Statistics
| Branch: | Revision:

ffmpeg / libavcodec / aac.c @ b5e2bb8c

History | View | Annotate | Download (72.7 KB)

1
/*
2
 * AAC decoder
3
 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4
 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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 libavcodec/aac.c
25
 * AAC decoder
26
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
27
 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
28
 */
29

    
30
/*
31
 * supported tools
32
 *
33
 * Support?             Name
34
 * N (code in SoC repo) gain control
35
 * Y                    block switching
36
 * Y                    window shapes - standard
37
 * N                    window shapes - Low Delay
38
 * Y                    filterbank - standard
39
 * N (code in SoC repo) filterbank - Scalable Sample Rate
40
 * Y                    Temporal Noise Shaping
41
 * N (code in SoC repo) Long Term Prediction
42
 * Y                    intensity stereo
43
 * Y                    channel coupling
44
 * Y                    frequency domain prediction
45
 * Y                    Perceptual Noise Substitution
46
 * Y                    Mid/Side stereo
47
 * N                    Scalable Inverse AAC Quantization
48
 * N                    Frequency Selective Switch
49
 * N                    upsampling filter
50
 * Y                    quantization & coding - AAC
51
 * N                    quantization & coding - TwinVQ
52
 * N                    quantization & coding - BSAC
53
 * N                    AAC Error Resilience tools
54
 * N                    Error Resilience payload syntax
55
 * N                    Error Protection tool
56
 * N                    CELP
57
 * N                    Silence Compression
58
 * N                    HVXC
59
 * N                    HVXC 4kbits/s VR
60
 * N                    Structured Audio tools
61
 * N                    Structured Audio Sample Bank Format
62
 * N                    MIDI
63
 * N                    Harmonic and Individual Lines plus Noise
64
 * N                    Text-To-Speech Interface
65
 * N (in progress)      Spectral Band Replication
66
 * Y (not in this code) Layer-1
67
 * Y (not in this code) Layer-2
68
 * Y (not in this code) Layer-3
69
 * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
70
 * N (planned)          Parametric Stereo
71
 * N                    Direct Stream Transfer
72
 *
73
 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
74
 *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
75
           Parametric Stereo.
76
 */
77

    
78

    
79
#include "avcodec.h"
80
#include "internal.h"
81
#include "get_bits.h"
82
#include "dsputil.h"
83
#include "lpc.h"
84

    
85
#include "aac.h"
86
#include "aactab.h"
87
#include "aacdectab.h"
88
#include "mpeg4audio.h"
89
#include "aac_parser.h"
90

    
91
#include <assert.h>
92
#include <errno.h>
93
#include <math.h>
94
#include <string.h>
95

    
96
#if ARCH_ARM
97
#   include "arm/aac.h"
98
#endif
99

    
100
union float754 {
101
    float f;
102
    uint32_t i;
103
};
104

    
105
static VLC vlc_scalefactors;
106
static VLC vlc_spectral[11];
107

    
108
static uint32_t cbrt_tab[1<<13];
109

    
110
static const char overread_err[] = "Input buffer exhausted before END element found\n";
111

    
112
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
113
{
114
    if (ac->tag_che_map[type][elem_id]) {
115
        return ac->tag_che_map[type][elem_id];
116
    }
117
    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
118
        return NULL;
119
    }
120
    switch (ac->m4ac.chan_config) {
121
    case 7:
122
        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
123
            ac->tags_mapped++;
124
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
125
        }
126
    case 6:
127
        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
128
           instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
129
           encountered such a stream, transfer the LFE[0] element to SCE[1] */
130
        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
131
            ac->tags_mapped++;
132
            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
133
        }
134
    case 5:
135
        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
136
            ac->tags_mapped++;
137
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
138
        }
139
    case 4:
140
        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
141
            ac->tags_mapped++;
142
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
143
        }
144
    case 3:
145
    case 2:
146
        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
147
            ac->tags_mapped++;
148
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
149
        } else if (ac->m4ac.chan_config == 2) {
150
            return NULL;
151
        }
152
    case 1:
153
        if (!ac->tags_mapped && type == TYPE_SCE) {
154
            ac->tags_mapped++;
155
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
156
        }
157
    default:
158
        return NULL;
159
    }
160
}
161

    
162
/**
163
 * Check for the channel element in the current channel position configuration.
164
 * If it exists, make sure the appropriate element is allocated and map the
165
 * channel order to match the internal FFmpeg channel layout.
166
 *
167
 * @param   che_pos current channel position configuration
168
 * @param   type channel element type
169
 * @param   id channel element id
170
 * @param   channels count of the number of channels in the configuration
171
 *
172
 * @return  Returns error status. 0 - OK, !0 - error
173
 */
174
static av_cold int che_configure(AACContext *ac,
175
                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
176
                         int type, int id,
177
                         int *channels)
178
{
179
    if (che_pos[type][id]) {
180
        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
181
            return AVERROR(ENOMEM);
182
        if (type != TYPE_CCE) {
183
            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
184
            if (type == TYPE_CPE) {
185
                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
186
            }
187
        }
188
    } else
189
        av_freep(&ac->che[type][id]);
190
    return 0;
191
}
192

    
193
/**
194
 * Configure output channel order based on the current program configuration element.
195
 *
196
 * @param   che_pos current channel position configuration
197
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
198
 *
199
 * @return  Returns error status. 0 - OK, !0 - error
200
 */
201
static av_cold int output_configure(AACContext *ac,
202
                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
203
                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
204
                            int channel_config, enum OCStatus oc_type)
205
{
206
    AVCodecContext *avctx = ac->avccontext;
207
    int i, type, channels = 0, ret;
208

    
209
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
210

    
211
    if (channel_config) {
212
        for (i = 0; i < tags_per_config[channel_config]; i++) {
213
            if ((ret = che_configure(ac, che_pos,
214
                                     aac_channel_layout_map[channel_config - 1][i][0],
215
                                     aac_channel_layout_map[channel_config - 1][i][1],
216
                                     &channels)))
217
                return ret;
218
        }
219

    
220
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
221
        ac->tags_mapped = 0;
222

    
223
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
224
    } else {
225
        /* Allocate or free elements depending on if they are in the
226
         * current program configuration.
227
         *
228
         * Set up default 1:1 output mapping.
229
         *
230
         * For a 5.1 stream the output order will be:
231
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
232
         */
233

    
234
        for (i = 0; i < MAX_ELEM_ID; i++) {
235
            for (type = 0; type < 4; type++) {
236
                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
237
                    return ret;
238
            }
239
        }
240

    
241
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
242
        ac->tags_mapped = 4 * MAX_ELEM_ID;
243

    
244
        avctx->channel_layout = 0;
245
    }
246

    
247
    avctx->channels = channels;
248

    
249
    ac->output_configured = oc_type;
250

    
251
    return 0;
252
}
253

    
254
/**
255
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
256
 *
257
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
258
 * @param sce_map mono (Single Channel Element) map
259
 * @param type speaker type/position for these channels
260
 */
261
static void decode_channel_map(enum ChannelPosition *cpe_map,
262
                               enum ChannelPosition *sce_map,
263
                               enum ChannelPosition type,
264
                               GetBitContext *gb, int n)
265
{
266
    while (n--) {
267
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
268
        map[get_bits(gb, 4)] = type;
269
    }
270
}
271

    
272
/**
273
 * Decode program configuration element; reference: table 4.2.
274
 *
275
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
276
 *
277
 * @return  Returns error status. 0 - OK, !0 - error
278
 */
279
static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
280
                      GetBitContext *gb)
281
{
282
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
283
    int comment_len;
284

    
285
    skip_bits(gb, 2);  // object_type
286

    
287
    sampling_index = get_bits(gb, 4);
288
    if (ac->m4ac.sampling_index != sampling_index)
289
        av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
290

