related variables. */
1125
6
        q->subpacket[s].samples_per_channel = samples_per_frame / avctx->channels;
1126
6
        q->subpacket[s].bits_per_subpacket = avctx->block_align * 8;
1127
1128
        /* Initialize default data states. */
1129
6
        q->subpacket[s].log2_numvector_size = 5;
1130
6
        q->subpacket[s].total_subbands = q->subpacket[s].subbands;
1131
6
        q->subpacket[s].num_channels = 1;
1132
1133
        /* Initialize version-dependent variables */
1134
1135
6
        av_log(avctx, AV_LOG_DEBUG, "subpacket[%i].cookversion=%x\n", s,
1136
               q->subpacket[s].cookversion);
1137
6
        q->subpacket[s].joint_stereo = 0;
1138

6
        switch (q->subpacket[s].cookversion) {
1139
        case MONO:
1140
            if (avctx->channels != 1) {
1141
                avpriv_request_sample(avctx, "Container channels != 1");
1142
                return AVERROR_PATCHWELCOME;
1143
            }
1144
            av_log(avctx, AV_LOG_DEBUG, "MONO\n");
1145
            break;
1146
2
        case STEREO:
1147
2
            if (avctx->channels != 1) {
1148
                q->subpacket[s].bits_per_subpdiv = 1;
1149
                q->subpacket[s].num_channels = 2;
1150
            }
1151
2
            av_log(avctx, AV_LOG_DEBUG, "STEREO\n");
1152
2
            break;
1153
4
        case JOINT_STEREO:
1154
4
            if (avctx->channels != 2) {
1155
                avpriv_request_sample(avctx, "Container channels != 2");
1156
                return AVERROR_PATCHWELCOME;
1157
            }
1158
4
            av_log(avctx, AV_LOG_DEBUG, "JOINT_STEREO\n");
1159
4
            if (avctx->extradata_size >= 16) {
1160
4
                q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1161
4
                                                 q->subpacket[s].js_subband_start;
1162
4
                q->subpacket[s].joint_stereo = 1;
1163
4
                q->subpacket[s].num_channels = 2;
1164
            }
1165
4
            if (q->subpacket[s].samples_per_channel > 256) {
1166
4
                q->subpacket[s].log2_numvector_size = 6;
1167
            }
1168
4
            if (q->subpacket[s].samples_per_channel > 512) {
1169
4
                q->subpacket[s].log2_numvector_size = 7;
1170
            }
1171
4
            break;
1172
        case MC_COOK:
1173
            av_log(avctx, AV_LOG_DEBUG, "MULTI_CHANNEL\n");
1174
            channel_mask |= q->subpacket[s].channel_mask = bytestream2_get_be32(&gb);
1175
1176
            if (av_get_channel_layout_nb_channels(q->subpacket[s].channel_mask) > 1) {
1177
                q->subpacket[s].total_subbands = q->subpacket[s].subbands +
1178
                                                 q->subpacket[s].js_subband_start;
1179
                q->subpacket[s].joint_stereo = 1;
1180
                q->subpacket[s].num_channels = 2;
1181
                q->subpacket[s].samples_per_channel = samples_per_frame >> 1;
1182
1183
                if (q->subpacket[s].samples_per_channel > 256) {
1184
                    q->subpacket[s].log2_numvector_size = 6;
1185
                }
1186
                if (q->subpacket[s].samples_per_channel > 512) {
1187
                    q->subpacket[s].log2_numvector_size = 7;
1188
                }
1189
            } else
1190
                q->subpacket[s].samples_per_channel = samples_per_frame;
1191
1192
            break;
1193
        default:
1194
            avpriv_request_sample(avctx, "Cook version %d",
1195
                                  q->subpacket[s].cookversion);
1196
            return AVERROR_PATCHWELCOME;
1197
        }
1198
1199

