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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
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
19
 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
22

    
23
/**
24
 * @file libavcodec/aac.c
25
 * AAC decoder
26
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
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 * @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
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 * N                    Error Protection tool
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 * 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
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 * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
70
 * N (planned)          Parametric Stereo
71
 * N                    Direct Stream Transfer
72
 *
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 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
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 *       - 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
union float754 {
97
    float f;
98
    uint32_t i;
99
};
100

    
101
static VLC vlc_scalefactors;
102
static VLC vlc_spectral[11];
103

    
104

    
105
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
106
{
107
    if (ac->tag_che_map[type][elem_id]) {
108
        return ac->tag_che_map[type][elem_id];
109
    }
110
    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
111
        return NULL;
112
    }
113
    switch (ac->m4ac.chan_config) {
114
    case 7:
115
        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
116
            ac->tags_mapped++;
117
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
118
        }
119
    case 6:
120
        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
121
           instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
122
           encountered such a stream, transfer the LFE[0] element to SCE[1] */
123
        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
124
            ac->tags_mapped++;
125
            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
126
        }
127
    case 5:
128
        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
129
            ac->tags_mapped++;
130
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
131
        }
132
    case 4:
133
        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
134
            ac->tags_mapped++;
135
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
136
        }
137
    case 3:
138
    case 2:
139
        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
140
            ac->tags_mapped++;
141
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
142
        } else if (ac->m4ac.chan_config == 2) {
143
            return NULL;
144
        }
145
    case 1:
146
        if (!ac->tags_mapped && type == TYPE_SCE) {
147
            ac->tags_mapped++;
148
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
149
        }
150
    default:
151
        return NULL;
152
    }
153
}
154

    
155
/**
156
 * Check for the channel element in the current channel position configuration.
157
 * If it exists, make sure the appropriate element is allocated and map the
158
 * channel order to match the internal FFmpeg channel layout.
159
 *
160
 * @param   che_pos current channel position configuration
161
 * @param   type channel element type
162
 * @param   id channel element id
163
 * @param   channels count of the number of channels in the configuration
164
 *
165
 * @return  Returns error status. 0 - OK, !0 - error
166
 */
167
static int che_configure(AACContext *ac,
168
                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
169
                         int type, int id,
170
                         int *channels)
171
{
172
    if (che_pos[type][id]) {
173
        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
174
            return AVERROR(ENOMEM);
175
        if (type != TYPE_CCE) {
176
            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
177
            if (type == TYPE_CPE) {
178
                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
179
            }
180
        }
181
    } else
182
        av_freep(&ac->che[type][id]);
183
    return 0;
184
}
185

    
186
/**
187
 * Configure output channel order based on the current program configuration element.
188
 *
189
 * @param   che_pos current channel position configuration
190
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
191
 *
192
 * @return  Returns error status. 0 - OK, !0 - error
193
 */
194
static int output_configure(AACContext *ac,
195
                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
196
                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
197
                            int channel_config, enum OCStatus oc_type)
198
{
199
    AVCodecContext *avctx = ac->avccontext;
200
    int i, type, channels = 0, ret;
201

    
202
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
203

    
204
    if (channel_config) {
205
        for (i = 0; i < tags_per_config[channel_config]; i++) {
206
            if ((ret = che_configure(ac, che_pos,
207
                                     aac_channel_layout_map[channel_config - 1][i][0],
208
                                     aac_channel_layout_map[channel_config - 1][i][1],
209
                                     &channels)))
210
                return ret;
211
        }
212

    
213
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
214
        ac->tags_mapped = 0;
215

    
216
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
217
    } else {
218
        /* Allocate or free elements depending on if they are in the
219
         * current program configuration.
220
         *
221
         * Set up default 1:1 output mapping.
222
         *
223
         * For a 5.1 stream the output order will be:
224
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
225
         */
226

    
227
        for (i = 0; i < MAX_ELEM_ID; i++) {
228
            for (type = 0; type < 4; type++) {
229
                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
230
                    return ret;
231
            }
232
        }
233

    
234
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
235
        ac->tags_mapped = 4 * MAX_ELEM_ID;
236

    
237
        avctx->channel_layout = 0;
238
    }
239

    
240
    avctx->channels = channels;
241

    
242
    ac->output_configured = oc_type;
243

    
244
    return 0;
245
}
246

    
247
/**
248
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
249
 *
250
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
251
 * @param sce_map mono (Single Channel Element) map
252
 * @param type speaker type/position for these channels
253
 */
254
static void decode_channel_map(enum ChannelPosition *cpe_map,
255
                               enum ChannelPosition *sce_map,
256
                               enum ChannelPosition type,
257
                               GetBitContext *gb, int n)
258
{
259
    while (n--) {
260
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
261
        map[get_bits(gb, 4)] = type;
262
    }
263
}
264

    
265
/**
266
 * Decode program configuration element; reference: table 4.2.
267
 *
268
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
269
 *
270
 * @return  Returns error status. 0 - OK, !0 - error
271
 */
272
static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
273
                      GetBitContext *gb)
274
{
275
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
276

    
277
    skip_bits(gb, 2);  // object_type
278

    
279
    sampling_index = get_bits(gb, 4);
280
    if (ac->m4ac.sampling_index != sampling_index)
281
        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");
282

    
283
    num_front       = get_bits(gb, 4);
284
    num_side        = get_bits(gb, 4);
285
    num_back        = get_bits(gb, 4);
286
    num_lfe         = get_bits(gb, 2);
287
    num_assoc_data  = get_bits(gb, 3);
288
    num_cc          = get_bits(gb, 4);
289

    
290
    if (get_bits1(gb))
291
        skip_bits(gb, 4); // mono_mixdown_tag
292
    if (get_bits1(gb))
293
        skip_bits(gb, 4); // stereo_mixdown_tag
294

    
295
    if (get_bits1(gb))
296
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
297

    
298
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
299
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
300
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
301
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
302

    
303
    skip_bits_long(gb, 4 * num_assoc_data);
304

    
305
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
306

    
307
    align_get_bits(gb);
308

    
309
    /* comment field, first byte is length */
310
    skip_bits_long(gb, 8 * get_bits(gb, 8));
311
    return 0;
312
}
313

    
314
/**
315
 * Set up channel positions based on a default channel configuration
316
 * as specified in table 1.17.
317
 *
318
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
319
 *
320
 * @return  Returns error status. 0 - OK, !0 - error
321
 */
322
static int set_default_channel_config(AACContext *ac,
323
                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
324
                                      int channel_config)
325
{
326
    if (channel_config < 1 || channel_config > 7) {
327
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
328
               channel_config);
329
        return -1;
330
    }
331

