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ffmpeg / libavcodec / aac.c @ 72415b2a

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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 )
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 *
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 * This file is part of FFmpeg.
7
 *
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 * 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.
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 *
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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
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 * 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
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 * N (code in SoC repo) filterbank - Scalable Sample Rate
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 * Y                    Temporal Noise Shaping
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 * N (code in SoC repo) Long Term Prediction
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 * Y                    intensity stereo
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 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * Y                    Perceptual Noise Substitution
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 * Y                    Mid/Side stereo
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 * N                    Scalable Inverse AAC Quantization
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 * N                    Frequency Selective Switch
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 * N                    upsampling filter
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 * Y                    quantization & coding - AAC
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 * 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
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 * N                    HVXC 4kbits/s VR
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 * 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
 * Y                    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
71
 * 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
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           Parametric Stereo.
76
 */
77

    
78

    
79
#include "avcodec.h"
80
#include "internal.h"
81
#include "get_bits.h"
82
#include "dsputil.h"
83
#include "fft.h"
84
#include "lpc.h"
85

    
86
#include "aac.h"
87
#include "aactab.h"
88
#include "aacdectab.h"
89
#include "cbrt_tablegen.h"
90
#include "sbr.h"
91
#include "aacsbr.h"
92
#include "mpeg4audio.h"
93
#include "aac_parser.h"
94

    
95
#include <assert.h>
96
#include <errno.h>
97
#include <math.h>
98
#include <string.h>
99

    
100
#if ARCH_ARM
101
#   include "arm/aac.h"
102
#endif
103

    
104
union float754 {
105
    float f;
106
    uint32_t i;
107
};
108

    
109
static VLC vlc_scalefactors;
110
static VLC vlc_spectral[11];
111

    
112
static const char overread_err[] = "Input buffer exhausted before END element found\n";
113

    
114
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
115
{
116
    if (ac->tag_che_map[type][elem_id]) {
117
        return ac->tag_che_map[type][elem_id];
118
    }
119
    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
120
        return NULL;
121
    }
122
    switch (ac->m4ac.chan_config) {
123
    case 7:
124
        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
125
            ac->tags_mapped++;
126
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
127
        }
128
    case 6:
129
        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
130
           instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
131
           encountered such a stream, transfer the LFE[0] element to SCE[1] */
132
        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
133
            ac->tags_mapped++;
134
            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
135
        }
136
    case 5:
137
        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
138
            ac->tags_mapped++;
139
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
140
        }
141
    case 4:
142
        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
143
            ac->tags_mapped++;
144
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
145
        }
146
    case 3:
147
    case 2:
148
        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
149
            ac->tags_mapped++;
150
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
151
        } else if (ac->m4ac.chan_config == 2) {
152
            return NULL;
153
        }
154
    case 1:
155
        if (!ac->tags_mapped && type == TYPE_SCE) {
156
            ac->tags_mapped++;
157
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
158
        }
159
    default:
160
        return NULL;
161
    }
162
}
163

    
164
/**
165
 * Check for the channel element in the current channel position configuration.
166
 * If it exists, make sure the appropriate element is allocated and map the
167
 * channel order to match the internal FFmpeg channel layout.
168
 *
169
 * @param   che_pos current channel position configuration
170
 * @param   type channel element type
171
 * @param   id channel element id
172
 * @param   channels count of the number of channels in the configuration
173
 *
174
 * @return  Returns error status. 0 - OK, !0 - error
175
 */
176
static av_cold int che_configure(AACContext *ac,
177
                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
178
                         int type, int id,
179
                         int *channels)
180
{
181
    if (che_pos[type][id]) {
182
        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
183
            return AVERROR(ENOMEM);
184
        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
185
        if (type != TYPE_CCE) {
186
            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
187
            if (type == TYPE_CPE) {
188
                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
189
            }
190
        }
191
    } else {
192
        if (ac->che[type][id])
193
            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
194
        av_freep(&ac->che[type][id]);
195
    }
196
    return 0;
197
}
198

    
199
/**
200
 * Configure output channel order based on the current program configuration element.
201
 *
202
 * @param   che_pos current channel position configuration
203
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
204
 *
205
 * @return  Returns error status. 0 - OK, !0 - error
206
 */
207
static av_cold int output_configure(AACContext *ac,
208
                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
209
                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
210
                            int channel_config, enum OCStatus oc_type)
211
{
212
    AVCodecContext *avctx = ac->avccontext;
213
    int i, type, channels = 0, ret;
214

    
215
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
216

    
217
    if (channel_config) {
218
        for (i = 0; i < tags_per_config[channel_config]; i++) {
219
            if ((ret = che_configure(ac, che_pos,
220
                                     aac_channel_layout_map[channel_config - 1][i][0],
221
                                     aac_channel_layout_map[channel_config - 1][i][1],
222
                                     &channels)))
223
                return ret;
224
        }
225

    
226
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
227
        ac->tags_mapped = 0;
228

    
229
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
230
    } else {
231
        /* Allocate or free elements depending on if they are in the
232
         * current program configuration.
233
         *
234
         * Set up default 1:1 output mapping.
235
         *
236
         * For a 5.1 stream the output order will be:
237
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
238
         */
239

    
240
        for (i = 0; i < MAX_ELEM_ID; i++) {
241
            for (type = 0; type < 4; type++) {
242
                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
243
                    return ret;
244
            }
245
        }
246

    
247
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
248
        ac->tags_mapped = 4 * MAX_ELEM_ID;
249

    
250
        avctx->channel_layout = 0;
251
    }
252

    
253
    avctx->channels = channels;
254

    
255
    ac->output_configured = oc_type;
256

    
257
    return 0;
258
}
259

    
260
/**
261
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
262
 *
263
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
264
 * @param sce_map mono (Single Channel Element) map
265
 * @param type speaker type/position for these channels
266
 */
267
static void decode_channel_map(enum ChannelPosition *cpe_map,
268
                               enum ChannelPosition *sce_map,
269
                               enum ChannelPosition type,
270
                               GetBitContext *gb, int n)
271
{
272
    while (n--) {
273
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
274
        map[get_bits(gb, 4)] = type;
275
    }
276
}
277

    
278
/**
279
 * Decode program configuration element; reference: table 4.2.
280
 *
281
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
282
 *
283
 * @return  Returns error status. 0 - OK, !0 - error
284
 */
285
static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
286
                      GetBitContext *gb)
287
{
288
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
289
    int comment_len;
290

    
291
    skip_bits(gb, 2);  // object_type
292

    
293
    sampling_index = get_bits(gb, 4);
294
    if (ac->m4ac.sampling_index != sampling_index)
295
        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");
296

    
297
    num_front       = get_bits(gb, 4);
298
    num_side        = get_bits(gb, 4);
299
    num_back        = get_bits(gb, 4);
300
    num_lfe         = get_bits(gb, 2);
301
    num_assoc_data  = get_bits(gb, 3);
302
    num_cc          = get_bits(gb, 4);
303

    
304
    if (get_bits1(gb))
305
        skip_bits(gb, 4); // mono_mixdown_tag
306
    if (get_bits1(gb))
307
        skip_bits(gb, 4); // stereo_mixdown_tag
308

    
309
    if (get_bits1(gb))
310
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
311

    
312
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
313
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
314
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
315
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
316

    
317
    skip_bits_long(gb, 4 * num_assoc_data);
318

    
319
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
320

    
321
    align_get_bits(gb);
322

    
323
    /* comment field, first byte is length */
324
    comment_len = get_bits(gb, 8) * 8;
325
    if (get_bits_left(gb) < comment_len) {
326
        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
327
        return -1;
328
    }
329
    skip_bits_long(gb, comment_len);
330
    return 0;
331
}
332

