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1
/*
2
 * AAC decoder
3
 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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 *
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 * AAC LATM decoder
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 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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 * Copyright (c) 2010      Janne Grunau <janne-ffmpeg@jannau.net>
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 *
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 * This file is part of Libav.
11
 *
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 * Libav is free software; you can redistribute it and/or
13
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * Libav is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
19
 * 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 Libav; 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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 */
26

    
27
/**
28
 * @file
29
 * AAC decoder
30
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
31
 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
32
 */
33

    
34
/*
35
 * supported tools
36
 *
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 * Support?             Name
38
 * N (code in SoC repo) gain control
39
 * Y                    block switching
40
 * Y                    window shapes - standard
41
 * 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
45
 * Y                    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
52
 * 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
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 * N                    Structured Audio Sample Bank Format
66
 * N                    MIDI
67
 * N                    Harmonic and Individual Lines plus Noise
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 * N                    Text-To-Speech Interface
69
 * Y                    Spectral Band Replication
70
 * Y (not in this code) Layer-1
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 * 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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 * Y                    Parametric Stereo
75
 * N                    Direct Stream Transfer
76
 *
77
 * 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.
80
 */
81

    
82

    
83
#include "avcodec.h"
84
#include "internal.h"
85
#include "get_bits.h"
86
#include "dsputil.h"
87
#include "fft.h"
88
#include "fmtconvert.h"
89
#include "lpc.h"
90
#include "kbdwin.h"
91
#include "sinewin.h"
92

    
93
#include "aac.h"
94
#include "aactab.h"
95
#include "aacdectab.h"
96
#include "cbrt_tablegen.h"
97
#include "sbr.h"
98
#include "aacsbr.h"
99
#include "mpeg4audio.h"
100
#include "aacadtsdec.h"
101

    
102
#include <assert.h>
103
#include <errno.h>
104
#include <math.h>
105
#include <string.h>
106

    
107
#if ARCH_ARM
108
#   include "arm/aac.h"
109
#endif
110

    
111
union float754 {
112
    float f;
113
    uint32_t i;
114
};
115

    
116
static VLC vlc_scalefactors;
117
static VLC vlc_spectral[11];
118

    
119
static const char overread_err[] = "Input buffer exhausted before END element found\n";
120

    
121
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
122
{
123
    // For PCE based channel configurations map the channels solely based on tags.
124
    if (!ac->m4ac.chan_config) {
125
        return ac->tag_che_map[type][elem_id];
126
    }
127
    // For indexed channel configurations map the channels solely based on position.
128
    switch (ac->m4ac.chan_config) {
129
    case 7:
130
        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
131
            ac->tags_mapped++;
132
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
133
        }
134
    case 6:
135
        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
136
           instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
137
           encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
138
        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
139
            ac->tags_mapped++;
140
            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
141
        }
142
    case 5:
143
        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
144
            ac->tags_mapped++;
145
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
146
        }
147
    case 4:
148
        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
149
            ac->tags_mapped++;
150
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
151
        }
152
    case 3:
153
    case 2:
154
        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
155
            ac->tags_mapped++;
156
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
157
        } else if (ac->m4ac.chan_config == 2) {
158
            return NULL;
159
        }
160
    case 1:
161
        if (!ac->tags_mapped && type == TYPE_SCE) {
162
            ac->tags_mapped++;
163
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
164
        }
165
    default:
166
        return NULL;
167
    }
168
}
169

    
170
/**
171
 * Check for the channel element in the current channel position configuration.
172
 * If it exists, make sure the appropriate element is allocated and map the
173
 * channel order to match the internal Libav channel layout.
174
 *
175
 * @param   che_pos current channel position configuration
176
 * @param   type channel element type
177
 * @param   id channel element id
178
 * @param   channels count of the number of channels in the configuration
179
 *
180
 * @return  Returns error status. 0 - OK, !0 - error
181
 */
182
static av_cold int che_configure(AACContext *ac,
183
                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
184
                         int type, int id,
185
                         int *channels)
186
{
187
    if (che_pos[type][id]) {
188
        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
189
            return AVERROR(ENOMEM);
190
        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
191
        if (type != TYPE_CCE) {
192
            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
193
            if (type == TYPE_CPE ||
194
                (type == TYPE_SCE && ac->m4ac.ps == 1)) {
195
                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
196
            }
197
        }
198
    } else {
199
        if (ac->che[type][id])
200
            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
201
        av_freep(&ac->che[type][id]);
202
    }
203
    return 0;
204
}
205

    
206
/**
207
 * Configure output channel order based on the current program configuration element.
208
 *
209
 * @param   che_pos current channel position configuration
210
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
211
 *
212
 * @return  Returns error status. 0 - OK, !0 - error
213
 */
214
static av_cold int output_configure(AACContext *ac,
215
                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
216
                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
217
                            int channel_config, enum OCStatus oc_type)
218
{
219
    AVCodecContext *avctx = ac->avctx;
220
    int i, type, channels = 0, ret;
221

    
222
    if (new_che_pos != che_pos)
223
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
224

    
225
    if (channel_config) {
226
        for (i = 0; i < tags_per_config[channel_config]; i++) {
227
            if ((ret = che_configure(ac, che_pos,
228
                                     aac_channel_layout_map[channel_config - 1][i][0],
229
                                     aac_channel_layout_map[channel_config - 1][i][1],
230
                                     &channels)))
231
                return ret;
232
        }
233

    
234
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
235

    
236
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
237
    } else {
238
        /* Allocate or free elements depending on if they are in the
239
         * current program configuration.
240
         *
241
         * Set up default 1:1 output mapping.
242
         *
243
         * For a 5.1 stream the output order will be:
244
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
245
         */
246

    
247
        for (i = 0; i < MAX_ELEM_ID; i++) {
248
            for (type = 0; type < 4; type++) {
249
                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
250
                    return ret;
251
            }
252
        }
253

    
254
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
255

    
256
        avctx->channel_layout = 0;
257
    }
258

    
259
    avctx->channels = channels;
260

    
261
    ac->output_configured = oc_type;
262

    
263
    return 0;
264
}
265

    
266
/**
267
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
268
 *
269
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
270
 * @param sce_map mono (Single Channel Element) map
271
 * @param type speaker type/position for these channels
272
 */
273
static void decode_channel_map(enum ChannelPosition *cpe_map,
274
                               enum ChannelPosition *sce_map,
275
                               enum ChannelPosition type,
276
                               GetBitContext *gb, int n)
277
{
278
    while (n--) {
279
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
280
        map[get_bits(gb, 4)] = type;
281
    }
282
}
283

    
284
/**
285
 * Decode program configuration element; reference: table 4.2.
286
 *
287
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
288
 *
289
 * @return  Returns error status. 0 - OK, !0 - error
290
 */
291
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
292
                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
293
                      GetBitContext *gb)
294
{
295
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
296
    int comment_len;
297

    
298
    skip_bits(gb, 2);  // object_type
299

    
300
    sampling_index = get_bits(gb, 4);
301
    if (m4ac->sampling_index != sampling_index)
302
        av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
303

    
304
    num_front       = get_bits(gb, 4);
305
    num_side        = get_bits(gb, 4);
306
    num_back        = get_bits(gb, 4);
307
    num_lfe         = get_bits(gb, 2);
308
    num_assoc_data  = get_bits(gb, 3);
309
    num_cc          = get_bits(gb, 4);
310

    
311
    if (get_bits1(gb))
312
        skip_bits(gb, 4); // mono_mixdown_tag
313
    if (get_bits1(gb))
314
        skip_bits(gb, 4); // stereo_mixdown_tag
315

    
316
    if (get_bits1(gb))
317
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
318

    
319
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
320
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
321
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
322
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
323

    
324
    skip_bits_long(gb, 4 * num_assoc_data);
325

    
326
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
327

    
328
    align_get_bits(gb);
329

    
330
    /* comment field, first byte is length */
331
    comment_len = get_bits(gb, 8) * 8;
332
    if (get_bits_left(gb) < comment_len) {
333
        av_log(avctx, AV_LOG_ERROR, overread_err);
334
        return -1;
335
    }
336
    skip_bits_long(gb, comment_len);
337
    return 0;
338
}
339

    
340
/**
341
 * Set up channel positions based on a default channel configuration
342
 * as specified in table 1.17.
343
 *
344
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
345
 *
346
 * @return  Returns error status. 0 - OK, !0 - error
347
 */
348
static av_cold int set_default_channel_config(AVCodecContext *avctx,
349
                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
350
                                      int channel_config)
351
{
352
    if (channel_config < 1 || channel_config > 7) {
353
        av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
354
               channel_config);
355
        return -1;
356
    }
357

    
358
    /* default channel configurations:
359
     *
360
     * 1ch : front center (mono)
361
     * 2ch : L + R (stereo)
362
     * 3ch : front center + L + R
363
     * 4ch : front center + L + R + back center
364
     * 5ch : front center + L + R + back stereo
365
     * 6ch : front center + L + R + back stereo + LFE
366
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
367
     */
368

    
369
    if (channel_config != 2)
370
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
371
    if (channel_config > 1)
372
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
373
    if (channel_config == 4)
374
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
375
    if (channel_config > 4)
376
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
377
        = AAC_CHANNEL_BACK;  // back stereo
378
    if (channel_config > 5)
379
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
380
    if (channel_config == 7)
381
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
382