    
332
    /* default channel configurations:
333
     *
334
     * 1ch : front center (mono)
335
     * 2ch : L + R (stereo)
336
     * 3ch : front center + L + R
337
     * 4ch : front center + L + R + back center
338
     * 5ch : front center + L + R + back stereo
339
     * 6ch : front center + L + R + back stereo + LFE
340
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
341
     */
342

    
343
    if (channel_config != 2)
344
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
345
    if (channel_config > 1)
346
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
347
    if (channel_config == 4)
348
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
349
    if (channel_config > 4)
350
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
351
        = AAC_CHANNEL_BACK;  // back stereo
352
    if (channel_config > 5)
353
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
354
    if (channel_config == 7)
355
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
356

    
357
    return 0;
358
}
359

    
360
/**
361
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
362
 *
363
 * @return  Returns error status. 0 - OK, !0 - error
364
 */
365
static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
366
                                     int channel_config)
367
{
368
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
369
    int extension_flag, ret;
370

    
371
    if (get_bits1(gb)) { // frameLengthFlag
372
        av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
373
        return -1;
374
    }
375

    
376
    if (get_bits1(gb))       // dependsOnCoreCoder
377
        skip_bits(gb, 14);   // coreCoderDelay
378
    extension_flag = get_bits1(gb);
379

    
380
    if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
381
        ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
382
        skip_bits(gb, 3);     // layerNr
383

    
384
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
385
    if (channel_config == 0) {
386
        skip_bits(gb, 4);  // element_instance_tag
387
        if ((ret = decode_pce(ac, new_che_pos, gb)))
388
            return ret;
389
    } else {
390
        if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
391
            return ret;
392
    }
393
    if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
394
        return ret;
395

    
396
    if (extension_flag) {
397
        switch (ac->m4ac.object_type) {
398
        case AOT_ER_BSAC:
399
            skip_bits(gb, 5);    // numOfSubFrame
400
            skip_bits(gb, 11);   // layer_length
401
            break;
402
        case AOT_ER_AAC_LC:
403
        case AOT_ER_AAC_LTP:
404
        case AOT_ER_AAC_SCALABLE:
405
        case AOT_ER_AAC_LD:
406
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
407
                                    * aacScalefactorDataResilienceFlag
408
                                    * aacSpectralDataResilienceFlag
409
                                    */
410
            break;
411
        }
412
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
413
    }
414
    return 0;
415
}
416

    
417
/**
418
 * Decode audio specific configuration; reference: table 1.13.
419
 *
420
 * @param   data        pointer to AVCodecContext extradata
421
 * @param   data_size   size of AVCCodecContext extradata
422
 *
423
 * @return  Returns error status. 0 - OK, !0 - error
424
 */
425
static int decode_audio_specific_config(AACContext *ac, void *data,
426
                                        int data_size)
427
{
428
    GetBitContext gb;
429
    int i;
430

    
431
    init_get_bits(&gb, data, data_size * 8);
432

    
433
    if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
434
        return -1;
435
    if (ac->m4ac.sampling_index > 12) {
436
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
437
        return -1;
438
    }
439

    
440
    skip_bits_long(&gb, i);
441

    
442
    switch (ac->m4ac.object_type) {
443
    case AOT_AAC_MAIN:
444
    case AOT_AAC_LC:
445
        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
446
            return -1;
447
        break;
448
    default:
449
        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
450
               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
451
        return -1;
452
    }
453
    return 0;
454
}
455

    
456
/**
457
 * linear congruential pseudorandom number generator
458
 *
459
 * @param   previous_val    pointer to the current state of the generator
460
 *
461
 * @return  Returns a 32-bit pseudorandom integer
462
 */
463
static av_always_inline int lcg_random(int previous_val)
464
{
465
    return previous_val * 1664525 + 1013904223;
466
}
467

    
468
static void reset_predict_state(PredictorState *ps)
469
{
470
    ps->r0   = 0.0f;
471
    ps->r1   = 0.0f;
472
    ps->cor0 = 0.0f;
473
    ps->cor1 = 0.0f;
474
    ps->var0 = 1.0f;
475
    ps->var1 = 1.0f;
476
}
477

    
478
static void reset_all_predictors(PredictorState *ps)
479
{
480
    int i;
481
    for (i = 0; i < MAX_PREDICTORS; i++)
482
        reset_predict_state(&ps[i]);
483
}
484

    
485
static void reset_predictor_group(PredictorState *ps, int group_num)
486
{
487
    int i;
488
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
489
        reset_predict_state(&ps[i]);
490
}
491

    
492
static av_cold int aac_decode_init(AVCodecContext *avccontext)
493
{
494
    AACContext *ac = avccontext->priv_data;
495
    int i;
496

    
497
    ac->avccontext = avccontext;
498

    
499
    if (avccontext->extradata_size > 0) {
500
        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
501
            return -1;
502
        avccontext->sample_rate = ac->m4ac.sample_rate;
503
    } else if (avccontext->channels > 0) {
504
        ac->m4ac.sample_rate = avccontext->sample_rate;
505
    }
506

    
507
    avccontext->sample_fmt = SAMPLE_FMT_S16;
508
    avccontext->frame_size = 1024;
509

    
510
    AAC_INIT_VLC_STATIC( 0, 144);
511
    AAC_INIT_VLC_STATIC( 1, 114);
512
    AAC_INIT_VLC_STATIC( 2, 188);
513
    AAC_INIT_VLC_STATIC( 3, 180);
514
    AAC_INIT_VLC_STATIC( 4, 172);
515
    AAC_INIT_VLC_STATIC( 5, 140);
516
    AAC_INIT_VLC_STATIC( 6, 168);
517
    AAC_INIT_VLC_STATIC( 7, 114);
518
    AAC_INIT_VLC_STATIC( 8, 262);
519
    AAC_INIT_VLC_STATIC( 9, 248);
520
    AAC_INIT_VLC_STATIC(10, 384);
521

    
522
    dsputil_init(&ac->dsp, avccontext);
523

    
524
    ac->random_state = 0x1f2e3d4c;
525

    
526
    // -1024 - Compensate wrong IMDCT method.
527
    // 32768 - Required to scale values to the correct range for the bias method
528
    //         for float to int16 conversion.
529

    
530
    if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
531
        ac->add_bias  = 385.0f;
532
        ac->sf_scale  = 1. / (-1024. * 32768.);
533
        ac->sf_offset = 0;
534
    } else {
535
        ac->add_bias  = 0.0f;
536
        ac->sf_scale  = 1. / -1024.;
537
        ac->sf_offset = 60;
538
    }
539

    
540
#if !CONFIG_HARDCODED_TABLES
541
    for (i = 0; i < 428; i++)
542
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
543
#endif /* CONFIG_HARDCODED_TABLES */
544

    
545
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
546
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
547
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
548
                    352);
549

    
550
    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
551
    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
552
    // window initialization
553
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
554
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
555
    ff_sine_window_init(ff_sine_1024, 1024);
556
    ff_sine_window_init(ff_sine_128, 128);
557

