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/*
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 FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * 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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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
26

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

    
34
/*
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 * supported tools
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 *
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 * Support?             Name
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 * N (code in SoC repo) gain control
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 * Y                    block switching
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 * Y                    window shapes - standard
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 * N                    window shapes - Low Delay
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 * Y                    filterbank - standard
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 * N (code in SoC repo) filterbank - Scalable Sample Rate
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 * Y                    Temporal Noise Shaping
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 * N (code in SoC repo) Long Term Prediction
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 * Y                    intensity stereo
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 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * Y                    Perceptual Noise Substitution
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 * Y                    Mid/Side stereo
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 * N                    Scalable Inverse AAC Quantization
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 * N                    Frequency Selective Switch
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 * N                    upsampling filter
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 * Y                    quantization & coding - AAC
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 * N                    quantization & coding - TwinVQ
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 * N                    quantization & coding - BSAC
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 * N                    AAC Error Resilience tools
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 * N                    Error Resilience payload syntax
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 * N                    Error Protection tool
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 * N                    CELP
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 * N                    Silence Compression
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 * N                    HVXC
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 * N                    HVXC 4kbits/s VR
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 * N                    Structured Audio tools
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 * N                    Structured Audio Sample Bank Format
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 * N                    MIDI
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 * N                    Harmonic and Individual Lines plus Noise
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 * N                    Text-To-Speech Interface
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 * Y                    Spectral Band Replication
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 * Y (not in this code) Layer-1
71
 * 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
79
           Parametric Stereo.
80
 */
81

    
82

    
83
#include "avcodec.h"
84
#include "internal.h"
85
#include "get_bits.h"
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#include "dsputil.h"
87
#include "fft.h"
88
#include "fmtconvert.h"
89
#include "lpc.h"
90

    
91
#include "aac.h"
92
#include "aactab.h"
93
#include "aacdectab.h"
94
#include "cbrt_tablegen.h"
95
#include "sbr.h"
96
#include "aacsbr.h"
97
#include "mpeg4audio.h"
98
#include "aacadtsdec.h"
99

    
100
#include <assert.h>
101
#include <errno.h>
102
#include <math.h>
103
#include <string.h>
104

    
105
#if ARCH_ARM
106
#   include "arm/aac.h"
107
#endif
108

    
109
union float754 {
110
    float f;
111
    uint32_t i;
112
};
113

    
114
static VLC vlc_scalefactors;
115
static VLC vlc_spectral[11];
116

    
117
static const char overread_err[] = "Input buffer exhausted before END element found\n";
118

    
119
static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
120
{
121
    // For PCE based channel configurations map the channels solely based on tags.
122
    if (!ac->m4ac.chan_config) {
123
        return ac->tag_che_map[type][elem_id];
124
    }
125
    // For indexed channel configurations map the channels solely based on position.
126
    switch (ac->m4ac.chan_config) {
127
    case 7:
128
        if (ac->tags_mapped == 3 && type == TYPE_CPE) {
129
            ac->tags_mapped++;
130
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
131
        }
132
    case 6:
133
        /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
134
           instead of SCE[0] CPE[0] CPE[1] LFE[0]. If we seem to have
135
           encountered such a stream, transfer the LFE[0] element to the SCE[1]'s mapping */
136
        if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
137
            ac->tags_mapped++;
138
            return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
139
        }
140
    case 5:
141
        if (ac->tags_mapped == 2 && type == TYPE_CPE) {
142
            ac->tags_mapped++;
143
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
144
        }
145
    case 4:
146
        if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
147
            ac->tags_mapped++;
148
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
149
        }
150
    case 3:
151
    case 2:
152
        if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
153
            ac->tags_mapped++;
154
            return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
155
        } else if (ac->m4ac.chan_config == 2) {
156
            return NULL;
157
        }
158
    case 1:
159
        if (!ac->tags_mapped && type == TYPE_SCE) {
160
            ac->tags_mapped++;
161
            return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
162
        }
163
    default:
164
        return NULL;
165
    }
166
}
167

    
168
/**
169
 * Check for the channel element in the current channel position configuration.
170
 * If it exists, make sure the appropriate element is allocated and map the
171
 * channel order to match the internal FFmpeg channel layout.
172
 *
173
 * @param   che_pos current channel position configuration
174
 * @param   type channel element type
175
 * @param   id channel element id
176
 * @param   channels count of the number of channels in the configuration
177
 *
178
 * @return  Returns error status. 0 - OK, !0 - error
179
 */
180
static av_cold int che_configure(AACContext *ac,
181
                         enum ChannelPosition che_pos[4][MAX_ELEM_ID],
182
                         int type, int id,
183
                         int *channels)
184
{
185
    if (che_pos[type][id]) {
186
        if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
187
            return AVERROR(ENOMEM);
188
        ff_aac_sbr_ctx_init(&ac->che[type][id]->sbr);
189
        if (type != TYPE_CCE) {
190
            ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
191
            if (type == TYPE_CPE ||
192
                (type == TYPE_SCE && ac->m4ac.ps == 1)) {
193
                ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
194
            }
195
        }
196
    } else {
197
        if (ac->che[type][id])
198
            ff_aac_sbr_ctx_close(&ac->che[type][id]->sbr);
199
        av_freep(&ac->che[type][id]);
200
    }
201
    return 0;
202
}
203

    
204
/**
205
 * Configure output channel order based on the current program configuration element.
206
 *
207
 * @param   che_pos current channel position configuration
208
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
209
 *
210
 * @return  Returns error status. 0 - OK, !0 - error
211
 */
212
static av_cold int output_configure(AACContext *ac,
213
                            enum ChannelPosition che_pos[4][MAX_ELEM_ID],
214
                            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
215
                            int channel_config, enum OCStatus oc_type)
216
{
217
    AVCodecContext *avctx = ac->avctx;
218
    int i, type, channels = 0, ret;
219

    
220
    if (new_che_pos != che_pos)
221
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
222

    
223
    if (channel_config) {
224
        for (i = 0; i < tags_per_config[channel_config]; i++) {
225
            if ((ret = che_configure(ac, che_pos,
226
                                     aac_channel_layout_map[channel_config - 1][i][0],
227
                                     aac_channel_layout_map[channel_config - 1][i][1],
228
                                     &channels)))
229
                return ret;
230
        }
231

    
232
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
233

    
234
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
235
    } else {
236
        /* Allocate or free elements depending on if they are in the
237
         * current program configuration.
238
         *
239
         * Set up default 1:1 output mapping.
240
         *
241
         * For a 5.1 stream the output order will be:
242
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
243
         */
244

    
245
        for (i = 0; i < MAX_ELEM_ID; i++) {
246
            for (type = 0; type < 4; type++) {
247
                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
248
                    return ret;
249
            }
250
        }
251

    
252
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
253

    
254
        avctx->channel_layout = 0;
255
    }
256

    
257
    avctx->channels = channels;
258

    
259
    ac->output_configured = oc_type;
260

    
261
    return 0;
262
}
263

    
264
/**
265
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
266
 *
267
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
268
 * @param sce_map mono (Single Channel Element) map
269
 * @param type speaker type/position for these channels
270
 */
271
static void decode_channel_map(enum ChannelPosition *cpe_map,
272
                               enum ChannelPosition *sce_map,
273
                               enum ChannelPosition type,
274
                               GetBitContext *gb, int n)
275
{
276
    while (n--) {
277
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
278
        map[get_bits(gb, 4)] = type;
279
    }
280
}
281