    
333
/**
334
 * Set up channel positions based on a default channel configuration
335
 * as specified in table 1.17.
336
 *
337
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
338
 *
339
 * @return  Returns error status. 0 - OK, !0 - error
340
 */
341
static av_cold int set_default_channel_config(AACContext *ac,
342
                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
343
                                      int channel_config)
344
{
345
    if (channel_config < 1 || channel_config > 7) {
346
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
347
               channel_config);
348
        return -1;
349
    }
350

    
351
    /* default channel configurations:
352
     *
353
     * 1ch : front center (mono)
354
     * 2ch : L + R (stereo)
355
     * 3ch : front center + L + R
356
     * 4ch : front center + L + R + back center
357
     * 5ch : front center + L + R + back stereo
358
     * 6ch : front center + L + R + back stereo + LFE
359
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
360
     */
361

    
362
    if (channel_config != 2)
363
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
364
    if (channel_config > 1)
365
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
366
    if (channel_config == 4)
367
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
368
    if (channel_config > 4)
369
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
370
        = AAC_CHANNEL_BACK;  // back stereo
371
    if (channel_config > 5)
372
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
373
    if (channel_config == 7)
374
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
375

    
376
    return 0;
377
}
378

    
379
/**
380
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
381
 *
382
 * @return  Returns error status. 0 - OK, !0 - error
383
 */
384
static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
385
                                     int channel_config)
386
{
387
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
388
    int extension_flag, ret;
389

    
390
    if (get_bits1(gb)) { // frameLengthFlag
391
        av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
392
        return -1;
393
    }
394

    
395
    if (get_bits1(gb))       // dependsOnCoreCoder
396
        skip_bits(gb, 14);   // coreCoderDelay
397
    extension_flag = get_bits1(gb);
398

    
399
    if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
400
        ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
401
        skip_bits(gb, 3);     // layerNr
402

    
403
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
404
    if (channel_config == 0) {
405
        skip_bits(gb, 4);  // element_instance_tag
406
        if ((ret = decode_pce(ac, new_che_pos, gb)))
407
            return ret;
408
    } else {
409
        if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
410
            return ret;
411
    }
412
    if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
413
        return ret;
414

    
415
    if (extension_flag) {
416
        switch (ac->m4ac.object_type) {
417
        case AOT_ER_BSAC:
418
            skip_bits(gb, 5);    // numOfSubFrame
419
            skip_bits(gb, 11);   // layer_length
420
            break;
421
        case AOT_ER_AAC_LC:
422
        case AOT_ER_AAC_LTP:
423
        case AOT_ER_AAC_SCALABLE:
424
        case AOT_ER_AAC_LD:
425
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
426
                                    * aacScalefactorDataResilienceFlag
427
                                    * aacSpectralDataResilienceFlag
428
                                    */
429
            break;
430
        }
431
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
432
    }
433
    return 0;
434
}
435

    
436
/**
437
 * Decode audio specific configuration; reference: table 1.13.
438
 *
439
 * @param   data        pointer to AVCodecContext extradata
440
 * @param   data_size   size of AVCCodecContext extradata
441
 *
442
 * @return  Returns error status. 0 - OK, !0 - error
443
 */
444
static int decode_audio_specific_config(AACContext *ac, void *data,
445
                                        int data_size)
446
{
447
    GetBitContext gb;
448
    int i;
449

    
450
    init_get_bits(&gb, data, data_size * 8);
451

    
452
    if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
453
        return -1;
454
    if (ac->m4ac.sampling_index > 12) {
455
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
456
        return -1;
457
    }
458

    
459
    skip_bits_long(&gb, i);
460

    
461
    switch (ac->m4ac.object_type) {
462
    case AOT_AAC_MAIN:
463
    case AOT_AAC_LC:
464
        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
465
            return -1;
466
        break;
467
    default:
468
        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
469
               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
470
        return -1;
471
    }
472
    return 0;
473
}
474

    
475
/**
476
 * linear congruential pseudorandom number generator
477
 *
478
 * @param   previous_val    pointer to the current state of the generator
479
 *
480
 * @return  Returns a 32-bit pseudorandom integer
481
 */
482
static av_always_inline int lcg_random(int previous_val)
483
{
484
    return previous_val * 1664525 + 1013904223;
485
}
486

    
487
static av_always_inline void reset_predict_state(PredictorState *ps)
488
{
489
    ps->r0   = 0.0f;
490
    ps->r1   = 0.0f;
491
    ps->cor0 = 0.0f;
492
    ps->cor1 = 0.0f;
493
    ps->var0 = 1.0f;
494
    ps->var1 = 1.0f;
495
}
496

    
497
static void reset_all_predictors(PredictorState *ps)
498
{
499
    int i;
500
    for (i = 0; i < MAX_PREDICTORS; i++)
501
        reset_predict_state(&ps[i]);
502
}
503

    
504
static void reset_predictor_group(PredictorState *ps, int group_num)
505
{
506
    int i;
507
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
508
        reset_predict_state(&ps[i]);
509
}
510

    
511
static av_cold int aac_decode_init(AVCodecContext *avccontext)
512
{
513
    AACContext *ac = avccontext->priv_data;
514
    int i;
515

    
516
    ac->avccontext = avccontext;
517
    ac->m4ac.sample_rate = avccontext->sample_rate;
518

    
519
    if (avccontext->extradata_size > 0) {
520
        if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
521
            return -1;
522
    }
523

    
524
    avccontext->sample_fmt = SAMPLE_FMT_S16;
525

    
526
    AAC_INIT_VLC_STATIC( 0, 304);
527
    AAC_INIT_VLC_STATIC( 1, 270);
528
    AAC_INIT_VLC_STATIC( 2, 550);
529
    AAC_INIT_VLC_STATIC( 3, 300);
530
    AAC_INIT_VLC_STATIC( 4, 328);
531
    AAC_INIT_VLC_STATIC( 5, 294);
532
    AAC_INIT_VLC_STATIC( 6, 306);
533
    AAC_INIT_VLC_STATIC( 7, 268);
534
    AAC_INIT_VLC_STATIC( 8, 510);
535
    AAC_INIT_VLC_STATIC( 9, 366);
536
    AAC_INIT_VLC_STATIC(10, 462);
537

    
538
    ff_aac_sbr_init();
539

    
540
    dsputil_init(&ac->dsp, avccontext);
541

    
542
    ac->random_state = 0x1f2e3d4c;
543

    
544
    // -1024 - Compensate wrong IMDCT method.
545
    // 32768 - Required to scale values to the correct range for the bias method
546
    //         for float to int16 conversion.
547

    
548
    if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
549
        ac->add_bias  = 385.0f;
550
        ac->sf_scale  = 1. / (-1024. * 32768.);
551
        ac->sf_offset = 0;
552
    } else {
553
        ac->add_bias  = 0.0f;
554
        ac->sf_scale  = 1. / -1024.;
555
        ac->sf_offset = 60;
556
    }
557

    
558
#if !CONFIG_HARDCODED_TABLES
559
    for (i = 0; i < 428; i++)
560
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
561
#endif /* CONFIG_HARDCODED_TABLES */
562

    
563
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
564
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
565
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
566
                    352);
567

    
568
    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
569
    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
570
    // window initialization
571
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
572
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
573
    ff_init_ff_sine_windows(10);
574
    ff_init_ff_sine_windows( 7);
575

    
576
    cbrt_tableinit();
577

    
578
    return 0;
579
}
580

    
581
/**
582
 * Skip data_stream_element; reference: table 4.10.
583
 */
584
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
585
{
586
    int byte_align = get_bits1(gb);
587
    int count = get_bits(gb, 8);
588
    if (count == 255)
589
        count += get_bits(gb, 8);
590
    if (byte_align)
591
        align_get_bits(gb);
592

    
593
    if (get_bits_left(gb) < 8 * count) {
594
        av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
595
        return -1;
596
    }
597
    skip_bits_long(gb, 8 * count);
598
    return 0;
599
}
600

