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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
 *
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 * 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
 *
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 * FFmpeg is distributed in the hope that it will be useful,
14
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * 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
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 * 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
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 * Y                    filterbank - standard
39
 * N (code in SoC repo) filterbank - Scalable Sample Rate
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 * Y                    Temporal Noise Shaping
41
 * N (code in SoC repo) Long Term Prediction
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 * Y                    intensity stereo
43
 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * Y                    Perceptual Noise Substitution
46
 * Y                    Mid/Side stereo
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 * N                    Scalable Inverse AAC Quantization
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 * N                    Frequency Selective Switch
49
 * N                    upsampling filter
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 * Y                    quantization & coding - AAC
51
 * N                    quantization & coding - TwinVQ
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 * N                    quantization & coding - BSAC
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 * N                    AAC Error Resilience tools
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 * N                    Error Resilience payload syntax
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 * N                    Error Protection tool
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 * N                    CELP
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 * N                    Silence Compression
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 * N                    HVXC
59
 * N                    HVXC 4kbits/s VR
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 * N                    Structured Audio tools
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 * 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
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 * Y (not in this code) Layer-3
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 * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
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 * N (planned)          Parametric Stereo
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 * N                    Direct Stream Transfer
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 *
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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
 * Configure output channel order based on the current program configuration element.
157
 *
158
 * @param   che_pos current channel position configuration
159
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
160
 *
161
 * @return  Returns error status. 0 - OK, !0 - error
162
 */
163
static int output_configure(AACContext *ac,
164
                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
165
                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
166
                            int channel_config)
167
{
168
    AVCodecContext *avctx = ac->avccontext;
169
    int i, type, channels = 0;
170

    
171
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
172

    
173
    if (channel_config) {
174
        for (i = 0; i < tags_per_config[channel_config]; i++) {
175
            const int id = aac_channel_layout_map[channel_config - 1][i][1];
176
            type         = aac_channel_layout_map[channel_config - 1][i][0];
177

    
178
            if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
179
                return AVERROR(ENOMEM);
180

    
181
            if (type != TYPE_CCE) {
182
                ac->output_data[channels++] = ac->che[type][id]->ch[0].ret;
183
                if (type == TYPE_CPE)
184
                    ac->output_data[channels++] = ac->che[type][id]->ch[1].ret;
185
            }
186
        }
187

    
188
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
189
        ac->tags_mapped = 0;
190

    
191
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
192
    } else {
193
        /* Allocate or free elements depending on if they are in the
194
         * current program configuration.
195
         *
196
         * Set up default 1:1 output mapping.
197
         *
198
         * For a 5.1 stream the output order will be:
199
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
200
         */
201

    
202
        for (i = 0; i < MAX_ELEM_ID; i++) {
203
            for (type = 0; type < 4; type++) {
204
                if (che_pos[type][i]) {
205
                    if (!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement))))
206
                        return AVERROR(ENOMEM);
207
                    if (type != TYPE_CCE) {
208
                        ac->output_data[channels++] = ac->che[type][i]->ch[0].ret;
209
                        if (type == TYPE_CPE) {
210
                            ac->output_data[channels++] = ac->che[type][i]->ch[1].ret;
211
                        }
212
                    }
213
                } else
214
                    av_freep(&ac->che[type][i]);
215
            }
216
        }
217

    
218
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
219
        ac->tags_mapped = 4 * MAX_ELEM_ID;
220

    
221
        avctx->channel_layout = 0;
222
    }
223

    
224
    avctx->channels = channels;
225

    
226
    ac->output_configured = 1;
227

    
228
    return 0;
229
}
230

    
231
/**
232
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
233
 *
234
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
235
 * @param sce_map mono (Single Channel Element) map
236
 * @param type speaker type/position for these channels
237
 */
238
static void decode_channel_map(enum ChannelPosition *cpe_map,
239
                               enum ChannelPosition *sce_map,
240
                               enum ChannelPosition type,
241
                               GetBitContext *gb, int n)
242
{
243
    while (n--) {
244
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
245
        map[get_bits(gb, 4)] = type;
246
    }
247
}
248

    
249
/**
250
 * Decode program configuration element; reference: table 4.2.
251
 *
252
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
253
 *
254
 * @return  Returns error status. 0 - OK, !0 - error
255
 */
256
static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
257
                      GetBitContext *gb)
258
{
259
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
260

    
261
    skip_bits(gb, 2);  // object_type
262

    
263
    sampling_index = get_bits(gb, 4);
264
    if (ac->m4ac.sampling_index != sampling_index)
265
        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");
266

    
267
    num_front       = get_bits(gb, 4);
268
    num_side        = get_bits(gb, 4);
269
    num_back        = get_bits(gb, 4);
270
    num_lfe         = get_bits(gb, 2);
271
    num_assoc_data  = get_bits(gb, 3);
272
    num_cc          = get_bits(gb, 4);
273

    
274
    if (get_bits1(gb))
275
        skip_bits(gb, 4); // mono_mixdown_tag
276
    if (get_bits1(gb))
277
        skip_bits(gb, 4); // stereo_mixdown_tag
278

    
279
    if (get_bits1(gb))
280
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
281

    
282
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
283
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
284
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
285
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
286

    
287
    skip_bits_long(gb, 4 * num_assoc_data);
288

    
289
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
290

    
291
    align_get_bits(gb);
292

    
293
    /* comment field, first byte is length */
294
    skip_bits_long(gb, 8 * get_bits(gb, 8));
295
    return 0;
296
}
297

    
298
/**
299
 * Set up channel positions based on a default channel configuration
300
 * as specified in table 1.17.
301
 *
302
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
303
 *
304
 * @return  Returns error status. 0 - OK, !0 - error
305
 */
306
static int set_default_channel_config(AACContext *ac,
307
                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
308
                                      int channel_config)
309
{
310
    if (channel_config < 1 || channel_config > 7) {
311
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
312
               channel_config);
313
        return -1;
314
    }
315

    
316
    /* default channel configurations:
317
     *
318
     * 1ch : front center (mono)
319
     * 2ch : L + R (stereo)
320
     * 3ch : front center + L + R
321
     * 4ch : front center + L + R + back center
322
     * 5ch : front center + L + R + back stereo
323
     * 6ch : front center + L + R + back stereo + LFE
324
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
325
     */
326

    
327
    if (channel_config != 2)
328
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
329
    if (channel_config > 1)
330
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
331
    if (channel_config == 4)
332
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
333
    if (channel_config > 4)
334
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
335
        = AAC_CHANNEL_BACK;  // back stereo
336
    if (channel_config > 5)
337
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
338
    if (channel_config == 7)
339
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
340