    
383
    return 0;
384
}
385

    
386
/**
387
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
388
 *
389
 * @param   ac          pointer to AACContext, may be null
390
 * @param   avctx       pointer to AVCCodecContext, used for logging
391
 *
392
 * @return  Returns error status. 0 - OK, !0 - error
393
 */
394
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
395
                                     GetBitContext *gb,
396
                                     MPEG4AudioConfig *m4ac,
397
                                     int channel_config)
398
{
399
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
400
    int extension_flag, ret;
401

    
402
    if (get_bits1(gb)) { // frameLengthFlag
403
        av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
404
        return -1;
405
    }
406

    
407
    if (get_bits1(gb))       // dependsOnCoreCoder
408
        skip_bits(gb, 14);   // coreCoderDelay
409
    extension_flag = get_bits1(gb);
410

    
411
    if (m4ac->object_type == AOT_AAC_SCALABLE ||
412
        m4ac->object_type == AOT_ER_AAC_SCALABLE)
413
        skip_bits(gb, 3);     // layerNr
414

    
415
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
416
    if (channel_config == 0) {
417
        skip_bits(gb, 4);  // element_instance_tag
418
        if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
419
            return ret;
420
    } else {
421
        if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
422
            return ret;
423
    }
424
    if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
425
        return ret;
426

    
427
    if (extension_flag) {
428
        switch (m4ac->object_type) {
429
        case AOT_ER_BSAC:
430
            skip_bits(gb, 5);    // numOfSubFrame
431
            skip_bits(gb, 11);   // layer_length
432
            break;
433
        case AOT_ER_AAC_LC:
434
        case AOT_ER_AAC_LTP:
435
        case AOT_ER_AAC_SCALABLE:
436
        case AOT_ER_AAC_LD:
437
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
438
                                    * aacScalefactorDataResilienceFlag
439
                                    * aacSpectralDataResilienceFlag
440
                                    */
441
            break;
442
        }
443
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
444
    }
445
    return 0;
446
}
447

    
448
/**
449
 * Decode audio specific configuration; reference: table 1.13.
450
 *
451
 * @param   ac          pointer to AACContext, may be null
452
 * @param   avctx       pointer to AVCCodecContext, used for logging
453
 * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
454
 * @param   data        pointer to AVCodecContext extradata
455
 * @param   data_size   size of AVCCodecContext extradata
456
 *
457
 * @return  Returns error status or number of consumed bits. <0 - error
458
 */
459
static int decode_audio_specific_config(AACContext *ac,
460
                                        AVCodecContext *avctx,
461
                                        MPEG4AudioConfig *m4ac,
462
                                        const uint8_t *data, int data_size)
463
{
464
    GetBitContext gb;
465
    int i;
466

    
467
    init_get_bits(&gb, data, data_size * 8);
468

    
469
    if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
470
        return -1;
471
    if (m4ac->sampling_index > 12) {
472
        av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
473
        return -1;
474
    }
475
    if (m4ac->sbr == 1 && m4ac->ps == -1)
476
        m4ac->ps = 1;
477

    
478
    skip_bits_long(&gb, i);
479

    
480
    switch (m4ac->object_type) {
481
    case AOT_AAC_MAIN:
482
    case AOT_AAC_LC:
483
    case AOT_AAC_LTP:
484
        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
485
            return -1;
486
        break;
487
    default:
488
        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
489
               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
490
        return -1;
491
    }
492

    
493
    return get_bits_count(&gb);
494
}
495

    
496
/**
497
 * linear congruential pseudorandom number generator
498
 *
499
 * @param   previous_val    pointer to the current state of the generator
500
 *
501
 * @return  Returns a 32-bit pseudorandom integer
502
 */
503
static av_always_inline int lcg_random(int previous_val)
504
{
505
    return previous_val * 1664525 + 1013904223;
506
}
507

    
508
static av_always_inline void reset_predict_state(PredictorState *ps)
509
{
510
    ps->r0   = 0.0f;
511
    ps->r1   = 0.0f;
512
    ps->cor0 = 0.0f;
513
    ps->cor1 = 0.0f;
514
    ps->var0 = 1.0f;
515
    ps->var1 = 1.0f;
516
}
517

    
518
static void reset_all_predictors(PredictorState *ps)
519
{
520
    int i;
521
    for (i = 0; i < MAX_PREDICTORS; i++)
522
        reset_predict_state(&ps[i]);
523
}
524

    
525
static void reset_predictor_group(PredictorState *ps, int group_num)
526
{
527
    int i;
528
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
529
        reset_predict_state(&ps[i]);
530
}
531

    
532
#define AAC_INIT_VLC_STATIC(num, size) \
533
    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
534
         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
535
        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
536
        size);
537

    
538
static av_cold int aac_decode_init(AVCodecContext *avctx)
539
{
540
    AACContext *ac = avctx->priv_data;
541

    
542
    ac->avctx = avctx;
543
    ac->m4ac.sample_rate = avctx->sample_rate;
544

    
545
    if (avctx->extradata_size > 0) {
546
        if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
547
                                         avctx->extradata,
548
                                         avctx->extradata_size) < 0)
549
            return -1;
550
    }
551

    
552
    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
553

    
554
    AAC_INIT_VLC_STATIC( 0, 304);
555
    AAC_INIT_VLC_STATIC( 1, 270);
556
    AAC_INIT_VLC_STATIC( 2, 550);
557
    AAC_INIT_VLC_STATIC( 3, 300);
558
    AAC_INIT_VLC_STATIC( 4, 328);
559
    AAC_INIT_VLC_STATIC( 5, 294);
560
    AAC_INIT_VLC_STATIC( 6, 306);
561
    AAC_INIT_VLC_STATIC( 7, 268);
562
    AAC_INIT_VLC_STATIC( 8, 510);
563
    AAC_INIT_VLC_STATIC( 9, 366);
564
    AAC_INIT_VLC_STATIC(10, 462);
565

    
566
    ff_aac_sbr_init();
567

    
568
    dsputil_init(&ac->dsp, avctx);
569
    ff_fmt_convert_init(&ac->fmt_conv, avctx);
570

    
571
    ac->random_state = 0x1f2e3d4c;
572

    
573
    // -1024 - Compensate wrong IMDCT method.
574
    // 60    - Required to scale values to the correct range [-32768,32767]
575
    //         for float to int16 conversion. (1 << (60 / 4)) == 32768
576
    ac->sf_scale  = 1. / -1024.;
577
    ac->sf_offset = 60;
578

    
579
    ff_aac_tableinit();
580

    
581
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
582
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
583
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
584
                    352);
585

    
586
    ff_mdct_init(&ac->mdct,       11, 1, 1.0);
587
    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0);
588
    ff_mdct_init(&ac->mdct_ltp,   11, 0, 1.0);
589
    // window initialization
590
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
591
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
592
    ff_init_ff_sine_windows(10);
593
    ff_init_ff_sine_windows( 7);
594

    
595
    cbrt_tableinit();
596

    
597
    return 0;
598
}
599

    
600
/**
601
 * Skip data_stream_element; reference: table 4.10.
602
 */
603
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
604
{
605
    int byte_align = get_bits1(gb);
606
    int count = get_bits(gb, 8);
607
    if (count == 255)
608
        count += get_bits(gb, 8);
609
    if (byte_align)
610
        align_get_bits(gb);
611

    
612
    if (get_bits_left(gb) < 8 * count) {
613
        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
614
        return -1;
615
    }
616
    skip_bits_long(gb, 8 * count);
617
    return 0;
618
}
619

    
620
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
621
                             GetBitContext *gb)
622
{
623
    int sfb;
624
    if (get_bits1(gb)) {
625
        ics->predictor_reset_group = get_bits(gb, 5);
626
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
627
            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
628
            return -1;
629
        }
630
    }
631
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
632
        ics->prediction_used[sfb] = get_bits1(gb);
633
    }
634
    return 0;
635
}
636

    
637
/**
638
 * Decode Long Term Prediction data; reference: table 4.xx.
639
 */
640
static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
641
                       GetBitContext *gb, uint8_t max_sfb)
642
{
643
    int sfb;
644

    
645
    ltp->lag  = get_bits(gb, 11);
646
    ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
647
    for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
648
        ltp->used[sfb] = get_bits1(gb);
649
}
650