    
601
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
602
                             GetBitContext *gb)
603
{
604
    int sfb;
605
    if (get_bits1(gb)) {
606
        ics->predictor_reset_group = get_bits(gb, 5);
607
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
608
            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
609
            return -1;
610
        }
611
    }
612
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
613
        ics->prediction_used[sfb] = get_bits1(gb);
614
    }
615
    return 0;
616
}
617

    
618
/**
619
 * Decode Individual Channel Stream info; reference: table 4.6.
620
 *
621
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
622
 */
623
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
624
                           GetBitContext *gb, int common_window)
625
{
626
    if (get_bits1(gb)) {
627
        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
628
        memset(ics, 0, sizeof(IndividualChannelStream));
629
        return -1;
630
    }
631
    ics->window_sequence[1] = ics->window_sequence[0];
632
    ics->window_sequence[0] = get_bits(gb, 2);
633
    ics->use_kb_window[1]   = ics->use_kb_window[0];
634
    ics->use_kb_window[0]   = get_bits1(gb);
635
    ics->num_window_groups  = 1;
636
    ics->group_len[0]       = 1;
637
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
638
        int i;
639
        ics->max_sfb = get_bits(gb, 4);
640
        for (i = 0; i < 7; i++) {
641
            if (get_bits1(gb)) {
642
                ics->group_len[ics->num_window_groups - 1]++;
643
            } else {
644
                ics->num_window_groups++;
645
                ics->group_len[ics->num_window_groups - 1] = 1;
646
            }
647
        }
648
        ics->num_windows       = 8;
649
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
650
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
651
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
652
        ics->predictor_present = 0;
653
    } else {
654
        ics->max_sfb               = get_bits(gb, 6);
655
        ics->num_windows           = 1;
656
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
657
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
658
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
659
        ics->predictor_present     = get_bits1(gb);
660
        ics->predictor_reset_group = 0;
661
        if (ics->predictor_present) {
662
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
663
                if (decode_prediction(ac, ics, gb)) {
664
                    memset(ics, 0, sizeof(IndividualChannelStream));
665
                    return -1;
666
                }
667
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
668
                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
669
                memset(ics, 0, sizeof(IndividualChannelStream));
670
                return -1;
671
            } else {
672
                av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
673
                memset(ics, 0, sizeof(IndividualChannelStream));
674
                return -1;
675
            }
676
        }
677
    }
678

    
679
    if (ics->max_sfb > ics->num_swb) {
680
        av_log(ac->avccontext, AV_LOG_ERROR,
681
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
682
               ics->max_sfb, ics->num_swb);
683
        memset(ics, 0, sizeof(IndividualChannelStream));
684
        return -1;
685
    }
686

    
687
    return 0;
688
}
689

    
690
/**
691
 * Decode band types (section_data payload); reference: table 4.46.
692
 *
693
 * @param   band_type           array of the used band type
694
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
695
 *
696
 * @return  Returns error status. 0 - OK, !0 - error
697
 */
698
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
699
                             int band_type_run_end[120], GetBitContext *gb,
700
                             IndividualChannelStream *ics)
701
{
702
    int g, idx = 0;
703
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
704
    for (g = 0; g < ics->num_window_groups; g++) {
705
        int k = 0;
706
        while (k < ics->max_sfb) {
707
            uint8_t sect_end = k;
708
            int sect_len_incr;
709
            int sect_band_type = get_bits(gb, 4);
710
            if (sect_band_type == 12) {
711
                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
712
                return -1;
713
            }
714
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
715
                sect_end += sect_len_incr;
716
            sect_end += sect_len_incr;
717
            if (get_bits_left(gb) < 0) {
718
                av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
719
                return -1;
720
            }
721
            if (sect_end > ics->max_sfb) {
722
                av_log(ac->avccontext, AV_LOG_ERROR,
723
                       "Number of bands (%d) exceeds limit (%d).\n",
724
                       sect_end, ics->max_sfb);
725
                return -1;
726
            }
727
            for (; k < sect_end; k++) {
728
                band_type        [idx]   = sect_band_type;
729
                band_type_run_end[idx++] = sect_end;
730
            }
731
        }
732
    }
733
    return 0;
734
}
735

    
736
/**
737
 * Decode scalefactors; reference: table 4.47.
738
 *
739
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
740
 * @param   band_type           array of the used band type
741
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
742
 * @param   sf                  array of scalefactors or intensity stereo positions
743
 *
744
 * @return  Returns error status. 0 - OK, !0 - error
745
 */
746
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
747
                               unsigned int global_gain,
748
                               IndividualChannelStream *ics,
749
                               enum BandType band_type[120],
750
                               int band_type_run_end[120])
751
{
752
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
753
    int g, i, idx = 0;
754
    int offset[3] = { global_gain, global_gain - 90, 100 };
755
    int noise_flag = 1;
756
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
757
    for (g = 0; g < ics->num_window_groups; g++) {
758
        for (i = 0; i < ics->max_sfb;) {
759
            int run_end = band_type_run_end[idx];
760
            if (band_type[idx] == ZERO_BT) {
761
                for (; i < run_end; i++, idx++)
762
                    sf[idx] = 0.;
763
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
764
                for (; i < run_end; i++, idx++) {
765
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
766
                    if (offset[2] > 255U) {
767
                        av_log(ac->avccontext, AV_LOG_ERROR,
768
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
769
                        return -1;
770
                    }
771
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
772
                }
773
            } else if (band_type[idx] == NOISE_BT) {
774
                for (; i < run_end; i++, idx++) {
775
                    if (noise_flag-- > 0)
776
                        offset[1] += get_bits(gb, 9) - 256;
777
                    else
778
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
779
                    if (offset[1] > 255U) {
780
                        av_log(ac->avccontext, AV_LOG_ERROR,
781
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
782
                        return -1;
783
                    }
784
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
785
                }
786
            } else {
787
                for (; i < run_end; i++, idx++) {
788
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
789
                    if (offset[0] > 255U) {
790
                        av_log(ac->avccontext, AV_LOG_ERROR,
791
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
792
                        return -1;
793
                    }
794
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
795
                }
796
            }
797
        }
798
    }
799
    return 0;
800
}
801

    
802
/**
803
 * Decode pulse data; reference: table 4.7.
804
 */
805
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
806
                         const uint16_t *swb_offset, int num_swb)
807
{
808
    int i, pulse_swb;
809
    pulse->num_pulse = get_bits(gb, 2) + 1;
810
    pulse_swb        = get_bits(gb, 6);
811
    if (pulse_swb >= num_swb)
812
        return -1;
813
    pulse->pos[0]    = swb_offset[pulse_swb];
814
    pulse->pos[0]   += get_bits(gb, 5);
815
    if (pulse->pos[0] > 1023)
816
        return -1;
817
    pulse->amp[0]    = get_bits(gb, 4);
818
    for (i = 1; i < pulse->num_pulse; i++) {
819
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
820
        if (pulse->pos[i] > 1023)
821
            return -1;
822
        pulse->amp[i] = get_bits(gb, 4);
823
    }
824
    return 0;
825
}
826

    
827
/**
828
 * Decode Temporal Noise Shaping data; reference: table 4.48.
829
 *
830
 * @return  Returns error status. 0 - OK, !0 - error
831
 */
832
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
833
                      GetBitContext *gb, const IndividualChannelStream *ics)
834
{
835
    int w, filt, i, coef_len, coef_res, coef_compress;
836
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
837
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
838
    for (w = 0; w < ics->num_windows; w++) {
839
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
840
            coef_res = get_bits1(gb);
841

    
842
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
843
                int tmp2_idx;
844
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
845

    
846
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
847
                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
848
                           tns->order[w][filt], tns_max_order);
849
                    tns->order[w][filt] = 0;
850
                    return -1;
851
                }
852
                if (tns->order[w][filt]) {
853
                    tns->direction[w][filt] = get_bits1(gb);
854
                    coef_compress = get_bits1(gb);
855
                    coef_len = coef_res + 3 - coef_compress;
856
                    tmp2_idx = 2 * coef_compress + coef_res;
857