    
341
    return 0;
342
}
343

    
344
/**
345
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
346
 *
347
 * @return  Returns error status. 0 - OK, !0 - error
348
 */
349
static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
350
                                     int channel_config)
351
{
352
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
353
    int extension_flag, ret;
354

    
355
    if (get_bits1(gb)) { // frameLengthFlag
356
        av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
357
        return -1;
358
    }
359

    
360
    if (get_bits1(gb))       // dependsOnCoreCoder
361
        skip_bits(gb, 14);   // coreCoderDelay
362
    extension_flag = get_bits1(gb);
363

    
364
    if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
365
        ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
366
        skip_bits(gb, 3);     // layerNr
367

    
368
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
369
    if (channel_config == 0) {
370
        skip_bits(gb, 4);  // element_instance_tag
371
        if ((ret = decode_pce(ac, new_che_pos, gb)))
372
            return ret;
373
    } else {
374
        if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
375
            return ret;
376
    }
377
    if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config)))
378
        return ret;
379

    
380
    if (extension_flag) {
381
        switch (ac->m4ac.object_type) {
382
        case AOT_ER_BSAC:
383
            skip_bits(gb, 5);    // numOfSubFrame
384
            skip_bits(gb, 11);   // layer_length
385
            break;
386
        case AOT_ER_AAC_LC:
387
        case AOT_ER_AAC_LTP:
388
        case AOT_ER_AAC_SCALABLE:
389
        case AOT_ER_AAC_LD:
390
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
391
                                    * aacScalefactorDataResilienceFlag
392
                                    * aacSpectralDataResilienceFlag
393
                                    */
394
            break;
395
        }
396
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
397
    }
398
    return 0;
399
}
400

    
401
/**
402
 * Decode audio specific configuration; reference: table 1.13.
403
 *
404
 * @param   data        pointer to AVCodecContext extradata
405
 * @param   data_size   size of AVCCodecContext extradata
406
 *
407
 * @return  Returns error status. 0 - OK, !0 - error
408
 */
409
static int decode_audio_specific_config(AACContext *ac, void *data,
410
                                        int data_size)
411
{
412
    GetBitContext gb;
413
    int i;
414

    
415
    init_get_bits(&gb, data, data_size * 8);
416

    
417
    if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
418
        return -1;
419
    if (ac->m4ac.sampling_index > 12) {
420
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
421
        return -1;
422
    }
423

    
424
    skip_bits_long(&gb, i);
425

    
426
    switch (ac->m4ac.object_type) {
427
    case AOT_AAC_MAIN:
428
    case AOT_AAC_LC:
429
        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
430
            return -1;
431
        break;
432
    default:
433
        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
434
               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
435
        return -1;
436
    }
437
    return 0;
438
}
439

    
440
/**
441
 * linear congruential pseudorandom number generator
442
 *
443
 * @param   previous_val    pointer to the current state of the generator
444
 *
445
 * @return  Returns a 32-bit pseudorandom integer
446
 */
447
static av_always_inline int lcg_random(int previous_val)
448
{
449
    return previous_val * 1664525 + 1013904223;
450
}
451

    
452
static void reset_predict_state(PredictorState *ps)
453
{
454
    ps->r0   = 0.0f;
455
    ps->r1   = 0.0f;
456
    ps->cor0 = 0.0f;
457
    ps->cor1 = 0.0f;
458
    ps->var0 = 1.0f;
459
    ps->var1 = 1.0f;
460
}
461

    
462
static void reset_all_predictors(PredictorState *ps)
463
{
464
    int i;
465
    for (i = 0; i < MAX_PREDICTORS; i++)
466
        reset_predict_state(&ps[i]);
467
}
468

    
469
static void reset_predictor_group(PredictorState *ps, int group_num)
470
{
471
    int i;
472
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
473
        reset_predict_state(&ps[i]);
474
}
475

    
476
static av_cold int aac_decode_init(AVCodecContext *avccontext)
477
{
478
    AACContext *ac = avccontext->priv_data;
479
    int i;
480

    
481
    ac->avccontext = avccontext;
482

    
483
    if (avccontext->extradata_size > 0) {
484
        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
485
            return -1;
486
        avccontext->sample_rate = ac->m4ac.sample_rate;
487
    } else if (avccontext->channels > 0) {
488
        ac->m4ac.sample_rate = avccontext->sample_rate;
489
    }
490

    
491
    avccontext->sample_fmt = SAMPLE_FMT_S16;
492
    avccontext->frame_size = 1024;
493

    
494
    AAC_INIT_VLC_STATIC( 0, 144);
495
    AAC_INIT_VLC_STATIC( 1, 114);
496
    AAC_INIT_VLC_STATIC( 2, 188);
497
    AAC_INIT_VLC_STATIC( 3, 180);
498
    AAC_INIT_VLC_STATIC( 4, 172);
499
    AAC_INIT_VLC_STATIC( 5, 140);
500
    AAC_INIT_VLC_STATIC( 6, 168);
501
    AAC_INIT_VLC_STATIC( 7, 114);
502
    AAC_INIT_VLC_STATIC( 8, 262);
503
    AAC_INIT_VLC_STATIC( 9, 248);
504
    AAC_INIT_VLC_STATIC(10, 384);
505

    
506
    dsputil_init(&ac->dsp, avccontext);
507

    
508
    ac->random_state = 0x1f2e3d4c;
509

    
510
    // -1024 - Compensate wrong IMDCT method.
511
    // 32768 - Required to scale values to the correct range for the bias method
512
    //         for float to int16 conversion.
513

    
514
    if (ac->dsp.float_to_int16 == ff_float_to_int16_c) {
515
        ac->add_bias  = 385.0f;
516
        ac->sf_scale  = 1. / (-1024. * 32768.);
517
        ac->sf_offset = 0;
518
    } else {
519
        ac->add_bias  = 0.0f;
520
        ac->sf_scale  = 1. / -1024.;
521
        ac->sf_offset = 60;
522
    }
523

    
524
#if !CONFIG_HARDCODED_TABLES
525
    for (i = 0; i < 428; i++)
526
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
527
#endif /* CONFIG_HARDCODED_TABLES */
528

    
529
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
530
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
531
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
532
                    352);
533

    
534
    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
535
    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
536
    // window initialization
537
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
538
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
539
    ff_sine_window_init(ff_sine_1024, 1024);
540
    ff_sine_window_init(ff_sine_128, 128);
541

    
542
    return 0;
543
}
544

    
545
/**
546
 * Skip data_stream_element; reference: table 4.10.
547
 */
548
static void skip_data_stream_element(GetBitContext *gb)
549
{
550
    int byte_align = get_bits1(gb);
551
    int count = get_bits(gb, 8);
552
    if (count == 255)
553
        count += get_bits(gb, 8);
554
    if (byte_align)
555
        align_get_bits(gb);
556
    skip_bits_long(gb, 8 * count);
557
}
558