    
651
/**
652
 * Decode Individual Channel Stream info; reference: table 4.6.
653
 *
654
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
655
 */
656
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
657
                           GetBitContext *gb, int common_window)
658
{
659
    if (get_bits1(gb)) {
660
        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
661
        memset(ics, 0, sizeof(IndividualChannelStream));
662
        return -1;
663
    }
664
    ics->window_sequence[1] = ics->window_sequence[0];
665
    ics->window_sequence[0] = get_bits(gb, 2);
666
    ics->use_kb_window[1]   = ics->use_kb_window[0];
667
    ics->use_kb_window[0]   = get_bits1(gb);
668
    ics->num_window_groups  = 1;
669
    ics->group_len[0]       = 1;
670
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
671
        int i;
672
        ics->max_sfb = get_bits(gb, 4);
673
        for (i = 0; i < 7; i++) {
674
            if (get_bits1(gb)) {
675
                ics->group_len[ics->num_window_groups - 1]++;
676
            } else {
677
                ics->num_window_groups++;
678
                ics->group_len[ics->num_window_groups - 1] = 1;
679
            }
680
        }
681
        ics->num_windows       = 8;
682
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
683
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
684
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
685
        ics->predictor_present = 0;
686
    } else {
687
        ics->max_sfb               = get_bits(gb, 6);
688
        ics->num_windows           = 1;
689
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
690
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
691
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
692
        ics->predictor_present     = get_bits1(gb);
693
        ics->predictor_reset_group = 0;
694
        if (ics->predictor_present) {
695
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
696
                if (decode_prediction(ac, ics, gb)) {
697
                    memset(ics, 0, sizeof(IndividualChannelStream));
698
                    return -1;
699
                }
700
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
701
                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
702
                memset(ics, 0, sizeof(IndividualChannelStream));
703
                return -1;
704
            } else {
705
                if ((ics->ltp.present = get_bits(gb, 1)))
706
                    decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
707
            }
708
        }
709
    }
710

    
711
    if (ics->max_sfb > ics->num_swb) {
712
        av_log(ac->avctx, AV_LOG_ERROR,
713
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
714
               ics->max_sfb, ics->num_swb);
715
        memset(ics, 0, sizeof(IndividualChannelStream));
716
        return -1;
717
    }
718

    
719
    return 0;
720
}
721

    
722
/**
723
 * Decode band types (section_data payload); reference: table 4.46.
724
 *
725
 * @param   band_type           array of the used band type
726
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
727
 *
728
 * @return  Returns error status. 0 - OK, !0 - error
729
 */
730
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
731
                             int band_type_run_end[120], GetBitContext *gb,
732
                             IndividualChannelStream *ics)
733
{
734
    int g, idx = 0;
735
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
736
    for (g = 0; g < ics->num_window_groups; g++) {
737
        int k = 0;
738
        while (k < ics->max_sfb) {
739
            uint8_t sect_end = k;
740
            int sect_len_incr;
741
            int sect_band_type = get_bits(gb, 4);
742
            if (sect_band_type == 12) {
743
                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
744
                return -1;
745
            }
746
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
747
                sect_end += sect_len_incr;
748
            sect_end += sect_len_incr;
749
            if (get_bits_left(gb) < 0) {
750
                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
751
                return -1;
752
            }
753
            if (sect_end > ics->max_sfb) {
754
                av_log(ac->avctx, AV_LOG_ERROR,
755
                       "Number of bands (%d) exceeds limit (%d).\n",
756
                       sect_end, ics->max_sfb);
757
                return -1;
758
            }
759
            for (; k < sect_end; k++) {
760
                band_type        [idx]   = sect_band_type;
761
                band_type_run_end[idx++] = sect_end;
762
            }
763
        }
764
    }
765
    return 0;
766
}
767

    
768
/**
769
 * Decode scalefactors; reference: table 4.47.
770
 *
771
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
772
 * @param   band_type           array of the used band type
773
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
774
 * @param   sf                  array of scalefactors or intensity stereo positions
775
 *
776
 * @return  Returns error status. 0 - OK, !0 - error
777
 */
778
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
779
                               unsigned int global_gain,
780
                               IndividualChannelStream *ics,
781
                               enum BandType band_type[120],
782
                               int band_type_run_end[120])
783
{
784
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
785
    int g, i, idx = 0;
786
    int offset[3] = { global_gain, global_gain - 90, 100 };
787
    int noise_flag = 1;
788
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
789
    for (g = 0; g < ics->num_window_groups; g++) {
790
        for (i = 0; i < ics->max_sfb;) {
791
            int run_end = band_type_run_end[idx];
792
            if (band_type[idx] == ZERO_BT) {
793
                for (; i < run_end; i++, idx++)
794
                    sf[idx] = 0.;
795
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
796
                for (; i < run_end; i++, idx++) {
797
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
798
                    if (offset[2] > 255U) {
799
                        av_log(ac->avctx, AV_LOG_ERROR,
800
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
801
                        return -1;
802
                    }
803
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
804
                }
805
            } else if (band_type[idx] == NOISE_BT) {
806
                for (; i < run_end; i++, idx++) {
807
                    if (noise_flag-- > 0)
808
                        offset[1] += get_bits(gb, 9) - 256;
809
                    else
810
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
811
                    if (offset[1] > 255U) {
812
                        av_log(ac->avctx, AV_LOG_ERROR,
813
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
814
                        return -1;
815
                    }
816
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
817
                }
818
            } else {
819
                for (; i < run_end; i++, idx++) {
820
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
821
                    if (offset[0] > 255U) {
822
                        av_log(ac->avctx, AV_LOG_ERROR,
823
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
824
                        return -1;
825
                    }
826
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
827
                }
828
            }
829
        }
830
    }
831
    return 0;
832
}
833

    
834
/**
835
 * Decode pulse data; reference: table 4.7.
836
 */
837
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
838
                         const uint16_t *swb_offset, int num_swb)
839
{
840
    int i, pulse_swb;
841
    pulse->num_pulse = get_bits(gb, 2) + 1;
842
    pulse_swb        = get_bits(gb, 6);
843
    if (pulse_swb >= num_swb)
844
        return -1;
845
    pulse->pos[0]    = swb_offset[pulse_swb];
846
    pulse->pos[0]   += get_bits(gb, 5);
847
    if (pulse->pos[0] > 1023)
848
        return -1;
849
    pulse->amp[0]    = get_bits(gb, 4);
850
    for (i = 1; i < pulse->num_pulse; i++) {
851
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
852
        if (pulse->pos[i] > 1023)
853
            return -1;
854
        pulse->amp[i] = get_bits(gb, 4);
855
    }
856
    return 0;
857
}
858

    
859
/**
860
 * Decode Temporal Noise Shaping data; reference: table 4.48.
861
 *
862
 * @return  Returns error status. 0 - OK, !0 - error
863
 */
864
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
865
                      GetBitContext *gb, const IndividualChannelStream *ics)
866
{
867
    int w, filt, i, coef_len, coef_res, coef_compress;
868
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
869
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
870
    for (w = 0; w < ics->num_windows; w++) {
871
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
872
            coef_res = get_bits1(gb);
873

    
874
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
875
                int tmp2_idx;
876
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
877

    
878
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
879
                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
880
                           tns->order[w][filt], tns_max_order);
881
                    tns->order[w][filt] = 0;
882
                    return -1;
883
                }
884
                if (tns->order[w][filt]) {
885
                    tns->direction[w][filt] = get_bits1(gb);
886
                    coef_compress = get_bits1(gb);
887
                    coef_len = coef_res + 3 - coef_compress;
888
                    tmp2_idx = 2 * coef_compress + coef_res;
889

    
890
                    for (i = 0; i < tns->order[w][filt]; i++)
891
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
892
                }
893
            }
894
        }
895
    }
896
    return 0;
897
}
898

    
899
/**
900
 * Decode Mid/Side data; reference: table 4.54.
901
 *
902
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
903
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
904
 *                      [3] reserved for scalable AAC
905
 */
906
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
907
                                   int ms_present)
908
{
909
    int idx;
910
    if (ms_present == 1) {
911
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
912
            cpe->ms_mask[idx] = get_bits1(gb);
913
    } else if (ms_present == 2) {
914
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
915
    }
916
}
917

    
918
#ifndef VMUL2
919
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
920
                           const float *scale)
921
{
922
    float s = *scale;
923
    *dst++ = v[idx    & 15] * s;
924
    *dst++ = v[idx>>4 & 15] * s;
925
    return dst;
926
}
927
#endif
928

    
929
#ifndef VMUL4
930
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
931
                           const float *scale)
932
{
933
    float s = *scale;
934
    *dst++ = v[idx    & 3] * s;
935
    *dst++ = v[idx>>2 & 3] * s;
936
    *dst++ = v[idx>>4 & 3] * s;
937
    *dst++ = v[idx>>6 & 3] * s;
938
    return dst;
939
}
940
#endif
941

    
942
#ifndef VMUL2S
943
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
944
                            unsigned sign, const float *scale)
945
{
946
    union float754 s0, s1;
947

    
948
    s0.f = s1.f = *scale;
949
    s0.i ^= sign >> 1 << 31;
950
    s1.i ^= sign      << 31;
951

    
952
    *dst++ = v[idx    & 15] * s0.f;
953
    *dst++ = v[idx>>4 & 15] * s1.f;
954

    
955
    return dst;
956
}
957
#endif
958

    
959
#ifndef VMUL4S
960
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
961
                            unsigned sign, const float *scale)
962
{
963
    unsigned nz = idx >> 12;
964
    union float754 s = { .f = *scale };
965
    union float754 t;
966

    
967
    t.i = s.i ^ (sign & 1U<<31);
968
    *dst++ = v[idx    & 3] * t.f;
969

    
970
    sign <<= nz & 1; nz >>= 1;
971
    t.i = s.i ^ (sign & 1U<<31);
972
    *dst++ = v[idx>>2 & 3] * t.f;
973

    
974
    sign <<= nz & 1; nz >>= 1;
975
    t.i = s.i ^ (sign & 1U<<31);
976
    *dst++ = v[idx>>4 & 3] * t.f;
977