    
858
                    for (i = 0; i < tns->order[w][filt]; i++)
859
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
860
                }
861
            }
862
        }
863
    }
864
    return 0;
865
}
866

    
867
/**
868
 * Decode Mid/Side data; reference: table 4.54.
869
 *
870
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
871
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
872
 *                      [3] reserved for scalable AAC
873
 */
874
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
875
                                   int ms_present)
876
{
877
    int idx;
878
    if (ms_present == 1) {
879
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
880
            cpe->ms_mask[idx] = get_bits1(gb);
881
    } else if (ms_present == 2) {
882
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
883
    }
884
}
885

    
886
#ifndef VMUL2
887
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
888
                           const float *scale)
889
{
890
    float s = *scale;
891
    *dst++ = v[idx    & 15] * s;
892
    *dst++ = v[idx>>4 & 15] * s;
893
    return dst;
894
}
895
#endif
896

    
897
#ifndef VMUL4
898
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
899
                           const float *scale)
900
{
901
    float s = *scale;
902
    *dst++ = v[idx    & 3] * s;
903
    *dst++ = v[idx>>2 & 3] * s;
904
    *dst++ = v[idx>>4 & 3] * s;
905
    *dst++ = v[idx>>6 & 3] * s;
906
    return dst;
907
}
908
#endif
909

    
910
#ifndef VMUL2S
911
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
912
                            unsigned sign, const float *scale)
913
{
914
    union float754 s0, s1;
915

    
916
    s0.f = s1.f = *scale;
917
    s0.i ^= sign >> 1 << 31;
918
    s1.i ^= sign      << 31;
919

    
920
    *dst++ = v[idx    & 15] * s0.f;
921
    *dst++ = v[idx>>4 & 15] * s1.f;
922

    
923
    return dst;
924
}
925
#endif
926

    
927
#ifndef VMUL4S
928
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
929
                            unsigned sign, const float *scale)
930
{
931
    unsigned nz = idx >> 12;
932
    union float754 s = { .f = *scale };
933
    union float754 t;
934

    
935
    t.i = s.i ^ (sign & 1<<31);
936
    *dst++ = v[idx    & 3] * t.f;
937

    
938
    sign <<= nz & 1; nz >>= 1;
939
    t.i = s.i ^ (sign & 1<<31);
940
    *dst++ = v[idx>>2 & 3] * t.f;
941

    
942
    sign <<= nz & 1; nz >>= 1;
943
    t.i = s.i ^ (sign & 1<<31);
944
    *dst++ = v[idx>>4 & 3] * t.f;
945

    
946
    sign <<= nz & 1; nz >>= 1;
947
    t.i = s.i ^ (sign & 1<<31);
948
    *dst++ = v[idx>>6 & 3] * t.f;
949

    
950
    return dst;
951
}
952
#endif
953

    
954
/**
955
 * Decode spectral data; reference: table 4.50.
956
 * Dequantize and scale spectral data; reference: 4.6.3.3.
957
 *
958
 * @param   coef            array of dequantized, scaled spectral data
959
 * @param   sf              array of scalefactors or intensity stereo positions
960
 * @param   pulse_present   set if pulses are present
961
 * @param   pulse           pointer to pulse data struct
962
 * @param   band_type       array of the used band type
963
 *
964
 * @return  Returns error status. 0 - OK, !0 - error
965
 */
966
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
967
                                       GetBitContext *gb, const float sf[120],
968
                                       int pulse_present, const Pulse *pulse,
969
                                       const IndividualChannelStream *ics,
970
                                       enum BandType band_type[120])
971
{
972
    int i, k, g, idx = 0;
973
    const int c = 1024 / ics->num_windows;
974
    const uint16_t *offsets = ics->swb_offset;
975
    float *coef_base = coef;
976
    int err_idx;
977

    
978
    for (g = 0; g < ics->num_windows; g++)
979
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
980

    
981
    for (g = 0; g < ics->num_window_groups; g++) {
982
        unsigned g_len = ics->group_len[g];
983

    
984
        for (i = 0; i < ics->max_sfb; i++, idx++) {
985
            const unsigned cbt_m1 = band_type[idx] - 1;
986
            float *cfo = coef + offsets[i];
987
            int off_len = offsets[i + 1] - offsets[i];
988
            int group;
989

    
990
            if (cbt_m1 >= INTENSITY_BT2 - 1) {
991
                for (group = 0; group < g_len; group++, cfo+=128) {
992
                    memset(cfo, 0, off_len * sizeof(float));
993
                }
994
            } else if (cbt_m1 == NOISE_BT - 1) {
995
                for (group = 0; group < g_len; group++, cfo+=128) {
996
                    float scale;
997
                    float band_energy;
998

    
999
                    for (k = 0; k < off_len; k++) {
1000
                        ac->random_state  = lcg_random(ac->random_state);
1001
                        cfo[k] = ac->random_state;
1002
                    }
1003

    
1004
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1005
                    scale = sf[idx] / sqrtf(band_energy);
1006
                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1007
                }
1008
            } else {
1009
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1010
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1011
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1012
                const int cb_size = ff_aac_spectral_sizes[cbt_m1];
1013
                OPEN_READER(re, gb);
1014

    
1015
                switch (cbt_m1 >> 1) {
1016
                case 0:
1017
                    for (group = 0; group < g_len; group++, cfo+=128) {
1018
                        float *cf = cfo;
1019
                        int len = off_len;
1020

    
1021
                        do {
1022
                            int code;
1023
                            unsigned cb_idx;
1024

    
1025
                            UPDATE_CACHE(re, gb);
1026
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1027

    
1028
                            if (code >= cb_size) {
1029
                                err_idx = code;
1030
                                goto err_cb_overflow;
1031
                            }
1032

    
1033
                            cb_idx = cb_vector_idx[code];
1034
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1035
                        } while (len -= 4);
1036
                    }
1037
                    break;
1038

    
1039
                case 1:
1040
                    for (group = 0; group < g_len; group++, cfo+=128) {
1041
                        float *cf = cfo;
1042
                        int len = off_len;
1043

    
1044
                        do {
1045
                            int code;
1046
                            unsigned nnz;
1047
                            unsigned cb_idx;
1048
                            uint32_t bits;
1049

    
1050
                            UPDATE_CACHE(re, gb);
1051
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1052

    
1053
                            if (code >= cb_size) {
1054
                                err_idx = code;
1055
                                goto err_cb_overflow;
1056
                            }
1057

    
1058
#if MIN_CACHE_BITS < 20
1059
                            UPDATE_CACHE(re, gb);
1060
#endif
1061
                            cb_idx = cb_vector_idx[code];
1062
                            nnz = cb_idx >> 8 & 15;
1063
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1064
                            LAST_SKIP_BITS(re, gb, nnz);
1065
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1066
                        } while (len -= 4);
1067
                    }
1068
                    break;
1069

    
1070
                case 2:
1071
                    for (group = 0; group < g_len; group++, cfo+=128) {
1072
                        float *cf = cfo;
1073
                        int len = off_len;
1074

    
1075
                        do {
1076
                            int code;
1077
                            unsigned cb_idx;
1078

    
1079
                            UPDATE_CACHE(re, gb);
1080
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1081

    
1082
                            if (code >= cb_size) {
1083
                                err_idx = code;
1084
                                goto err_cb_overflow;
1085
                            }
1086

    
1087
                            cb_idx = cb_vector_idx[code];
1088
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1089
                        } while (len -= 2);
1090
                    }
1091
                    break;
1092

    
1093
                case 3:
1094
                case 4:
1095
                    for (group = 0; group < g_len; group++, cfo+=128) {
1096
                        float *cf = cfo;
1097
                        int len = off_len;
1098