    
282
/**
283
 * Decode program configuration element; reference: table 4.2.
284
 *
285
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
286
 *
287
 * @return  Returns error status. 0 - OK, !0 - error
288
 */
289
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
290
                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
291
                      GetBitContext *gb)
292
{
293
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
294
    int comment_len;
295

    
296
    skip_bits(gb, 2);  // object_type
297

    
298
    sampling_index = get_bits(gb, 4);
299
    if (m4ac->sampling_index != sampling_index)
300
        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");
301

    
302
    num_front       = get_bits(gb, 4);
303
    num_side        = get_bits(gb, 4);
304
    num_back        = get_bits(gb, 4);
305
    num_lfe         = get_bits(gb, 2);
306
    num_assoc_data  = get_bits(gb, 3);
307
    num_cc          = get_bits(gb, 4);
308

    
309
    if (get_bits1(gb))
310
        skip_bits(gb, 4); // mono_mixdown_tag
311
    if (get_bits1(gb))
312
        skip_bits(gb, 4); // stereo_mixdown_tag
313

    
314
    if (get_bits1(gb))
315
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
316

    
317
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
318
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
319
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
320
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
321

    
322
    skip_bits_long(gb, 4 * num_assoc_data);
323

    
324
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
325

    
326
    align_get_bits(gb);
327

    
328
    /* comment field, first byte is length */
329
    comment_len = get_bits(gb, 8) * 8;
330
    if (get_bits_left(gb) < comment_len) {
331
        av_log(avctx, AV_LOG_ERROR, overread_err);
332
        return -1;
333
    }
334
    skip_bits_long(gb, comment_len);
335
    return 0;
336
}
337

    
338
/**
339
 * Set up channel positions based on a default channel configuration
340
 * as specified in table 1.17.
341
 *
342
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
343
 *
344
 * @return  Returns error status. 0 - OK, !0 - error
345
 */
346
static av_cold int set_default_channel_config(AVCodecContext *avctx,
347
                                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
348
                                      int channel_config)
349
{
350
    if (channel_config < 1 || channel_config > 7) {
351
        av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
352
               channel_config);
353
        return -1;
354
    }
355

    
356
    /* default channel configurations:
357
     *
358
     * 1ch : front center (mono)
359
     * 2ch : L + R (stereo)
360
     * 3ch : front center + L + R
361
     * 4ch : front center + L + R + back center
362
     * 5ch : front center + L + R + back stereo
363
     * 6ch : front center + L + R + back stereo + LFE
364
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
365
     */
366

    
367
    if (channel_config != 2)
368
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
369
    if (channel_config > 1)
370
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
371
    if (channel_config == 4)
372
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
373
    if (channel_config > 4)
374
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
375
        = AAC_CHANNEL_BACK;  // back stereo
376
    if (channel_config > 5)
377
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
378
    if (channel_config == 7)
379
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
380

    
381
    return 0;
382
}
383

    
384
/**
385
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
386
 *
387
 * @param   ac          pointer to AACContext, may be null
388
 * @param   avctx       pointer to AVCCodecContext, used for logging
389
 *
390
 * @return  Returns error status. 0 - OK, !0 - error
391
 */
392
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
393
                                     GetBitContext *gb,
394
                                     MPEG4AudioConfig *m4ac,
395
                                     int channel_config)
396
{
397
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
398
    int extension_flag, ret;
399

    
400
    if (get_bits1(gb)) { // frameLengthFlag
401
        av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
402
        return -1;
403
    }
404

    
405
    if (get_bits1(gb))       // dependsOnCoreCoder
406
        skip_bits(gb, 14);   // coreCoderDelay
407
    extension_flag = get_bits1(gb);
408

    
409
    if (m4ac->object_type == AOT_AAC_SCALABLE ||
410
        m4ac->object_type == AOT_ER_AAC_SCALABLE)
411
        skip_bits(gb, 3);     // layerNr
412

    
413
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
414
    if (channel_config == 0) {
415
        skip_bits(gb, 4);  // element_instance_tag
416
        if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
417
            return ret;
418
    } else {
419
        if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
420
            return ret;
421
    }
422
    if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
423
        return ret;
424

    
425
    if (extension_flag) {
426
        switch (m4ac->object_type) {
427
        case AOT_ER_BSAC:
428
            skip_bits(gb, 5);    // numOfSubFrame
429
            skip_bits(gb, 11);   // layer_length
430
            break;
431
        case AOT_ER_AAC_LC:
432
        case AOT_ER_AAC_LTP:
433
        case AOT_ER_AAC_SCALABLE:
434
        case AOT_ER_AAC_LD:
435
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
436
                                    * aacScalefactorDataResilienceFlag
437
                                    * aacSpectralDataResilienceFlag
438
                                    */
439
            break;
440
        }
441
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
442
    }
443
    return 0;
444
}
445

    
446
/**
447
 * Decode audio specific configuration; reference: table 1.13.
448
 *
449
 * @param   ac          pointer to AACContext, may be null
450
 * @param   avctx       pointer to AVCCodecContext, used for logging
451
 * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
452
 * @param   data        pointer to AVCodecContext extradata
453
 * @param   data_size   size of AVCCodecContext extradata
454
 *
455
 * @return  Returns error status or number of consumed bits. <0 - error
456
 */
457
static int decode_audio_specific_config(AACContext *ac,
458
                                        AVCodecContext *avctx,
459
                                        MPEG4AudioConfig *m4ac,
460
                                        const uint8_t *data, int data_size)
461
{
462
    GetBitContext gb;
463
    int i;
464

    
465
    init_get_bits(&gb, data, data_size * 8);
466

    
467
    if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
468
        return -1;
469
    if (m4ac->sampling_index > 12) {
470
        av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
471
        return -1;
472
    }
473
    if (m4ac->sbr == 1 && m4ac->ps == -1)
474
        m4ac->ps = 1;
475

    
476
    skip_bits_long(&gb, i);
477

    
478
    switch (m4ac->object_type) {
479
    case AOT_AAC_MAIN:
480
    case AOT_AAC_LC:
481
        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
482
            return -1;
483
        break;
484
    default:
485
        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
486
               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
487
        return -1;
488
    }
489

    
490
    return get_bits_count(&gb);
491
}
492

    
493
/**
494
 * linear congruential pseudorandom number generator
495
 *
496
 * @param   previous_val    pointer to the current state of the generator
497
 *
498
 * @return  Returns a 32-bit pseudorandom integer
499
 */
500
static av_always_inline int lcg_random(int previous_val)
501
{
502
    return previous_val * 1664525 + 1013904223;
503
}
504

    
505
static av_always_inline void reset_predict_state(PredictorState *ps)
506
{
507
    ps->r0   = 0.0f;
508
    ps->r1   = 0.0f;
509
    ps->cor0 = 0.0f;
510
    ps->cor1 = 0.0f;
511
    ps->var0 = 1.0f;
512
    ps->var1 = 1.0f;
513
}
514

    
515
static void reset_all_predictors(PredictorState *ps)
516
{
517
    int i;
518
    for (i = 0; i < MAX_PREDICTORS; i++)
519
        reset_predict_state(&ps[i]);
520
}
521

    
522
static void reset_predictor_group(PredictorState *ps, int group_num)
523
{
524
    int i;
525
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
526
        reset_predict_state(&ps[i]);
527
}
528

    
529
#define AAC_INIT_VLC_STATIC(num, size) \
530
    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
531
         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
532
        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
533
        size);
534

    
535
static av_cold int aac_decode_init(AVCodecContext *avctx)
536
{
537
    AACContext *ac = avctx->priv_data;
538

    
539
    ac->avctx = avctx;
540
    ac->m4ac.sample_rate = avctx->sample_rate;
541