    
559
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
560
                             GetBitContext *gb)
561
{
562
    int sfb;
563
    if (get_bits1(gb)) {
564
        ics->predictor_reset_group = get_bits(gb, 5);
565
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
566
            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
567
            return -1;
568
        }
569
    }
570
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
571
        ics->prediction_used[sfb] = get_bits1(gb);
572
    }
573
    return 0;
574
}
575

    
576
/**
577
 * Decode Individual Channel Stream info; reference: table 4.6.
578
 *
579
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
580
 */
581
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
582
                           GetBitContext *gb, int common_window)
583
{
584
    if (get_bits1(gb)) {
585
        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
586
        memset(ics, 0, sizeof(IndividualChannelStream));
587
        return -1;
588
    }
589
    ics->window_sequence[1] = ics->window_sequence[0];
590
    ics->window_sequence[0] = get_bits(gb, 2);
591
    ics->use_kb_window[1]   = ics->use_kb_window[0];
592
    ics->use_kb_window[0]   = get_bits1(gb);
593
    ics->num_window_groups  = 1;
594
    ics->group_len[0]       = 1;
595
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
596
        int i;
597
        ics->max_sfb = get_bits(gb, 4);
598
        for (i = 0; i < 7; i++) {
599
            if (get_bits1(gb)) {
600
                ics->group_len[ics->num_window_groups - 1]++;
601
            } else {
602
                ics->num_window_groups++;
603
                ics->group_len[ics->num_window_groups - 1] = 1;
604
            }
605
        }
606
        ics->num_windows       = 8;
607
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
608
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
609
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
610
        ics->predictor_present = 0;
611
    } else {
612
        ics->max_sfb               = get_bits(gb, 6);
613
        ics->num_windows           = 1;
614
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
615
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
616
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
617
        ics->predictor_present     = get_bits1(gb);
618
        ics->predictor_reset_group = 0;
619
        if (ics->predictor_present) {
620
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
621
                if (decode_prediction(ac, ics, gb)) {
622
                    memset(ics, 0, sizeof(IndividualChannelStream));
623
                    return -1;
624
                }
625
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
626
                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
627
                memset(ics, 0, sizeof(IndividualChannelStream));
628
                return -1;
629
            } else {
630
                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
631
                memset(ics, 0, sizeof(IndividualChannelStream));
632
                return -1;
633
            }
634
        }
635
    }
636

    
637
    if (ics->max_sfb > ics->num_swb) {
638
        av_log(ac->avccontext, AV_LOG_ERROR,
639
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
640
               ics->max_sfb, ics->num_swb);
641
        memset(ics, 0, sizeof(IndividualChannelStream));
642
        return -1;
643
    }
644

    
645
    return 0;
646
}
647

    
648
/**
649
 * Decode band types (section_data payload); reference: table 4.46.
650
 *
651
 * @param   band_type           array of the used band type
652
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
653
 *
654
 * @return  Returns error status. 0 - OK, !0 - error
655
 */
656
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
657
                             int band_type_run_end[120], GetBitContext *gb,
658
                             IndividualChannelStream *ics)
659
{
660
    int g, idx = 0;
661
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
662
    for (g = 0; g < ics->num_window_groups; g++) {
663
        int k = 0;
664
        while (k < ics->max_sfb) {
665
            uint8_t sect_len = k;
666
            int sect_len_incr;
667
            int sect_band_type = get_bits(gb, 4);
668
            if (sect_band_type == 12) {
669
                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
670
                return -1;
671
            }
672
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
673
                sect_len += sect_len_incr;
674
            sect_len += sect_len_incr;
675
            if (sect_len > ics->max_sfb) {
676
                av_log(ac->avccontext, AV_LOG_ERROR,
677
                       "Number of bands (%d) exceeds limit (%d).\n",
678
                       sect_len, ics->max_sfb);
679
                return -1;
680
            }
681
            for (; k < sect_len; k++) {
682
                band_type        [idx]   = sect_band_type;
683
                band_type_run_end[idx++] = sect_len;
684
            }
685
        }
686
    }
687
    return 0;
688
}
689

    
690
/**
691
 * Decode scalefactors; reference: table 4.47.
692
 *
693
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
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
 * @param   sf                  array of scalefactors or intensity stereo positions
697
 *
698
 * @return  Returns error status. 0 - OK, !0 - error
699
 */
700
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
701
                               unsigned int global_gain,
702
                               IndividualChannelStream *ics,
703
                               enum BandType band_type[120],
704
                               int band_type_run_end[120])
705
{
706
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
707
    int g, i, idx = 0;
708
    int offset[3] = { global_gain, global_gain - 90, 100 };
709
    int noise_flag = 1;
710
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
711
    for (g = 0; g < ics->num_window_groups; g++) {
712
        for (i = 0; i < ics->max_sfb;) {
713
            int run_end = band_type_run_end[idx];
714
            if (band_type[idx] == ZERO_BT) {
715
                for (; i < run_end; i++, idx++)
716
                    sf[idx] = 0.;
717
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
718
                for (; i < run_end; i++, idx++) {
719
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
720
                    if (offset[2] > 255U) {
721
                        av_log(ac->avccontext, AV_LOG_ERROR,
722
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
723
                        return -1;
724
                    }
725
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
726
                }
727
            } else if (band_type[idx] == NOISE_BT) {
728
                for (; i < run_end; i++, idx++) {
729
                    if (noise_flag-- > 0)
730
                        offset[1] += get_bits(gb, 9) - 256;
731
                    else
732
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
733
                    if (offset[1] > 255U) {
734
                        av_log(ac->avccontext, AV_LOG_ERROR,
735
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
736
                        return -1;
737
                    }
738
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
739
                }
740
            } else {
741
                for (; i < run_end; i++, idx++) {
742
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
743
                    if (offset[0] > 255U) {
744
                        av_log(ac->avccontext, AV_LOG_ERROR,
745
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
746
                        return -1;
747
                    }
748
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
749
                }
750
            }
751
        }
752
    }
753
    return 0;
754
}
755

    
756
/**
757
 * Decode pulse data; reference: table 4.7.
758
 */
759
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
760
                         const uint16_t *swb_offset, int num_swb)
761
{
762
    int i, pulse_swb;
763
    pulse->num_pulse = get_bits(gb, 2) + 1;
764
    pulse_swb        = get_bits(gb, 6);
765
    if (pulse_swb >= num_swb)
766
        return -1;
767
    pulse->pos[0]    = swb_offset[pulse_swb];
768
    pulse->pos[0]   += get_bits(gb, 5);
769
    if (pulse->pos[0] > 1023)
770
        return -1;
771
    pulse->amp[0]    = get_bits(gb, 4);
772
    for (i = 1; i < pulse->num_pulse; i++) {
773
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
774
        if (pulse->pos[i] > 1023)
775
            return -1;
776
        pulse->amp[i] = get_bits(gb, 4);
777
    }
778
    return 0;
779
}
780