    
1099
                        do {
1100
                            int code;
1101
                            unsigned nnz;
1102
                            unsigned cb_idx;
1103
                            unsigned sign;
1104

    
1105
                            UPDATE_CACHE(re, gb);
1106
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1107

    
1108
                            if (code >= cb_size) {
1109
                                err_idx = code;
1110
                                goto err_cb_overflow;
1111
                            }
1112

    
1113
                            cb_idx = cb_vector_idx[code];
1114
                            nnz = cb_idx >> 8 & 15;
1115
                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1116
                            LAST_SKIP_BITS(re, gb, nnz);
1117
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1118
                        } while (len -= 2);
1119
                    }
1120
                    break;
1121

    
1122
                default:
1123
                    for (group = 0; group < g_len; group++, cfo+=128) {
1124
                        float *cf = cfo;
1125
                        uint32_t *icf = (uint32_t *) cf;
1126
                        int len = off_len;
1127

    
1128
                        do {
1129
                            int code;
1130
                            unsigned nzt, nnz;
1131
                            unsigned cb_idx;
1132
                            uint32_t bits;
1133
                            int j;
1134

    
1135
                            UPDATE_CACHE(re, gb);
1136
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1137

    
1138
                            if (!code) {
1139
                                *icf++ = 0;
1140
                                *icf++ = 0;
1141
                                continue;
1142
                            }
1143

    
1144
                            if (code >= cb_size) {
1145
                                err_idx = code;
1146
                                goto err_cb_overflow;
1147
                            }
1148

    
1149
                            cb_idx = cb_vector_idx[code];
1150
                            nnz = cb_idx >> 12;
1151
                            nzt = cb_idx >> 8;
1152
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1153
                            LAST_SKIP_BITS(re, gb, nnz);
1154

    
1155
                            for (j = 0; j < 2; j++) {
1156
                                if (nzt & 1<<j) {
1157
                                    uint32_t b;
1158
                                    int n;
1159
                                    /* The total length of escape_sequence must be < 22 bits according
1160
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1161
                                    UPDATE_CACHE(re, gb);
1162
                                    b = GET_CACHE(re, gb);
1163
                                    b = 31 - av_log2(~b);
1164

    
1165
                                    if (b > 8) {
1166
                                        av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1167
                                        return -1;
1168
                                    }
1169

    
1170
#if MIN_CACHE_BITS < 21
1171
                                    LAST_SKIP_BITS(re, gb, b + 1);
1172
                                    UPDATE_CACHE(re, gb);
1173
#else
1174
                                    SKIP_BITS(re, gb, b + 1);
1175
#endif
1176
                                    b += 4;
1177
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
1178
                                    LAST_SKIP_BITS(re, gb, b);
1179
                                    *icf++ = cbrt_tab[n] | (bits & 1<<31);
1180
                                    bits <<= 1;
1181
                                } else {
1182
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1183
                                    *icf++ = (bits & 1<<31) | v;
1184
                                    bits <<= !!v;
1185
                                }
1186
                                cb_idx >>= 4;
1187
                            }
1188
                        } while (len -= 2);
1189

    
1190
                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1191
                    }
1192
                }
1193

    
1194
                CLOSE_READER(re, gb);
1195
            }
1196
        }
1197
        coef += g_len << 7;
1198
    }
1199

    
1200
    if (pulse_present) {
1201
        idx = 0;
1202
        for (i = 0; i < pulse->num_pulse; i++) {
1203
            float co = coef_base[ pulse->pos[i] ];
1204
            while (offsets[idx + 1] <= pulse->pos[i])
1205
                idx++;
1206
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1207
                float ico = -pulse->amp[i];
1208
                if (co) {
1209
                    co /= sf[idx];
1210
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1211
                }
1212
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1213
            }
1214
        }
1215
    }
1216
    return 0;
1217

    
1218
err_cb_overflow:
1219
    av_log(ac->avccontext, AV_LOG_ERROR,
1220
           "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
1221
           band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
1222
    return -1;
1223
}
1224

    
1225
static av_always_inline float flt16_round(float pf)
1226
{
1227
    union float754 tmp;
1228
    tmp.f = pf;
1229
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1230
    return tmp.f;
1231
}
1232

    
1233
static av_always_inline float flt16_even(float pf)
1234
{
1235
    union float754 tmp;
1236
    tmp.f = pf;
1237
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1238
    return tmp.f;
1239
}
1240

    
1241
static av_always_inline float flt16_trunc(float pf)
1242
{
1243
    union float754 pun;
1244
    pun.f = pf;
1245
    pun.i &= 0xFFFF0000U;
1246
    return pun.f;
1247
}
1248

    
1249
static av_always_inline void predict(AACContext *ac, PredictorState *ps, float *coef,
1250
                    int output_enable)
1251
{
1252
    const float a     = 0.953125; // 61.0 / 64
1253
    const float alpha = 0.90625;  // 29.0 / 32
1254
    float e0, e1;
1255
    float pv;
1256
    float k1, k2;
1257

    
1258
    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
1259
    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
1260

    
1261
    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
1262
    if (output_enable)
1263
        *coef += pv * ac->sf_scale;
1264

    
1265
    e0 = *coef / ac->sf_scale;
1266
    e1 = e0 - k1 * ps->r0;
1267

    
1268
    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
1269
    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
1270
    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
1271
    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
1272

    
1273
    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
1274
    ps->r0 = flt16_trunc(a * e0);
1275
}
1276

    
1277
/**
1278
 * Apply AAC-Main style frequency domain prediction.
1279
 */
1280
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1281
{
1282
    int sfb, k;
1283

    
1284
    if (!sce->ics.predictor_initialized) {
1285
        reset_all_predictors(sce->predictor_state);
1286
        sce->ics.predictor_initialized = 1;
1287
    }
1288

    
1289
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1290
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1291
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1292
                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
1293
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1294
            }
1295
        }
1296
        if (sce->ics.predictor_reset_group)
1297
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1298
    } else
1299
        reset_all_predictors(sce->predictor_state);
1300
}
1301

    
1302
/**
1303
 * Decode an individual_channel_stream payload; reference: table 4.44.
1304
 *
1305
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1306
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1307
 *
1308
 * @return  Returns error status. 0 - OK, !0 - error
1309
 */
1310
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1311
                      GetBitContext *gb, int common_window, int scale_flag)
1312
{
1313
    Pulse pulse;
1314
    TemporalNoiseShaping    *tns = &sce->tns;
1315
    IndividualChannelStream *ics = &sce->ics;
1316
    float *out = sce->coeffs;
1317
    int global_gain, pulse_present = 0;
1318

    
1319
    /* This assignment is to silence a GCC warning about the variable being used
1320
     * uninitialized when in fact it always is.
1321
     */
1322
    pulse.num_pulse = 0;
1323

    
1324
    global_gain = get_bits(gb, 8);
1325

    
1326
    if (!common_window && !scale_flag) {
1327
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1328
            return -1;
1329
    }
1330

    
1331
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1332
        return -1;
1333
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1334
        return -1;
1335

    
1336
    pulse_present = 0;
1337
    if (!scale_flag) {
1338
        if ((pulse_present = get_bits1(gb))) {
1339
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1340
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1341
                return -1;
1342
            }
1343
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1344
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1345
                return -1;
1346
            }
1347
        }
1348
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1349
            return -1;
1350
        if (get_bits1(gb)) {
1351
            av_log_missing_feature(ac->avccontext, "SSR", 1);
1352
            return -1;
1353
        }
1354
    }
1355

    
1356
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1357
        return -1;
1358

    
1359
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1360
        apply_prediction(ac, sce);
1361