6
        if (s > 1 && q->subpacket[s].samples_per_channel != q->samples_per_channel) {
1200
            av_log(avctx, AV_LOG_ERROR, "different number of samples per channel!\n");
1201
            return AVERROR_INVALIDDATA;
1202
        } else
1203
6
            q->samples_per_channel = q->subpacket[0].samples_per_channel;
1204
1205
1206
        /* Initialize variable relations */
1207
6
        q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);
1208
1209
        /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1210
6
        if (q->subpacket[s].total_subbands > 53) {
1211
            avpriv_request_sample(avctx, "total_subbands > 53");
1212
            return AVERROR_PATCHWELCOME;
1213
        }
1214
1215
6
        if ((q->subpacket[s].js_vlc_bits > 6) ||
1216
6
            (q->subpacket[s].js_vlc_bits < 2 * q->subpacket[s].joint_stereo)) {
1217
            av_log(avctx, AV_LOG_ERROR, "js_vlc_bits = %d, only >= %d and <= 6 allowed!\n",
1218
                   q->subpacket[s].js_vlc_bits, 2 * q->subpacket[s].joint_stereo);
1219
            return AVERROR_INVALIDDATA;
1220
        }
1221
1222
6
        if (q->subpacket[s].subbands > 50) {
1223
            avpriv_request_sample(avctx, "subbands > 50");
1224
            return AVERROR_PATCHWELCOME;
1225
        }
1226
6
        if (q->subpacket[s].subbands == 0) {
1227
            avpriv_request_sample(avctx, "subbands = 0");
1228
            return AVERROR_PATCHWELCOME;
1229
        }
1230
6
        q->subpacket[s].gains1.now      = q->subpacket[s].gain_1;
1231
6
        q->subpacket[s].gains1.previous = q->subpacket[s].gain_2;
1232
6
        q->subpacket[s].gains2.now      = q->subpacket[s].gain_3;
1233
6
        q->subpacket[s].gains2.previous = q->subpacket[s].gain_4;
1234
1235
6
        if (q->num_subpackets + q->subpacket[s].num_channels > q->avctx->channels) {
1236
            av_log(avctx, AV_LOG_ERROR, "Too many subpackets %d for channels %d\n", q->num_subpackets, q->avctx->channels);
1237
            return AVERROR_INVALIDDATA;
1238
        }
1239
1240
6
        q->num_subpackets++;
1241
6
        s++;
1242
    }
1243
1244
    /* Try to catch some obviously faulty streams, otherwise it might be exploitable */
1245

6
    if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&
1246
6
        q->samples_per_channel != 1024) {
1247
        avpriv_request_sample(avctx, "samples_per_channel = %d",
1248
                              q->samples_per_channel);
1249
        return AVERROR_PATCHWELCOME;
1250
    }
1251
1252
    /* Generate tables */
1253
6
    init_pow2table();
1254
6
    init_gain_table(q);
1255
6
    init_cplscales_table(q);
1256
1257
6
    if ((ret = init_cook_vlc_tables(q)))
1258
        return ret;
1259
1260
    /* Pad the databuffer with:
1261
       DECODE_BYTES_PAD1 or DECODE_BYTES_PAD2 for decode_bytes(),
1262
       AV_INPUT_BUFFER_PADDING_SIZE, for the bitstreamreader. */
1263
6
    q->decoded_bytes_buffer =
1264
6
        av_mallocz(avctx->block_align
1265
6
                   + DECODE_BYTES_PAD1(avctx->block_align)
1266
6
                   + AV_INPUT_BUFFER_PADDING_SIZE);
1267
6
    if (!q->decoded_bytes_buffer)
1268
        return AVERROR(ENOMEM);
1269
1270
    /* Initialize transform. */
1271
6
    if ((ret = init_cook_mlt(q)))
1272
        return ret;
1273
1274
    /* Initialize COOK signal arithmetic handling */
1275
    if (1) {
1276
6
        q->scalar_dequant  = scalar_dequant_float;
1277
6
        q->decouple        = decouple_float;
1278
6
        q->imlt_window     = imlt_window_float;
1279
6
        q->interpolate     = interpolate_float;
1280
6
        q->saturate_output = saturate_output_float;
1281
    }
1282
1283
6
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1284
6
    if (channel_mask)
1285
        avctx->channel_layout = channel_mask;
1286
    else
1287
6
        avctx->channel_layout = (avctx->channels == 2) ? AV_CH_LAYOUT_STEREO : AV_CH_LAYOUT_MONO;
1288
1289
1290
6
    dump_cook_context(q);
1291
1292
6
    return 0;
1293
}
1294
1295
AVCodec ff_cook_decoder = {
1296
    .name           = "cook",
1297
    .long_name      = NULL_IF_CONFIG_SMALL("Cook / Cooker / Gecko (RealAudio G2)"),
1298
    .type           = AVMEDIA_TYPE_AUDIO,
1299
    .id             = AV_CODEC_ID_COOK,
1300
    .priv_data_size = sizeof(COOKContext),
1301
    .init           = cook_decode_init,
1302
    .close          = cook_decode_close,
1303
    .decode         = cook_decode_frame,
1304
    .capabilities   = AV_CODEC_CAP_DR1,
1305
    .caps_internal  = FF_CODEC_CAP_INIT_CLEANUP,
1306
    .sample_fmts    = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
1307
                                                      AV_SAMPLE_FMT_NONE },
1308
};