    
542
    if (avctx->extradata_size > 0) {
543
        if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
544
                                         avctx->extradata,
545
                                         avctx->extradata_size) < 0)
546
            return -1;
547
    }
548

    
549
    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
550

    
551
    AAC_INIT_VLC_STATIC( 0, 304);
552
    AAC_INIT_VLC_STATIC( 1, 270);
553
    AAC_INIT_VLC_STATIC( 2, 550);
554
    AAC_INIT_VLC_STATIC( 3, 300);
555
    AAC_INIT_VLC_STATIC( 4, 328);
556
    AAC_INIT_VLC_STATIC( 5, 294);
557
    AAC_INIT_VLC_STATIC( 6, 306);
558
    AAC_INIT_VLC_STATIC( 7, 268);
559
    AAC_INIT_VLC_STATIC( 8, 510);
560
    AAC_INIT_VLC_STATIC( 9, 366);
561
    AAC_INIT_VLC_STATIC(10, 462);
562

    
563
    ff_aac_sbr_init();
564

    
565
    dsputil_init(&ac->dsp, avctx);
566
    ff_fmt_convert_init(&ac->fmt_conv, avctx);
567

    
568
    ac->random_state = 0x1f2e3d4c;
569

    
570
    // -1024 - Compensate wrong IMDCT method.
571
    // 60    - Required to scale values to the correct range [-32768,32767]
572
    //         for float to int16 conversion. (1 << (60 / 4)) == 32768
573
    ac->sf_scale  = 1. / -1024.;
574
    ac->sf_offset = 60;
575

    
576
    ff_aac_tableinit();
577

    
578
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
579
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
580
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
581
                    352);
582

    
583
    ff_mdct_init(&ac->mdct, 11, 1, 1.0);
584
    ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
585
    // window initialization
586
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
587
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
588
    ff_init_ff_sine_windows(10);
589
    ff_init_ff_sine_windows( 7);
590

    
591
    cbrt_tableinit();
592

    
593
    return 0;
594
}
595

    
596
/**
597
 * Skip data_stream_element; reference: table 4.10.
598
 */
599
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
600
{
601
    int byte_align = get_bits1(gb);
602
    int count = get_bits(gb, 8);
603
    if (count == 255)
604
        count += get_bits(gb, 8);
605
    if (byte_align)
606
        align_get_bits(gb);
607

    
608
    if (get_bits_left(gb) < 8 * count) {
609
        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
610
        return -1;
611
    }
612
    skip_bits_long(gb, 8 * count);
613
    return 0;
614
}
615

    
616
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
617
                             GetBitContext *gb)
618
{
619
    int sfb;
620
    if (get_bits1(gb)) {
621
        ics->predictor_reset_group = get_bits(gb, 5);
622
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
623
            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
624
            return -1;
625
        }
626
    }
627
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
628
        ics->prediction_used[sfb] = get_bits1(gb);
629
    }
630
    return 0;
631
}
632

    
633
/**
634
 * Decode Individual Channel Stream info; reference: table 4.6.
635
 *
636
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
637
 */
638
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
639
                           GetBitContext *gb, int common_window)
640
{
641
    if (get_bits1(gb)) {
642
        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
643
        memset(ics, 0, sizeof(IndividualChannelStream));
644
        return -1;
645
    }
646
    ics->window_sequence[1] = ics->window_sequence[0];
647
    ics->window_sequence[0] = get_bits(gb, 2);
648
    ics->use_kb_window[1]   = ics->use_kb_window[0];
649
    ics->use_kb_window[0]   = get_bits1(gb);
650
    ics->num_window_groups  = 1;
651
    ics->group_len[0]       = 1;
652
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
653
        int i;
654
        ics->max_sfb = get_bits(gb, 4);
655
        for (i = 0; i < 7; i++) {
656
            if (get_bits1(gb)) {
657
                ics->group_len[ics->num_window_groups - 1]++;
658
            } else {
659
                ics->num_window_groups++;
660
                ics->group_len[ics->num_window_groups - 1] = 1;
661
            }
662
        }
663
        ics->num_windows       = 8;
664
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
665
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
666
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
667
        ics->predictor_present = 0;
668
    } else {
669
        ics->max_sfb               = get_bits(gb, 6);
670
        ics->num_windows           = 1;
671
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
672
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
673
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
674
        ics->predictor_present     = get_bits1(gb);
675
        ics->predictor_reset_group = 0;
676
        if (ics->predictor_present) {
677
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
678
                if (decode_prediction(ac, ics, gb)) {
679
                    memset(ics, 0, sizeof(IndividualChannelStream));
680
                    return -1;
681
                }
682
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
683
                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
684
                memset(ics, 0, sizeof(IndividualChannelStream));
685
                return -1;
686
            } else {
687
                av_log_missing_feature(ac->avctx, "Predictor bit set but LTP is", 1);
688
                memset(ics, 0, sizeof(IndividualChannelStream));
689
                return -1;
690
            }
691
        }
692
    }
693

    
694
    if (ics->max_sfb > ics->num_swb) {
695
        av_log(ac->avctx, AV_LOG_ERROR,
696
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
697
               ics->max_sfb, ics->num_swb);
698
        memset(ics, 0, sizeof(IndividualChannelStream));
699
        return -1;
700
    }
701

    
702
    return 0;
703
}
704

    
705
/**
706
 * Decode band types (section_data payload); reference: table 4.46.
707
 *
708
 * @param   band_type           array of the used band type
709
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
710
 *
711
 * @return  Returns error status. 0 - OK, !0 - error
712
 */
713
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
714
                             int band_type_run_end[120], GetBitContext *gb,
715
                             IndividualChannelStream *ics)
716
{
717
    int g, idx = 0;
718
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
719
    for (g = 0; g < ics->num_window_groups; g++) {
720
        int k = 0;
721
        while (k < ics->max_sfb) {
722
            uint8_t sect_end = k;
723
            int sect_len_incr;
724
            int sect_band_type = get_bits(gb, 4);
725
            if (sect_band_type == 12) {
726
                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
727
                return -1;
728
            }
729
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
730
                sect_end += sect_len_incr;
731
            sect_end += sect_len_incr;
732
            if (get_bits_left(gb) < 0) {
733
                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
734
                return -1;
735
            }
736
            if (sect_end > ics->max_sfb) {
737
                av_log(ac->avctx, AV_LOG_ERROR,
738
                       "Number of bands (%d) exceeds limit (%d).\n",
739
                       sect_end, ics->max_sfb);
740
                return -1;
741
            }
742
            for (; k < sect_end; k++) {
743
                band_type        [idx]   = sect_band_type;
744
                band_type_run_end[idx++] = sect_end;
745
            }
746
        }
747
    }
748
    return 0;
749
}
750