    
291
    num_front       = get_bits(gb, 4);
292
    num_side        = get_bits(gb, 4);
293
    num_back        = get_bits(gb, 4);
294
    num_lfe         = get_bits(gb, 2);
295
    num_assoc_data  = get_bits(gb, 3);
296
    num_cc          = get_bits(gb, 4);
297

    
298
    if (get_bits1(gb))
299
        skip_bits(gb, 4); // mono_mixdown_tag
300
    if (get_bits1(gb))
301
        skip_bits(gb, 4); // stereo_mixdown_tag
302

    
303
    if (get_bits1(gb))
304
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
305

    
306
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
307
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
308
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
309
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
310

    
311
    skip_bits_long(gb, 4 * num_assoc_data);
312

    
313
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
314

    
315
    align_get_bits(gb);
316

    
317
    /* comment field, first byte is length */
318
    comment_len = get_bits(gb, 8) * 8;
319
    if (get_bits_left(gb) < comment_len) {
320
        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
321
        return -1;
322
    }
323
    skip_bits_long(gb, comment_len);
324
    return 0;
325
}
326

    
327
/**
328
 * Set up channel positions based on a default channel configuration
329
 * as specified in table 1.17.
330
 *
331
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
332
 *
333
 * @return  Returns error status. 0 - OK, !0 - error
334
 */
335
static av_cold int set_default_channel_config(AACContext *ac,
336
                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
337
                                      int channel_config)
338
{
339
    if (channel_config < 1 || channel_config > 7) {
340
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
341
               channel_config);
342
        return -1;
343
    }
344

    
345
    /* default channel configurations:
346
     *
347
     * 1ch : front center (mono)
348
     * 2ch : L + R (stereo)
349
     * 3ch : front center + L + R
350
     * 4ch : front center + L + R + back center
351
     * 5ch : front center + L + R + back stereo
352
     * 6ch : front center + L + R + back stereo + LFE
353
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
354
     */
355

    
356
    if (channel_config != 2)
357
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
358
    if (channel_config > 1)
359
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
360
    if (channel_config == 4)
361
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
362
    if (channel_config > 4)
363
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
364
        = AAC_CHANNEL_BACK;  // back stereo
365
    if (channel_config > 5)
366
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
367
    if (channel_config == 7)
368
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
369

    
370
    return 0;
371
}
372

    
373
/**
374
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
375
 *
376
 * @return  Returns error status. 0 - OK, !0 - error
377
 */
378
static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
379
                                     int channel_config)
380
{
381
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
382
    int extension_flag, ret;
383

    
384
    if (get_bits1(gb)) { // frameLengthFlag
385
        av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
386
        return -1;
387
    }
388

    
389
    if (get_bits1(gb))       // dependsOnCoreCoder
390
        skip_bits(gb, 14);   // coreCoderDelay
391
    extension_flag = get_bits1(gb);
392

    
393
    if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
394
        ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
395
        skip_bits(gb, 3);     // layerNr
396

    
397
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
398
    if (channel_config == 0) {
399
        skip_bits(gb, 4);  // element_instance_tag
400
        if ((ret = decode_pce(ac, new_che_pos, gb)))
401
            return ret;
402
    } else {
403
        if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
404
            return ret;
405
    }
406
    if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
407
        return ret;
408

    
409
    if (extension_flag) {
410
        switch (ac->m4ac.object_type) {
411
        case AOT_ER_BSAC:
412
            skip_bits(gb, 5);    // numOfSubFrame
413
            skip_bits(gb, 11);   // layer_length
414
            break;
415
        case AOT_ER_AAC_LC:
416
        case AOT_ER_AAC_LTP:
417
        case AOT_ER_AAC_SCALABLE:
418
        case AOT_ER_AAC_LD:
419
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
420
                                    * aacScalefactorDataResilienceFlag
421
                                    * aacSpectralDataResilienceFlag
422
                                    */
423
            break;
424
        }
425
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
426
    }
427
    return 0;
428
}
429

    
430
/**
431
 * Decode audio specific configuration; reference: table 1.13.
432
 *
433
 * @param   data        pointer to AVCodecContext extradata
434
 * @param   data_size   size of AVCCodecContext extradata
435
 *
436
 * @return  Returns error status. 0 - OK, !0 - error
437
 */
438
static int decode_audio_specific_config(AACContext *ac, void *data,
439
                                        int data_size)
440
{
441
    GetBitContext gb;
442
    int i;
443

    
444
    init_get_bits(&gb, data, data_size * 8);
445

    
446
    if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
447
        return -1;
448
    if (ac->m4ac.sampling_index > 12) {
449
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
450
        return -1;
451
    }
452

    
453
    skip_bits_long(&gb, i);
454

    
455
    switch (ac->m4ac.object_type) {
456
    case AOT_AAC_MAIN:
457
    case AOT_AAC_LC:
458
        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
459
            return -1;
460
        break;
461
    default:
462
        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
463
               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
464
        return -1;
465
    }
466
    return 0;
467
}
468

    
469
/**
470
 * linear congruential pseudorandom number generator
471
 *
472
 * @param   previous_val    pointer to the current state of the generator
473
 *
474
 * @return  Returns a 32-bit pseudorandom integer
475
 */
476
static av_always_inline int lcg_random(int previous_val)
477
{
478
    return previous_val * 1664525 + 1013904223;
479
}
480

    
481
static av_always_inline void reset_predict_state(PredictorState *ps)
482
{
483
    ps->r0   = 0.0f;
484
    ps->r1   = 0.0f;
485
    ps->cor0 = 0.0f;
486
    ps->cor1 = 0.0f;
487
    ps->var0 = 1.0f;
488
    ps->var1 = 1.0f;
489
}
490

    
491
static void reset_all_predictors(PredictorState *ps)
492
{
493
    int i;
494
    for (i = 0; i < MAX_PREDICTORS; i++)
495
        reset_predict_state(&ps[i]);
496
}
497

    
498
static void reset_predictor_group(PredictorState *ps, int group_num)
499
{
500
    int i;
501
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
502
        reset_predict_state(&ps[i]);
503
}
504

    
505
static av_cold int aac_decode_init(AVCodecContext *avccontext)
506
{
507
    AACContext *ac = avccontext->priv_data;
508
    int i;
509

    
510
    ac->avccontext = avccontext;
511

    
512
    if (avccontext->extradata_size > 0) {
513
        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
514
            return -1;
515
        avccontext->sample_rate = ac->m4ac.sample_rate;
516
    } else if (avccontext->channels > 0) {
517
        ac->m4ac.sample_rate = avccontext->sample_rate;
518
    }
519

    
520
    avccontext->sample_fmt = SAMPLE_FMT_S16;
521
    avccontext->frame_size = 1024;
522

    
523
    AAC_INIT_VLC_STATIC( 0, 304);
524
    AAC_INIT_VLC_STATIC( 1, 270);
525
    AAC_INIT_VLC_STATIC( 2, 550);
526
    AAC_INIT_VLC_STATIC( 3, 300);
527
    AAC_INIT_VLC_STATIC( 4, 328);
528
    AAC_INIT_VLC_STATIC( 5, 294);
529
    AAC_INIT_VLC_STATIC( 6, 306);
530
    AAC_INIT_VLC_STATIC( 7, 268);
531
    AAC_INIT_VLC_STATIC( 8, 510);
532
    AAC_INIT_VLC_STATIC( 9, 366);
533
    AAC_INIT_VLC_STATIC(10, 462);
534

    
535
    dsputil_init(&ac->dsp, avccontext);
536

    
537
    ac->random_state = 0x1f2e3d4c;
538

    
539
    // -1024 - Compensate wrong IMDCT method.
540
    // 32768 - Required to scale values to the correct range for the bias method
541
    //         for float to int16 conversion.
542

    
543
    if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
544
        ac->add_bias  = 385.0f;
545
        ac->sf_scale  = 1. / (-1024. * 32768.);
546
        ac->sf_offset = 0;
547
    } else {
548
        ac->add_bias  = 0.0f;
549
        ac->sf_scale  = 1. / -1024.;
550
        ac->sf_offset = 60;
551
    }
552