    
558
    return 0;
559
}
560

    
561
/**
562
 * Skip data_stream_element; reference: table 4.10.
563
 */
564
static void skip_data_stream_element(GetBitContext *gb)
565
{
566
    int byte_align = get_bits1(gb);
567
    int count = get_bits(gb, 8);
568
    if (count == 255)
569
        count += get_bits(gb, 8);
570
    if (byte_align)
571
        align_get_bits(gb);
572
    skip_bits_long(gb, 8 * count);
573
}
574

    
575
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
576
                             GetBitContext *gb)
577
{
578
    int sfb;
579
    if (get_bits1(gb)) {
580
        ics->predictor_reset_group = get_bits(gb, 5);
581
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
582
            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
583
            return -1;
584
        }
585
    }
586
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
587
        ics->prediction_used[sfb] = get_bits1(gb);
588
    }
589
    return 0;
590
}
591

    
592
/**
593
 * Decode Individual Channel Stream info; reference: table 4.6.
594
 *
595
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
596
 */
597
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
598
                           GetBitContext *gb, int common_window)
599
{
600
    if (get_bits1(gb)) {
601
        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
602
        memset(ics, 0, sizeof(IndividualChannelStream));
603
        return -1;
604
    }
605
    ics->window_sequence[1] = ics->window_sequence[0];
606
    ics->window_sequence[0] = get_bits(gb, 2);
607
    ics->use_kb_window[1]   = ics->use_kb_window[0];
608
    ics->use_kb_window[0]   = get_bits1(gb);
609
    ics->num_window_groups  = 1;
610
    ics->group_len[0]       = 1;
611
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
612
        int i;
613
        ics->max_sfb = get_bits(gb, 4);
614
        for (i = 0; i < 7; i++) {
615
            if (get_bits1(gb)) {
616
                ics->group_len[ics->num_window_groups - 1]++;
617
            } else {
618
                ics->num_window_groups++;
619
                ics->group_len[ics->num_window_groups - 1] = 1;
620
            }
621
        }
622
        ics->num_windows       = 8;
623
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
624
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
625
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
626
        ics->predictor_present = 0;
627
    } else {
628
        ics->max_sfb               = get_bits(gb, 6);
629
        ics->num_windows           = 1;
630
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
631
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
632
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
633
        ics->predictor_present     = get_bits1(gb);
634
        ics->predictor_reset_group = 0;
635
        if (ics->predictor_present) {
636
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
637
                if (decode_prediction(ac, ics, gb)) {
638
                    memset(ics, 0, sizeof(IndividualChannelStream));
639
                    return -1;
640
                }
641
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
642
                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
643
                memset(ics, 0, sizeof(IndividualChannelStream));
644
                return -1;
645
            } else {
646
                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
647
                memset(ics, 0, sizeof(IndividualChannelStream));
648
                return -1;
649
            }
650
        }
651
    }
652

    
653
    if (ics->max_sfb > ics->num_swb) {
654
        av_log(ac->avccontext, AV_LOG_ERROR,
655
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
656
               ics->max_sfb, ics->num_swb);
657
        memset(ics, 0, sizeof(IndividualChannelStream));
658
        return -1;
659
    }
660

    
661
    return 0;
662
}
663

    
664
/**
665
 * Decode band types (section_data payload); reference: table 4.46.
666
 *
667
 * @param   band_type           array of the used band type
668
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
669
 *
670
 * @return  Returns error status. 0 - OK, !0 - error
671
 */
672
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
673
                             int band_type_run_end[120], GetBitContext *gb,
674
                             IndividualChannelStream *ics)
675
{
676
    int g, idx = 0;
677
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
678
    for (g = 0; g < ics->num_window_groups; g++) {
679
        int k = 0;
680
        while (k < ics->max_sfb) {
681
            uint8_t sect_end = k;
682
            int sect_len_incr;
683
            int sect_band_type = get_bits(gb, 4);
684
            if (sect_band_type == 12) {
685
                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
686
                return -1;
687
            }
688
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
689
                sect_end += sect_len_incr;
690
            sect_end += sect_len_incr;
691
            if (sect_end > ics->max_sfb) {
692
                av_log(ac->avccontext, AV_LOG_ERROR,
693
                       "Number of bands (%d) exceeds limit (%d).\n",
694
                       sect_end, ics->max_sfb);
695
                return -1;
696
            }
697
            for (; k < sect_end; k++) {
698
                band_type        [idx]   = sect_band_type;
699
                band_type_run_end[idx++] = sect_end;
700
            }
701
        }
702
    }
703
    return 0;
704
}
705

    
706
/**
707
 * Decode scalefactors; reference: table 4.47.
708
 *
709
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
710
 * @param   band_type           array of the used band type
711
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
712
 * @param   sf                  array of scalefactors or intensity stereo positions
713
 *
714
 * @return  Returns error status. 0 - OK, !0 - error
715
 */
716
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
717
                               unsigned int global_gain,
718
                               IndividualChannelStream *ics,
719
                               enum BandType band_type[120],
720
                               int band_type_run_end[120])
721
{
722
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
723
    int g, i, idx = 0;
724
    int offset[3] = { global_gain, global_gain - 90, 100 };
725
    int noise_flag = 1;
726
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
727
    for (g = 0; g < ics->num_window_groups; g++) {
728
        for (i = 0; i < ics->max_sfb;) {
729
            int run_end = band_type_run_end[idx];
730
            if (band_type[idx] == ZERO_BT) {
731
                for (; i < run_end; i++, idx++)
732
                    sf[idx] = 0.;
733
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
734
                for (; i < run_end; i++, idx++) {
735
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
736
                    if (offset[2] > 255U) {
737
                        av_log(ac->avccontext, AV_LOG_ERROR,
738
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
739
                        return -1;
740
                    }
741
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
742
                }
743
            } else if (band_type[idx] == NOISE_BT) {
744
                for (; i < run_end; i++, idx++) {
745
                    if (noise_flag-- > 0)
746
                        offset[1] += get_bits(gb, 9) - 256;
747
                    else
748
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
749
                    if (offset[1] > 255U) {
750
                        av_log(ac->avccontext, AV_LOG_ERROR,
751
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
752
                        return -1;
753
                    }
754
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
755
                }
756
            } else {
757
                for (; i < run_end; i++, idx++) {
758
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
759
                    if (offset[0] > 255U) {
760
                        av_log(ac->avccontext, AV_LOG_ERROR,
761
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
762
                        return -1;
763
                    }
764
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
765
                }
766
            }
767
        }
768
    }
769
    return 0;
770
}
771