    
751
/**
752
 * Decode scalefactors; reference: table 4.47.
753
 *
754
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
755
 * @param   band_type           array of the used band type
756
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
757
 * @param   sf                  array of scalefactors or intensity stereo positions
758
 *
759
 * @return  Returns error status. 0 - OK, !0 - error
760
 */
761
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
762
                               unsigned int global_gain,
763
                               IndividualChannelStream *ics,
764
                               enum BandType band_type[120],
765
                               int band_type_run_end[120])
766
{
767
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
768
    int g, i, idx = 0;
769
    int offset[3] = { global_gain, global_gain - 90, 100 };
770
    int noise_flag = 1;
771
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
772
    for (g = 0; g < ics->num_window_groups; g++) {
773
        for (i = 0; i < ics->max_sfb;) {
774
            int run_end = band_type_run_end[idx];
775
            if (band_type[idx] == ZERO_BT) {
776
                for (; i < run_end; i++, idx++)
777
                    sf[idx] = 0.;
778
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
779
                for (; i < run_end; i++, idx++) {
780
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
781
                    if (offset[2] > 255U) {
782
                        av_log(ac->avctx, AV_LOG_ERROR,
783
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
784
                        return -1;
785
                    }
786
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
787
                }
788
            } else if (band_type[idx] == NOISE_BT) {
789
                for (; i < run_end; i++, idx++) {
790
                    if (noise_flag-- > 0)
791
                        offset[1] += get_bits(gb, 9) - 256;
792
                    else
793
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
794
                    if (offset[1] > 255U) {
795
                        av_log(ac->avctx, AV_LOG_ERROR,
796
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
797
                        return -1;
798
                    }
799
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
800
                }
801
            } else {
802
                for (; i < run_end; i++, idx++) {
803
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
804
                    if (offset[0] > 255U) {
805
                        av_log(ac->avctx, AV_LOG_ERROR,
806
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
807
                        return -1;
808
                    }
809
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
810
                }
811
            }
812
        }
813
    }
814
    return 0;
815
}
816

    
817
/**
818
 * Decode pulse data; reference: table 4.7.
819
 */
820
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
821
                         const uint16_t *swb_offset, int num_swb)
822
{
823
    int i, pulse_swb;
824
    pulse->num_pulse = get_bits(gb, 2) + 1;
825
    pulse_swb        = get_bits(gb, 6);
826
    if (pulse_swb >= num_swb)
827
        return -1;
828
    pulse->pos[0]    = swb_offset[pulse_swb];
829
    pulse->pos[0]   += get_bits(gb, 5);
830
    if (pulse->pos[0] > 1023)
831
        return -1;
832
    pulse->amp[0]    = get_bits(gb, 4);
833
    for (i = 1; i < pulse->num_pulse; i++) {
834
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
835
        if (pulse->pos[i] > 1023)
836
            return -1;
837
        pulse->amp[i] = get_bits(gb, 4);
838
    }
839
    return 0;
840
}
841

    
842
/**
843
 * Decode Temporal Noise Shaping data; reference: table 4.48.
844
 *
845
 * @return  Returns error status. 0 - OK, !0 - error
846
 */
847
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
848
                      GetBitContext *gb, const IndividualChannelStream *ics)
849
{
850
    int w, filt, i, coef_len, coef_res, coef_compress;
851
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
852
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
853
    for (w = 0; w < ics->num_windows; w++) {
854
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
855
            coef_res = get_bits1(gb);
856

    
857
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
858
                int tmp2_idx;
859
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
860

    
861
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
862
                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
863
                           tns->order[w][filt], tns_max_order);
864
                    tns->order[w][filt] = 0;
865
                    return -1;
866
                }
867
                if (tns->order[w][filt]) {
868
                    tns->direction[w][filt] = get_bits1(gb);
869
                    coef_compress = get_bits1(gb);
870
                    coef_len = coef_res + 3 - coef_compress;
871
                    tmp2_idx = 2 * coef_compress + coef_res;
872

    
873
                    for (i = 0; i < tns->order[w][filt]; i++)
874
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
875
                }
876
            }
877
        }
878
    }
879
    return 0;
880
}
881

    
882
/**
883
 * Decode Mid/Side data; reference: table 4.54.
884
 *
885
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
886
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
887
 *                      [3] reserved for scalable AAC
888
 */
889
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
890
                                   int ms_present)
891
{
892
    int idx;
893
    if (ms_present == 1) {
894
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
895
            cpe->ms_mask[idx] = get_bits1(gb);
896
    } else if (ms_present == 2) {
897
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
898
    }
899
}
900

    
901
#ifndef VMUL2
902
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
903
                           const float *scale)
904
{
905
    float s = *scale;
906
    *dst++ = v[idx    & 15] * s;
907
    *dst++ = v[idx>>4 & 15] * s;
908
    return dst;
909
}
910
#endif
911

    
912
#ifndef VMUL4
913
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
914
                           const float *scale)
915
{
916
    float s = *scale;
917
    *dst++ = v[idx    & 3] * s;
918
    *dst++ = v[idx>>2 & 3] * s;
919
    *dst++ = v[idx>>4 & 3] * s;
920
    *dst++ = v[idx>>6 & 3] * s;
921
    return dst;
922
}
923
#endif
924

    
925
#ifndef VMUL2S
926
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
927
                            unsigned sign, const float *scale)
928
{
929
    union float754 s0, s1;
930

    
931
    s0.f = s1.f = *scale;
932
    s0.i ^= sign >> 1 << 31;
933
    s1.i ^= sign      << 31;
934

    
935
    *dst++ = v[idx    & 15] * s0.f;
936
    *dst++ = v[idx>>4 & 15] * s1.f;
937

    
938
    return dst;
939
}
940
#endif
941

    
942
#ifndef VMUL4S
943
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
944
                            unsigned sign, const float *scale)
945
{
946
    unsigned nz = idx >> 12;
947
    union float754 s = { .f = *scale };
948
    union float754 t;
949

    
950
    t.i = s.i ^ (sign & 1<<31);
951
    *dst++ = v[idx    & 3] * t.f;
952

    
953
    sign <<= nz & 1; nz >>= 1;
954
    t.i = s.i ^ (sign & 1<<31);
955
    *dst++ = v[idx>>2 & 3] * t.f;
956

    
957
    sign <<= nz & 1; nz >>= 1;
958
    t.i = s.i ^ (sign & 1<<31);
959
    *dst++ = v[idx>>4 & 3] * t.f;
960

    
961
    sign <<= nz & 1; nz >>= 1;
962
    t.i = s.i ^ (sign & 1<<31);
963
    *dst++ = v[idx>>6 & 3] * t.f;
964

    
965
    return dst;
966
}
967
#endif
968

    
969
/**
970
 * Decode spectral data; reference: table 4.50.
971
 * Dequantize and scale spectral data; reference: 4.6.3.3.
972
 *
973
 * @param   coef            array of dequantized, scaled spectral data
974
 * @param   sf              array of scalefactors or intensity stereo positions
975
 * @param   pulse_present   set if pulses are present
976
 * @param   pulse           pointer to pulse data struct
977
 * @param   band_type       array of the used band type
978
 *
979
 * @return  Returns error status. 0 - OK, !0 - error
980
 */
981
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
982
                                       GetBitContext *gb, const float sf[120],
983
                                       int pulse_present, const Pulse *pulse,
984
                                       const IndividualChannelStream *ics,
985
                                       enum BandType band_type[120])
986
{
987
    int i, k, g, idx = 0;
988
    const int c = 1024 / ics->num_windows;
989
    const uint16_t *offsets = ics->swb_offset;
990
    float *coef_base = coef;
991

    
992
    for (g = 0; g < ics->num_windows; g++)
993
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
994

    
995
    for (g = 0; g < ics->num_window_groups; g++) {
996
        unsigned g_len = ics->group_len[g];
997

    
998
        for (i = 0; i < ics->max_sfb; i++, idx++) {
999
            const unsigned cbt_m1 = band_type[idx] - 1;
1000
            float *cfo = coef + offsets[i];
1001
            int off_len = offsets[i + 1] - offsets[i];
1002
            int group;
1003

    
1004
            if (cbt_m1 >= INTENSITY_BT2 - 1) {
1005
                for (group = 0; group < g_len; group++, cfo+=128) {
1006
                    memset(cfo, 0, off_len * sizeof(float));
1007
                }
1008
            } else if (cbt_m1 == NOISE_BT - 1) {
1009
                for (group = 0; group < g_len; group++, cfo+=128) {
1010
                    float scale;
1011
                    float band_energy;
1012