    
553
#if !CONFIG_HARDCODED_TABLES
554
    for (i = 0; i < 428; i++)
555
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
556
#endif /* CONFIG_HARDCODED_TABLES */
557

    
558
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
559
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
560
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
561
                    352);
562

    
563
    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
564
    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
565
    // window initialization
566
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
567
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
568
    ff_init_ff_sine_windows(10);
569
    ff_init_ff_sine_windows( 7);
570

    
571
    if (!cbrt_tab[(1<<13) - 1]) {
572
        for (i = 0; i < 1<<13; i++) {
573
            union float754 f;
574
            f.f = cbrtf(i) * i;
575
            cbrt_tab[i] = f.i;
576
        }
577
    }
578

    
579
    return 0;
580
}
581

    
582
/**
583
 * Skip data_stream_element; reference: table 4.10.
584
 */
585
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
586
{
587
    int byte_align = get_bits1(gb);
588
    int count = get_bits(gb, 8);
589
    if (count == 255)
590
        count += get_bits(gb, 8);
591
    if (byte_align)
592
        align_get_bits(gb);
593

    
594
    if (get_bits_left(gb) < 8 * count) {
595
        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
596
        return -1;
597
    }
598
    skip_bits_long(gb, 8 * count);
599
    return 0;
600
}
601

    
602
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
603
                             GetBitContext *gb)
604
{
605
    int sfb;
606
    if (get_bits1(gb)) {
607
        ics->predictor_reset_group = get_bits(gb, 5);
608
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
609
            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
610
            return -1;
611
        }
612
    }
613
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
614
        ics->prediction_used[sfb] = get_bits1(gb);
615
    }
616
    return 0;
617
}
618

    
619
/**
620
 * Decode Individual Channel Stream info; reference: table 4.6.
621
 *
622
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
623
 */
624
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
625
                           GetBitContext *gb, int common_window)
626
{
627
    if (get_bits1(gb)) {
628
        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
629
        memset(ics, 0, sizeof(IndividualChannelStream));
630
        return -1;
631
    }
632
    ics->window_sequence[1] = ics->window_sequence[0];
633
    ics->window_sequence[0] = get_bits(gb, 2);
634
    ics->use_kb_window[1]   = ics->use_kb_window[0];
635
    ics->use_kb_window[0]   = get_bits1(gb);
636
    ics->num_window_groups  = 1;
637
    ics->group_len[0]       = 1;
638
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
639
        int i;
640
        ics->max_sfb = get_bits(gb, 4);
641
        for (i = 0; i < 7; i++) {
642
            if (get_bits1(gb)) {
643
                ics->group_len[ics->num_window_groups - 1]++;
644
            } else {
645
                ics->num_window_groups++;
646
                ics->group_len[ics->num_window_groups - 1] = 1;
647
            }
648
        }
649
        ics->num_windows       = 8;
650
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
651
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
652
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
653
        ics->predictor_present = 0;
654
    } else {
655
        ics->max_sfb               = get_bits(gb, 6);
656
        ics->num_windows           = 1;
657
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
658
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
659
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
660
        ics->predictor_present     = get_bits1(gb);
661
        ics->predictor_reset_group = 0;
662
        if (ics->predictor_present) {
663
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
664
                if (decode_prediction(ac, ics, gb)) {
665
                    memset(ics, 0, sizeof(IndividualChannelStream));
666
                    return -1;
667
                }
668
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
669
                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
670
                memset(ics, 0, sizeof(IndividualChannelStream));
671
                return -1;
672
            } else {
673
                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
674
                memset(ics, 0, sizeof(IndividualChannelStream));
675
                return -1;
676
            }
677
        }
678
    }
679

    
680
    if (ics->max_sfb > ics->num_swb) {
681
        av_log(ac->avccontext, AV_LOG_ERROR,
682
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
683
               ics->max_sfb, ics->num_swb);
684
        memset(ics, 0, sizeof(IndividualChannelStream));
685
        return -1;
686
    }
687

    
688
    return 0;
689
}
690

    
691
/**
692
 * Decode band types (section_data payload); reference: table 4.46.
693
 *
694
 * @param   band_type           array of the used band type
695
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
696
 *
697
 * @return  Returns error status. 0 - OK, !0 - error
698
 */
699
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
700
                             int band_type_run_end[120], GetBitContext *gb,
701
                             IndividualChannelStream *ics)
702
{
703
    int g, idx = 0;
704
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
705
    for (g = 0; g < ics->num_window_groups; g++) {
706
        int k = 0;
707
        while (k < ics->max_sfb) {
708
            uint8_t sect_end = k;
709
            int sect_len_incr;
710
            int sect_band_type = get_bits(gb, 4);
711
            if (sect_band_type == 12) {
712
                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
713
                return -1;
714
            }
715
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
716
                sect_end += sect_len_incr;
717
            sect_end += sect_len_incr;
718
            if (get_bits_left(gb) < 0) {
719
                av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
720
                return -1;
721
            }
722
            if (sect_end > ics->max_sfb) {
723
                av_log(ac->avccontext, AV_LOG_ERROR,
724
                       "Number of bands (%d) exceeds limit (%d).\n",
725
                       sect_end, ics->max_sfb);
726
                return -1;
727
            }
728
            for (; k < sect_end; k++) {
729
                band_type        [idx]   = sect_band_type;
730
                band_type_run_end[idx++] = sect_end;
731
            }
732
        }
733
    }
734
    return 0;
735
}
736

    
737
/**
738
 * Decode scalefactors; reference: table 4.47.
739
 *
740
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
741
 * @param   band_type           array of the used band type
742
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
743
 * @param   sf                  array of scalefactors or intensity stereo positions
744
 *
745
 * @return  Returns error status. 0 - OK, !0 - error
746
 */
747
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
748
                               unsigned int global_gain,
749
                               IndividualChannelStream *ics,
750
                               enum BandType band_type[120],
751
                               int band_type_run_end[120])
752
{
753
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
754
    int g, i, idx = 0;
755
    int offset[3] = { global_gain, global_gain - 90, 100 };
756
    int noise_flag = 1;
757
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
758
    for (g = 0; g < ics->num_window_groups; g++) {
759
        for (i = 0; i < ics->max_sfb;) {
760
            int run_end = band_type_run_end[idx];
761
            if (band_type[idx] == ZERO_BT) {
762
                for (; i < run_end; i++, idx++)
763
                    sf[idx] = 0.;
764
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
765
                for (; i < run_end; i++, idx++) {
766
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
767
                    if (offset[2] > 255U) {
768
                        av_log(ac->avccontext, AV_LOG_ERROR,
769
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
770
                        return -1;
771
                    }
772
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
773
                }
774
            } else if (band_type[idx] == NOISE_BT) {
775
                for (; i < run_end; i++, idx++) {
776
                    if (noise_flag-- > 0)
777
                        offset[1] += get_bits(gb, 9) - 256;
778
                    else
779
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
780
                    if (offset[1] > 255U) {
781
                        av_log(ac->avccontext, AV_LOG_ERROR,
782
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
783
                        return -1;
784
                    }
785
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
786
                }
787
            } else {
788
                for (; i < run_end; i++, idx++) {
789
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
790
                    if (offset[0] > 255U) {
791
                        av_log(ac->avccontext, AV_LOG_ERROR,
792
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
793
                        return -1;
794
                    }
795
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
796
                }
797
            }
798
        }
799
    }
800
    return 0;
801
}
802