    
772
/**
773
 * Decode pulse data; reference: table 4.7.
774
 */
775
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
776
                         const uint16_t *swb_offset, int num_swb)
777
{
778
    int i, pulse_swb;
779
    pulse->num_pulse = get_bits(gb, 2) + 1;
780
    pulse_swb        = get_bits(gb, 6);
781
    if (pulse_swb >= num_swb)
782
        return -1;
783
    pulse->pos[0]    = swb_offset[pulse_swb];
784
    pulse->pos[0]   += get_bits(gb, 5);
785
    if (pulse->pos[0] > 1023)
786
        return -1;
787
    pulse->amp[0]    = get_bits(gb, 4);
788
    for (i = 1; i < pulse->num_pulse; i++) {
789
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
790
        if (pulse->pos[i] > 1023)
791
            return -1;
792
        pulse->amp[i] = get_bits(gb, 4);
793
    }
794
    return 0;
795
}
796

    
797
/**
798
 * Decode Temporal Noise Shaping data; reference: table 4.48.
799
 *
800
 * @return  Returns error status. 0 - OK, !0 - error
801
 */
802
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
803
                      GetBitContext *gb, const IndividualChannelStream *ics)
804
{
805
    int w, filt, i, coef_len, coef_res, coef_compress;
806
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
807
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
808
    for (w = 0; w < ics->num_windows; w++) {
809
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
810
            coef_res = get_bits1(gb);
811

    
812
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
813
                int tmp2_idx;
814
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
815

    
816
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
817
                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
818
                           tns->order[w][filt], tns_max_order);
819
                    tns->order[w][filt] = 0;
820
                    return -1;
821
                }
822
                if (tns->order[w][filt]) {
823
                    tns->direction[w][filt] = get_bits1(gb);
824
                    coef_compress = get_bits1(gb);
825
                    coef_len = coef_res + 3 - coef_compress;
826
                    tmp2_idx = 2 * coef_compress + coef_res;
827

    
828
                    for (i = 0; i < tns->order[w][filt]; i++)
829
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
830
                }
831
            }
832
        }
833
    }
834
    return 0;
835
}
836

    
837
/**
838
 * Decode Mid/Side data; reference: table 4.54.
839
 *
840
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
841
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
842
 *                      [3] reserved for scalable AAC
843
 */
844
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
845
                                   int ms_present)
846
{
847
    int idx;
848
    if (ms_present == 1) {
849
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
850
            cpe->ms_mask[idx] = get_bits1(gb);
851
    } else if (ms_present == 2) {
852
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
853
    }
854
}
855

    
856
/**
857
 * Decode spectral data; reference: table 4.50.
858
 * Dequantize and scale spectral data; reference: 4.6.3.3.
859
 *
860
 * @param   coef            array of dequantized, scaled spectral data
861
 * @param   sf              array of scalefactors or intensity stereo positions
862
 * @param   pulse_present   set if pulses are present
863
 * @param   pulse           pointer to pulse data struct
864
 * @param   band_type       array of the used band type
865
 *
866
 * @return  Returns error status. 0 - OK, !0 - error
867
 */
868
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
869
                                       GetBitContext *gb, float sf[120],
870
                                       int pulse_present, const Pulse *pulse,
871
                                       const IndividualChannelStream *ics,
872
                                       enum BandType band_type[120])
873
{
874
    int i, k, g, idx = 0;
875
    const int c = 1024 / ics->num_windows;
876
    const uint16_t *offsets = ics->swb_offset;
877
    float *coef_base = coef;
878
    static const float sign_lookup[] = { 1.0f, -1.0f };
879

    
880
    for (g = 0; g < ics->num_windows; g++)
881
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
882

    
883
    for (g = 0; g < ics->num_window_groups; g++) {
884
        for (i = 0; i < ics->max_sfb; i++, idx++) {
885
            const int cur_band_type = band_type[idx];
886
            const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4;
887
            const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type);
888
            int group;
889
            if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) {
890
                for (group = 0; group < ics->group_len[g]; group++) {
891
                    memset(coef + group * 128 + offsets[i], 0, (offsets[i + 1] - offsets[i]) * sizeof(float));
892
                }
893
            } else if (cur_band_type == NOISE_BT) {
894
                for (group = 0; group < ics->group_len[g]; group++) {
895
                    float scale;
896
                    float band_energy;
897
                    float *cf = coef + group * 128 + offsets[i];
898
                    int len = offsets[i+1] - offsets[i];
899

    
900
                    for (k = 0; k < len; k++) {
901
                        ac->random_state  = lcg_random(ac->random_state);
902
                        cf[k] = ac->random_state;
903
                    }
904

    
905
                    band_energy = ac->dsp.scalarproduct_float(cf, cf, len);
906
                    scale = sf[idx] / sqrtf(band_energy);
907
                    ac->dsp.vector_fmul_scalar(cf, cf, scale, len);
908
                }
909
            } else {
910
                for (group = 0; group < ics->group_len[g]; group++) {
911
                    const float *vq[96];
912
                    const float **vqp = vq;
913
                    float *cf = coef + (group << 7) + offsets[i];
914
                    int len = offsets[i + 1] - offsets[i];
915

    
916
                    for (k = offsets[i]; k < offsets[i + 1]; k += dim) {
917
                        const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3);
918
                        const int coef_tmp_idx = (group << 7) + k;
919
                        const float *vq_ptr;
920
                        int j;
921
                        if (index >= ff_aac_spectral_sizes[cur_band_type - 1]) {
922
                            av_log(ac->avccontext, AV_LOG_ERROR,
923
                                   "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
924
                                   cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]);
925
                            return -1;
926
                        }
927
                        vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
928
                        *vqp++ = vq_ptr;
929
                        if (is_cb_unsigned) {
930
                            if (vq_ptr[0])
931
                                coef[coef_tmp_idx    ] = sign_lookup[get_bits1(gb)];
932
                            if (vq_ptr[1])
933
                                coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)];
934
                            if (dim == 4) {
935
                                if (vq_ptr[2])
936
                                    coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)];
937
                                if (vq_ptr[3])
938
                                    coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)];
939
                            }
940
                            if (cur_band_type == ESC_BT) {
941
                                for (j = 0; j < 2; j++) {
942
                                    if (vq_ptr[j] == 64.0f) {
943
                                        int n = 4;
944
                                        /* The total length of escape_sequence must be < 22 bits according
945
                                           to the specification (i.e. max is 11111111110xxxxxxxxxx). */
946
                                        while (get_bits1(gb) && n < 15) n++;
947
                                        if (n == 15) {
948
                                            av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
949
                                            return -1;
950
                                        }
951
                                        n = (1 << n) + get_bits(gb, n);
952
                                        coef[coef_tmp_idx + j] *= cbrtf(n) * n;
953
                                    } else
954
                                        coef[coef_tmp_idx + j] *= vq_ptr[j];
955
                                }
956
                            }
957
                        }
958
                    }
959