    
978
    sign <<= nz & 1; nz >>= 1;
979
    t.i = s.i ^ (sign & 1U<<31);
980
    *dst++ = v[idx>>6 & 3] * t.f;
981

    
982
    return dst;
983
}
984
#endif
985

    
986
/**
987
 * Decode spectral data; reference: table 4.50.
988
 * Dequantize and scale spectral data; reference: 4.6.3.3.
989
 *
990
 * @param   coef            array of dequantized, scaled spectral data
991
 * @param   sf              array of scalefactors or intensity stereo positions
992
 * @param   pulse_present   set if pulses are present
993
 * @param   pulse           pointer to pulse data struct
994
 * @param   band_type       array of the used band type
995
 *
996
 * @return  Returns error status. 0 - OK, !0 - error
997
 */
998
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
999
                                       GetBitContext *gb, const float sf[120],
1000
                                       int pulse_present, const Pulse *pulse,
1001
                                       const IndividualChannelStream *ics,
1002
                                       enum BandType band_type[120])
1003
{
1004
    int i, k, g, idx = 0;
1005
    const int c = 1024 / ics->num_windows;
1006
    const uint16_t *offsets = ics->swb_offset;
1007
    float *coef_base = coef;
1008

    
1009
    for (g = 0; g < ics->num_windows; g++)
1010
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
1011

    
1012
    for (g = 0; g < ics->num_window_groups; g++) {
1013
        unsigned g_len = ics->group_len[g];
1014

    
1015
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1016
            const unsigned cbt_m1 = band_type[idx] - 1;
1017
            float *cfo = coef + offsets[i];
1018
            int off_len = offsets[i + 1] - offsets[i];
1019
            int group;
1020

    
1021
            if (cbt_m1 >= INTENSITY_BT2 - 1) {
1022
                for (group = 0; group < g_len; group++, cfo+=128) {
1023
                    memset(cfo, 0, off_len * sizeof(float));
1024
                }
1025
            } else if (cbt_m1 == NOISE_BT - 1) {
1026
                for (group = 0; group < g_len; group++, cfo+=128) {
1027
                    float scale;
1028
                    float band_energy;
1029

    
1030
                    for (k = 0; k < off_len; k++) {
1031
                        ac->random_state  = lcg_random(ac->random_state);
1032
                        cfo[k] = ac->random_state;
1033
                    }
1034

    
1035
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1036
                    scale = sf[idx] / sqrtf(band_energy);
1037
                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1038
                }
1039
            } else {
1040
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1041
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1042
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1043
                OPEN_READER(re, gb);
1044

    
1045
                switch (cbt_m1 >> 1) {
1046
                case 0:
1047
                    for (group = 0; group < g_len; group++, cfo+=128) {
1048
                        float *cf = cfo;
1049
                        int len = off_len;
1050

    
1051
                        do {
1052
                            int code;
1053
                            unsigned cb_idx;
1054

    
1055
                            UPDATE_CACHE(re, gb);
1056
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1057
                            cb_idx = cb_vector_idx[code];
1058
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1059
                        } while (len -= 4);
1060
                    }
1061
                    break;
1062

    
1063
                case 1:
1064
                    for (group = 0; group < g_len; group++, cfo+=128) {
1065
                        float *cf = cfo;
1066
                        int len = off_len;
1067

    
1068
                        do {
1069
                            int code;
1070
                            unsigned nnz;
1071
                            unsigned cb_idx;
1072
                            uint32_t bits;
1073

    
1074
                            UPDATE_CACHE(re, gb);
1075
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1076
                            cb_idx = cb_vector_idx[code];
1077
                            nnz = cb_idx >> 8 & 15;
1078
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1079
                            LAST_SKIP_BITS(re, gb, nnz);
1080
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1081
                        } while (len -= 4);
1082
                    }
1083
                    break;
1084

    
1085
                case 2:
1086
                    for (group = 0; group < g_len; group++, cfo+=128) {
1087
                        float *cf = cfo;
1088
                        int len = off_len;
1089

    
1090
                        do {
1091
                            int code;
1092
                            unsigned cb_idx;
1093

    
1094
                            UPDATE_CACHE(re, gb);
1095
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1096
                            cb_idx = cb_vector_idx[code];
1097
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1098
                        } while (len -= 2);
1099
                    }
1100
                    break;
1101

    
1102
                case 3:
1103
                case 4:
1104
                    for (group = 0; group < g_len; group++, cfo+=128) {
1105
                        float *cf = cfo;
1106
                        int len = off_len;
1107

    
1108
                        do {
1109
                            int code;
1110
                            unsigned nnz;
1111
                            unsigned cb_idx;
1112
                            unsigned sign;
1113

    
1114
                            UPDATE_CACHE(re, gb);
1115
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1116
                            cb_idx = cb_vector_idx[code];
1117
                            nnz = cb_idx >> 8 & 15;
1118
                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1119
                            LAST_SKIP_BITS(re, gb, nnz);
1120
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1121
                        } while (len -= 2);
1122
                    }
1123
                    break;
1124

    
1125
                default:
1126
                    for (group = 0; group < g_len; group++, cfo+=128) {
1127
                        float *cf = cfo;
1128
                        uint32_t *icf = (uint32_t *) cf;
1129
                        int len = off_len;
1130

    
1131
                        do {
1132
                            int code;
1133
                            unsigned nzt, nnz;
1134
                            unsigned cb_idx;
1135
                            uint32_t bits;
1136
                            int j;
1137

    
1138
                            UPDATE_CACHE(re, gb);
1139
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1140

    
1141
                            if (!code) {
1142
                                *icf++ = 0;
1143
                                *icf++ = 0;
1144
                                continue;
1145
                            }
1146

    
1147
                            cb_idx = cb_vector_idx[code];
1148
                            nnz = cb_idx >> 12;
1149
                            nzt = cb_idx >> 8;
1150
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1151
                            LAST_SKIP_BITS(re, gb, nnz);
1152

    
1153
                            for (j = 0; j < 2; j++) {
1154
                                if (nzt & 1<<j) {
1155
                                    uint32_t b;
1156
                                    int n;
1157
                                    /* The total length of escape_sequence must be < 22 bits according
1158
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1159
                                    UPDATE_CACHE(re, gb);
1160
                                    b = GET_CACHE(re, gb);
1161
                                    b = 31 - av_log2(~b);
1162

    
1163
                                    if (b > 8) {
1164
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1165
                                        return -1;
1166
                                    }
1167

    
1168
                                    SKIP_BITS(re, gb, b + 1);
1169
                                    b += 4;
1170
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
1171
                                    LAST_SKIP_BITS(re, gb, b);
1172
                                    *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1173
                                    bits <<= 1;
1174
                                } else {
1175
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1176
                                    *icf++ = (bits & 1U<<31) | v;
1177
                                    bits <<= !!v;
1178
                                }
1179
                                cb_idx >>= 4;
1180
                            }
1181
                        } while (len -= 2);
1182

    
1183
                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1184
                    }
1185
                }
1186

    
1187
                CLOSE_READER(re, gb);
1188
            }
1189
        }
1190
        coef += g_len << 7;
1191
    }
1192

    
1193
    if (pulse_present) {
1194
        idx = 0;
1195
        for (i = 0; i < pulse->num_pulse; i++) {
1196
            float co = coef_base[ pulse->pos[i] ];
1197
            while (offsets[idx + 1] <= pulse->pos[i])
1198
                idx++;
1199
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1200
                float ico = -pulse->amp[i];
1201
                if (co) {
1202
                    co /= sf[idx];
1203
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1204
                }
1205
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1206
            }
1207
        }
1208
    }
1209
    return 0;
1210
}
1211

    
1212
static av_always_inline float flt16_round(float pf)
1213
{
1214
    union float754 tmp;
1215
    tmp.f = pf;
1216
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1217
    return tmp.f;
1218
}
1219

    
1220
static av_always_inline float flt16_even(float pf)
1221
{
1222
    union float754 tmp;
1223
    tmp.f = pf;
1224
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1225
    return tmp.f;
1226
}
1227

    
1228
static av_always_inline float flt16_trunc(float pf)
1229
{
1230
    union float754 pun;
1231
    pun.f = pf;
1232
    pun.i &= 0xFFFF0000U;
1233
    return pun.f;
1234
}
1235

    
1236
static av_always_inline void predict(PredictorState *ps, float *coef,
1237
                                     float sf_scale, float inv_sf_scale,
1238
                    int output_enable)
1239
{
1240
    const float a     = 0.953125; // 61.0 / 64
1241
    const float alpha = 0.90625;  // 29.0 / 32
1242
    float e0, e1;
1243
    float pv;
1244
    float k1, k2;
1245
    float   r0 = ps->r0,     r1 = ps->r1;
1246
    float cor0 = ps->cor0, cor1 = ps->cor1;
1247
    float var0 = ps->var0, var1 = ps->var1;
1248

    
1249
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
1250
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1251

    
1252
    pv = flt16_round(k1 * r0 + k2 * r1);
1253
    if (output_enable)
1254
        *coef += pv * sf_scale;
1255

    
1256
    e0 = *coef * inv_sf_scale;
1257
    e1 = e0 - k1 * r0;
1258

    
1259
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
1260
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
1261
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
1262
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1263