    
803
/**
804
 * Decode pulse data; reference: table 4.7.
805
 */
806
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
807
                         const uint16_t *swb_offset, int num_swb)
808
{
809
    int i, pulse_swb;
810
    pulse->num_pulse = get_bits(gb, 2) + 1;
811
    pulse_swb        = get_bits(gb, 6);
812
    if (pulse_swb >= num_swb)
813
        return -1;
814
    pulse->pos[0]    = swb_offset[pulse_swb];
815
    pulse->pos[0]   += get_bits(gb, 5);
816
    if (pulse->pos[0] > 1023)
817
        return -1;
818
    pulse->amp[0]    = get_bits(gb, 4);
819
    for (i = 1; i < pulse->num_pulse; i++) {
820
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
821
        if (pulse->pos[i] > 1023)
822
            return -1;
823
        pulse->amp[i] = get_bits(gb, 4);
824
    }
825
    return 0;
826
}
827

    
828
/**
829
 * Decode Temporal Noise Shaping data; reference: table 4.48.
830
 *
831
 * @return  Returns error status. 0 - OK, !0 - error
832
 */
833
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
834
                      GetBitContext *gb, const IndividualChannelStream *ics)
835
{
836
    int w, filt, i, coef_len, coef_res, coef_compress;
837
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
838
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
839
    for (w = 0; w < ics->num_windows; w++) {
840
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
841
            coef_res = get_bits1(gb);
842

    
843
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
844
                int tmp2_idx;
845
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
846

    
847
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
848
                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
849
                           tns->order[w][filt], tns_max_order);
850
                    tns->order[w][filt] = 0;
851
                    return -1;
852
                }
853
                if (tns->order[w][filt]) {
854
                    tns->direction[w][filt] = get_bits1(gb);
855
                    coef_compress = get_bits1(gb);
856
                    coef_len = coef_res + 3 - coef_compress;
857
                    tmp2_idx = 2 * coef_compress + coef_res;
858

    
859
                    for (i = 0; i < tns->order[w][filt]; i++)
860
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
861
                }
862
            }
863
        }
864
    }
865
    return 0;
866
}
867

    
868
/**
869
 * Decode Mid/Side data; reference: table 4.54.
870
 *
871
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
872
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
873
 *                      [3] reserved for scalable AAC
874
 */
875
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
876
                                   int ms_present)
877
{
878
    int idx;
879
    if (ms_present == 1) {
880
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
881
            cpe->ms_mask[idx] = get_bits1(gb);
882
    } else if (ms_present == 2) {
883
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
884
    }
885
}
886

    
887
#ifndef VMUL2
888
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
889
                           const float *scale)
890
{
891
    float s = *scale;
892
    *dst++ = v[idx    & 15] * s;
893
    *dst++ = v[idx>>4 & 15] * s;
894
    return dst;
895
}
896
#endif
897

    
898
#ifndef VMUL4
899
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
900
                           const float *scale)
901
{
902
    float s = *scale;
903
    *dst++ = v[idx    & 3] * s;
904
    *dst++ = v[idx>>2 & 3] * s;
905
    *dst++ = v[idx>>4 & 3] * s;
906
    *dst++ = v[idx>>6 & 3] * s;
907
    return dst;
908
}
909
#endif
910

    
911
#ifndef VMUL2S
912
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
913
                            unsigned sign, const float *scale)
914
{
915
    union float754 s0, s1;
916

    
917
    s0.f = s1.f = *scale;
918
    s0.i ^= sign >> 1 << 31;
919
    s1.i ^= sign      << 31;
920

    
921
    *dst++ = v[idx    & 15] * s0.f;
922
    *dst++ = v[idx>>4 & 15] * s1.f;
923

    
924
    return dst;
925
}
926
#endif
927

    
928
#ifndef VMUL4S
929
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
930
                            unsigned sign, const float *scale)
931
{
932
    unsigned nz = idx >> 12;
933
    union float754 s = { .f = *scale };
934
    union float754 t;
935

    
936
    t.i = s.i ^ (sign & 1<<31);
937
    *dst++ = v[idx    & 3] * t.f;
938

    
939
    sign <<= nz & 1; nz >>= 1;
940
    t.i = s.i ^ (sign & 1<<31);
941
    *dst++ = v[idx>>2 & 3] * t.f;
942

    
943
    sign <<= nz & 1; nz >>= 1;
944
    t.i = s.i ^ (sign & 1<<31);
945
    *dst++ = v[idx>>4 & 3] * t.f;
946

    
947
    sign <<= nz & 1; nz >>= 1;
948
    t.i = s.i ^ (sign & 1<<31);
949
    *dst++ = v[idx>>6 & 3] * t.f;
950

    
951
    return dst;
952
}
953
#endif
954

    
955
/**
956
 * Decode spectral data; reference: table 4.50.
957
 * Dequantize and scale spectral data; reference: 4.6.3.3.
958
 *
959
 * @param   coef            array of dequantized, scaled spectral data
960
 * @param   sf              array of scalefactors or intensity stereo positions
961
 * @param   pulse_present   set if pulses are present
962
 * @param   pulse           pointer to pulse data struct
963
 * @param   band_type       array of the used band type
964
 *
965
 * @return  Returns error status. 0 - OK, !0 - error
966
 */
967
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
968
                                       GetBitContext *gb, const float sf[120],
969
                                       int pulse_present, const Pulse *pulse,
970
                                       const IndividualChannelStream *ics,
971
                                       enum BandType band_type[120])
972
{
973
    int i, k, g, idx = 0;
974
    const int c = 1024 / ics->num_windows;
975
    const uint16_t *offsets = ics->swb_offset;
976
    float *coef_base = coef;
977
    int err_idx;
978

    
979
    for (g = 0; g < ics->num_windows; g++)
980
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
981

    
982
    for (g = 0; g < ics->num_window_groups; g++) {
983
        unsigned g_len = ics->group_len[g];
984

    
985
        for (i = 0; i < ics->max_sfb; i++, idx++) {
986
            const unsigned cbt_m1 = band_type[idx] - 1;
987
            float *cfo = coef + offsets[i];
988
            int off_len = offsets[i + 1] - offsets[i];
989
            int group;
990

    
991
            if (cbt_m1 >= INTENSITY_BT2 - 1) {
992
                for (group = 0; group < g_len; group++, cfo+=128) {
993
                    memset(cfo, 0, off_len * sizeof(float));
994
                }
995
            } else if (cbt_m1 == NOISE_BT - 1) {
996
                for (group = 0; group < g_len; group++, cfo+=128) {
997
                    float scale;
998
                    float band_energy;
999

    
1000
                    for (k = 0; k < off_len; k++) {
1001
                        ac->random_state  = lcg_random(ac->random_state);
1002
                        cfo[k] = ac->random_state;
1003
                    }
1004

    
1005
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1006
                    scale = sf[idx] / sqrtf(band_energy);
1007
                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1008
                }
1009
            } else {
1010
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1011
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1012
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1013
                const int cb_size = ff_aac_spectral_sizes[cbt_m1];
1014
                OPEN_READER(re, gb);
1015

    
1016
                switch (cbt_m1 >> 1) {
1017
                case 0:
1018
                    for (group = 0; group < g_len; group++, cfo+=128) {
1019
                        float *cf = cfo;
1020
                        int len = off_len;
1021

    
1022
                        do {
1023
                            int code;
1024
                            unsigned cb_idx;
1025

    
1026
                            UPDATE_CACHE(re, gb);
1027
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1028

    
1029
                            if (code >= cb_size) {
1030
                                err_idx = code;
1031
                                goto err_cb_overflow;
1032
                            }
1033

    
1034
                            cb_idx = cb_vector_idx[code];
1035
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1036
                        } while (len -= 4);
1037
                    }
1038
                    break;
1039

    
1040
                case 1:
1041
                    for (group = 0; group < g_len; group++, cfo+=128) {
1042
                        float *cf = cfo;
1043
                        int len = off_len;
1044

    
1045
                        do {
1046
                            int code;
1047
                            unsigned nnz;
1048
                            unsigned cb_idx;
1049
                            uint32_t bits;
1050

    
1051
                            UPDATE_CACHE(re, gb);
1052
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1053

    
1054
                            if (code >= cb_size) {
1055
                                err_idx = code;
1056
                                goto err_cb_overflow;
1057
                            }
1058