    
781
/**
782
 * Decode Temporal Noise Shaping data; reference: table 4.48.
783
 *
784
 * @return  Returns error status. 0 - OK, !0 - error
785
 */
786
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
787
                      GetBitContext *gb, const IndividualChannelStream *ics)
788
{
789
    int w, filt, i, coef_len, coef_res, coef_compress;
790
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
791
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
792
    for (w = 0; w < ics->num_windows; w++) {
793
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
794
            coef_res = get_bits1(gb);
795

    
796
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
797
                int tmp2_idx;
798
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
799

    
800
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
801
                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
802
                           tns->order[w][filt], tns_max_order);
803
                    tns->order[w][filt] = 0;
804
                    return -1;
805
                }
806
                if (tns->order[w][filt]) {
807
                    tns->direction[w][filt] = get_bits1(gb);
808
                    coef_compress = get_bits1(gb);
809
                    coef_len = coef_res + 3 - coef_compress;
810
                    tmp2_idx = 2 * coef_compress + coef_res;
811

    
812
                    for (i = 0; i < tns->order[w][filt]; i++)
813
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
814
                }
815
            }
816
        }
817
    }
818
    return 0;
819
}
820

    
821
/**
822
 * Decode Mid/Side data; reference: table 4.54.
823
 *
824
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
825
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
826
 *                      [3] reserved for scalable AAC
827
 */
828
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
829
                                   int ms_present)
830
{
831
    int idx;
832
    if (ms_present == 1) {
833
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
834
            cpe->ms_mask[idx] = get_bits1(gb);
835
    } else if (ms_present == 2) {
836
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
837
    }
838
}
839

    
840
/**
841
 * Decode spectral data; reference: table 4.50.
842
 * Dequantize and scale spectral data; reference: 4.6.3.3.
843
 *
844
 * @param   coef            array of dequantized, scaled spectral data
845
 * @param   sf              array of scalefactors or intensity stereo positions
846
 * @param   pulse_present   set if pulses are present
847
 * @param   pulse           pointer to pulse data struct
848
 * @param   band_type       array of the used band type
849
 *
850
 * @return  Returns error status. 0 - OK, !0 - error
851
 */
852
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
853
                                       GetBitContext *gb, float sf[120],
854
                                       int pulse_present, const Pulse *pulse,
855
                                       const IndividualChannelStream *ics,
856
                                       enum BandType band_type[120])
857
{
858
    int i, k, g, idx = 0;
859
    const int c = 1024 / ics->num_windows;
860
    const uint16_t *offsets = ics->swb_offset;
861
    float *coef_base = coef;
862
    static const float sign_lookup[] = { 1.0f, -1.0f };
863

    
864
    for (g = 0; g < ics->num_windows; g++)
865
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
866

    
867
    for (g = 0; g < ics->num_window_groups; g++) {
868
        for (i = 0; i < ics->max_sfb; i++, idx++) {
869
            const int cur_band_type = band_type[idx];
870
            const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4;
871
            const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type);
872
            int group;
873
            if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) {
874
                for (group = 0; group < ics->group_len[g]; group++) {
875
                    memset(coef + group * 128 + offsets[i], 0, (offsets[i + 1] - offsets[i]) * sizeof(float));
876
                }
877
            } else if (cur_band_type == NOISE_BT) {
878
                for (group = 0; group < ics->group_len[g]; group++) {
879
                    float scale;
880
                    float band_energy;
881
                    float *cf = coef + group * 128 + offsets[i];
882
                    int len = offsets[i+1] - offsets[i];
883

    
884
                    for (k = 0; k < len; k++) {
885
                        ac->random_state  = lcg_random(ac->random_state);
886
                        cf[k] = ac->random_state;
887
                    }
888

    
889
                    band_energy = ac->dsp.scalarproduct_float(cf, cf, len);
890
                    scale = sf[idx] / sqrtf(band_energy);
891
                    ac->dsp.vector_fmul_scalar(cf, cf, scale, len);
892
                }
893
            } else {
894
                for (group = 0; group < ics->group_len[g]; group++) {
895
                    const float *vq[96];
896
                    const float **vqp = vq;
897
                    float *cf = coef + (group << 7) + offsets[i];
898
                    int len = offsets[i + 1] - offsets[i];
899

    
900
                    for (k = offsets[i]; k < offsets[i + 1]; k += dim) {
901
                        const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3);
902
                        const int coef_tmp_idx = (group << 7) + k;
903
                        const float *vq_ptr;
904
                        int j;
905
                        if (index >= ff_aac_spectral_sizes[cur_band_type - 1]) {
906
                            av_log(ac->avccontext, AV_LOG_ERROR,
907
                                   "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
908
                                   cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]);
909
                            return -1;
910
                        }
911
                        vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
912
                        *vqp++ = vq_ptr;
913
                        if (is_cb_unsigned) {
914
                            if (vq_ptr[0])
915
                                coef[coef_tmp_idx    ] = sign_lookup[get_bits1(gb)];
916
                            if (vq_ptr[1])
917
                                coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)];
918
                            if (dim == 4) {
919
                                if (vq_ptr[2])
920
                                    coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)];
921
                                if (vq_ptr[3])
922
                                    coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)];
923
                            }
924
                            if (cur_band_type == ESC_BT) {
925
                                for (j = 0; j < 2; j++) {
926
                                    if (vq_ptr[j] == 64.0f) {
927
                                        int n = 4;
928
                                        /* The total length of escape_sequence must be < 22 bits according
929
                                           to the specification (i.e. max is 11111111110xxxxxxxxxx). */
930
                                        while (get_bits1(gb) && n < 15) n++;
931
                                        if (n == 15) {
932
                                            av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
933
                                            return -1;
934
                                        }
935
                                        n = (1 << n) + get_bits(gb, n);
936
                                        coef[coef_tmp_idx + j] *= cbrtf(n) * n;
937
                                    } else
938
                                        coef[coef_tmp_idx + j] *= vq_ptr[j];
939
                                }
940
                            }
941
                        }
942
                    }
943