    
960
                    if (is_cb_unsigned && cur_band_type != ESC_BT) {
961
                        ac->dsp.vector_fmul_sv_scalar[dim>>2](
962
                            cf, cf, vq, sf[idx], len);
963
                    } else if (cur_band_type == ESC_BT) {
964
                        ac->dsp.vector_fmul_scalar(cf, cf, sf[idx], len);
965
                    } else {    /* !is_cb_unsigned */
966
                        ac->dsp.sv_fmul_scalar[dim>>2](cf, vq, sf[idx], len);
967
                    }
968
                }
969
            }
970
        }
971
        coef += ics->group_len[g] << 7;
972
    }
973

    
974
    if (pulse_present) {
975
        idx = 0;
976
        for (i = 0; i < pulse->num_pulse; i++) {
977
            float co = coef_base[ pulse->pos[i] ];
978
            while (offsets[idx + 1] <= pulse->pos[i])
979
                idx++;
980
            if (band_type[idx] != NOISE_BT && sf[idx]) {
981
                float ico = -pulse->amp[i];
982
                if (co) {
983
                    co /= sf[idx];
984
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
985
                }
986
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
987
            }
988
        }
989
    }
990
    return 0;
991
}
992

    
993
static av_always_inline float flt16_round(float pf)
994
{
995
    union float754 tmp;
996
    tmp.f = pf;
997
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
998
    return tmp.f;
999
}
1000

    
1001
static av_always_inline float flt16_even(float pf)
1002
{
1003
    union float754 tmp;
1004
    tmp.f = pf;
1005
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1006
    return tmp.f;
1007
}
1008

    
1009
static av_always_inline float flt16_trunc(float pf)
1010
{
1011
    union float754 pun;
1012
    pun.f = pf;
1013
    pun.i &= 0xFFFF0000U;
1014
    return pun.f;
1015
}
1016

    
1017
static void predict(AACContext *ac, PredictorState *ps, float *coef,
1018
                    int output_enable)
1019
{
1020
    const float a     = 0.953125; // 61.0 / 64
1021
    const float alpha = 0.90625;  // 29.0 / 32
1022
    float e0, e1;
1023
    float pv;
1024
    float k1, k2;
1025

    
1026
    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
1027
    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
1028

    
1029
    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
1030
    if (output_enable)
1031
        *coef += pv * ac->sf_scale;
1032

    
1033
    e0 = *coef / ac->sf_scale;
1034
    e1 = e0 - k1 * ps->r0;
1035

    
1036
    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
1037
    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
1038
    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
1039
    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
1040

    
1041
    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
1042
    ps->r0 = flt16_trunc(a * e0);
1043
}
1044

    
1045
/**
1046
 * Apply AAC-Main style frequency domain prediction.
1047
 */
1048
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1049
{
1050
    int sfb, k;
1051

    
1052
    if (!sce->ics.predictor_initialized) {
1053
        reset_all_predictors(sce->predictor_state);
1054
        sce->ics.predictor_initialized = 1;
1055
    }
1056

    
1057
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1058
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1059
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1060
                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
1061
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1062
            }
1063
        }
1064
        if (sce->ics.predictor_reset_group)
1065
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1066
    } else
1067
        reset_all_predictors(sce->predictor_state);
1068
}
1069

    
1070
/**
1071
 * Decode an individual_channel_stream payload; reference: table 4.44.
1072
 *
1073
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1074
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1075
 *
1076
 * @return  Returns error status. 0 - OK, !0 - error
1077
 */
1078
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1079
                      GetBitContext *gb, int common_window, int scale_flag)
1080
{
1081
    Pulse pulse;
1082
    TemporalNoiseShaping    *tns = &sce->tns;
1083
    IndividualChannelStream *ics = &sce->ics;
1084
    float *out = sce->coeffs;
1085
    int global_gain, pulse_present = 0;
1086

    
1087
    /* This assignment is to silence a GCC warning about the variable being used
1088
     * uninitialized when in fact it always is.
1089
     */
1090
    pulse.num_pulse = 0;
1091

    
1092
    global_gain = get_bits(gb, 8);
1093

    
1094
    if (!common_window && !scale_flag) {
1095
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1096
            return -1;
1097
    }
1098

    
1099
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1100
        return -1;
1101
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1102
        return -1;
1103

    
1104
    pulse_present = 0;
1105
    if (!scale_flag) {
1106
        if ((pulse_present = get_bits1(gb))) {
1107
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1108
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1109
                return -1;
1110
            }
1111
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1112
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1113
                return -1;
1114
            }
1115
        }
1116
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1117
            return -1;
1118
        if (get_bits1(gb)) {
1119
            av_log_missing_feature(ac->avccontext, "SSR", 1);
1120
            return -1;
1121
        }
1122
    }
1123

    
1124
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1125
        return -1;
1126

    
1127
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1128
        apply_prediction(ac, sce);
1129

    
1130
    return 0;
1131
}
1132

    
1133
/**
1134
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1135
 */
1136
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1137
{
1138
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1139
    float *ch0 = cpe->ch[0].coeffs;
1140
    float *ch1 = cpe->ch[1].coeffs;
1141
    int g, i, group, idx = 0;
1142
    const uint16_t *offsets = ics->swb_offset;
1143
    for (g = 0; g < ics->num_window_groups; g++) {
1144
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1145
            if (cpe->ms_mask[idx] &&
1146
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1147
                for (group = 0; group < ics->group_len[g]; group++) {
1148
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1149
                                              ch1 + group * 128 + offsets[i],
1150
                                              offsets[i+1] - offsets[i]);
1151
                }
1152
            }
1153
        }
1154
        ch0 += ics->group_len[g] * 128;
1155
        ch1 += ics->group_len[g] * 128;
1156
    }
1157
}
1158

    
1159
/**
1160
 * intensity stereo decoding; reference: 4.6.8.2.3
1161
 *
1162
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1163
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1164
 *                      [3] reserved for scalable AAC
1165
 */
1166
static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
1167
{
1168
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1169
    SingleChannelElement         *sce1 = &cpe->ch[1];
1170
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1171
    const uint16_t *offsets = ics->swb_offset;
1172
    int g, group, i, k, idx = 0;
1173
    int c;
1174
    float scale;
1175
    for (g = 0; g < ics->num_window_groups; g++) {
1176
        for (i = 0; i < ics->max_sfb;) {
1177
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1178
                const int bt_run_end = sce1->band_type_run_end[idx];
1179
                for (; i < bt_run_end; i++, idx++) {
1180
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1181
                    if (ms_present)
1182
                        c *= 1 - 2 * cpe->ms_mask[idx];
1183
                    scale = c * sce1->sf[idx];
1184
                    for (group = 0; group < ics->group_len[g]; group++)
1185
                        for (k = offsets[i]; k < offsets[i + 1]; k++)
1186
                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
1187
                }
1188
            } else {
1189
                int bt_run_end = sce1->band_type_run_end[idx];
1190
                idx += bt_run_end - i;
1191
                i    = bt_run_end;
1192
            }
1193
        }
1194
        coef0 += ics->group_len[g] * 128;
1195
        coef1 += ics->group_len[g] * 128;
1196
    }
1197
}
1198