    
1059
#if MIN_CACHE_BITS < 20
1060
                            UPDATE_CACHE(re, gb);
1061
#endif
1062
                            cb_idx = cb_vector_idx[code];
1063
                            nnz = cb_idx >> 8 & 15;
1064
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1065
                            LAST_SKIP_BITS(re, gb, nnz);
1066
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1067
                        } while (len -= 4);
1068
                    }
1069
                    break;
1070

    
1071
                case 2:
1072
                    for (group = 0; group < g_len; group++, cfo+=128) {
1073
                        float *cf = cfo;
1074
                        int len = off_len;
1075

    
1076
                        do {
1077
                            int code;
1078
                            unsigned cb_idx;
1079

    
1080
                            UPDATE_CACHE(re, gb);
1081
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1082

    
1083
                            if (code >= cb_size) {
1084
                                err_idx = code;
1085
                                goto err_cb_overflow;
1086
                            }
1087

    
1088
                            cb_idx = cb_vector_idx[code];
1089
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1090
                        } while (len -= 2);
1091
                    }
1092
                    break;
1093

    
1094
                case 3:
1095
                case 4:
1096
                    for (group = 0; group < g_len; group++, cfo+=128) {
1097
                        float *cf = cfo;
1098
                        int len = off_len;
1099

    
1100
                        do {
1101
                            int code;
1102
                            unsigned nnz;
1103
                            unsigned cb_idx;
1104
                            unsigned sign;
1105

    
1106
                            UPDATE_CACHE(re, gb);
1107
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1108

    
1109
                            if (code >= cb_size) {
1110
                                err_idx = code;
1111
                                goto err_cb_overflow;
1112
                            }
1113

    
1114
                            cb_idx = cb_vector_idx[code];
1115
                            nnz = cb_idx >> 8 & 15;
1116
                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1117
                            LAST_SKIP_BITS(re, gb, nnz);
1118
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1119
                        } while (len -= 2);
1120
                    }
1121
                    break;
1122

    
1123
                default:
1124
                    for (group = 0; group < g_len; group++, cfo+=128) {
1125
                        float *cf = cfo;
1126
                        uint32_t *icf = (uint32_t *) cf;
1127
                        int len = off_len;
1128

    
1129
                        do {
1130
                            int code;
1131
                            unsigned nzt, nnz;
1132
                            unsigned cb_idx;
1133
                            uint32_t bits;
1134
                            int j;
1135

    
1136
                            UPDATE_CACHE(re, gb);
1137
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1138

    
1139
                            if (!code) {
1140
                                *icf++ = 0;
1141
                                *icf++ = 0;
1142
                                continue;
1143
                            }
1144

    
1145
                            if (code >= cb_size) {
1146
                                err_idx = code;
1147
                                goto err_cb_overflow;
1148
                            }
1149

    
1150
                            cb_idx = cb_vector_idx[code];
1151
                            nnz = cb_idx >> 12;
1152
                            nzt = cb_idx >> 8;
1153
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1154
                            LAST_SKIP_BITS(re, gb, nnz);
1155

    
1156
                            for (j = 0; j < 2; j++) {
1157
                                if (nzt & 1<<j) {
1158
                                    uint32_t b;
1159
                                    int n;
1160
                                    /* The total length of escape_sequence must be < 22 bits according
1161
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1162
                                    UPDATE_CACHE(re, gb);
1163
                                    b = GET_CACHE(re, gb);
1164
                                    b = 31 - av_log2(~b);
1165

    
1166
                                    if (b > 8) {
1167
                                        av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1168
                                        return -1;
1169
                                    }
1170

    
1171
#if MIN_CACHE_BITS < 21
1172
                                    LAST_SKIP_BITS(re, gb, b + 1);
1173
                                    UPDATE_CACHE(re, gb);
1174
#else
1175
                                    SKIP_BITS(re, gb, b + 1);
1176
#endif
1177
                                    b += 4;
1178
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
1179
                                    LAST_SKIP_BITS(re, gb, b);
1180
                                    *icf++ = cbrt_tab[n] | (bits & 1<<31);
1181
                                    bits <<= 1;
1182
                                } else {
1183
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1184
                                    *icf++ = (bits & 1<<31) | v;
1185
                                    bits <<= !!v;
1186
                                }
1187
                                cb_idx >>= 4;
1188
                            }
1189
                        } while (len -= 2);
1190

    
1191
                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1192
                    }
1193
                }
1194

    
1195
                CLOSE_READER(re, gb);
1196
            }
1197
        }
1198
        coef += g_len << 7;
1199
    }
1200

    
1201
    if (pulse_present) {
1202
        idx = 0;
1203
        for (i = 0; i < pulse->num_pulse; i++) {
1204
            float co = coef_base[ pulse->pos[i] ];
1205
            while (offsets[idx + 1] <= pulse->pos[i])
1206
                idx++;
1207
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1208
                float ico = -pulse->amp[i];
1209
                if (co) {
1210
                    co /= sf[idx];
1211
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1212
                }
1213
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1214
            }
1215
        }
1216
    }
1217
    return 0;
1218

    
1219
err_cb_overflow:
1220
    av_log(ac->avccontext, AV_LOG_ERROR,
1221
           "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
1222
           band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
1223
    return -1;
1224
}
1225

    
1226
static av_always_inline float flt16_round(float pf)
1227
{
1228
    union float754 tmp;
1229
    tmp.f = pf;
1230
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1231
    return tmp.f;
1232
}
1233

    
1234
static av_always_inline float flt16_even(float pf)
1235
{
1236
    union float754 tmp;
1237
    tmp.f = pf;
1238
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1239
    return tmp.f;
1240
}
1241

    
1242
static av_always_inline float flt16_trunc(float pf)
1243
{
1244
    union float754 pun;
1245
    pun.f = pf;
1246
    pun.i &= 0xFFFF0000U;
1247
    return pun.f;
1248
}
1249

    
1250
static av_always_inline void predict(AACContext *ac, PredictorState *ps, float *coef,
1251
                    int output_enable)
1252
{
1253
    const float a     = 0.953125; // 61.0 / 64
1254
    const float alpha = 0.90625;  // 29.0 / 32
1255
    float e0, e1;
1256
    float pv;
1257
    float k1, k2;
1258

    
1259
    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
1260
    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
1261

    
1262
    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
1263
    if (output_enable)
1264
        *coef += pv * ac->sf_scale;
1265

    
1266
    e0 = *coef / ac->sf_scale;
1267
    e1 = e0 - k1 * ps->r0;
1268

    
1269
    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
1270
    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
1271
    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
1272
    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
1273

    
1274
    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
1275
    ps->r0 = flt16_trunc(a * e0);
1276
}
1277

    
1278
/**
1279
 * Apply AAC-Main style frequency domain prediction.
1280
 */
1281
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1282
{
1283
    int sfb, k;
1284

    
1285
    if (!sce->ics.predictor_initialized) {
1286
        reset_all_predictors(sce->predictor_state);
1287
        sce->ics.predictor_initialized = 1;
1288
    }
1289

    
1290
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1291
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1292
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1293
                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
1294
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1295
            }
1296
        }
1297
        if (sce->ics.predictor_reset_group)
1298
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1299
    } else
1300
        reset_all_predictors(sce->predictor_state);
1301
}
1302

    
1303
/**
1304
 * Decode an individual_channel_stream payload; reference: table 4.44.
1305
 *
1306
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1307
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1308
 *
1309
 * @return  Returns error status. 0 - OK, !0 - error
1310
 */
1311
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1312
                      GetBitContext *gb, int common_window, int scale_flag)
1313
{
1314
    Pulse pulse;
1315
    TemporalNoiseShaping    *tns = &sce->tns;
1316
    IndividualChannelStream *ics = &sce->ics;
1317
    float *out = sce->coeffs;
1318
    int global_gain, pulse_present = 0;
1319