    
1013
                    for (k = 0; k < off_len; k++) {
1014
                        ac->random_state  = lcg_random(ac->random_state);
1015
                        cfo[k] = ac->random_state;
1016
                    }
1017

    
1018
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1019
                    scale = sf[idx] / sqrtf(band_energy);
1020
                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1021
                }
1022
            } else {
1023
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1024
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1025
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1026
                OPEN_READER(re, gb);
1027

    
1028
                switch (cbt_m1 >> 1) {
1029
                case 0:
1030
                    for (group = 0; group < g_len; group++, cfo+=128) {
1031
                        float *cf = cfo;
1032
                        int len = off_len;
1033

    
1034
                        do {
1035
                            int code;
1036
                            unsigned cb_idx;
1037

    
1038
                            UPDATE_CACHE(re, gb);
1039
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1040
                            cb_idx = cb_vector_idx[code];
1041
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1042
                        } while (len -= 4);
1043
                    }
1044
                    break;
1045

    
1046
                case 1:
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 nnz;
1054
                            unsigned cb_idx;
1055
                            uint32_t bits;
1056

    
1057
                            UPDATE_CACHE(re, gb);
1058
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1059
                            cb_idx = cb_vector_idx[code];
1060
                            nnz = cb_idx >> 8 & 15;
1061
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1062
                            LAST_SKIP_BITS(re, gb, nnz);
1063
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1064
                        } while (len -= 4);
1065
                    }
1066
                    break;
1067

    
1068
                case 2:
1069
                    for (group = 0; group < g_len; group++, cfo+=128) {
1070
                        float *cf = cfo;
1071
                        int len = off_len;
1072

    
1073
                        do {
1074
                            int code;
1075
                            unsigned cb_idx;
1076

    
1077
                            UPDATE_CACHE(re, gb);
1078
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1079
                            cb_idx = cb_vector_idx[code];
1080
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1081
                        } while (len -= 2);
1082
                    }
1083
                    break;
1084

    
1085
                case 3:
1086
                case 4:
1087
                    for (group = 0; group < g_len; group++, cfo+=128) {
1088
                        float *cf = cfo;
1089
                        int len = off_len;
1090

    
1091
                        do {
1092
                            int code;
1093
                            unsigned nnz;
1094
                            unsigned cb_idx;
1095
                            unsigned sign;
1096

    
1097
                            UPDATE_CACHE(re, gb);
1098
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1099
                            cb_idx = cb_vector_idx[code];
1100
                            nnz = cb_idx >> 8 & 15;
1101
                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1102
                            LAST_SKIP_BITS(re, gb, nnz);
1103
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1104
                        } while (len -= 2);
1105
                    }
1106
                    break;
1107

    
1108
                default:
1109
                    for (group = 0; group < g_len; group++, cfo+=128) {
1110
                        float *cf = cfo;
1111
                        uint32_t *icf = (uint32_t *) cf;
1112
                        int len = off_len;
1113

    
1114
                        do {
1115
                            int code;
1116
                            unsigned nzt, nnz;
1117
                            unsigned cb_idx;
1118
                            uint32_t bits;
1119
                            int j;
1120

    
1121
                            UPDATE_CACHE(re, gb);
1122
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1123

    
1124
                            if (!code) {
1125
                                *icf++ = 0;
1126
                                *icf++ = 0;
1127
                                continue;
1128
                            }
1129

    
1130
                            cb_idx = cb_vector_idx[code];
1131
                            nnz = cb_idx >> 12;
1132
                            nzt = cb_idx >> 8;
1133
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1134
                            LAST_SKIP_BITS(re, gb, nnz);
1135

    
1136
                            for (j = 0; j < 2; j++) {
1137
                                if (nzt & 1<<j) {
1138
                                    uint32_t b;
1139
                                    int n;
1140
                                    /* The total length of escape_sequence must be < 22 bits according
1141
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1142
                                    UPDATE_CACHE(re, gb);
1143
                                    b = GET_CACHE(re, gb);
1144
                                    b = 31 - av_log2(~b);
1145

    
1146
                                    if (b > 8) {
1147
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1148
                                        return -1;
1149
                                    }
1150

    
1151
                                    SKIP_BITS(re, gb, b + 1);
1152
                                    b += 4;
1153
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
1154
                                    LAST_SKIP_BITS(re, gb, b);
1155
                                    *icf++ = cbrt_tab[n] | (bits & 1<<31);
1156
                                    bits <<= 1;
1157
                                } else {
1158
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1159
                                    *icf++ = (bits & 1<<31) | v;
1160
                                    bits <<= !!v;
1161
                                }
1162
                                cb_idx >>= 4;
1163
                            }
1164
                        } while (len -= 2);
1165

    
1166
                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1167
                    }
1168
                }
1169

    
1170
                CLOSE_READER(re, gb);
1171
            }
1172
        }
1173
        coef += g_len << 7;
1174
    }
1175

    
1176
    if (pulse_present) {
1177
        idx = 0;
1178
        for (i = 0; i < pulse->num_pulse; i++) {
1179
            float co = coef_base[ pulse->pos[i] ];
1180
            while (offsets[idx + 1] <= pulse->pos[i])
1181
                idx++;
1182
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1183
                float ico = -pulse->amp[i];
1184
                if (co) {
1185
                    co /= sf[idx];
1186
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1187
                }
1188
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1189
            }
1190
        }
1191
    }
1192
    return 0;
1193
}
1194

    
1195
static av_always_inline float flt16_round(float pf)
1196
{
1197
    union float754 tmp;
1198
    tmp.f = pf;
1199
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1200
    return tmp.f;
1201
}
1202

    
1203
static av_always_inline float flt16_even(float pf)
1204
{
1205
    union float754 tmp;
1206
    tmp.f = pf;
1207
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1208
    return tmp.f;
1209
}
1210

    
1211
static av_always_inline float flt16_trunc(float pf)
1212
{
1213
    union float754 pun;
1214
    pun.f = pf;
1215
    pun.i &= 0xFFFF0000U;
1216
    return pun.f;
1217
}
1218

    
1219
static av_always_inline void predict(PredictorState *ps, float *coef,
1220
                                     float sf_scale, float inv_sf_scale,
1221
                    int output_enable)
1222
{
1223
    const float a     = 0.953125; // 61.0 / 64
1224
    const float alpha = 0.90625;  // 29.0 / 32
1225
    float e0, e1;
1226
    float pv;
1227
    float k1, k2;
1228
    float   r0 = ps->r0,     r1 = ps->r1;
1229
    float cor0 = ps->cor0, cor1 = ps->cor1;
1230
    float var0 = ps->var0, var1 = ps->var1;
1231

    
1232
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
1233
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1234

    
1235
    pv = flt16_round(k1 * r0 + k2 * r1);
1236
    if (output_enable)
1237
        *coef += pv * sf_scale;
1238

    
1239
    e0 = *coef * inv_sf_scale;
1240
    e1 = e0 - k1 * r0;
1241

    
1242
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
1243
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
1244
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
1245
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1246

    
1247
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1248
    ps->r0 = flt16_trunc(a * e0);
1249
}
1250

    
1251
/**
1252
 * Apply AAC-Main style frequency domain prediction.
1253
 */
1254
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1255
{
1256
    int sfb, k;
1257
    float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
1258

    
1259
    if (!sce->ics.predictor_initialized) {
1260
        reset_all_predictors(sce->predictor_state);
1261
        sce->ics.predictor_initialized = 1;
1262
    }
1263