    
1199
/**
1200
 * Decode a channel_pair_element; reference: table 4.4.
1201
 *
1202
 * @param   elem_id Identifies the instance of a syntax element.
1203
 *
1204
 * @return  Returns error status. 0 - OK, !0 - error
1205
 */
1206
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1207
{
1208
    int i, ret, common_window, ms_present = 0;
1209

    
1210
    common_window = get_bits1(gb);
1211
    if (common_window) {
1212
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1213
            return -1;
1214
        i = cpe->ch[1].ics.use_kb_window[0];
1215
        cpe->ch[1].ics = cpe->ch[0].ics;
1216
        cpe->ch[1].ics.use_kb_window[1] = i;
1217
        ms_present = get_bits(gb, 2);
1218
        if (ms_present == 3) {
1219
            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1220
            return -1;
1221
        } else if (ms_present)
1222
            decode_mid_side_stereo(cpe, gb, ms_present);
1223
    }
1224
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1225
        return ret;
1226
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1227
        return ret;
1228

    
1229
    if (common_window) {
1230
        if (ms_present)
1231
            apply_mid_side_stereo(ac, cpe);
1232
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1233
            apply_prediction(ac, &cpe->ch[0]);
1234
            apply_prediction(ac, &cpe->ch[1]);
1235
        }
1236
    }
1237

    
1238
    apply_intensity_stereo(cpe, ms_present);
1239
    return 0;
1240
}
1241

    
1242
/**
1243
 * Decode coupling_channel_element; reference: table 4.8.
1244
 *
1245
 * @param   elem_id Identifies the instance of a syntax element.
1246
 *
1247
 * @return  Returns error status. 0 - OK, !0 - error
1248
 */
1249
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1250
{
1251
    int num_gain = 0;
1252
    int c, g, sfb, ret;
1253
    int sign;
1254
    float scale;
1255
    SingleChannelElement *sce = &che->ch[0];
1256
    ChannelCoupling     *coup = &che->coup;
1257

    
1258
    coup->coupling_point = 2 * get_bits1(gb);
1259
    coup->num_coupled = get_bits(gb, 3);
1260
    for (c = 0; c <= coup->num_coupled; c++) {
1261
        num_gain++;
1262
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1263
        coup->id_select[c] = get_bits(gb, 4);
1264
        if (coup->type[c] == TYPE_CPE) {
1265
            coup->ch_select[c] = get_bits(gb, 2);
1266
            if (coup->ch_select[c] == 3)
1267
                num_gain++;
1268
        } else
1269
            coup->ch_select[c] = 2;
1270
    }
1271
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1272

    
1273
    sign  = get_bits(gb, 1);
1274
    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1275

    
1276
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1277
        return ret;
1278

    
1279
    for (c = 0; c < num_gain; c++) {
1280
        int idx  = 0;
1281
        int cge  = 1;
1282
        int gain = 0;
1283
        float gain_cache = 1.;
1284
        if (c) {
1285
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1286
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1287
            gain_cache = pow(scale, -gain);
1288
        }
1289
        if (coup->coupling_point == AFTER_IMDCT) {
1290
            coup->gain[c][0] = gain_cache;
1291
        } else {
1292
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1293
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1294
                    if (sce->band_type[idx] != ZERO_BT) {
1295
                        if (!cge) {
1296
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1297
                            if (t) {
1298
                                int s = 1;
1299
                                t = gain += t;
1300
                                if (sign) {
1301
                                    s  -= 2 * (t & 0x1);
1302
                                    t >>= 1;
1303
                                }
1304
                                gain_cache = pow(scale, -t) * s;
1305
                            }
1306
                        }
1307
                        coup->gain[c][idx] = gain_cache;
1308
                    }
1309
                }
1310
            }
1311
        }
1312
    }
1313
    return 0;
1314
}
1315

    
1316
/**
1317
 * Decode Spectral Band Replication extension data; reference: table 4.55.
1318
 *
1319
 * @param   crc flag indicating the presence of CRC checksum
1320
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1321
 *
1322
 * @return  Returns number of bytes consumed from the TYPE_FIL element.
1323
 */
1324
static int decode_sbr_extension(AACContext *ac, GetBitContext *gb,
1325
                                int crc, int cnt)
1326
{
1327
    // TODO : sbr_extension implementation
1328
    av_log_missing_feature(ac->avccontext, "SBR", 0);
1329
    skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type
1330
    return cnt;
1331
}
1332

    
1333
/**
1334
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1335
 *
1336
 * @return  Returns number of bytes consumed.
1337
 */
1338
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1339
                                         GetBitContext *gb)
1340
{
1341
    int i;
1342
    int num_excl_chan = 0;
1343

    
1344
    do {
1345
        for (i = 0; i < 7; i++)
1346
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1347
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1348

    
1349
    return num_excl_chan / 7;
1350
}
1351

    
1352
/**
1353
 * Decode dynamic range information; reference: table 4.52.
1354
 *
1355
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1356
 *
1357
 * @return  Returns number of bytes consumed.
1358
 */
1359
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1360
                                GetBitContext *gb, int cnt)
1361
{
1362
    int n             = 1;
1363
    int drc_num_bands = 1;
1364
    int i;
1365

    
1366
    /* pce_tag_present? */
1367
    if (get_bits1(gb)) {
1368
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1369
        skip_bits(gb, 4); // tag_reserved_bits
1370
        n++;
1371
    }
1372

    
1373
    /* excluded_chns_present? */
1374
    if (get_bits1(gb)) {
1375
        n += decode_drc_channel_exclusions(che_drc, gb);
1376
    }
1377

    
1378
    /* drc_bands_present? */
1379
    if (get_bits1(gb)) {
1380
        che_drc->band_incr            = get_bits(gb, 4);
1381
        che_drc->interpolation_scheme = get_bits(gb, 4);
1382
        n++;
1383
        drc_num_bands += che_drc->band_incr;
1384
        for (i = 0; i < drc_num_bands; i++) {
1385
            che_drc->band_top[i] = get_bits(gb, 8);
1386
            n++;
1387
        }
1388
    }
1389

    
1390
    /* prog_ref_level_present? */
1391
    if (get_bits1(gb)) {
1392
        che_drc->prog_ref_level = get_bits(gb, 7);
1393
        skip_bits1(gb); // prog_ref_level_reserved_bits
1394
        n++;
1395
    }
1396

    
1397
    for (i = 0; i < drc_num_bands; i++) {
1398
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1399
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1400
        n++;
1401
    }
1402