    
944
                    if (is_cb_unsigned && cur_band_type != ESC_BT) {
945
                        ac->dsp.vector_fmul_sv_scalar[dim>>2](
946
                            cf, cf, vq, sf[idx], len);
947
                    } else if (cur_band_type == ESC_BT) {
948
                        ac->dsp.vector_fmul_scalar(cf, cf, sf[idx], len);
949
                    } else {    /* !is_cb_unsigned */
950
                        ac->dsp.sv_fmul_scalar[dim>>2](cf, vq, sf[idx], len);
951
                    }
952
                }
953
            }
954
        }
955
        coef += ics->group_len[g] << 7;
956
    }
957

    
958
    if (pulse_present) {
959
        idx = 0;
960
        for (i = 0; i < pulse->num_pulse; i++) {
961
            float co = coef_base[ pulse->pos[i] ];
962
            while (offsets[idx + 1] <= pulse->pos[i])
963
                idx++;
964
            if (band_type[idx] != NOISE_BT && sf[idx]) {
965
                float ico = -pulse->amp[i];
966
                if (co) {
967
                    co /= sf[idx];
968
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
969
                }
970
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
971
            }
972
        }
973
    }
974
    return 0;
975
}
976

    
977
static av_always_inline float flt16_round(float pf)
978
{
979
    union float754 tmp;
980
    tmp.f = pf;
981
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
982
    return tmp.f;
983
}
984

    
985
static av_always_inline float flt16_even(float pf)
986
{
987
    union float754 tmp;
988
    tmp.f = pf;
989
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
990
    return tmp.f;
991
}
992

    
993
static av_always_inline float flt16_trunc(float pf)
994
{
995
    union float754 pun;
996
    pun.f = pf;
997
    pun.i &= 0xFFFF0000U;
998
    return pun.f;
999
}
1000

    
1001
static void predict(AACContext *ac, PredictorState *ps, float *coef,
1002
                    int output_enable)
1003
{
1004
    const float a     = 0.953125; // 61.0 / 64
1005
    const float alpha = 0.90625;  // 29.0 / 32
1006
    float e0, e1;
1007
    float pv;
1008
    float k1, k2;
1009

    
1010
    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
1011
    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
1012

    
1013
    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
1014
    if (output_enable)
1015
        *coef += pv * ac->sf_scale;
1016

    
1017
    e0 = *coef / ac->sf_scale;
1018
    e1 = e0 - k1 * ps->r0;
1019

    
1020
    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
1021
    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
1022
    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
1023
    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
1024

    
1025
    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
1026
    ps->r0 = flt16_trunc(a * e0);
1027
}
1028

    
1029
/**
1030
 * Apply AAC-Main style frequency domain prediction.
1031
 */
1032
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1033
{
1034
    int sfb, k;
1035

    
1036
    if (!sce->ics.predictor_initialized) {
1037
        reset_all_predictors(sce->predictor_state);
1038
        sce->ics.predictor_initialized = 1;
1039
    }
1040

    
1041
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1042
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1043
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1044
                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
1045
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1046
            }
1047
        }
1048
        if (sce->ics.predictor_reset_group)
1049
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1050
    } else
1051
        reset_all_predictors(sce->predictor_state);
1052
}
1053

    
1054
/**
1055
 * Decode an individual_channel_stream payload; reference: table 4.44.
1056
 *
1057
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1058
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1059
 *
1060
 * @return  Returns error status. 0 - OK, !0 - error
1061
 */
1062
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1063
                      GetBitContext *gb, int common_window, int scale_flag)
1064
{
1065
    Pulse pulse;
1066
    TemporalNoiseShaping    *tns = &sce->tns;
1067
    IndividualChannelStream *ics = &sce->ics;
1068
    float *out = sce->coeffs;
1069
    int global_gain, pulse_present = 0;
1070

    
1071
    /* This assignment is to silence a GCC warning about the variable being used
1072
     * uninitialized when in fact it always is.
1073
     */
1074
    pulse.num_pulse = 0;
1075

    
1076
    global_gain = get_bits(gb, 8);
1077

    
1078
    if (!common_window && !scale_flag) {
1079
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1080
            return -1;
1081
    }
1082

    
1083
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1084
        return -1;
1085
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1086
        return -1;
1087

    
1088
    pulse_present = 0;
1089
    if (!scale_flag) {
1090
        if ((pulse_present = get_bits1(gb))) {
1091
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1092
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1093
                return -1;
1094
            }
1095
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1096
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1097
                return -1;
1098
            }
1099
        }
1100
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1101
            return -1;
1102
        if (get_bits1(gb)) {
1103
            av_log_missing_feature(ac->avccontext, "SSR", 1);
1104
            return -1;
1105
        }
1106
    }
1107

    
1108
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1109
        return -1;
1110

    
1111
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1112
        apply_prediction(ac, sce);
1113

    
1114
    return 0;
1115
}
1116

    
1117
/**
1118
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1119
 */
1120
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1121
{
1122
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1123
    float *ch0 = cpe->ch[0].coeffs;
1124
    float *ch1 = cpe->ch[1].coeffs;
1125
    int g, i, group, idx = 0;
1126
    const uint16_t *offsets = ics->swb_offset;
1127
    for (g = 0; g < ics->num_window_groups; g++) {
1128
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1129
            if (cpe->ms_mask[idx] &&
1130
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1131
                for (group = 0; group < ics->group_len[g]; group++) {
1132
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1133
                                              ch1 + group * 128 + offsets[i],
1134
                                              offsets[i+1] - offsets[i]);
1135
                }
1136
            }
1137
        }
1138
        ch0 += ics->group_len[g] * 128;
1139
        ch1 += ics->group_len[g] * 128;
1140
    }
1141
}
1142

    
1143
/**
1144
 * intensity stereo decoding; reference: 4.6.8.2.3
1145
 *
1146
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1147
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1148
 *                      [3] reserved for scalable AAC
1149
 */
1150
static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
1151
{
1152
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1153
    SingleChannelElement         *sce1 = &cpe->ch[1];
1154
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1155
    const uint16_t *offsets = ics->swb_offset;
1156
    int g, group, i, k, idx = 0;
1157
    int c;
1158
    float scale;
1159
    for (g = 0; g < ics->num_window_groups; g++) {
1160
        for (i = 0; i < ics->max_sfb;) {
1161
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1162
                const int bt_run_end = sce1->band_type_run_end[idx];
1163
                for (; i < bt_run_end; i++, idx++) {
1164
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1165
                    if (ms_present)
1166
                        c *= 1 - 2 * cpe->ms_mask[idx];
1167
                    scale = c * sce1->sf[idx];
1168
                    for (group = 0; group < ics->group_len[g]; group++)
1169
                        for (k = offsets[i]; k < offsets[i + 1]; k++)
1170
                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
1171
                }
1172
            } else {
1173
                int bt_run_end = sce1->band_type_run_end[idx];
1174
                idx += bt_run_end - i;
1175
                i    = bt_run_end;
1176
            }
1177
        }
1178
        coef0 += ics->group_len[g] * 128;
1179
        coef1 += ics->group_len[g] * 128;
1180
    }
1181
}
1182