    
1264
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1265
    ps->r0 = flt16_trunc(a * e0);
1266
}
1267

    
1268
/**
1269
 * Apply AAC-Main style frequency domain prediction.
1270
 */
1271
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1272
{
1273
    int sfb, k;
1274
    float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
1275

    
1276
    if (!sce->ics.predictor_initialized) {
1277
        reset_all_predictors(sce->predictor_state);
1278
        sce->ics.predictor_initialized = 1;
1279
    }
1280

    
1281
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1282
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1283
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1284
                predict(&sce->predictor_state[k], &sce->coeffs[k],
1285
                        sf_scale, inv_sf_scale,
1286
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1287
            }
1288
        }
1289
        if (sce->ics.predictor_reset_group)
1290
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1291
    } else
1292
        reset_all_predictors(sce->predictor_state);
1293
}
1294

    
1295
/**
1296
 * Decode an individual_channel_stream payload; reference: table 4.44.
1297
 *
1298
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1299
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1300
 *
1301
 * @return  Returns error status. 0 - OK, !0 - error
1302
 */
1303
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1304
                      GetBitContext *gb, int common_window, int scale_flag)
1305
{
1306
    Pulse pulse;
1307
    TemporalNoiseShaping    *tns = &sce->tns;
1308
    IndividualChannelStream *ics = &sce->ics;
1309
    float *out = sce->coeffs;
1310
    int global_gain, pulse_present = 0;
1311

    
1312
    /* This assignment is to silence a GCC warning about the variable being used
1313
     * uninitialized when in fact it always is.
1314
     */
1315
    pulse.num_pulse = 0;
1316

    
1317
    global_gain = get_bits(gb, 8);
1318

    
1319
    if (!common_window && !scale_flag) {
1320
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1321
            return -1;
1322
    }
1323

    
1324
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1325
        return -1;
1326
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1327
        return -1;
1328

    
1329
    pulse_present = 0;
1330
    if (!scale_flag) {
1331
        if ((pulse_present = get_bits1(gb))) {
1332
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1333
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1334
                return -1;
1335
            }
1336
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1337
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1338
                return -1;
1339
            }
1340
        }
1341
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1342
            return -1;
1343
        if (get_bits1(gb)) {
1344
            av_log_missing_feature(ac->avctx, "SSR", 1);
1345
            return -1;
1346
        }
1347
    }
1348

    
1349
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1350
        return -1;
1351

    
1352
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1353
        apply_prediction(ac, sce);
1354

    
1355
    return 0;
1356
}
1357

    
1358
/**
1359
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1360
 */
1361
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1362
{
1363
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1364
    float *ch0 = cpe->ch[0].coeffs;
1365
    float *ch1 = cpe->ch[1].coeffs;
1366
    int g, i, group, idx = 0;
1367
    const uint16_t *offsets = ics->swb_offset;
1368
    for (g = 0; g < ics->num_window_groups; g++) {
1369
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1370
            if (cpe->ms_mask[idx] &&
1371
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1372
                for (group = 0; group < ics->group_len[g]; group++) {
1373
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1374
                                              ch1 + group * 128 + offsets[i],
1375
                                              offsets[i+1] - offsets[i]);
1376
                }
1377
            }
1378
        }
1379
        ch0 += ics->group_len[g] * 128;
1380
        ch1 += ics->group_len[g] * 128;
1381
    }
1382
}
1383

    
1384
/**
1385
 * intensity stereo decoding; reference: 4.6.8.2.3
1386
 *
1387
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1388
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1389
 *                      [3] reserved for scalable AAC
1390
 */
1391
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
1392
{
1393
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1394
    SingleChannelElement         *sce1 = &cpe->ch[1];
1395
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1396
    const uint16_t *offsets = ics->swb_offset;
1397
    int g, group, i, idx = 0;
1398
    int c;
1399
    float scale;
1400
    for (g = 0; g < ics->num_window_groups; g++) {
1401
        for (i = 0; i < ics->max_sfb;) {
1402
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1403
                const int bt_run_end = sce1->band_type_run_end[idx];
1404
                for (; i < bt_run_end; i++, idx++) {
1405
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1406
                    if (ms_present)
1407
                        c *= 1 - 2 * cpe->ms_mask[idx];
1408
                    scale = c * sce1->sf[idx];
1409
                    for (group = 0; group < ics->group_len[g]; group++)
1410
                        ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
1411
                                                   coef0 + group * 128 + offsets[i],
1412
                                                   scale,
1413
                                                   offsets[i + 1] - offsets[i]);
1414
                }
1415
            } else {
1416
                int bt_run_end = sce1->band_type_run_end[idx];
1417
                idx += bt_run_end - i;
1418
                i    = bt_run_end;
1419
            }
1420
        }
1421
        coef0 += ics->group_len[g] * 128;
1422
        coef1 += ics->group_len[g] * 128;
1423
    }
1424
}
1425

    
1426
/**
1427
 * Decode a channel_pair_element; reference: table 4.4.
1428
 *
1429
 * @return  Returns error status. 0 - OK, !0 - error
1430
 */
1431
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1432
{
1433
    int i, ret, common_window, ms_present = 0;
1434

    
1435
    common_window = get_bits1(gb);
1436
    if (common_window) {
1437
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1438
            return -1;
1439
        i = cpe->ch[1].ics.use_kb_window[0];
1440
        cpe->ch[1].ics = cpe->ch[0].ics;
1441
        cpe->ch[1].ics.use_kb_window[1] = i;
1442
        if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
1443
            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
1444
                decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
1445
        ms_present = get_bits(gb, 2);
1446
        if (ms_present == 3) {
1447
            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1448
            return -1;
1449
        } else if (ms_present)
1450
            decode_mid_side_stereo(cpe, gb, ms_present);
1451
    }
1452
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1453
        return ret;
1454
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1455
        return ret;
1456

    
1457
    if (common_window) {
1458
        if (ms_present)
1459
            apply_mid_side_stereo(ac, cpe);
1460
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1461
            apply_prediction(ac, &cpe->ch[0]);
1462
            apply_prediction(ac, &cpe->ch[1]);
1463
        }
1464
    }
1465

    
1466
    apply_intensity_stereo(ac, cpe, ms_present);
1467
    return 0;
1468
}
1469

    
1470
static const float cce_scale[] = {
1471
    1.09050773266525765921, //2^(1/8)
1472
    1.18920711500272106672, //2^(1/4)
1473
    M_SQRT2,
1474
    2,
1475
};
1476

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

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

    
1506
    sign  = get_bits(gb, 1);
1507
    scale = cce_scale[get_bits(gb, 2)];
1508

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

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

    
1549
/**
1550
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1551
 *
1552
 * @return  Returns number of bytes consumed.
1553
 */
1554
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1555
                                         GetBitContext *gb)
1556
{
1557
    int i;
1558
    int num_excl_chan = 0;
1559

    
1560
    do {
1561
        for (i = 0; i < 7; i++)
1562
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1563
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1564

    
1565
    return num_excl_chan / 7;
1566
}
1567

    
1568
/**
1569
 * Decode dynamic range information; reference: table 4.52.
1570
 *
1571
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1572
 *
1573
 * @return  Returns number of bytes consumed.
1574
 */
1575
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1576
                                GetBitContext *gb, int cnt)
1577
{
1578
    int n             = 1;
1579
    int drc_num_bands = 1;
1580
    int i;
1581

    
1582
    /* pce_tag_present? */
1583
    if (get_bits1(gb)) {
1584
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1585
        skip_bits(gb, 4); // tag_reserved_bits
1586
        n++;
1587
    }
1588

    
1589
    /* excluded_chns_present? */
1590
    if (get_bits1(gb)) {
1591
        n += decode_drc_channel_exclusions(che_drc, gb);
1592
    }
1593

    
1594
    /* drc_bands_present? */
1595
    if (get_bits1(gb)) {
1596
        che_drc->band_incr            = get_bits(gb, 4);
1597
        che_drc->interpolation_scheme = get_bits(gb, 4);
1598
        n++;
1599
        drc_num_bands += che_drc->band_incr;
1600
        for (i = 0; i < drc_num_bands; i++) {
1601
            che_drc->band_top[i] = get_bits(gb, 8);
1602
            n++;
1603
        }
1604
    }
1605

    
1606
    /* prog_ref_level_present? */
1607
    if (get_bits1(gb)) {
1608
        che_drc->prog_ref_level = get_bits(gb, 7);
1609
        skip_bits1(gb); // prog_ref_level_reserved_bits
1610
        n++;
1611
    }
1612

    
1613
    for (i = 0; i < drc_num_bands; i++) {
1614
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1615
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1616
        n++;
1617
    }
1618