    
1403
    return n;
1404
}
1405

    
1406
/**
1407
 * Decode extension data (incomplete); reference: table 4.51.
1408
 *
1409
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1410
 *
1411
 * @return Returns number of bytes consumed
1412
 */
1413
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt)
1414
{
1415
    int crc_flag = 0;
1416
    int res = cnt;
1417
    switch (get_bits(gb, 4)) { // extension type
1418
    case EXT_SBR_DATA_CRC:
1419
        crc_flag++;
1420
    case EXT_SBR_DATA:
1421
        res = decode_sbr_extension(ac, gb, crc_flag, cnt);
1422
        break;
1423
    case EXT_DYNAMIC_RANGE:
1424
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1425
        break;
1426
    case EXT_FILL:
1427
    case EXT_FILL_DATA:
1428
    case EXT_DATA_ELEMENT:
1429
    default:
1430
        skip_bits_long(gb, 8 * cnt - 4);
1431
        break;
1432
    };
1433
    return res;
1434
}
1435

    
1436
/**
1437
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1438
 *
1439
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1440
 * @param   coef    spectral coefficients
1441
 */
1442
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1443
                      IndividualChannelStream *ics, int decode)
1444
{
1445
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1446
    int w, filt, m, i;
1447
    int bottom, top, order, start, end, size, inc;
1448
    float lpc[TNS_MAX_ORDER];
1449

    
1450
    for (w = 0; w < ics->num_windows; w++) {
1451
        bottom = ics->num_swb;
1452
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1453
            top    = bottom;
1454
            bottom = FFMAX(0, top - tns->length[w][filt]);
1455
            order  = tns->order[w][filt];
1456
            if (order == 0)
1457
                continue;
1458

    
1459
            // tns_decode_coef
1460
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1461

    
1462
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1463
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1464
            if ((size = end - start) <= 0)
1465
                continue;
1466
            if (tns->direction[w][filt]) {
1467
                inc = -1;
1468
                start = end - 1;
1469
            } else {
1470
                inc = 1;
1471
            }
1472
            start += w * 128;
1473

    
1474
            // ar filter
1475
            for (m = 0; m < size; m++, start += inc)
1476
                for (i = 1; i <= FFMIN(m, order); i++)
1477
                    coef[start] -= coef[start - i * inc] * lpc[i - 1];
1478
        }
1479
    }
1480
}
1481

    
1482
/**
1483
 * Conduct IMDCT and windowing.
1484
 */
1485
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1486
{
1487
    IndividualChannelStream *ics = &sce->ics;
1488
    float *in    = sce->coeffs;
1489
    float *out   = sce->ret;
1490
    float *saved = sce->saved;
1491
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1492
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1493
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1494
    float *buf  = ac->buf_mdct;
1495
    float *temp = ac->temp;
1496
    int i;
1497

    
1498
    // imdct
1499
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1500
        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1501
            av_log(ac->avccontext, AV_LOG_WARNING,
1502
                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1503
                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1504
        for (i = 0; i < 1024; i += 128)
1505
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1506
    } else
1507
        ff_imdct_half(&ac->mdct, buf, in);
1508

    
1509
    /* window overlapping
1510
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1511
     * and long to short transitions are considered to be short to short
1512
     * transitions. This leaves just two cases (long to long and short to short)
1513
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1514
     */
1515
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1516
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1517
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, ac->add_bias, 512);
1518
    } else {
1519
        for (i = 0; i < 448; i++)
1520
            out[i] = saved[i] + ac->add_bias;
1521

    
1522
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1523
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, ac->add_bias, 64);
1524
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      ac->add_bias, 64);
1525
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      ac->add_bias, 64);
1526
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      ac->add_bias, 64);
1527
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      ac->add_bias, 64);
1528
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1529
        } else {
1530
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, ac->add_bias, 64);
1531
            for (i = 576; i < 1024; i++)
1532
                out[i] = buf[i-512] + ac->add_bias;
1533
        }
1534
    }
1535

    
1536
    // buffer update
1537
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1538
        for (i = 0; i < 64; i++)
1539
            saved[i] = temp[64 + i] - ac->add_bias;
1540
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1541
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1542
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1543
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1544
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1545
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1546
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1547
    } else { // LONG_STOP or ONLY_LONG
1548
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1549
    }
1550
}
1551

    
1552
/**
1553
 * Apply dependent channel coupling (applied before IMDCT).
1554
 *
1555
 * @param   index   index into coupling gain array
1556
 */
1557
static void apply_dependent_coupling(AACContext *ac,
1558
                                     SingleChannelElement *target,
1559
                                     ChannelElement *cce, int index)
1560
{
1561
    IndividualChannelStream *ics = &cce->ch[0].ics;
1562
    const uint16_t *offsets = ics->swb_offset;
1563
    float *dest = target->coeffs;
1564
    const float *src = cce->ch[0].coeffs;
1565
    int g, i, group, k, idx = 0;
1566
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1567
        av_log(ac->avccontext, AV_LOG_ERROR,
1568
               "Dependent coupling is not supported together with LTP\n");
1569
        return;
1570
    }
1571
    for (g = 0; g < ics->num_window_groups; g++) {
1572
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1573
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1574
                const float gain = cce->coup.gain[index][idx];
1575
                for (group = 0; group < ics->group_len[g]; group++) {
1576
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1577
                        // XXX dsputil-ize
1578
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1579
                    }
1580
                }
1581
            }
1582
        }
1583
        dest += ics->group_len[g] * 128;
1584
        src  += ics->group_len[g] * 128;
1585
    }
1586
}
1587

    
1588
/**
1589
 * Apply independent channel coupling (applied after IMDCT).
1590
 *
1591
 * @param   index   index into coupling gain array
1592
 */
1593
static void apply_independent_coupling(AACContext *ac,
1594
                                       SingleChannelElement *target,
1595
                                       ChannelElement *cce, int index)
1596
{
1597
    int i;
1598
    const float gain = cce->coup.gain[index][0];
1599
    const float bias = ac->add_bias;
1600
    const float *src = cce->ch[0].ret;
1601
    float *dest = target->ret;
1602

    
1603
    for (i = 0; i < 1024; i++)
1604
        dest[i] += gain * (src[i] - bias);
1605
}
1606

    
1607
/**
1608
 * channel coupling transformation interface
1609
 *
1610
 * @param   index   index into coupling gain array
1611
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1612
 */
1613
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1614
                                   enum RawDataBlockType type, int elem_id,
1615
                                   enum CouplingPoint coupling_point,
1616
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1617
{
1618
    int i, c;
1619

    
1620
    for (i = 0; i < MAX_ELEM_ID; i++) {
1621
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1622
        int index = 0;
1623

    
1624
        if (cce && cce->coup.coupling_point == coupling_point) {
1625
            ChannelCoupling *coup = &cce->coup;
1626