    
1264
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1265
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1266
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1267
                predict(&sce->predictor_state[k], &sce->coeffs[k],
1268
                        sf_scale, inv_sf_scale,
1269
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1270
            }
1271
        }
1272
        if (sce->ics.predictor_reset_group)
1273
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1274
    } else
1275
        reset_all_predictors(sce->predictor_state);
1276
}
1277

    
1278
/**
1279
 * Decode an individual_channel_stream payload; reference: table 4.44.
1280
 *
1281
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1282
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1283
 *
1284
 * @return  Returns error status. 0 - OK, !0 - error
1285
 */
1286
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1287
                      GetBitContext *gb, int common_window, int scale_flag)
1288
{
1289
    Pulse pulse;
1290
    TemporalNoiseShaping    *tns = &sce->tns;
1291
    IndividualChannelStream *ics = &sce->ics;
1292
    float *out = sce->coeffs;
1293
    int global_gain, pulse_present = 0;
1294

    
1295
    /* This assignment is to silence a GCC warning about the variable being used
1296
     * uninitialized when in fact it always is.
1297
     */
1298
    pulse.num_pulse = 0;
1299

    
1300
    global_gain = get_bits(gb, 8);
1301

    
1302
    if (!common_window && !scale_flag) {
1303
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1304
            return -1;
1305
    }
1306

    
1307
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1308
        return -1;
1309
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1310
        return -1;
1311

    
1312
    pulse_present = 0;
1313
    if (!scale_flag) {
1314
        if ((pulse_present = get_bits1(gb))) {
1315
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1316
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1317
                return -1;
1318
            }
1319
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1320
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1321
                return -1;
1322
            }
1323
        }
1324
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1325
            return -1;
1326
        if (get_bits1(gb)) {
1327
            av_log_missing_feature(ac->avctx, "SSR", 1);
1328
            return -1;
1329
        }
1330
    }
1331

    
1332
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1333
        return -1;
1334

    
1335
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1336
        apply_prediction(ac, sce);
1337

    
1338
    return 0;
1339
}
1340

    
1341
/**
1342
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1343
 */
1344
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1345
{
1346
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1347
    float *ch0 = cpe->ch[0].coeffs;
1348
    float *ch1 = cpe->ch[1].coeffs;
1349
    int g, i, group, idx = 0;
1350
    const uint16_t *offsets = ics->swb_offset;
1351
    for (g = 0; g < ics->num_window_groups; g++) {
1352
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1353
            if (cpe->ms_mask[idx] &&
1354
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1355
                for (group = 0; group < ics->group_len[g]; group++) {
1356
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1357
                                              ch1 + group * 128 + offsets[i],
1358
                                              offsets[i+1] - offsets[i]);
1359
                }
1360
            }
1361
        }
1362
        ch0 += ics->group_len[g] * 128;
1363
        ch1 += ics->group_len[g] * 128;
1364
    }
1365
}
1366

    
1367
/**
1368
 * intensity stereo decoding; reference: 4.6.8.2.3
1369
 *
1370
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1371
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1372
 *                      [3] reserved for scalable AAC
1373
 */
1374
static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
1375
{
1376
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1377
    SingleChannelElement         *sce1 = &cpe->ch[1];
1378
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1379
    const uint16_t *offsets = ics->swb_offset;
1380
    int g, group, i, k, idx = 0;
1381
    int c;
1382
    float scale;
1383
    for (g = 0; g < ics->num_window_groups; g++) {
1384
        for (i = 0; i < ics->max_sfb;) {
1385
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1386
                const int bt_run_end = sce1->band_type_run_end[idx];
1387
                for (; i < bt_run_end; i++, idx++) {
1388
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1389
                    if (ms_present)
1390
                        c *= 1 - 2 * cpe->ms_mask[idx];
1391
                    scale = c * sce1->sf[idx];
1392
                    for (group = 0; group < ics->group_len[g]; group++)
1393
                        for (k = offsets[i]; k < offsets[i + 1]; k++)
1394
                            coef1[group * 128 + k] = scale * coef0[group * 128 + k];
1395
                }
1396
            } else {
1397
                int bt_run_end = sce1->band_type_run_end[idx];
1398
                idx += bt_run_end - i;
1399
                i    = bt_run_end;
1400
            }
1401
        }
1402
        coef0 += ics->group_len[g] * 128;
1403
        coef1 += ics->group_len[g] * 128;
1404
    }
1405
}
1406

    
1407
/**
1408
 * Decode a channel_pair_element; reference: table 4.4.
1409
 *
1410
 * @return  Returns error status. 0 - OK, !0 - error
1411
 */
1412
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1413
{
1414
    int i, ret, common_window, ms_present = 0;
1415

    
1416
    common_window = get_bits1(gb);
1417
    if (common_window) {
1418
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1419
            return -1;
1420
        i = cpe->ch[1].ics.use_kb_window[0];
1421
        cpe->ch[1].ics = cpe->ch[0].ics;
1422
        cpe->ch[1].ics.use_kb_window[1] = i;
1423
        ms_present = get_bits(gb, 2);
1424
        if (ms_present == 3) {
1425
            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1426
            return -1;
1427
        } else if (ms_present)
1428
            decode_mid_side_stereo(cpe, gb, ms_present);
1429
    }
1430
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1431
        return ret;
1432
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1433
        return ret;
1434

    
1435
    if (common_window) {
1436
        if (ms_present)
1437
            apply_mid_side_stereo(ac, cpe);
1438
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1439
            apply_prediction(ac, &cpe->ch[0]);
1440
            apply_prediction(ac, &cpe->ch[1]);
1441
        }
1442
    }
1443

    
1444
    apply_intensity_stereo(cpe, ms_present);
1445
    return 0;
1446
}
1447

    
1448
static const float cce_scale[] = {
1449
    1.09050773266525765921, //2^(1/8)
1450
    1.18920711500272106672, //2^(1/4)
1451
    M_SQRT2,
1452
    2,
1453
};
1454

    
1455
/**
1456
 * Decode coupling_channel_element; reference: table 4.8.
1457
 *
1458
 * @return  Returns error status. 0 - OK, !0 - error
1459
 */
1460
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1461
{
1462
    int num_gain = 0;
1463
    int c, g, sfb, ret;
1464
    int sign;
1465
    float scale;
1466
    SingleChannelElement *sce = &che->ch[0];
1467
    ChannelCoupling     *coup = &che->coup;
1468

    
1469
    coup->coupling_point = 2 * get_bits1(gb);
1470
    coup->num_coupled = get_bits(gb, 3);
1471
    for (c = 0; c <= coup->num_coupled; c++) {
1472
        num_gain++;
1473
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1474
        coup->id_select[c] = get_bits(gb, 4);
1475
        if (coup->type[c] == TYPE_CPE) {
1476
            coup->ch_select[c] = get_bits(gb, 2);
1477
            if (coup->ch_select[c] == 3)
1478
                num_gain++;
1479
        } else
1480
            coup->ch_select[c] = 2;
1481
    }
1482
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1483

    
1484
    sign  = get_bits(gb, 1);
1485
    scale = cce_scale[get_bits(gb, 2)];
1486

    
1487
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1488
        return ret;
1489