    
1183
/**
1184
 * Decode a channel_pair_element; reference: table 4.4.
1185
 *
1186
 * @param   elem_id Identifies the instance of a syntax element.
1187
 *
1188
 * @return  Returns error status. 0 - OK, !0 - error
1189
 */
1190
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1191
{
1192
    int i, ret, common_window, ms_present = 0;
1193

    
1194
    common_window = get_bits1(gb);
1195
    if (common_window) {
1196
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1197
            return -1;
1198
        i = cpe->ch[1].ics.use_kb_window[0];
1199
        cpe->ch[1].ics = cpe->ch[0].ics;
1200
        cpe->ch[1].ics.use_kb_window[1] = i;
1201
        ms_present = get_bits(gb, 2);
1202
        if (ms_present == 3) {
1203
            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1204
            return -1;
1205
        } else if (ms_present)
1206
            decode_mid_side_stereo(cpe, gb, ms_present);
1207
    }
1208
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1209
        return ret;
1210
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1211
        return ret;
1212

    
1213
    if (common_window) {
1214
        if (ms_present)
1215
            apply_mid_side_stereo(ac, cpe);
1216
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1217
            apply_prediction(ac, &cpe->ch[0]);
1218
            apply_prediction(ac, &cpe->ch[1]);
1219
        }
1220
    }
1221

    
1222
    apply_intensity_stereo(cpe, ms_present);
1223
    return 0;
1224
}
1225

    
1226
/**
1227
 * Decode coupling_channel_element; reference: table 4.8.
1228
 *
1229
 * @param   elem_id Identifies the instance of a syntax element.
1230
 *
1231
 * @return  Returns error status. 0 - OK, !0 - error
1232
 */
1233
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1234
{
1235
    int num_gain = 0;
1236
    int c, g, sfb, ret;
1237
    int sign;
1238
    float scale;
1239
    SingleChannelElement *sce = &che->ch[0];
1240
    ChannelCoupling     *coup = &che->coup;
1241

    
1242
    coup->coupling_point = 2 * get_bits1(gb);
1243
    coup->num_coupled = get_bits(gb, 3);
1244
    for (c = 0; c <= coup->num_coupled; c++) {
1245
        num_gain++;
1246
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1247
        coup->id_select[c] = get_bits(gb, 4);
1248
        if (coup->type[c] == TYPE_CPE) {
1249
            coup->ch_select[c] = get_bits(gb, 2);
1250
            if (coup->ch_select[c] == 3)
1251
                num_gain++;
1252
        } else
1253
            coup->ch_select[c] = 2;
1254
    }
1255
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1256

    
1257
    sign  = get_bits(gb, 1);
1258
    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1259

    
1260
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1261
        return ret;
1262

    
1263
    for (c = 0; c < num_gain; c++) {
1264
        int idx  = 0;
1265
        int cge  = 1;
1266
        int gain = 0;
1267
        float gain_cache = 1.;
1268
        if (c) {
1269
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1270
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1271
            gain_cache = pow(scale, -gain);
1272
        }
1273
        if (coup->coupling_point == AFTER_IMDCT) {
1274
            coup->gain[c][0] = gain_cache;
1275
        } else {
1276
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1277
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1278
                    if (sce->band_type[idx] != ZERO_BT) {
1279
                        if (!cge) {
1280
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1281
                            if (t) {
1282
                                int s = 1;
1283
                                t = gain += t;
1284
                                if (sign) {
1285
                                    s  -= 2 * (t & 0x1);
1286
                                    t >>= 1;
1287
                                }
1288
                                gain_cache = pow(scale, -t) * s;
1289
                            }
1290
                        }
1291
                        coup->gain[c][idx] = gain_cache;
1292
                    }
1293
                }
1294
            }
1295
        }
1296
    }
1297
    return 0;
1298
}
1299

    
1300
/**
1301
 * Decode Spectral Band Replication extension data; reference: table 4.55.
1302
 *
1303
 * @param   crc flag indicating the presence of CRC checksum
1304
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1305
 *
1306
 * @return  Returns number of bytes consumed from the TYPE_FIL element.
1307
 */
1308
static int decode_sbr_extension(AACContext *ac, GetBitContext *gb,
1309
                                int crc, int cnt)
1310
{
1311
    // TODO : sbr_extension implementation
1312
    av_log_missing_feature(ac->avccontext, "SBR", 0);
1313
    skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type
1314
    return cnt;
1315
}
1316

    
1317
/**
1318
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1319
 *
1320
 * @return  Returns number of bytes consumed.
1321
 */
1322
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1323
                                         GetBitContext *gb)
1324
{
1325
    int i;
1326
    int num_excl_chan = 0;
1327

    
1328
    do {
1329
        for (i = 0; i < 7; i++)
1330
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1331
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1332

    
1333
    return num_excl_chan / 7;
1334
}
1335

    
1336
/**
1337
 * Decode dynamic range information; reference: table 4.52.
1338
 *
1339
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1340
 *
1341
 * @return  Returns number of bytes consumed.
1342
 */
1343
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1344
                                GetBitContext *gb, int cnt)
1345
{
1346
    int n             = 1;
1347
    int drc_num_bands = 1;
1348
    int i;
1349

    
1350
    /* pce_tag_present? */
1351
    if (get_bits1(gb)) {
1352
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1353
        skip_bits(gb, 4); // tag_reserved_bits
1354
        n++;
1355
    }
1356

    
1357
    /* excluded_chns_present? */
1358
    if (get_bits1(gb)) {
1359
        n += decode_drc_channel_exclusions(che_drc, gb);
1360
    }
1361

    
1362
    /* drc_bands_present? */
1363
    if (get_bits1(gb)) {
1364
        che_drc->band_incr            = get_bits(gb, 4);
1365
        che_drc->interpolation_scheme = get_bits(gb, 4);
1366
        n++;
1367
        drc_num_bands += che_drc->band_incr;
1368
        for (i = 0; i < drc_num_bands; i++) {
1369
            che_drc->band_top[i] = get_bits(gb, 8);
1370
            n++;
1371
        }
1372
    }
1373

    
1374
    /* prog_ref_level_present? */
1375
    if (get_bits1(gb)) {
1376
        che_drc->prog_ref_level = get_bits(gb, 7);
1377
        skip_bits1(gb); // prog_ref_level_reserved_bits
1378
        n++;
1379
    }
1380

    
1381
    for (i = 0; i < drc_num_bands; i++) {
1382
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1383
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1384
        n++;
1385
    }
1386