    
1627
            for (c = 0; c <= coup->num_coupled; c++) {
1628
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1629
                    if (coup->ch_select[c] != 1) {
1630
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1631
                        if (coup->ch_select[c] != 0)
1632
                            index++;
1633
                    }
1634
                    if (coup->ch_select[c] != 2)
1635
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1636
                } else
1637
                    index += 1 + (coup->ch_select[c] == 3);
1638
            }
1639
        }
1640
    }
1641
}
1642

    
1643
/**
1644
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1645
 */
1646
static void spectral_to_sample(AACContext *ac)
1647
{
1648
    int i, type;
1649
    for (type = 3; type >= 0; type--) {
1650
        for (i = 0; i < MAX_ELEM_ID; i++) {
1651
            ChannelElement *che = ac->che[type][i];
1652
            if (che) {
1653
                if (type <= TYPE_CPE)
1654
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1655
                if (che->ch[0].tns.present)
1656
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1657
                if (che->ch[1].tns.present)
1658
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1659
                if (type <= TYPE_CPE)
1660
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1661
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
1662
                    imdct_and_windowing(ac, &che->ch[0]);
1663
                if (type == TYPE_CPE)
1664
                    imdct_and_windowing(ac, &che->ch[1]);
1665
                if (type <= TYPE_CCE)
1666
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1667
            }
1668
        }
1669
    }
1670
}
1671

    
1672
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
1673
{
1674
    int size;
1675
    AACADTSHeaderInfo hdr_info;
1676

    
1677
    size = ff_aac_parse_header(gb, &hdr_info);
1678
    if (size > 0) {
1679
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
1680
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1681
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1682
            ac->m4ac.chan_config = hdr_info.chan_config;
1683
            if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
1684
                return -7;
1685
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
1686
                return -7;
1687
        } else if (ac->output_configured != OC_LOCKED) {
1688
            ac->output_configured = OC_NONE;
1689
        }
1690
        if (ac->output_configured != OC_LOCKED)
1691
            ac->m4ac.sbr = -1;
1692
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1693
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1694
        ac->m4ac.object_type     = hdr_info.object_type;
1695
        if (!ac->avccontext->sample_rate)
1696
            ac->avccontext->sample_rate = hdr_info.sample_rate;
1697
        if (hdr_info.num_aac_frames == 1) {
1698
            if (!hdr_info.crc_absent)
1699
                skip_bits(gb, 16);
1700
        } else {
1701
            av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1702
            return -1;
1703
        }
1704
    }
1705
    return size;
1706
}
1707

    
1708
static int aac_decode_frame(AVCodecContext *avccontext, void *data,
1709
                            int *data_size, AVPacket *avpkt)
1710
{
1711
    const uint8_t *buf = avpkt->data;
1712
    int buf_size = avpkt->size;
1713
    AACContext *ac = avccontext->priv_data;
1714
    ChannelElement *che = NULL;
1715
    GetBitContext gb;
1716
    enum RawDataBlockType elem_type;
1717
    int err, elem_id, data_size_tmp;
1718

    
1719
    init_get_bits(&gb, buf, buf_size * 8);
1720

    
1721
    if (show_bits(&gb, 12) == 0xfff) {
1722
        if (parse_adts_frame_header(ac, &gb) < 0) {
1723
            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1724
            return -1;
1725
        }
1726
        if (ac->m4ac.sampling_index > 12) {
1727
            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1728
            return -1;
1729
        }
1730
    }
1731

    
1732
    // parse
1733
    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1734
        elem_id = get_bits(&gb, 4);
1735

    
1736
        if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
1737
            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1738
            return -1;
1739
        }
1740

    
1741
        switch (elem_type) {
1742

    
1743
        case TYPE_SCE:
1744
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1745
            break;
1746

    
1747
        case TYPE_CPE:
1748
            err = decode_cpe(ac, &gb, che);
1749
            break;
1750

    
1751
        case TYPE_CCE:
1752
            err = decode_cce(ac, &gb, che);
1753
            break;
1754

    
1755
        case TYPE_LFE:
1756
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1757
            break;
1758

    
1759
        case TYPE_DSE:
1760
            skip_data_stream_element(&gb);
1761
            err = 0;
1762
            break;
1763

    
1764
        case TYPE_PCE: {
1765
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1766
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1767
            if ((err = decode_pce(ac, new_che_pos, &gb)))
1768
                break;
1769
            if (ac->output_configured > OC_TRIAL_PCE)
1770
                av_log(avccontext, AV_LOG_ERROR,
1771
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
1772
            else
1773
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
1774
            break;
1775
        }
1776

    
1777
        case TYPE_FIL:
1778
            if (elem_id == 15)
1779
                elem_id += get_bits(&gb, 8) - 1;
1780
            while (elem_id > 0)
1781
                elem_id -= decode_extension_payload(ac, &gb, elem_id);
1782
            err = 0; /* FIXME */
1783
            break;
1784

    
1785
        default:
1786
            err = -1; /* should not happen, but keeps compiler happy */
1787
            break;
1788
        }
1789

    
1790
        if (err)
1791
            return err;
1792
    }
1793

    
1794
    spectral_to_sample(ac);
1795

    
1796
    if (!ac->is_saved) {
1797
        ac->is_saved = 1;
1798
        *data_size = 0;
1799
        return buf_size;
1800
    }
1801

    
1802
    data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
1803
    if (*data_size < data_size_tmp) {
1804
        av_log(avccontext, AV_LOG_ERROR,
1805
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
1806
               *data_size, data_size_tmp);
1807
        return -1;
1808
    }
1809
    *data_size = data_size_tmp;
1810

    
1811
    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
1812

    
1813
    if (ac->output_configured)
1814
        ac->output_configured = OC_LOCKED;
1815

    
1816
    return buf_size;
1817
}
1818

    
1819
static av_cold int aac_decode_close(AVCodecContext *avccontext)
1820
{
1821
    AACContext *ac = avccontext->priv_data;
1822
    int i, type;
1823

    
1824
    for (i = 0; i < MAX_ELEM_ID; i++) {
1825
        for (type = 0; type < 4; type++)
1826
            av_freep(&ac->che[type][i]);
1827
    }
1828

    
1829
    ff_mdct_end(&ac->mdct);
1830
    ff_mdct_end(&ac->mdct_small);
1831
    return 0;
1832
}
1833

    
1834
AVCodec aac_decoder = {
1835
    "aac",
1836
    CODEC_TYPE_AUDIO,
1837
    CODEC_ID_AAC,
1838
    sizeof(AACContext),
1839
    aac_decode_init,
1840
    NULL,
1841
    aac_decode_close,
1842
    aac_decode_frame,
1843
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
1844
    .sample_fmts = (const enum SampleFormat[]) {
1845
        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
1846
    },
1847
    .channel_layouts = aac_channel_layout,
1848
};