    
1490
    for (c = 0; c < num_gain; c++) {
1491
        int idx  = 0;
1492
        int cge  = 1;
1493
        int gain = 0;
1494
        float gain_cache = 1.;
1495
        if (c) {
1496
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1497
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1498
            gain_cache = powf(scale, -gain);
1499
        }
1500
        if (coup->coupling_point == AFTER_IMDCT) {
1501
            coup->gain[c][0] = gain_cache;
1502
        } else {
1503
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1504
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1505
                    if (sce->band_type[idx] != ZERO_BT) {
1506
                        if (!cge) {
1507
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1508
                            if (t) {
1509
                                int s = 1;
1510
                                t = gain += t;
1511
                                if (sign) {
1512
                                    s  -= 2 * (t & 0x1);
1513
                                    t >>= 1;
1514
                                }
1515
                                gain_cache = powf(scale, -t) * s;
1516
                            }
1517
                        }
1518
                        coup->gain[c][idx] = gain_cache;
1519
                    }
1520
                }
1521
            }
1522
        }
1523
    }
1524
    return 0;
1525
}
1526

    
1527
/**
1528
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1529
 *
1530
 * @return  Returns number of bytes consumed.
1531
 */
1532
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1533
                                         GetBitContext *gb)
1534
{
1535
    int i;
1536
    int num_excl_chan = 0;
1537

    
1538
    do {
1539
        for (i = 0; i < 7; i++)
1540
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1541
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1542

    
1543
    return num_excl_chan / 7;
1544
}
1545

    
1546
/**
1547
 * Decode dynamic range information; reference: table 4.52.
1548
 *
1549
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1550
 *
1551
 * @return  Returns number of bytes consumed.
1552
 */
1553
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1554
                                GetBitContext *gb, int cnt)
1555
{
1556
    int n             = 1;
1557
    int drc_num_bands = 1;
1558
    int i;
1559

    
1560
    /* pce_tag_present? */
1561
    if (get_bits1(gb)) {
1562
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1563
        skip_bits(gb, 4); // tag_reserved_bits
1564
        n++;
1565
    }
1566

    
1567
    /* excluded_chns_present? */
1568
    if (get_bits1(gb)) {
1569
        n += decode_drc_channel_exclusions(che_drc, gb);
1570
    }
1571

    
1572
    /* drc_bands_present? */
1573
    if (get_bits1(gb)) {
1574
        che_drc->band_incr            = get_bits(gb, 4);
1575
        che_drc->interpolation_scheme = get_bits(gb, 4);
1576
        n++;
1577
        drc_num_bands += che_drc->band_incr;
1578
        for (i = 0; i < drc_num_bands; i++) {
1579
            che_drc->band_top[i] = get_bits(gb, 8);
1580
            n++;
1581
        }
1582
    }
1583

    
1584
    /* prog_ref_level_present? */
1585
    if (get_bits1(gb)) {
1586
        che_drc->prog_ref_level = get_bits(gb, 7);
1587
        skip_bits1(gb); // prog_ref_level_reserved_bits
1588
        n++;
1589
    }
1590

    
1591
    for (i = 0; i < drc_num_bands; i++) {
1592
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1593
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1594
        n++;
1595
    }
1596

    
1597
    return n;
1598
}
1599

    
1600
/**
1601
 * Decode extension data (incomplete); reference: table 4.51.
1602
 *
1603
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1604
 *
1605
 * @return Returns number of bytes consumed
1606
 */
1607
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
1608
                                    ChannelElement *che, enum RawDataBlockType elem_type)
1609
{
1610
    int crc_flag = 0;
1611
    int res = cnt;
1612
    switch (get_bits(gb, 4)) { // extension type
1613
    case EXT_SBR_DATA_CRC:
1614
        crc_flag++;
1615
    case EXT_SBR_DATA:
1616
        if (!che) {
1617
            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
1618
            return res;
1619
        } else if (!ac->m4ac.sbr) {
1620
            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
1621
            skip_bits_long(gb, 8 * cnt - 4);
1622
            return res;
1623
        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
1624
            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
1625
            skip_bits_long(gb, 8 * cnt - 4);
1626
            return res;
1627
        } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
1628
            ac->m4ac.sbr = 1;
1629
            ac->m4ac.ps = 1;
1630
            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
1631
        } else {
1632
            ac->m4ac.sbr = 1;
1633
        }
1634
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
1635
        break;
1636
    case EXT_DYNAMIC_RANGE:
1637
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1638
        break;
1639
    case EXT_FILL:
1640
    case EXT_FILL_DATA:
1641
    case EXT_DATA_ELEMENT:
1642
    default:
1643
        skip_bits_long(gb, 8 * cnt - 4);
1644
        break;
1645
    };
1646
    return res;
1647
}
1648

    
1649
/**
1650
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1651
 *
1652
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1653
 * @param   coef    spectral coefficients
1654
 */
1655
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1656
                      IndividualChannelStream *ics, int decode)
1657
{
1658
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1659
    int w, filt, m, i;
1660
    int bottom, top, order, start, end, size, inc;
1661
    float lpc[TNS_MAX_ORDER];
1662

    
1663
    for (w = 0; w < ics->num_windows; w++) {
1664
        bottom = ics->num_swb;
1665
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1666
            top    = bottom;
1667
            bottom = FFMAX(0, top - tns->length[w][filt]);
1668
            order  = tns->order[w][filt];
1669
            if (order == 0)
1670
                continue;
1671

    
1672
            // tns_decode_coef
1673
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1674

    
1675
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1676
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1677
            if ((size = end - start) <= 0)
1678
                continue;
1679
            if (tns->direction[w][filt]) {
1680
                inc = -1;
1681
                start = end - 1;
1682
            } else {
1683
                inc = 1;
1684
            }
1685
            start += w * 128;
1686

    
1687
            // ar filter
1688
            for (m = 0; m < size; m++, start += inc)
1689
                for (i = 1; i <= FFMIN(m, order); i++)
1690
                    coef[start] -= coef[start - i * inc] * lpc[i - 1];
1691
        }
1692
    }
1693
}
1694

    
1695
/**
1696
 * Conduct IMDCT and windowing.
1697
 */
1698
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1699
{
1700
    IndividualChannelStream *ics = &sce->ics;
1701
    float *in    = sce->coeffs;
1702
    float *out   = sce->ret;
1703
    float *saved = sce->saved;
1704
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1705
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1706
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1707
    float *buf  = ac->buf_mdct;
1708
    float *temp = ac->temp;
1709
    int i;
1710

    
1711
    // imdct
1712
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1713
        for (i = 0; i < 1024; i += 128)
1714
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1715
    } else
1716
        ff_imdct_half(&ac->mdct, buf, in);
1717

    
1718
    /* window overlapping
1719
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1720
     * and long to short transitions are considered to be short to short
1721
     * transitions. This leaves just two cases (long to long and short to short)
1722
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1723
     */
1724
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1725
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1726
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
1727
    } else {
1728
        memcpy(                        out,               saved,            448 * sizeof(float));
1729

    
1730
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1731
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
1732
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
1733
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
1734
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
1735
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
1736
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1737
        } else {
1738
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
1739
            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
1740
        }
1741
    }
1742

    
1743
    // buffer update
1744
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1745
        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
1746
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
1747
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
1748
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
1749
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1750
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1751
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1752
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1753
    } else { // LONG_STOP or ONLY_LONG
1754
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1755
    }
1756
}
1757