    
1320
    /* This assignment is to silence a GCC warning about the variable being used
1321
     * uninitialized when in fact it always is.
1322
     */
1323
    pulse.num_pulse = 0;
1324

    
1325
    global_gain = get_bits(gb, 8);
1326

    
1327
    if (!common_window && !scale_flag) {
1328
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1329
            return -1;
1330
    }
1331

    
1332
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1333
        return -1;
1334
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1335
        return -1;
1336

    
1337
    pulse_present = 0;
1338
    if (!scale_flag) {
1339
        if ((pulse_present = get_bits1(gb))) {
1340
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1341
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1342
                return -1;
1343
            }
1344
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1345
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1346
                return -1;
1347
            }
1348
        }
1349
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1350
            return -1;
1351
        if (get_bits1(gb)) {
1352
            av_log_missing_feature(ac->avccontext, "SSR", 1);
1353
            return -1;
1354
        }
1355
    }
1356

    
1357
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1358
        return -1;
1359

    
1360
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1361
        apply_prediction(ac, sce);
1362

    
1363
    return 0;
1364
}
1365

    
1366
/**
1367
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1368
 */
1369
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1370
{
1371
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1372
    float *ch0 = cpe->ch[0].coeffs;
1373
    float *ch1 = cpe->ch[1].coeffs;
1374
    int g, i, group, idx = 0;
1375
    const uint16_t *offsets = ics->swb_offset;
1376
    for (g = 0; g < ics->num_window_groups; g++) {
1377
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1378
            if (cpe->ms_mask[idx] &&
1379
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1380
                for (group = 0; group < ics->group_len[g]; group++) {
1381
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1382
                                              ch1 + group * 128 + offsets[i],
1383
                                              offsets[i+1] - offsets[i]);
1384
                }
1385
            }
1386
        }
1387
        ch0 += ics->group_len[g] * 128;
1388
        ch1 += ics->group_len[g] * 128;
1389
    }
1390
}
1391

    
1392
/**
1393
 * intensity stereo decoding; reference: 4.6.8.2.3
1394
 *
1395
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1396
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1397
 *                      [3] reserved for scalable AAC
1398
 */
1399
static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
1400
{
1401
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1402
    SingleChannelElement         *sce1 = &cpe->ch[1];
1403
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1404
    const uint16_t *offsets = ics->swb_offset;
1405
    int g, group, i, k, idx = 0;
1406
    int c;
1407
    float scale;
1408
    for (g = 0; g < ics->num_window_groups; g++) {
1409
        for (i = 0; i < ics->max_sfb;) {
1410
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1411
                const int bt_run_end = sce1->band_type_run_end[idx];
1412
                for (; i < bt_run_end; i++, idx++) {
1413
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1414
                    if (ms_present)
1415
                        c *= 1 - 2 * cpe->ms_mask[idx];
1416
                    scale = c * sce1->sf[idx];
1417
                    for (group = 0; group < ics->group_len[g]; group++)
1418
                        for (k = offsets[i]; k < offsets[i + 1]; k++)
1419
                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
1420
                }
1421
            } else {
1422
                int bt_run_end = sce1->band_type_run_end[idx];
1423
                idx += bt_run_end - i;
1424
                i    = bt_run_end;
1425
            }
1426
        }
1427
        coef0 += ics->group_len[g] * 128;
1428
        coef1 += ics->group_len[g] * 128;
1429
    }
1430
}
1431

    
1432
/**
1433
 * Decode a channel_pair_element; reference: table 4.4.
1434
 *
1435
 * @param   elem_id Identifies the instance of a syntax element.
1436
 *
1437
 * @return  Returns error status. 0 - OK, !0 - error
1438
 */
1439
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1440
{
1441
    int i, ret, common_window, ms_present = 0;
1442

    
1443
    common_window = get_bits1(gb);
1444
    if (common_window) {
1445
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1446
            return -1;
1447
        i = cpe->ch[1].ics.use_kb_window[0];
1448
        cpe->ch[1].ics = cpe->ch[0].ics;
1449
        cpe->ch[1].ics.use_kb_window[1] = i;
1450
        ms_present = get_bits(gb, 2);
1451
        if (ms_present == 3) {
1452
            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1453
            return -1;
1454
        } else if (ms_present)
1455
            decode_mid_side_stereo(cpe, gb, ms_present);
1456
    }
1457
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1458
        return ret;
1459
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1460
        return ret;
1461

    
1462
    if (common_window) {
1463
        if (ms_present)
1464
            apply_mid_side_stereo(ac, cpe);
1465
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1466
            apply_prediction(ac, &cpe->ch[0]);
1467
            apply_prediction(ac, &cpe->ch[1]);
1468
        }
1469
    }
1470

    
1471
    apply_intensity_stereo(cpe, ms_present);
1472
    return 0;
1473
}
1474

    
1475
/**
1476
 * Decode coupling_channel_element; reference: table 4.8.
1477
 *
1478
 * @param   elem_id Identifies the instance of a syntax element.
1479
 *
1480
 * @return  Returns error status. 0 - OK, !0 - error
1481
 */
1482
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1483
{
1484
    int num_gain = 0;
1485
    int c, g, sfb, ret;
1486
    int sign;
1487
    float scale;
1488
    SingleChannelElement *sce = &che->ch[0];
1489
    ChannelCoupling     *coup = &che->coup;
1490

    
1491
    coup->coupling_point = 2 * get_bits1(gb);
1492
    coup->num_coupled = get_bits(gb, 3);
1493
    for (c = 0; c <= coup->num_coupled; c++) {
1494
        num_gain++;
1495
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1496
        coup->id_select[c] = get_bits(gb, 4);
1497
        if (coup->type[c] == TYPE_CPE) {
1498
            coup->ch_select[c] = get_bits(gb, 2);
1499
            if (coup->ch_select[c] == 3)
1500
                num_gain++;
1501
        } else
1502
            coup->ch_select[c] = 2;
1503
    }
1504
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1505

    
1506
    sign  = get_bits(gb, 1);
1507
    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1508

    
1509
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1510
        return ret;
1511

    
1512
    for (c = 0; c < num_gain; c++) {
1513
        int idx  = 0;
1514
        int cge  = 1;
1515
        int gain = 0;
1516
        float gain_cache = 1.;
1517
        if (c) {
1518
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1519
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1520
            gain_cache = pow(scale, -gain);
1521
        }
1522
        if (coup->coupling_point == AFTER_IMDCT) {
1523
            coup->gain[c][0] = gain_cache;
1524
        } else {
1525
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1526
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1527
                    if (sce->band_type[idx] != ZERO_BT) {
1528
                        if (!cge) {
1529
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1530
                            if (t) {
1531
                                int s = 1;
1532
                                t = gain += t;
1533
                                if (sign) {
1534
                                    s  -= 2 * (t & 0x1);
1535
                                    t >>= 1;
1536
                                }
1537
                                gain_cache = pow(scale, -t) * s;
1538
                            }
1539
                        }
1540
                        coup->gain[c][idx] = gain_cache;
1541
                    }
1542
                }
1543
            }
1544
        }
1545
    }
1546
    return 0;
1547
}
1548

    
1549
/**
1550
 * Decode Spectral Band Replication extension data; reference: table 4.55.
1551
 *
1552
 * @param   crc flag indicating the presence of CRC checksum
1553
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1554
 *
1555
 * @return  Returns number of bytes consumed from the TYPE_FIL element.
1556
 */
1557
static int decode_sbr_extension(AACContext *ac, GetBitContext *gb,
1558
                                int crc, int cnt)
1559
{
1560
    // TODO : sbr_extension implementation
1561
    av_log_missing_feature(ac->avccontext, "SBR", 0);
1562
    skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type
1563
    return cnt;
1564
}
1565

    
1566
/**
1567
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1568
 *
1569
 * @return  Returns number of bytes consumed.
1570
 */
1571
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1572
                                         GetBitContext *gb)
1573
{
1574
    int i;
1575
    int num_excl_chan = 0;
1576