    
1362
    return 0;
1363
}
1364

    
1365
/**
1366
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1367
 */
1368
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1369
{
1370
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1371
    float *ch0 = cpe->ch[0].coeffs;
1372
    float *ch1 = cpe->ch[1].coeffs;
1373
    int g, i, group, idx = 0;
1374
    const uint16_t *offsets = ics->swb_offset;
1375
    for (g = 0; g < ics->num_window_groups; g++) {
1376
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1377
            if (cpe->ms_mask[idx] &&
1378
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1379
                for (group = 0; group < ics->group_len[g]; group++) {
1380
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1381
                                              ch1 + group * 128 + offsets[i],
1382
                                              offsets[i+1] - offsets[i]);
1383
                }
1384
            }
1385
        }
1386
        ch0 += ics->group_len[g] * 128;
1387
        ch1 += ics->group_len[g] * 128;
1388
    }
1389
}
1390

    
1391
/**
1392
 * intensity stereo decoding; reference: 4.6.8.2.3
1393
 *
1394
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1395
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1396
 *                      [3] reserved for scalable AAC
1397
 */
1398
static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
1399
{
1400
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1401
    SingleChannelElement         *sce1 = &cpe->ch[1];
1402
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1403
    const uint16_t *offsets = ics->swb_offset;
1404
    int g, group, i, k, idx = 0;
1405
    int c;
1406
    float scale;
1407
    for (g = 0; g < ics->num_window_groups; g++) {
1408
        for (i = 0; i < ics->max_sfb;) {
1409
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1410
                const int bt_run_end = sce1->band_type_run_end[idx];
1411
                for (; i < bt_run_end; i++, idx++) {
1412
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1413
                    if (ms_present)
1414
                        c *= 1 - 2 * cpe->ms_mask[idx];
1415
                    scale = c * sce1->sf[idx];
1416
                    for (group = 0; group < ics->group_len[g]; group++)
1417
                        for (k = offsets[i]; k < offsets[i + 1]; k++)
1418
                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
1419
                }
1420
            } else {
1421
                int bt_run_end = sce1->band_type_run_end[idx];
1422
                idx += bt_run_end - i;
1423
                i    = bt_run_end;
1424
            }
1425
        }
1426
        coef0 += ics->group_len[g] * 128;
1427
        coef1 += ics->group_len[g] * 128;
1428
    }
1429
}
1430

    
1431
/**
1432
 * Decode a channel_pair_element; reference: table 4.4.
1433
 *
1434
 * @param   elem_id Identifies the instance of a syntax element.
1435
 *
1436
 * @return  Returns error status. 0 - OK, !0 - error
1437
 */
1438
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1439
{
1440
    int i, ret, common_window, ms_present = 0;
1441

    
1442
    common_window = get_bits1(gb);
1443
    if (common_window) {
1444
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1445
            return -1;
1446
        i = cpe->ch[1].ics.use_kb_window[0];
1447
        cpe->ch[1].ics = cpe->ch[0].ics;
1448
        cpe->ch[1].ics.use_kb_window[1] = i;
1449
        ms_present = get_bits(gb, 2);
1450
        if (ms_present == 3) {
1451
            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1452
            return -1;
1453
        } else if (ms_present)
1454
            decode_mid_side_stereo(cpe, gb, ms_present);
1455
    }
1456
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1457
        return ret;
1458
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1459
        return ret;
1460

    
1461
    if (common_window) {
1462
        if (ms_present)
1463
            apply_mid_side_stereo(ac, cpe);
1464
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1465
            apply_prediction(ac, &cpe->ch[0]);
1466
            apply_prediction(ac, &cpe->ch[1]);
1467
        }
1468
    }
1469

    
1470
    apply_intensity_stereo(cpe, ms_present);
1471
    return 0;
1472
}
1473

    
1474
/**
1475
 * Decode coupling_channel_element; reference: table 4.8.
1476
 *
1477
 * @param   elem_id Identifies the instance of a syntax element.
1478
 *
1479
 * @return  Returns error status. 0 - OK, !0 - error
1480
 */
1481
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1482
{
1483
    int num_gain = 0;
1484
    int c, g, sfb, ret;
1485
    int sign;
1486
    float scale;
1487
    SingleChannelElement *sce = &che->ch[0];
1488
    ChannelCoupling     *coup = &che->coup;
1489

    
1490
    coup->coupling_point = 2 * get_bits1(gb);
1491
    coup->num_coupled = get_bits(gb, 3);
1492
    for (c = 0; c <= coup->num_coupled; c++) {
1493
        num_gain++;
1494
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1495
        coup->id_select[c] = get_bits(gb, 4);
1496
        if (coup->type[c] == TYPE_CPE) {
1497
            coup->ch_select[c] = get_bits(gb, 2);
1498
            if (coup->ch_select[c] == 3)
1499
                num_gain++;
1500
        } else
1501
            coup->ch_select[c] = 2;
1502
    }
1503
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1504

    
1505
    sign  = get_bits(gb, 1);
1506
    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1507

    
1508
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1509
        return ret;
1510

    
1511
    for (c = 0; c < num_gain; c++) {
1512
        int idx  = 0;
1513
        int cge  = 1;
1514
        int gain = 0;
1515
        float gain_cache = 1.;
1516
        if (c) {
1517
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1518
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1519
            gain_cache = pow(scale, -gain);
1520
        }
1521
        if (coup->coupling_point == AFTER_IMDCT) {
1522
            coup->gain[c][0] = gain_cache;
1523
        } else {
1524
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1525
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1526
                    if (sce->band_type[idx] != ZERO_BT) {
1527
                        if (!cge) {
1528
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1529
                            if (t) {
1530
                                int s = 1;
1531
                                t = gain += t;
1532
                                if (sign) {
1533
                                    s  -= 2 * (t & 0x1);
1534
                                    t >>= 1;
1535
                                }
1536
                                gain_cache = pow(scale, -t) * s;
1537
                            }
1538
                        }
1539
                        coup->gain[c][idx] = gain_cache;
1540
                    }
1541
                }
1542
            }
1543
        }
1544
    }
1545
    return 0;
1546
}
1547

    
1548
/**
1549
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1550
 *
1551
 * @return  Returns number of bytes consumed.
1552
 */
1553
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1554
                                         GetBitContext *gb)
1555
{
1556
    int i;
1557
    int num_excl_chan = 0;
1558

    
1559
    do {
1560
        for (i = 0; i < 7; i++)
1561
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1562
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1563

    
1564
    return num_excl_chan / 7;
1565
}
1566

    
1567
/**
1568
 * Decode dynamic range information; reference: table 4.52.
1569
 *
1570
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1571
 *
1572
 * @return  Returns number of bytes consumed.
1573
 */
1574
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1575
                                GetBitContext *gb, int cnt)
1576
{
1577
    int n             = 1;
1578
    int drc_num_bands = 1;
1579
    int i;
1580

    
1581
    /* pce_tag_present? */
1582
    if (get_bits1(gb)) {
1583
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1584
        skip_bits(gb, 4); // tag_reserved_bits
1585
        n++;
1586
    }
1587

    
1588
    /* excluded_chns_present? */
1589
    if (get_bits1(gb)) {
1590
        n += decode_drc_channel_exclusions(che_drc, gb);
1591
    }
1592

    
1593
    /* drc_bands_present? */
1594
    if (get_bits1(gb)) {
1595
        che_drc->band_incr            = get_bits(gb, 4);
1596
        che_drc->interpolation_scheme = get_bits(gb, 4);
1597
        n++;
1598
        drc_num_bands += che_drc->band_incr;
1599
        for (i = 0; i < drc_num_bands; i++) {
1600
            che_drc->band_top[i] = get_bits(gb, 8);
1601
            n++;
1602
        }
1603
    }
1604

    
1605
    /* prog_ref_level_present? */
1606
    if (get_bits1(gb)) {
1607
        che_drc->prog_ref_level = get_bits(gb, 7);
1608
        skip_bits1(gb); // prog_ref_level_reserved_bits
1609
        n++;
1610
    }
1611

    
1612
    for (i = 0; i < drc_num_bands; i++) {
1613
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1614
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1615
        n++;
1616
    }
1617