    
1387
    return n;
1388
}
1389

    
1390
/**
1391
 * Decode extension data (incomplete); reference: table 4.51.
1392
 *
1393
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1394
 *
1395
 * @return Returns number of bytes consumed
1396
 */
1397
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt)
1398
{
1399
    int crc_flag = 0;
1400
    int res = cnt;
1401
    switch (get_bits(gb, 4)) { // extension type
1402
    case EXT_SBR_DATA_CRC:
1403
        crc_flag++;
1404
    case EXT_SBR_DATA:
1405
        res = decode_sbr_extension(ac, gb, crc_flag, cnt);
1406
        break;
1407
    case EXT_DYNAMIC_RANGE:
1408
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1409
        break;
1410
    case EXT_FILL:
1411
    case EXT_FILL_DATA:
1412
    case EXT_DATA_ELEMENT:
1413
    default:
1414
        skip_bits_long(gb, 8 * cnt - 4);
1415
        break;
1416
    };
1417
    return res;
1418
}
1419

    
1420
/**
1421
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1422
 *
1423
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1424
 * @param   coef    spectral coefficients
1425
 */
1426
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1427
                      IndividualChannelStream *ics, int decode)
1428
{
1429
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1430
    int w, filt, m, i;
1431
    int bottom, top, order, start, end, size, inc;
1432
    float lpc[TNS_MAX_ORDER];
1433

    
1434
    for (w = 0; w < ics->num_windows; w++) {
1435
        bottom = ics->num_swb;
1436
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1437
            top    = bottom;
1438
            bottom = FFMAX(0, top - tns->length[w][filt]);
1439
            order  = tns->order[w][filt];
1440
            if (order == 0)
1441
                continue;
1442

    
1443
            // tns_decode_coef
1444
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1445

    
1446
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1447
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1448
            if ((size = end - start) <= 0)
1449
                continue;
1450
            if (tns->direction[w][filt]) {
1451
                inc = -1;
1452
                start = end - 1;
1453
            } else {
1454
                inc = 1;
1455
            }
1456
            start += w * 128;
1457

    
1458
            // ar filter
1459
            for (m = 0; m < size; m++, start += inc)
1460
                for (i = 1; i <= FFMIN(m, order); i++)
1461
                    coef[start] -= coef[start - i * inc] * lpc[i - 1];
1462
        }
1463
    }
1464
}
1465

    
1466
/**
1467
 * Conduct IMDCT and windowing.
1468
 */
1469
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1470
{
1471
    IndividualChannelStream *ics = &sce->ics;
1472
    float *in    = sce->coeffs;
1473
    float *out   = sce->ret;
1474
    float *saved = sce->saved;
1475
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1476
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1477
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1478
    float *buf  = ac->buf_mdct;
1479
    float *temp = ac->temp;
1480
    int i;
1481

    
1482
    // imdct
1483
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1484
        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1485
            av_log(ac->avccontext, AV_LOG_WARNING,
1486
                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1487
                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1488
        for (i = 0; i < 1024; i += 128)
1489
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1490
    } else
1491
        ff_imdct_half(&ac->mdct, buf, in);
1492

    
1493
    /* window overlapping
1494
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1495
     * and long to short transitions are considered to be short to short
1496
     * transitions. This leaves just two cases (long to long and short to short)
1497
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1498
     */
1499
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1500
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1501
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, ac->add_bias, 512);
1502
    } else {
1503
        for (i = 0; i < 448; i++)
1504
            out[i] = saved[i] + ac->add_bias;
1505

    
1506
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1507
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, ac->add_bias, 64);
1508
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      ac->add_bias, 64);
1509
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      ac->add_bias, 64);
1510
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      ac->add_bias, 64);
1511
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      ac->add_bias, 64);
1512
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1513
        } else {
1514
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, ac->add_bias, 64);
1515
            for (i = 576; i < 1024; i++)
1516
                out[i] = buf[i-512] + ac->add_bias;
1517
        }
1518
    }
1519

    
1520
    // buffer update
1521
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1522
        for (i = 0; i < 64; i++)
1523
            saved[i] = temp[64 + i] - ac->add_bias;
1524
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1525
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1526
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1527
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1528
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1529
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1530
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1531
    } else { // LONG_STOP or ONLY_LONG
1532
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1533
    }
1534
}
1535

    
1536
/**
1537
 * Apply dependent channel coupling (applied before IMDCT).
1538
 *
1539
 * @param   index   index into coupling gain array
1540
 */
1541
static void apply_dependent_coupling(AACContext *ac,
1542
                                     SingleChannelElement *target,
1543
                                     ChannelElement *cce, int index)
1544
{
1545
    IndividualChannelStream *ics = &cce->ch[0].ics;
1546
    const uint16_t *offsets = ics->swb_offset;
1547
    float *dest = target->coeffs;
1548
    const float *src = cce->ch[0].coeffs;
1549
    int g, i, group, k, idx = 0;
1550
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1551
        av_log(ac->avccontext, AV_LOG_ERROR,
1552
               "Dependent coupling is not supported together with LTP\n");
1553
        return;
1554
    }
1555
    for (g = 0; g < ics->num_window_groups; g++) {
1556
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1557
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1558
                const float gain = cce->coup.gain[index][idx];
1559
                for (group = 0; group < ics->group_len[g]; group++) {
1560
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1561
                        // XXX dsputil-ize
1562
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1563
                    }
1564
                }
1565
            }
1566
        }
1567
        dest += ics->group_len[g] * 128;
1568
        src  += ics->group_len[g] * 128;
1569
    }
1570
}
1571

    
1572
/**
1573
 * Apply independent channel coupling (applied after IMDCT).
1574
 *
1575
 * @param   index   index into coupling gain array
1576
 */
1577
static void apply_independent_coupling(AACContext *ac,
1578
                                       SingleChannelElement *target,
1579
                                       ChannelElement *cce, int index)
1580
{
1581
    int i;
1582
    const float gain = cce->coup.gain[index][0];
1583
    const float bias = ac->add_bias;
1584
    const float *src = cce->ch[0].ret;
1585
    float *dest = target->ret;
1586

    
1587
    for (i = 0; i < 1024; i++)
1588
        dest[i] += gain * (src[i] - bias);
1589
}
1590

    
1591
/**
1592
 * channel coupling transformation interface
1593
 *
1594
 * @param   index   index into coupling gain array
1595
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1596
 */
1597
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1598
                                   enum RawDataBlockType type, int elem_id,
1599
                                   enum CouplingPoint coupling_point,
1600
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1601
{
1602
    int i, c;
1603

    
1604
    for (i = 0; i < MAX_ELEM_ID; i++) {
1605
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1606
        int index = 0;
1607