    
1577
    do {
1578
        for (i = 0; i < 7; i++)
1579
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1580
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1581

    
1582
    return num_excl_chan / 7;
1583
}
1584

    
1585
/**
1586
 * Decode dynamic range information; reference: table 4.52.
1587
 *
1588
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1589
 *
1590
 * @return  Returns number of bytes consumed.
1591
 */
1592
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1593
                                GetBitContext *gb, int cnt)
1594
{
1595
    int n             = 1;
1596
    int drc_num_bands = 1;
1597
    int i;
1598

    
1599
    /* pce_tag_present? */
1600
    if (get_bits1(gb)) {
1601
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1602
        skip_bits(gb, 4); // tag_reserved_bits
1603
        n++;
1604
    }
1605

    
1606
    /* excluded_chns_present? */
1607
    if (get_bits1(gb)) {
1608
        n += decode_drc_channel_exclusions(che_drc, gb);
1609
    }
1610

    
1611
    /* drc_bands_present? */
1612
    if (get_bits1(gb)) {
1613
        che_drc->band_incr            = get_bits(gb, 4);
1614
        che_drc->interpolation_scheme = get_bits(gb, 4);
1615
        n++;
1616
        drc_num_bands += che_drc->band_incr;
1617
        for (i = 0; i < drc_num_bands; i++) {
1618
            che_drc->band_top[i] = get_bits(gb, 8);
1619
            n++;
1620
        }
1621
    }
1622

    
1623
    /* prog_ref_level_present? */
1624
    if (get_bits1(gb)) {
1625
        che_drc->prog_ref_level = get_bits(gb, 7);
1626
        skip_bits1(gb); // prog_ref_level_reserved_bits
1627
        n++;
1628
    }
1629

    
1630
    for (i = 0; i < drc_num_bands; i++) {
1631
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1632
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1633
        n++;
1634
    }
1635

    
1636
    return n;
1637
}
1638

    
1639
/**
1640
 * Decode extension data (incomplete); reference: table 4.51.
1641
 *
1642
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1643
 *
1644
 * @return Returns number of bytes consumed
1645
 */
1646
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt)
1647
{
1648
    int crc_flag = 0;
1649
    int res = cnt;
1650
    switch (get_bits(gb, 4)) { // extension type
1651
    case EXT_SBR_DATA_CRC:
1652
        crc_flag++;
1653
    case EXT_SBR_DATA:
1654
        res = decode_sbr_extension(ac, gb, crc_flag, cnt);
1655
        break;
1656
    case EXT_DYNAMIC_RANGE:
1657
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1658
        break;
1659
    case EXT_FILL:
1660
    case EXT_FILL_DATA:
1661
    case EXT_DATA_ELEMENT:
1662
    default:
1663
        skip_bits_long(gb, 8 * cnt - 4);
1664
        break;
1665
    };
1666
    return res;
1667
}
1668

    
1669
/**
1670
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1671
 *
1672
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1673
 * @param   coef    spectral coefficients
1674
 */
1675
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1676
                      IndividualChannelStream *ics, int decode)
1677
{
1678
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1679
    int w, filt, m, i;
1680
    int bottom, top, order, start, end, size, inc;
1681
    float lpc[TNS_MAX_ORDER];
1682

    
1683
    for (w = 0; w < ics->num_windows; w++) {
1684
        bottom = ics->num_swb;
1685
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1686
            top    = bottom;
1687
            bottom = FFMAX(0, top - tns->length[w][filt]);
1688
            order  = tns->order[w][filt];
1689
            if (order == 0)
1690
                continue;
1691

    
1692
            // tns_decode_coef
1693
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1694

    
1695
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1696
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1697
            if ((size = end - start) <= 0)
1698
                continue;
1699
            if (tns->direction[w][filt]) {
1700
                inc = -1;
1701
                start = end - 1;
1702
            } else {
1703
                inc = 1;
1704
            }
1705
            start += w * 128;
1706

    
1707
            // ar filter
1708
            for (m = 0; m < size; m++, start += inc)
1709
                for (i = 1; i <= FFMIN(m, order); i++)
1710
                    coef[start] -= coef[start - i * inc] * lpc[i - 1];
1711
        }
1712
    }
1713
}
1714

    
1715
/**
1716
 * Conduct IMDCT and windowing.
1717
 */
1718
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1719
{
1720
    IndividualChannelStream *ics = &sce->ics;
1721
    float *in    = sce->coeffs;
1722
    float *out   = sce->ret;
1723
    float *saved = sce->saved;
1724
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1725
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1726
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1727
    float *buf  = ac->buf_mdct;
1728
    float *temp = ac->temp;
1729
    int i;
1730

    
1731
    // imdct
1732
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1733
        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1734
            av_log(ac->avccontext, AV_LOG_WARNING,
1735
                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1736
                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1737
        for (i = 0; i < 1024; i += 128)
1738
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1739
    } else
1740
        ff_imdct_half(&ac->mdct, buf, in);
1741

    
1742
    /* window overlapping
1743
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1744
     * and long to short transitions are considered to be short to short
1745
     * transitions. This leaves just two cases (long to long and short to short)
1746
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1747
     */
1748
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1749
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1750
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, ac->add_bias, 512);
1751
    } else {
1752
        for (i = 0; i < 448; i++)
1753
            out[i] = saved[i] + ac->add_bias;
1754

    
1755
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1756
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, ac->add_bias, 64);
1757
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      ac->add_bias, 64);
1758
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      ac->add_bias, 64);
1759
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      ac->add_bias, 64);
1760
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      ac->add_bias, 64);
1761
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1762
        } else {
1763
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, ac->add_bias, 64);
1764
            for (i = 576; i < 1024; i++)
1765
                out[i] = buf[i-512] + ac->add_bias;
1766
        }
1767
    }
1768

    
1769
    // buffer update
1770
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1771
        for (i = 0; i < 64; i++)
1772
            saved[i] = temp[64 + i] - ac->add_bias;
1773
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1774
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1775
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1776
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1777
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1778
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1779
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1780
    } else { // LONG_STOP or ONLY_LONG
1781
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1782
    }
1783
}
1784

    
1785
/**
1786
 * Apply dependent channel coupling (applied before IMDCT).
1787
 *
1788
 * @param   index   index into coupling gain array
1789
 */
1790
static void apply_dependent_coupling(AACContext *ac,
1791
                                     SingleChannelElement *target,
1792
                                     ChannelElement *cce, int index)
1793
{
1794
    IndividualChannelStream *ics = &cce->ch[0].ics;
1795
    const uint16_t *offsets = ics->swb_offset;
1796
    float *dest = target->coeffs;
1797
    const float *src = cce->ch[0].coeffs;
1798
    int g, i, group, k, idx = 0;
1799
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1800
        av_log(ac->avccontext, AV_LOG_ERROR,
1801
               "Dependent coupling is not supported together with LTP\n");
1802
        return;
1803
    }
1804
    for (g = 0; g < ics->num_window_groups; g++) {
1805
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1806
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1807
                const float gain = cce->coup.gain[index][idx];
1808
                for (group = 0; group < ics->group_len[g]; group++) {
1809
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1810
                        // XXX dsputil-ize
1811
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1812
                    }
1813
                }
1814
            }
1815
        }
1816
        dest += ics->group_len[g] * 128;
1817
        src  += ics->group_len[g] * 128;
1818
    }
1819
}
1820

    
1821
/**
1822
 * Apply independent channel coupling (applied after IMDCT).
1823
 *
1824
 * @param   index   index into coupling gain array
1825
 */
1826
static void apply_independent_coupling(AACContext *ac,
1827
                                       SingleChannelElement *target,
1828
                                       ChannelElement *cce, int index)
1829
{
1830
    int i;
1831
    const float gain = cce->coup.gain[index][0];
1832
    const float bias = ac->add_bias;
1833
    const float *src = cce->ch[0].ret;
1834
    float *dest = target->ret;
1835