    
1618
    return n;
1619
}
1620

    
1621
/**
1622
 * Decode extension data (incomplete); reference: table 4.51.
1623
 *
1624
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1625
 *
1626
 * @return Returns number of bytes consumed
1627
 */
1628
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
1629
                                    ChannelElement *che, enum RawDataBlockType elem_type)
1630
{
1631
    int crc_flag = 0;
1632
    int res = cnt;
1633
    switch (get_bits(gb, 4)) { // extension type
1634
    case EXT_SBR_DATA_CRC:
1635
        crc_flag++;
1636
    case EXT_SBR_DATA:
1637
        if (!che) {
1638
            av_log(ac->avccontext, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
1639
            return res;
1640
        } else if (!ac->m4ac.sbr) {
1641
            av_log(ac->avccontext, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
1642
            skip_bits_long(gb, 8 * cnt - 4);
1643
            return res;
1644
        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
1645
            av_log(ac->avccontext, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
1646
            skip_bits_long(gb, 8 * cnt - 4);
1647
            return res;
1648
        } else {
1649
            ac->m4ac.sbr = 1;
1650
        }
1651
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
1652
        break;
1653
    case EXT_DYNAMIC_RANGE:
1654
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1655
        break;
1656
    case EXT_FILL:
1657
    case EXT_FILL_DATA:
1658
    case EXT_DATA_ELEMENT:
1659
    default:
1660
        skip_bits_long(gb, 8 * cnt - 4);
1661
        break;
1662
    };
1663
    return res;
1664
}
1665

    
1666
/**
1667
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1668
 *
1669
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1670
 * @param   coef    spectral coefficients
1671
 */
1672
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1673
                      IndividualChannelStream *ics, int decode)
1674
{
1675
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1676
    int w, filt, m, i;
1677
    int bottom, top, order, start, end, size, inc;
1678
    float lpc[TNS_MAX_ORDER];
1679

    
1680
    for (w = 0; w < ics->num_windows; w++) {
1681
        bottom = ics->num_swb;
1682
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1683
            top    = bottom;
1684
            bottom = FFMAX(0, top - tns->length[w][filt]);
1685
            order  = tns->order[w][filt];
1686
            if (order == 0)
1687
                continue;
1688

    
1689
            // tns_decode_coef
1690
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1691

    
1692
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1693
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1694
            if ((size = end - start) <= 0)
1695
                continue;
1696
            if (tns->direction[w][filt]) {
1697
                inc = -1;
1698
                start = end - 1;
1699
            } else {
1700
                inc = 1;
1701
            }
1702
            start += w * 128;
1703

    
1704
            // ar filter
1705
            for (m = 0; m < size; m++, start += inc)
1706
                for (i = 1; i <= FFMIN(m, order); i++)
1707
                    coef[start] -= coef[start - i * inc] * lpc[i - 1];
1708
        }
1709
    }
1710
}
1711

    
1712
/**
1713
 * Conduct IMDCT and windowing.
1714
 */
1715
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce, float bias)
1716
{
1717
    IndividualChannelStream *ics = &sce->ics;
1718
    float *in    = sce->coeffs;
1719
    float *out   = sce->ret;
1720
    float *saved = sce->saved;
1721
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1722
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1723
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1724
    float *buf  = ac->buf_mdct;
1725
    float *temp = ac->temp;
1726
    int i;
1727

    
1728
    // imdct
1729
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1730
        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1731
            av_log(ac->avccontext, AV_LOG_WARNING,
1732
                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1733
                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1734
        for (i = 0; i < 1024; i += 128)
1735
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1736
    } else
1737
        ff_imdct_half(&ac->mdct, buf, in);
1738

    
1739
    /* window overlapping
1740
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1741
     * and long to short transitions are considered to be short to short
1742
     * transitions. This leaves just two cases (long to long and short to short)
1743
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1744
     */
1745
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1746
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1747
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, bias, 512);
1748
    } else {
1749
        for (i = 0; i < 448; i++)
1750
            out[i] = saved[i] + bias;
1751

    
1752
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1753
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, bias, 64);
1754
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      bias, 64);
1755
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      bias, 64);
1756
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      bias, 64);
1757
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      bias, 64);
1758
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1759
        } else {
1760
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, bias, 64);
1761
            for (i = 576; i < 1024; i++)
1762
                out[i] = buf[i-512] + bias;
1763
        }
1764
    }
1765

    
1766
    // buffer update
1767
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1768
        for (i = 0; i < 64; i++)
1769
            saved[i] = temp[64 + i] - bias;
1770
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1771
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1772
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1773
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1774
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1775
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1776
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1777
    } else { // LONG_STOP or ONLY_LONG
1778
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1779
    }
1780
}
1781

    
1782
/**
1783
 * Apply dependent channel coupling (applied before IMDCT).
1784
 *
1785
 * @param   index   index into coupling gain array
1786
 */
1787
static void apply_dependent_coupling(AACContext *ac,
1788
                                     SingleChannelElement *target,
1789
                                     ChannelElement *cce, int index)
1790
{
1791
    IndividualChannelStream *ics = &cce->ch[0].ics;
1792
    const uint16_t *offsets = ics->swb_offset;
1793
    float *dest = target->coeffs;
1794
    const float *src = cce->ch[0].coeffs;
1795
    int g, i, group, k, idx = 0;
1796
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1797
        av_log(ac->avccontext, AV_LOG_ERROR,
1798
               "Dependent coupling is not supported together with LTP\n");
1799
        return;
1800
    }
1801
    for (g = 0; g < ics->num_window_groups; g++) {
1802
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1803
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1804
                const float gain = cce->coup.gain[index][idx];
1805
                for (group = 0; group < ics->group_len[g]; group++) {
1806
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1807
                        // XXX dsputil-ize
1808
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1809
                    }
1810
                }
1811
            }
1812
        }
1813
        dest += ics->group_len[g] * 128;
1814
        src  += ics->group_len[g] * 128;
1815
    }
1816
}
1817

    
1818
/**
1819
 * Apply independent channel coupling (applied after IMDCT).
1820
 *
1821
 * @param   index   index into coupling gain array
1822
 */
1823
static void apply_independent_coupling(AACContext *ac,
1824
                                       SingleChannelElement *target,
1825
                                       ChannelElement *cce, int index)
1826
{
1827
    int i;
1828
    const float gain = cce->coup.gain[index][0];
1829
    const float bias = ac->add_bias;
1830
    const float *src = cce->ch[0].ret;
1831
    float *dest = target->ret;
1832
    const int len = 1024 << (ac->m4ac.sbr == 1);
1833

    
1834
    for (i = 0; i < len; i++)
1835
        dest[i] += gain * (src[i] - bias);
1836
}
1837

    
1838
/**
1839
 * channel coupling transformation interface
1840
 *
1841
 * @param   index   index into coupling gain array
1842
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1843
 */
1844
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1845
                                   enum RawDataBlockType type, int elem_id,
1846
                                   enum CouplingPoint coupling_point,
1847
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1848
{
1849
    int i, c;
1850

    
1851
    for (i = 0; i < MAX_ELEM_ID; i++) {
1852
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1853
        int index = 0;
1854

    
1855
        if (cce && cce->coup.coupling_point == coupling_point) {
1856
            ChannelCoupling *coup = &cce->coup;
1857

    
1858
            for (c = 0; c <= coup->num_coupled; c++) {
1859
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1860
                    if (coup->ch_select[c] != 1) {
1861
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1862
                        if (coup->ch_select[c] != 0)
1863
                            index++;
1864
                    }
1865
                    if (coup->ch_select[c] != 2)
1866
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1867
                } else
1868
                    index += 1 + (coup->ch_select[c] == 3);
1869
            }
1870
        }
1871
    }
1872
}
1873