    
1608
        if (cce && cce->coup.coupling_point == coupling_point) {
1609
            ChannelCoupling *coup = &cce->coup;
1610

    
1611
            for (c = 0; c <= coup->num_coupled; c++) {
1612
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1613
                    if (coup->ch_select[c] != 1) {
1614
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1615
                        if (coup->ch_select[c] != 0)
1616
                            index++;
1617
                    }
1618
                    if (coup->ch_select[c] != 2)
1619
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1620
                } else
1621
                    index += 1 + (coup->ch_select[c] == 3);
1622
            }
1623
        }
1624
    }
1625
}
1626

    
1627
/**
1628
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1629
 */
1630
static void spectral_to_sample(AACContext *ac)
1631
{
1632
    int i, type;
1633
    for (type = 3; type >= 0; type--) {
1634
        for (i = 0; i < MAX_ELEM_ID; i++) {
1635
            ChannelElement *che = ac->che[type][i];
1636
            if (che) {
1637
                if (type <= TYPE_CPE)
1638
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1639
                if (che->ch[0].tns.present)
1640
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1641
                if (che->ch[1].tns.present)
1642
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1643
                if (type <= TYPE_CPE)
1644
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1645
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
1646
                    imdct_and_windowing(ac, &che->ch[0]);
1647
                if (type == TYPE_CPE)
1648
                    imdct_and_windowing(ac, &che->ch[1]);
1649
                if (type <= TYPE_CCE)
1650
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1651
            }
1652
        }
1653
    }
1654
}
1655

    
1656
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
1657
{
1658
    int size;
1659
    AACADTSHeaderInfo hdr_info;
1660

    
1661
    size = ff_aac_parse_header(gb, &hdr_info);
1662
    if (size > 0) {
1663
        if (!ac->output_configured && hdr_info.chan_config) {
1664
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1665
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1666
            ac->m4ac.chan_config = hdr_info.chan_config;
1667
            if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
1668
                return -7;
1669
            if (output_configure(ac, ac->che_pos, new_che_pos, 1))
1670
                return -7;
1671
        }
1672
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1673
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1674
        ac->m4ac.object_type     = hdr_info.object_type;
1675
        if (hdr_info.num_aac_frames == 1) {
1676
            if (!hdr_info.crc_absent)
1677
                skip_bits(gb, 16);
1678
        } else {
1679
            av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1680
            return -1;
1681
        }
1682
    }
1683
    return size;
1684
}
1685

    
1686
static int aac_decode_frame(AVCodecContext *avccontext, void *data,
1687
                            int *data_size, AVPacket *avpkt)
1688
{
1689
    const uint8_t *buf = avpkt->data;
1690
    int buf_size = avpkt->size;
1691
    AACContext *ac = avccontext->priv_data;
1692
    ChannelElement *che = NULL;
1693
    GetBitContext gb;
1694
    enum RawDataBlockType elem_type;
1695
    int err, elem_id, data_size_tmp;
1696

    
1697
    init_get_bits(&gb, buf, buf_size * 8);
1698

    
1699
    if (show_bits(&gb, 12) == 0xfff) {
1700
        if (parse_adts_frame_header(ac, &gb) < 0) {
1701
            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1702
            return -1;
1703
        }
1704
        if (ac->m4ac.sampling_index > 12) {
1705
            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1706
            return -1;
1707
        }
1708
    }
1709

    
1710
    // parse
1711
    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1712
        elem_id = get_bits(&gb, 4);
1713

    
1714
        if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
1715
            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1716
            return -1;
1717
        }
1718

    
1719
        switch (elem_type) {
1720

    
1721
        case TYPE_SCE:
1722
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1723
            break;
1724

    
1725
        case TYPE_CPE:
1726
            err = decode_cpe(ac, &gb, che);
1727
            break;
1728

    
1729
        case TYPE_CCE:
1730
            err = decode_cce(ac, &gb, che);
1731
            break;
1732

    
1733
        case TYPE_LFE:
1734
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1735
            break;
1736

    
1737
        case TYPE_DSE:
1738
            skip_data_stream_element(&gb);
1739
            err = 0;
1740
            break;
1741

    
1742
        case TYPE_PCE: {
1743
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1744
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1745
            if ((err = decode_pce(ac, new_che_pos, &gb)))
1746
                break;
1747
            if (ac->output_configured)
1748
                av_log(avccontext, AV_LOG_ERROR,
1749
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
1750
            else
1751
                err = output_configure(ac, ac->che_pos, new_che_pos, 0);
1752
            break;
1753
        }
1754

    
1755
        case TYPE_FIL:
1756
            if (elem_id == 15)
1757
                elem_id += get_bits(&gb, 8) - 1;
1758
            while (elem_id > 0)
1759
                elem_id -= decode_extension_payload(ac, &gb, elem_id);
1760
            err = 0; /* FIXME */
1761
            break;
1762

    
1763
        default:
1764
            err = -1; /* should not happen, but keeps compiler happy */
1765
            break;
1766
        }
1767

    
1768
        if (err)
1769
            return err;
1770
    }
1771

    
1772
    spectral_to_sample(ac);
1773

    
1774
    if (!ac->is_saved) {
1775
        ac->is_saved = 1;
1776
        *data_size = 0;
1777
        return buf_size;
1778
    }
1779

    
1780
    data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
1781
    if (*data_size < data_size_tmp) {
1782
        av_log(avccontext, AV_LOG_ERROR,
1783
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
1784
               *data_size, data_size_tmp);
1785
        return -1;
1786
    }
1787
    *data_size = data_size_tmp;
1788

    
1789
    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
1790

    
1791
    return buf_size;
1792
}
1793

    
1794
static av_cold int aac_decode_close(AVCodecContext *avccontext)
1795
{
1796
    AACContext *ac = avccontext->priv_data;
1797
    int i, type;
1798

    
1799
    for (i = 0; i < MAX_ELEM_ID; i++) {
1800
        for (type = 0; type < 4; type++)
1801
            av_freep(&ac->che[type][i]);
1802
    }
1803

    
1804
    ff_mdct_end(&ac->mdct);
1805
    ff_mdct_end(&ac->mdct_small);
1806
    return 0;
1807
}
1808

    
1809
AVCodec aac_decoder = {
1810
    "aac",
1811
    CODEC_TYPE_AUDIO,
1812
    CODEC_ID_AAC,
1813
    sizeof(AACContext),
1814
    aac_decode_init,
1815
    NULL,
1816
    aac_decode_close,
1817
    aac_decode_frame,
1818
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
1819
    .sample_fmts = (const enum SampleFormat[]) {
1820
        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
1821
    },
1822
    .channel_layouts = aac_channel_layout,
1823
};