    
1619
    return n;
1620
}
1621

    
1622
/**
1623
 * Decode extension data (incomplete); reference: table 4.51.
1624
 *
1625
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1626
 *
1627
 * @return Returns number of bytes consumed
1628
 */
1629
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
1630
                                    ChannelElement *che, enum RawDataBlockType elem_type)
1631
{
1632
    int crc_flag = 0;
1633
    int res = cnt;
1634
    switch (get_bits(gb, 4)) { // extension type
1635
    case EXT_SBR_DATA_CRC:
1636
        crc_flag++;
1637
    case EXT_SBR_DATA:
1638
        if (!che) {
1639
            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
1640
            return res;
1641
        } else if (!ac->m4ac.sbr) {
1642
            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
1643
            skip_bits_long(gb, 8 * cnt - 4);
1644
            return res;
1645
        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
1646
            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
1647
            skip_bits_long(gb, 8 * cnt - 4);
1648
            return res;
1649
        } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
1650
            ac->m4ac.sbr = 1;
1651
            ac->m4ac.ps = 1;
1652
            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
1653
        } else {
1654
            ac->m4ac.sbr = 1;
1655
        }
1656
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
1657
        break;
1658
    case EXT_DYNAMIC_RANGE:
1659
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1660
        break;
1661
    case EXT_FILL:
1662
    case EXT_FILL_DATA:
1663
    case EXT_DATA_ELEMENT:
1664
    default:
1665
        skip_bits_long(gb, 8 * cnt - 4);
1666
        break;
1667
    };
1668
    return res;
1669
}
1670

    
1671
/**
1672
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1673
 *
1674
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1675
 * @param   coef    spectral coefficients
1676
 */
1677
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1678
                      IndividualChannelStream *ics, int decode)
1679
{
1680
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1681
    int w, filt, m, i;
1682
    int bottom, top, order, start, end, size, inc;
1683
    float lpc[TNS_MAX_ORDER];
1684
    float tmp[TNS_MAX_ORDER];
1685

    
1686
    for (w = 0; w < ics->num_windows; w++) {
1687
        bottom = ics->num_swb;
1688
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1689
            top    = bottom;
1690
            bottom = FFMAX(0, top - tns->length[w][filt]);
1691
            order  = tns->order[w][filt];
1692
            if (order == 0)
1693
                continue;
1694

    
1695
            // tns_decode_coef
1696
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1697

    
1698
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1699
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1700
            if ((size = end - start) <= 0)
1701
                continue;
1702
            if (tns->direction[w][filt]) {
1703
                inc = -1;
1704
                start = end - 1;
1705
            } else {
1706
                inc = 1;
1707
            }
1708
            start += w * 128;
1709

    
1710
            if (decode) {
1711
                // ar filter
1712
                for (m = 0; m < size; m++, start += inc)
1713
                    for (i = 1; i <= FFMIN(m, order); i++)
1714
                        coef[start] -= coef[start - i * inc] * lpc[i - 1];
1715
            } else {
1716
                // ma filter
1717
                for (m = 0; m < size; m++, start += inc) {
1718
                    tmp[0] = coef[start];
1719
                    for (i = 1; i <= FFMIN(m, order); i++)
1720
                        coef[start] += tmp[i] * lpc[i - 1];
1721
                    for (i = order; i > 0; i--)
1722
                        tmp[i] = tmp[i - 1];
1723
                }
1724
            }
1725
        }
1726
    }
1727
}
1728

    
1729
/**
1730
 *  Apply windowing and MDCT to obtain the spectral
1731
 *  coefficient from the predicted sample by LTP.
1732
 */
1733
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
1734
                                   float *in, IndividualChannelStream *ics)
1735
{
1736
    const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1737
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1738
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1739
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1740

    
1741
    if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
1742
        ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
1743
    } else {
1744
        memset(in, 0, 448 * sizeof(float));
1745
        ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
1746
        memcpy(in + 576, in + 576, 448 * sizeof(float));
1747
    }
1748
    if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
1749
        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
1750
    } else {
1751
        memcpy(in + 1024, in + 1024, 448 * sizeof(float));
1752
        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
1753
        memset(in + 1024 + 576, 0, 448 * sizeof(float));
1754
    }
1755
    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
1756
}
1757

    
1758
/**
1759
 * Apply the long term prediction
1760
 */
1761
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
1762
{
1763
    const LongTermPrediction *ltp = &sce->ics.ltp;
1764
    const uint16_t *offsets = sce->ics.swb_offset;
1765
    int i, sfb;
1766

    
1767
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1768
        float *predTime = sce->ret;
1769
        float *predFreq = ac->buf_mdct;
1770
        int16_t num_samples = 2048;
1771

    
1772
        if (ltp->lag < 1024)
1773
            num_samples = ltp->lag + 1024;
1774
        for (i = 0; i < num_samples; i++)
1775
            predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
1776
        memset(&predTime[i], 0, (2048 - i) * sizeof(float));
1777

    
1778
        windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
1779

    
1780
        if (sce->tns.present)
1781
            apply_tns(predFreq, &sce->tns, &sce->ics, 0);
1782

    
1783
        for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
1784
            if (ltp->used[sfb])
1785
                for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
1786
                    sce->coeffs[i] += predFreq[i];
1787
    }
1788
}
1789

    
1790
/**
1791
 * Update the LTP buffer for next frame
1792
 */
1793
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
1794
{
1795
    IndividualChannelStream *ics = &sce->ics;
1796
    float *saved     = sce->saved;
1797
    float *saved_ltp = sce->coeffs;
1798
    const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1799
    const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1800
    int i;
1801

    
1802
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1803
        memcpy(saved_ltp,       saved, 512 * sizeof(float));
1804
        memset(saved_ltp + 576, 0,     448 * sizeof(float));
1805
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
1806
        for (i = 0; i < 64; i++)
1807
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
1808
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1809
        memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
1810
        memset(saved_ltp + 576, 0,                  448 * sizeof(float));
1811
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
1812
        for (i = 0; i < 64; i++)
1813
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
1814
    } else { // LONG_STOP or ONLY_LONG
1815
        ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
1816
        for (i = 0; i < 512; i++)
1817
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
1818
    }
1819

    
1820
    memcpy(sce->ltp_state, &sce->ltp_state[1024], 1024 * sizeof(int16_t));
1821
    ac->fmt_conv.float_to_int16(&(sce->ltp_state[1024]), sce->ret,  1024);
1822
    ac->fmt_conv.float_to_int16(&(sce->ltp_state[2048]), saved_ltp, 1024);
1823
}
1824

    
1825
/**
1826
 * Conduct IMDCT and windowing.
1827
 */
1828
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1829
{
1830
    IndividualChannelStream *ics = &sce->ics;
1831
    float *in    = sce->coeffs;
1832
    float *out   = sce->ret;
1833
    float *saved = sce->saved;
1834
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1835
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1836
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1837
    float *buf  = ac->buf_mdct;
1838
    float *temp = ac->temp;
1839
    int i;
1840

    
1841
    // imdct
1842
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1843
        for (i = 0; i < 1024; i += 128)
1844
            ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
1845
    } else
1846
        ac->mdct.imdct_half(&ac->mdct, buf, in);
1847

    
1848
    /* window overlapping
1849
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1850
     * and long to short transitions are considered to be short to short
1851
     * transitions. This leaves just two cases (long to long and short to short)
1852
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1853
     */
1854
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1855
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1856
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
1857
    } else {
1858
        memcpy(                        out,               saved,            448 * sizeof(float));
1859

    
1860
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1861
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
1862
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
1863
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
1864
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
1865
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
1866
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1867
        } else {
1868
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
1869
            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
1870
        }
1871
    }
1872

    
1873
    // buffer update
1874
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1875
        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
1876
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
1877
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
1878
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
1879
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1880
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1881
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1882
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1883
    } else { // LONG_STOP or ONLY_LONG
1884
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1885
    }
1886
}
1887

    
1888
/**
1889
 * Apply dependent channel coupling (applied before IMDCT).
1890
 *
1891
 * @param   index   index into coupling gain array
1892
 */
1893
static void apply_dependent_coupling(AACContext *ac,
1894
                                     SingleChannelElement *target,
1895
                                     ChannelElement *cce, int index)
1896
{
1897
    IndividualChannelStream *ics = &cce->ch[0].ics;
1898
    const uint16_t *offsets = ics->swb_offset;
1899
    float *dest = target->coeffs;
1900
    const float *src = cce->ch[0].coeffs;
1901
    int g, i, group, k, idx = 0;
1902
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1903
        av_log(ac->avctx, AV_LOG_ERROR,
1904
               "Dependent coupling is not supported together with LTP\n");
1905
        return;
1906
    }
1907
    for (g = 0; g < ics->num_window_groups; g++) {
1908
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1909
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1910
                const float gain = cce->coup.gain[index][idx];
1911
                for (group = 0; group < ics->group_len[g]; group++) {
1912
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1913
                        // XXX dsputil-ize
1914
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1915
                    }
1916
                }
1917
            }
1918
        }
1919
        dest += ics->group_len[g] * 128;
1920
        src  += ics->group_len[g] * 128;
1921
    }
1922
}
1923

    
1924
/**
1925
 * Apply independent channel coupling (applied after IMDCT).
1926
 *
1927
 * @param   index   index into coupling gain array
1928
 */
1929
static void apply_independent_coupling(AACContext *ac,
1930
                                       SingleChannelElement *target,
1931
                                       ChannelElement *cce, int index)
1932
{
1933
    int i;
1934
    const float gain = cce->coup.gain[index][0];
1935
    const float *src = cce->ch[0].ret;
1936
    float *dest = target->ret;
1937
    const int len = 1024 << (ac->m4ac.sbr == 1);
1938

    
1939
    for (i = 0; i < len; i++)
1940
        dest[i] += gain * src[i];
1941
}
1942