    
1874
/**
1875
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1876
 */
1877
static void spectral_to_sample(AACContext *ac)
1878
{
1879
    int i, type;
1880
    float imdct_bias = (ac->m4ac.sbr <= 0) ? ac->add_bias : 0.0f;
1881
    for (type = 3; type >= 0; type--) {
1882
        for (i = 0; i < MAX_ELEM_ID; i++) {
1883
            ChannelElement *che = ac->che[type][i];
1884
            if (che) {
1885
                if (type <= TYPE_CPE)
1886
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1887
                if (che->ch[0].tns.present)
1888
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1889
                if (che->ch[1].tns.present)
1890
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1891
                if (type <= TYPE_CPE)
1892
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1893
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
1894
                    imdct_and_windowing(ac, &che->ch[0], imdct_bias);
1895
                    if (ac->m4ac.sbr > 0) {
1896
                        ff_sbr_dequant(ac, &che->sbr, type == TYPE_CPE ? TYPE_CPE : TYPE_SCE);
1897
                        ff_sbr_apply(ac, &che->sbr, 0, che->ch[0].ret, che->ch[0].ret);
1898
                    }
1899
                }
1900
                if (type == TYPE_CPE) {
1901
                    imdct_and_windowing(ac, &che->ch[1], imdct_bias);
1902
                    if (ac->m4ac.sbr > 0)
1903
                        ff_sbr_apply(ac, &che->sbr, 1, che->ch[1].ret, che->ch[1].ret);
1904
                }
1905
                if (type <= TYPE_CCE)
1906
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1907
            }
1908
        }
1909
    }
1910
}
1911

    
1912
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
1913
{
1914
    int size;
1915
    AACADTSHeaderInfo hdr_info;
1916

    
1917
    size = ff_aac_parse_header(gb, &hdr_info);
1918
    if (size > 0) {
1919
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
1920
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1921
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1922
            ac->m4ac.chan_config = hdr_info.chan_config;
1923
            if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
1924
                return -7;
1925
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
1926
                return -7;
1927
        } else if (ac->output_configured != OC_LOCKED) {
1928
            ac->output_configured = OC_NONE;
1929
        }
1930
        if (ac->output_configured != OC_LOCKED)
1931
            ac->m4ac.sbr = -1;
1932
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1933
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1934
        ac->m4ac.object_type     = hdr_info.object_type;
1935
        if (!ac->avccontext->sample_rate)
1936
            ac->avccontext->sample_rate = hdr_info.sample_rate;
1937
        if (hdr_info.num_aac_frames == 1) {
1938
            if (!hdr_info.crc_absent)
1939
                skip_bits(gb, 16);
1940
        } else {
1941
            av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1942
            return -1;
1943
        }
1944
    }
1945
    return size;
1946
}
1947

    
1948
static int aac_decode_frame(AVCodecContext *avccontext, void *data,
1949
                            int *data_size, AVPacket *avpkt)
1950
{
1951
    const uint8_t *buf = avpkt->data;
1952
    int buf_size = avpkt->size;
1953
    AACContext *ac = avccontext->priv_data;
1954
    ChannelElement *che = NULL, *che_prev = NULL;
1955
    GetBitContext gb;
1956
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
1957
    int err, elem_id, data_size_tmp;
1958
    int buf_consumed;
1959
    int samples = 1024, multiplier;
1960

    
1961
    init_get_bits(&gb, buf, buf_size * 8);
1962

    
1963
    if (show_bits(&gb, 12) == 0xfff) {
1964
        if (parse_adts_frame_header(ac, &gb) < 0) {
1965
            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1966
            return -1;
1967
        }
1968
        if (ac->m4ac.sampling_index > 12) {
1969
            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1970
            return -1;
1971
        }
1972
    }
1973

    
1974
    // parse
1975
    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1976
        elem_id = get_bits(&gb, 4);
1977

    
1978
        if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
1979
            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1980
            return -1;
1981
        }
1982

    
1983
        switch (elem_type) {
1984

    
1985
        case TYPE_SCE:
1986
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1987
            break;
1988

    
1989
        case TYPE_CPE:
1990
            err = decode_cpe(ac, &gb, che);
1991
            break;
1992

    
1993
        case TYPE_CCE:
1994
            err = decode_cce(ac, &gb, che);
1995
            break;
1996

    
1997
        case TYPE_LFE:
1998
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1999
            break;
2000

    
2001
        case TYPE_DSE:
2002
            err = skip_data_stream_element(ac, &gb);
2003
            break;
2004

    
2005
        case TYPE_PCE: {
2006
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2007
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2008
            if ((err = decode_pce(ac, new_che_pos, &gb)))
2009
                break;
2010
            if (ac->output_configured > OC_TRIAL_PCE)
2011
                av_log(avccontext, AV_LOG_ERROR,
2012
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2013
            else
2014
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
2015
            break;
2016
        }
2017

    
2018
        case TYPE_FIL:
2019
            if (elem_id == 15)
2020
                elem_id += get_bits(&gb, 8) - 1;
2021
            if (get_bits_left(&gb) < 8 * elem_id) {
2022
                    av_log(avccontext, AV_LOG_ERROR, overread_err);
2023
                    return -1;
2024
            }
2025
            while (elem_id > 0)
2026
                elem_id -= decode_extension_payload(ac, &gb, elem_id, che_prev, elem_type_prev);
2027
            err = 0; /* FIXME */
2028
            break;
2029

    
2030
        default:
2031
            err = -1; /* should not happen, but keeps compiler happy */
2032
            break;
2033
        }
2034

    
2035
        che_prev       = che;
2036
        elem_type_prev = elem_type;
2037

    
2038
        if (err)
2039
            return err;
2040

    
2041
        if (get_bits_left(&gb) < 3) {
2042
            av_log(avccontext, AV_LOG_ERROR, overread_err);
2043
            return -1;
2044
        }
2045
    }
2046

    
2047
    spectral_to_sample(ac);
2048

    
2049
    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
2050
    samples <<= multiplier;
2051
    if (ac->output_configured < OC_LOCKED) {
2052
        avccontext->sample_rate = ac->m4ac.sample_rate << multiplier;
2053
        avccontext->frame_size = samples;
2054
    }
2055

    
2056
    data_size_tmp = samples * avccontext->channels * sizeof(int16_t);
2057
    if (*data_size < data_size_tmp) {
2058
        av_log(avccontext, AV_LOG_ERROR,
2059
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2060
               *data_size, data_size_tmp);
2061
        return -1;
2062
    }
2063
    *data_size = data_size_tmp;
2064

    
2065
    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avccontext->channels);
2066

    
2067
    if (ac->output_configured)
2068
        ac->output_configured = OC_LOCKED;
2069

    
2070
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2071
    return buf_size > buf_consumed ? buf_consumed : buf_size;
2072
}
2073

    
2074
static av_cold int aac_decode_close(AVCodecContext *avccontext)
2075
{
2076
    AACContext *ac = avccontext->priv_data;
2077
    int i, type;
2078

    
2079
    for (i = 0; i < MAX_ELEM_ID; i++) {
2080
        for (type = 0; type < 4; type++) {
2081
            if (ac->che[type][i])
2082
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
2083
            av_freep(&ac->che[type][i]);
2084
        }
2085
    }
2086

    
2087
    ff_mdct_end(&ac->mdct);
2088
    ff_mdct_end(&ac->mdct_small);
2089
    return 0;
2090
}
2091

    
2092
AVCodec aac_decoder = {
2093
    "aac",
2094
    AVMEDIA_TYPE_AUDIO,
2095
    CODEC_ID_AAC,
2096
    sizeof(AACContext),
2097
    aac_decode_init,
2098
    NULL,
2099
    aac_decode_close,
2100
    aac_decode_frame,
2101
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2102
    .sample_fmts = (const enum SampleFormat[]) {
2103
        SAMPLE_FMT_S16,SAMPLE_FMT_NONE
2104
    },
2105
    .channel_layouts = aac_channel_layout,
2106
};