    
1836
    for (i = 0; i < 1024; i++)
1837
        dest[i] += gain * (src[i] - bias);
1838
}
1839

    
1840
/**
1841
 * channel coupling transformation interface
1842
 *
1843
 * @param   index   index into coupling gain array
1844
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1845
 */
1846
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1847
                                   enum RawDataBlockType type, int elem_id,
1848
                                   enum CouplingPoint coupling_point,
1849
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1850
{
1851
    int i, c;
1852

    
1853
    for (i = 0; i < MAX_ELEM_ID; i++) {
1854
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1855
        int index = 0;
1856

    
1857
        if (cce && cce->coup.coupling_point == coupling_point) {
1858
            ChannelCoupling *coup = &cce->coup;
1859

    
1860
            for (c = 0; c <= coup->num_coupled; c++) {
1861
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1862
                    if (coup->ch_select[c] != 1) {
1863
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1864
                        if (coup->ch_select[c] != 0)
1865
                            index++;
1866
                    }
1867
                    if (coup->ch_select[c] != 2)
1868
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1869
                } else
1870
                    index += 1 + (coup->ch_select[c] == 3);
1871
            }
1872
        }
1873
    }
1874
}
1875

    
1876
/**
1877
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1878
 */
1879
static void spectral_to_sample(AACContext *ac)
1880
{
1881
    int i, type;
1882
    for (type = 3; type >= 0; type--) {
1883
        for (i = 0; i < MAX_ELEM_ID; i++) {
1884
            ChannelElement *che = ac->che[type][i];
1885
            if (che) {
1886
                if (type <= TYPE_CPE)
1887
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1888
                if (che->ch[0].tns.present)
1889
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1890
                if (che->ch[1].tns.present)
1891
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1892
                if (type <= TYPE_CPE)
1893
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1894
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
1895
                    imdct_and_windowing(ac, &che->ch[0]);
1896
                if (type == TYPE_CPE)
1897
                    imdct_and_windowing(ac, &che->ch[1]);
1898
                if (type <= TYPE_CCE)
1899
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1900
            }
1901
        }
1902
    }
1903
}
1904

    
1905
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
1906
{
1907
    int size;
1908
    AACADTSHeaderInfo hdr_info;
1909

    
1910
    size = ff_aac_parse_header(gb, &hdr_info);
1911
    if (size > 0) {
1912
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
1913
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1914
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1915
            ac->m4ac.chan_config = hdr_info.chan_config;
1916
            if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
1917
                return -7;
1918
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
1919
                return -7;
1920
        } else if (ac->output_configured != OC_LOCKED) {
1921
            ac->output_configured = OC_NONE;
1922
        }
1923
        if (ac->output_configured != OC_LOCKED)
1924
            ac->m4ac.sbr = -1;
1925
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1926
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1927
        ac->m4ac.object_type     = hdr_info.object_type;
1928
        if (!ac->avccontext->sample_rate)
1929
            ac->avccontext->sample_rate = hdr_info.sample_rate;
1930
        if (hdr_info.num_aac_frames == 1) {
1931
            if (!hdr_info.crc_absent)
1932
                skip_bits(gb, 16);
1933
        } else {
1934
            av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1935
            return -1;
1936
        }
1937
    }
1938
    return size;
1939
}
1940

    
1941
static int aac_decode_frame(AVCodecContext *avccontext, void *data,
1942
                            int *data_size, AVPacket *avpkt)
1943
{
1944
    const uint8_t *buf = avpkt->data;
1945
    int buf_size = avpkt->size;
1946
    AACContext *ac = avccontext->priv_data;
1947
    ChannelElement *che = NULL;
1948
    GetBitContext gb;
1949
    enum RawDataBlockType elem_type;
1950
    int err, elem_id, data_size_tmp;
1951
    int buf_consumed;
1952

    
1953
    init_get_bits(&gb, buf, buf_size * 8);
1954

    
1955
    if (show_bits(&gb, 12) == 0xfff) {
1956
        if (parse_adts_frame_header(ac, &gb) < 0) {
1957
            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1958
            return -1;
1959
        }
1960
        if (ac->m4ac.sampling_index > 12) {
1961
            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1962
            return -1;
1963
        }
1964
    }
1965

    
1966
    // parse
1967
    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1968
        elem_id = get_bits(&gb, 4);
1969

    
1970
        if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
1971
            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1972
            return -1;
1973
        }
1974

    
1975
        switch (elem_type) {
1976

    
1977
        case TYPE_SCE:
1978
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1979
            break;
1980

    
1981
        case TYPE_CPE:
1982
            err = decode_cpe(ac, &gb, che);
1983
            break;
1984

    
1985
        case TYPE_CCE:
1986
            err = decode_cce(ac, &gb, che);
1987
            break;
1988

    
1989
        case TYPE_LFE:
1990
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1991
            break;
1992

    
1993
        case TYPE_DSE:
1994
            err = skip_data_stream_element(ac, &gb);
1995
            break;
1996

    
1997
        case TYPE_PCE: {
1998
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1999
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2000
            if ((err = decode_pce(ac, new_che_pos, &gb)))
2001
                break;
2002
            if (ac->output_configured > OC_TRIAL_PCE)
2003
                av_log(avccontext, AV_LOG_ERROR,
2004
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2005
            else
2006
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
2007
            break;
2008
        }
2009

    
2010
        case TYPE_FIL:
2011
            if (elem_id == 15)
2012
                elem_id += get_bits(&gb, 8) - 1;
2013
            if (get_bits_left(&gb) < 8 * elem_id) {
2014
                    av_log(avccontext, AV_LOG_ERROR, overread_err);
2015
                    return -1;
2016
            }
2017
            while (elem_id > 0)
2018
                elem_id -= decode_extension_payload(ac, &gb, elem_id);
2019
            err = 0; /* FIXME */
2020
            break;
2021

    
2022
        default:
2023
            err = -1; /* should not happen, but keeps compiler happy */
2024
            break;
2025
        }
2026

    
2027
        if (err)
2028
            return err;
2029

    
2030
        if (get_bits_left(&gb) < 3) {
2031
            av_log(avccontext, AV_LOG_ERROR, overread_err);
2032
            return -1;
2033
        }
2034
    }
2035

    
2036
    spectral_to_sample(ac);
2037

    
2038
    data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
2039
    if (*data_size < data_size_tmp) {
2040
        av_log(avccontext, AV_LOG_ERROR,
2041
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2042
               *data_size, data_size_tmp);
2043
        return -1;
2044
    }
2045
    *data_size = data_size_tmp;
2046

    
2047
    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
2048

    
2049
    if (ac->output_configured)
2050
        ac->output_configured = OC_LOCKED;
2051

    
2052
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2053
    return buf_size > buf_consumed ? buf_consumed : buf_size;
2054
}
2055

    
2056
static av_cold int aac_decode_close(AVCodecContext *avccontext)
2057
{
2058
    AACContext *ac = avccontext->priv_data;
2059
    int i, type;
2060

    
2061
    for (i = 0; i < MAX_ELEM_ID; i++) {
2062
        for (type = 0; type < 4; type++)
2063
            av_freep(&ac->che[type][i]);
2064
    }
2065

    
2066
    ff_mdct_end(&ac->mdct);
2067
    ff_mdct_end(&ac->mdct_small);
2068
    return 0;
2069
}
2070

    
2071
AVCodec aac_decoder = {
2072
    "aac",
2073
    CODEC_TYPE_AUDIO,
2074
    CODEC_ID_AAC,
2075
    sizeof(AACContext),
2076
    aac_decode_init,
2077
    NULL,
2078
    aac_decode_close,
2079
    aac_decode_frame,
2080
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2081
    .sample_fmts = (const enum SampleFormat[]) {
2082
        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
2083
    },
2084
    .channel_layouts = aac_channel_layout,
2085
};