    
1943
/**
1944
 * channel coupling transformation interface
1945
 *
1946
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1947
 */
1948
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1949
                                   enum RawDataBlockType type, int elem_id,
1950
                                   enum CouplingPoint coupling_point,
1951
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1952
{
1953
    int i, c;
1954

    
1955
    for (i = 0; i < MAX_ELEM_ID; i++) {
1956
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1957
        int index = 0;
1958

    
1959
        if (cce && cce->coup.coupling_point == coupling_point) {
1960
            ChannelCoupling *coup = &cce->coup;
1961

    
1962
            for (c = 0; c <= coup->num_coupled; c++) {
1963
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1964
                    if (coup->ch_select[c] != 1) {
1965
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1966
                        if (coup->ch_select[c] != 0)
1967
                            index++;
1968
                    }
1969
                    if (coup->ch_select[c] != 2)
1970
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1971
                } else
1972
                    index += 1 + (coup->ch_select[c] == 3);
1973
            }
1974
        }
1975
    }
1976
}
1977

    
1978
/**
1979
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1980
 */
1981
static void spectral_to_sample(AACContext *ac)
1982
{
1983
    int i, type;
1984
    for (type = 3; type >= 0; type--) {
1985
        for (i = 0; i < MAX_ELEM_ID; i++) {
1986
            ChannelElement *che = ac->che[type][i];
1987
            if (che) {
1988
                if (type <= TYPE_CPE)
1989
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1990
                if (ac->m4ac.object_type == AOT_AAC_LTP) {
1991
                    if (che->ch[0].ics.predictor_present) {
1992
                        if (che->ch[0].ics.ltp.present)
1993
                            apply_ltp(ac, &che->ch[0]);
1994
                        if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
1995
                            apply_ltp(ac, &che->ch[1]);
1996
                    }
1997
                }
1998
                if (che->ch[0].tns.present)
1999
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2000
                if (che->ch[1].tns.present)
2001
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2002
                if (type <= TYPE_CPE)
2003
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
2004
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2005
                    imdct_and_windowing(ac, &che->ch[0]);
2006
                    if (ac->m4ac.object_type == AOT_AAC_LTP)
2007
                        update_ltp(ac, &che->ch[0]);
2008
                    if (type == TYPE_CPE) {
2009
                        imdct_and_windowing(ac, &che->ch[1]);
2010
                        if (ac->m4ac.object_type == AOT_AAC_LTP)
2011
                            update_ltp(ac, &che->ch[1]);
2012
                    }
2013
                    if (ac->m4ac.sbr > 0) {
2014
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2015
                    }
2016
                }
2017
                if (type <= TYPE_CCE)
2018
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
2019
            }
2020
        }
2021
    }
2022
}
2023

    
2024
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2025
{
2026
    int size;
2027
    AACADTSHeaderInfo hdr_info;
2028

    
2029
    size = ff_aac_parse_header(gb, &hdr_info);
2030
    if (size > 0) {
2031
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
2032
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2033
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2034
            ac->m4ac.chan_config = hdr_info.chan_config;
2035
            if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
2036
                return -7;
2037
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
2038
                return -7;
2039
        } else if (ac->output_configured != OC_LOCKED) {
2040
            ac->output_configured = OC_NONE;
2041
        }
2042
        if (ac->output_configured != OC_LOCKED) {
2043
            ac->m4ac.sbr = -1;
2044
            ac->m4ac.ps  = -1;
2045
        }
2046
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
2047
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
2048
        ac->m4ac.object_type     = hdr_info.object_type;
2049
        if (!ac->avctx->sample_rate)
2050
            ac->avctx->sample_rate = hdr_info.sample_rate;
2051
        if (hdr_info.num_aac_frames == 1) {
2052
            if (!hdr_info.crc_absent)
2053
                skip_bits(gb, 16);
2054
        } else {
2055
            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
2056
            return -1;
2057
        }
2058
    }
2059
    return size;
2060
}
2061

    
2062
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2063
                                int *data_size, GetBitContext *gb)
2064
{
2065
    AACContext *ac = avctx->priv_data;
2066
    ChannelElement *che = NULL, *che_prev = NULL;
2067
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2068
    int err, elem_id, data_size_tmp;
2069
    int samples = 0, multiplier;
2070

    
2071
    if (show_bits(gb, 12) == 0xfff) {
2072
        if (parse_adts_frame_header(ac, gb) < 0) {
2073
            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2074
            return -1;
2075
        }
2076
        if (ac->m4ac.sampling_index > 12) {
2077
            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
2078
            return -1;
2079
        }
2080
    }
2081

    
2082
    ac->tags_mapped = 0;
2083
    // parse
2084
    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2085
        elem_id = get_bits(gb, 4);
2086

    
2087
        if (elem_type < TYPE_DSE) {
2088
            if (!(che=get_che(ac, elem_type, elem_id))) {
2089
                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2090
                       elem_type, elem_id);
2091
                return -1;
2092
            }
2093
            samples = 1024;
2094
        }
2095

    
2096
        switch (elem_type) {
2097

    
2098
        case TYPE_SCE:
2099
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2100
            break;
2101

    
2102
        case TYPE_CPE:
2103
            err = decode_cpe(ac, gb, che);
2104
            break;
2105

    
2106
        case TYPE_CCE:
2107
            err = decode_cce(ac, gb, che);
2108
            break;
2109

    
2110
        case TYPE_LFE:
2111
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2112
            break;
2113

    
2114
        case TYPE_DSE:
2115
            err = skip_data_stream_element(ac, gb);
2116
            break;
2117

    
2118
        case TYPE_PCE: {
2119
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2120
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2121
            if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
2122
                break;
2123
            if (ac->output_configured > OC_TRIAL_PCE)
2124
                av_log(avctx, AV_LOG_ERROR,
2125
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2126
            else
2127
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
2128
            break;
2129
        }
2130

    
2131
        case TYPE_FIL:
2132
            if (elem_id == 15)
2133
                elem_id += get_bits(gb, 8) - 1;
2134
            if (get_bits_left(gb) < 8 * elem_id) {
2135
                    av_log(avctx, AV_LOG_ERROR, overread_err);
2136
                    return -1;
2137
            }
2138
            while (elem_id > 0)
2139
                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
2140
            err = 0; /* FIXME */
2141
            break;
2142

    
2143
        default:
2144
            err = -1; /* should not happen, but keeps compiler happy */
2145
            break;
2146
        }
2147

    
2148
        che_prev       = che;
2149
        elem_type_prev = elem_type;
2150

    
2151
        if (err)
2152
            return err;
2153

    
2154
        if (get_bits_left(gb) < 3) {
2155
            av_log(avctx, AV_LOG_ERROR, overread_err);
2156
            return -1;
2157
        }
2158
    }
2159

    
2160
    spectral_to_sample(ac);
2161

    
2162
    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
2163
    samples <<= multiplier;
2164
    if (ac->output_configured < OC_LOCKED) {
2165
        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
2166
        avctx->frame_size = samples;
2167
    }
2168

    
2169
    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
2170
    if (*data_size < data_size_tmp) {
2171
        av_log(avctx, AV_LOG_ERROR,
2172
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2173
               *data_size, data_size_tmp);
2174
        return -1;
2175
    }
2176
    *data_size = data_size_tmp;
2177

    
2178
    if (samples)
2179
        ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
2180

    
2181
    if (ac->output_configured)
2182
        ac->output_configured = OC_LOCKED;
2183

    
2184
    return 0;
2185
}
2186

    
2187
static int aac_decode_frame(AVCodecContext *avctx, void *data,
2188
                            int *data_size, AVPacket *avpkt)
2189
{
2190
    const uint8_t *buf = avpkt->data;
2191
    int buf_size = avpkt->size;
2192
    GetBitContext gb;
2193
    int buf_consumed;
2194
    int buf_offset;
2195
    int err;
2196

    
2197
    init_get_bits(&gb, buf, buf_size * 8);
2198

    
2199
    if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
2200
        return err;
2201

    
2202
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2203
    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
2204
        if (buf[buf_offset])
2205
            break;
2206

    
2207
    return buf_size > buf_offset ? buf_consumed : buf_size;
2208
}
2209

    
2210
static av_cold int aac_decode_close(AVCodecContext *avctx)
2211
{
2212
    AACContext *ac = avctx->priv_data;
2213
    int i, type;
2214

    
2215
    for (i = 0; i < MAX_ELEM_ID; i++) {
2216
        for (type = 0; type < 4; type++) {
2217
            if (ac->che[type][i])
2218
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
2219
            av_freep(&ac->che[type][i]);
2220
        }
2221
    }
2222

    
2223
    ff_mdct_end(&ac->mdct);
2224
    ff_mdct_end(&ac->mdct_small);
2225
    ff_mdct_end(&ac->mdct_ltp);
2226
    return 0;
2227
}
2228

    
2229

    
2230
#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
2231

    
2232
struct LATMContext {
2233
    AACContext      aac_ctx;             ///< containing AACContext
2234
    int             initialized;         ///< initilized after a valid extradata was seen
2235

    
2236
    // parser data
2237
    int             audio_mux_version_A; ///< LATM syntax version
2238
    int             frame_length_type;   ///< 0/1 variable/fixed frame length
2239
    int             frame_length;        ///< frame length for fixed frame length
2240
};
2241

    
2242
static inline uint32_t latm_get_value(GetBitContext *b)
2243
{
2244
    int length = get_bits(b, 2);
2245

    
2246
    return get_bits_long(b, (length+1)*8);
2247
}
2248

    
2249
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
2250
                                             GetBitContext *gb)
2251
{
2252
    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
2253
    MPEG4AudioConfig m4ac;
2254
    int  config_start_bit = get_bits_count(gb);
2255
    int     bits_consumed, esize;
2256

    
2257
    if (config_start_bit % 8) {
2258
        av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
2259
                               "config not byte aligned.\n", 1);
2260
        return AVERROR_INVALIDDATA;
2261
    } else {
2262
        bits_consumed =
2263
            decode_audio_specific_config(NULL, avctx, &m4ac,
2264
                                         gb->buffer + (config_start_bit / 8),
2265
                                         get_bits_left(gb) / 8);
2266

    
2267
        if (bits_consumed < 0)
2268
            return AVERROR_INVALIDDATA;
2269

    
2270
        esize = (bits_consumed+7) / 8;
2271

    
2272
        if (avctx->extradata_size <= esize) {
2273
            av_free(avctx->extradata);
2274
            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
2275
            if (!avctx->extradata)
2276
                return AVERROR(ENOMEM);
2277
        }
2278

    
2279
        avctx->extradata_size = esize;
2280
        memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
2281
        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
2282

    
2283
        skip_bits_long(gb, bits_consumed);
2284
    }
2285

    
2286
    return bits_consumed;
2287
}
2288

    
2289
static int read_stream_mux_config(struct LATMContext *latmctx,
2290
                                  GetBitContext *gb)
2291
{
2292
    int ret, audio_mux_version = get_bits(gb, 1);
2293

    
2294
    latmctx->audio_mux_version_A = 0;
2295
    if (audio_mux_version)
2296
        latmctx->audio_mux_version_A = get_bits(gb, 1);
2297

    
2298
    if (!latmctx->audio_mux_version_A) {
2299

    
2300
        if (audio_mux_version)
2301
            latm_get_value(gb);                 // taraFullness
2302

    
2303
        skip_bits(gb, 1);                       // allStreamSameTimeFraming
2304
        skip_bits(gb, 6);                       // numSubFrames
2305
        // numPrograms
2306
        if (get_bits(gb, 4)) {                  // numPrograms
2307
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2308
                                   "multiple programs are not supported\n", 1);
2309
            return AVERROR_PATCHWELCOME;
2310
        }
2311

    
2312
        // for each program (which there is only on in DVB)
2313

    
2314
        // for each layer (which there is only on in DVB)
2315
        if (get_bits(gb, 3)) {                   // numLayer
2316
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2317
                                   "multiple layers are not supported\n", 1);
2318
            return AVERROR_PATCHWELCOME;
2319
        }
2320

    
2321
        // for all but first stream: use_same_config = get_bits(gb, 1);
2322
        if (!audio_mux_version) {
2323
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2324
                return ret;
2325
        } else {
2326
            int ascLen = latm_get_value(gb);
2327
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2328
                return ret;
2329
            ascLen -= ret;
2330
            skip_bits_long(gb, ascLen);
2331
        }
2332

    
2333
        latmctx->frame_length_type = get_bits(gb, 3);
2334
        switch (latmctx->frame_length_type) {
2335
        case 0:
2336
            skip_bits(gb, 8);       // latmBufferFullness
2337
            break;
2338
        case 1:
2339
            latmctx->frame_length = get_bits(gb, 9);
2340
            break;
2341
        case 3:
2342
        case 4:
2343
        case 5:
2344
            skip_bits(gb, 6);       // CELP frame length table index
2345
            break;
2346
        case 6:
2347
        case 7:
2348
            skip_bits(gb, 1);       // HVXC frame length table index
2349
            break;
2350
        }
2351

    
2352
        if (get_bits(gb, 1)) {                  // other data
2353
            if (audio_mux_version) {
2354
                latm_get_value(gb);             // other_data_bits
2355
            } else {
2356
                int esc;
2357
                do {
2358
                    esc = get_bits(gb, 1);
2359
                    skip_bits(gb, 8);
2360
                } while (esc);
2361
            }
2362
        }
2363

    
2364
        if (get_bits(gb, 1))                     // crc present
2365
            skip_bits(gb, 8);                    // config_crc
2366
    }
2367

    
2368
    return 0;
2369
}
2370

    
2371
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
2372
{
2373
    uint8_t tmp;
2374

    
2375
    if (ctx->frame_length_type == 0) {
2376
        int mux_slot_length = 0;
2377
        do {
2378
            tmp = get_bits(gb, 8);
2379
            mux_slot_length += tmp;
2380
        } while (tmp == 255);
2381
        return mux_slot_length;
2382
    } else if (ctx->frame_length_type == 1) {
2383
        return ctx->frame_length;
2384
    } else if (ctx->frame_length_type == 3 ||
2385
               ctx->frame_length_type == 5 ||
2386
               ctx->frame_length_type == 7) {
2387
        skip_bits(gb, 2);          // mux_slot_length_coded
2388
    }
2389
    return 0;
2390
}
2391

    
2392
static int read_audio_mux_element(struct LATMContext *latmctx,
2393
                                  GetBitContext *gb)
2394
{
2395
    int err;
2396
    uint8_t use_same_mux = get_bits(gb, 1);
2397
    if (!use_same_mux) {
2398
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
2399
            return err;
2400
    } else if (!latmctx->aac_ctx.avctx->extradata) {
2401
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
2402
               "no decoder config found\n");
2403
        return AVERROR(EAGAIN);
2404
    }
2405
    if (latmctx->audio_mux_version_A == 0) {
2406
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
2407
        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
2408
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
2409
            return AVERROR_INVALIDDATA;
2410
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
2411
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2412
                   "frame length mismatch %d << %d\n",
2413
                   mux_slot_length_bytes * 8, get_bits_left(gb));
2414
            return AVERROR_INVALIDDATA;
2415
        }
2416
    }
2417
    return 0;
2418
}
2419

    
2420

    
2421
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
2422
                             AVPacket *avpkt)
2423
{
2424
    struct LATMContext *latmctx = avctx->priv_data;
2425
    int                 muxlength, err;
2426
    GetBitContext       gb;
2427

    
2428
    if (avpkt->size == 0)
2429
        return 0;
2430

    
2431
    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
2432

    
2433
    // check for LOAS sync word
2434
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
2435
        return AVERROR_INVALIDDATA;
2436

    
2437
    muxlength = get_bits(&gb, 13) + 3;
2438
    // not enough data, the parser should have sorted this
2439
    if (muxlength > avpkt->size)
2440
        return AVERROR_INVALIDDATA;
2441

    
2442
    if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
2443
        return err;
2444

    
2445
    if (!latmctx->initialized) {
2446
        if (!avctx->extradata) {
2447
            *out_size = 0;
2448
            return avpkt->size;
2449
        } else {
2450
            if ((err = aac_decode_init(avctx)) < 0)
2451
                return err;
2452
            latmctx->initialized = 1;
2453
        }
2454
    }
2455

    
2456
    if (show_bits(&gb, 12) == 0xfff) {
2457
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2458
               "ADTS header detected, probably as result of configuration "
2459
               "misparsing\n");
2460
        return AVERROR_INVALIDDATA;
2461
    }
2462

    
2463
    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
2464
        return err;
2465

    
2466
    return muxlength;
2467
}
2468

    
2469
av_cold static int latm_decode_init(AVCodecContext *avctx)
2470
{
2471
    struct LATMContext *latmctx = avctx->priv_data;
2472
    int ret;
2473

    
2474
    ret = aac_decode_init(avctx);
2475

    
2476
    if (avctx->extradata_size > 0) {
2477
        latmctx->initialized = !ret;
2478
    } else {
2479
        latmctx->initialized = 0;
2480
    }
2481

    
2482
    return ret;
2483
}
2484

    
2485

    
2486
AVCodec ff_aac_decoder = {
2487
    "aac",
2488
    AVMEDIA_TYPE_AUDIO,
2489
    CODEC_ID_AAC,
2490
    sizeof(AACContext),
2491
    aac_decode_init,
2492
    NULL,
2493
    aac_decode_close,
2494
    aac_decode_frame,
2495
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2496
    .sample_fmts = (const enum AVSampleFormat[]) {
2497
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
2498
    },
2499
    .channel_layouts = aac_channel_layout,
2500
};
2501

    
2502
/*
2503
    Note: This decoder filter is intended to decode LATM streams transferred
2504
    in MPEG transport streams which only contain one program.
2505
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
2506
*/
2507
AVCodec ff_aac_latm_decoder = {
2508
    .name = "aac_latm",
2509
    .type = AVMEDIA_TYPE_AUDIO,
2510
    .id   = CODEC_ID_AAC_LATM,
2511
    .priv_data_size = sizeof(struct LATMContext),
2512
    .init   = latm_decode_init,
2513
    .close  = aac_decode_close,
2514
    .decode = latm_decode_frame,
2515
    .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
2516
    .sample_fmts = (const enum AVSampleFormat[]) {
2517
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
2518
    },
2519
    .channel_layouts = aac_channel_layout,
2520
};