    
1758
/**
1759
 * Apply dependent channel coupling (applied before IMDCT).
1760
 *
1761
 * @param   index   index into coupling gain array
1762
 */
1763
static void apply_dependent_coupling(AACContext *ac,
1764
                                     SingleChannelElement *target,
1765
                                     ChannelElement *cce, int index)
1766
{
1767
    IndividualChannelStream *ics = &cce->ch[0].ics;
1768
    const uint16_t *offsets = ics->swb_offset;
1769
    float *dest = target->coeffs;
1770
    const float *src = cce->ch[0].coeffs;
1771
    int g, i, group, k, idx = 0;
1772
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1773
        av_log(ac->avctx, AV_LOG_ERROR,
1774
               "Dependent coupling is not supported together with LTP\n");
1775
        return;
1776
    }
1777
    for (g = 0; g < ics->num_window_groups; g++) {
1778
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1779
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1780
                const float gain = cce->coup.gain[index][idx];
1781
                for (group = 0; group < ics->group_len[g]; group++) {
1782
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1783
                        // XXX dsputil-ize
1784
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1785
                    }
1786
                }
1787
            }
1788
        }
1789
        dest += ics->group_len[g] * 128;
1790
        src  += ics->group_len[g] * 128;
1791
    }
1792
}
1793

    
1794
/**
1795
 * Apply independent channel coupling (applied after IMDCT).
1796
 *
1797
 * @param   index   index into coupling gain array
1798
 */
1799
static void apply_independent_coupling(AACContext *ac,
1800
                                       SingleChannelElement *target,
1801
                                       ChannelElement *cce, int index)
1802
{
1803
    int i;
1804
    const float gain = cce->coup.gain[index][0];
1805
    const float *src = cce->ch[0].ret;
1806
    float *dest = target->ret;
1807
    const int len = 1024 << (ac->m4ac.sbr == 1);
1808

    
1809
    for (i = 0; i < len; i++)
1810
        dest[i] += gain * src[i];
1811
}
1812

    
1813
/**
1814
 * channel coupling transformation interface
1815
 *
1816
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1817
 */
1818
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1819
                                   enum RawDataBlockType type, int elem_id,
1820
                                   enum CouplingPoint coupling_point,
1821
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1822
{
1823
    int i, c;
1824

    
1825
    for (i = 0; i < MAX_ELEM_ID; i++) {
1826
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1827
        int index = 0;
1828

    
1829
        if (cce && cce->coup.coupling_point == coupling_point) {
1830
            ChannelCoupling *coup = &cce->coup;
1831

    
1832
            for (c = 0; c <= coup->num_coupled; c++) {
1833
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1834
                    if (coup->ch_select[c] != 1) {
1835
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1836
                        if (coup->ch_select[c] != 0)
1837
                            index++;
1838
                    }
1839
                    if (coup->ch_select[c] != 2)
1840
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1841
                } else
1842
                    index += 1 + (coup->ch_select[c] == 3);
1843
            }
1844
        }
1845
    }
1846
}
1847

    
1848
/**
1849
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1850
 */
1851
static void spectral_to_sample(AACContext *ac)
1852
{
1853
    int i, type;
1854
    for (type = 3; type >= 0; type--) {
1855
        for (i = 0; i < MAX_ELEM_ID; i++) {
1856
            ChannelElement *che = ac->che[type][i];
1857
            if (che) {
1858
                if (type <= TYPE_CPE)
1859
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1860
                if (che->ch[0].tns.present)
1861
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1862
                if (che->ch[1].tns.present)
1863
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1864
                if (type <= TYPE_CPE)
1865
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1866
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
1867
                    imdct_and_windowing(ac, &che->ch[0]);
1868
                    if (type == TYPE_CPE) {
1869
                        imdct_and_windowing(ac, &che->ch[1]);
1870
                    }
1871
                    if (ac->m4ac.sbr > 0) {
1872
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
1873
                    }
1874
                }
1875
                if (type <= TYPE_CCE)
1876
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1877
            }
1878
        }
1879
    }
1880
}
1881

    
1882
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
1883
{
1884
    int size;
1885
    AACADTSHeaderInfo hdr_info;
1886

    
1887
    size = ff_aac_parse_header(gb, &hdr_info);
1888
    if (size > 0) {
1889
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
1890
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1891
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1892
            ac->m4ac.chan_config = hdr_info.chan_config;
1893
            if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
1894
                return -7;
1895
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
1896
                return -7;
1897
        } else if (ac->output_configured != OC_LOCKED) {
1898
            ac->output_configured = OC_NONE;
1899
        }
1900
        if (ac->output_configured != OC_LOCKED) {
1901
            ac->m4ac.sbr = -1;
1902
            ac->m4ac.ps  = -1;
1903
        }
1904
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1905
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1906
        ac->m4ac.object_type     = hdr_info.object_type;
1907
        if (!ac->avctx->sample_rate)
1908
            ac->avctx->sample_rate = hdr_info.sample_rate;
1909
        if (hdr_info.num_aac_frames == 1) {
1910
            if (!hdr_info.crc_absent)
1911
                skip_bits(gb, 16);
1912
        } else {
1913
            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
1914
            return -1;
1915
        }
1916
    }
1917
    return size;
1918
}
1919

    
1920
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
1921
                                int *data_size, GetBitContext *gb)
1922
{
1923
    AACContext *ac = avctx->priv_data;
1924
    ChannelElement *che = NULL, *che_prev = NULL;
1925
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
1926
    int err, elem_id, data_size_tmp;
1927
    int samples = 0, multiplier;
1928

    
1929
    if (show_bits(gb, 12) == 0xfff) {
1930
        if (parse_adts_frame_header(ac, gb) < 0) {
1931
            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1932
            return -1;
1933
        }
1934
        if (ac->m4ac.sampling_index > 12) {
1935
            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1936
            return -1;
1937
        }
1938
    }
1939

    
1940
    ac->tags_mapped = 0;
1941
    // parse
1942
    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
1943
        elem_id = get_bits(gb, 4);
1944

    
1945
        if (elem_type < TYPE_DSE) {
1946
            if (!(che=get_che(ac, elem_type, elem_id))) {
1947
                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
1948
                       elem_type, elem_id);
1949
                return -1;
1950
            }
1951
            samples = 1024;
1952
        }
1953

    
1954
        switch (elem_type) {
1955

    
1956
        case TYPE_SCE:
1957
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
1958
            break;
1959

    
1960
        case TYPE_CPE:
1961
            err = decode_cpe(ac, gb, che);
1962
            break;
1963

    
1964
        case TYPE_CCE:
1965
            err = decode_cce(ac, gb, che);
1966
            break;
1967

    
1968
        case TYPE_LFE:
1969
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
1970
            break;
1971

    
1972
        case TYPE_DSE:
1973
            err = skip_data_stream_element(ac, gb);
1974
            break;
1975

    
1976
        case TYPE_PCE: {
1977
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1978
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1979
            if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
1980
                break;
1981
            if (ac->output_configured > OC_TRIAL_PCE)
1982
                av_log(avctx, AV_LOG_ERROR,
1983
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
1984
            else
1985
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
1986
            break;
1987
        }
1988

    
1989
        case TYPE_FIL:
1990
            if (elem_id == 15)
1991
                elem_id += get_bits(gb, 8) - 1;
1992
            if (get_bits_left(gb) < 8 * elem_id) {
1993
                    av_log(avctx, AV_LOG_ERROR, overread_err);
1994
                    return -1;
1995
            }
1996
            while (elem_id > 0)
1997
                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
1998
            err = 0; /* FIXME */
1999
            break;
2000

    
2001
        default:
2002
            err = -1; /* should not happen, but keeps compiler happy */
2003
            break;
2004
        }
2005

    
2006
        che_prev       = che;
2007
        elem_type_prev = elem_type;
2008

    
2009
        if (err)
2010
            return err;
2011

    
2012
        if (get_bits_left(gb) < 3) {
2013
            av_log(avctx, AV_LOG_ERROR, overread_err);
2014
            return -1;
2015
        }
2016
    }
2017

    
2018
    spectral_to_sample(ac);
2019

    
2020
    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
2021
    samples <<= multiplier;
2022
    if (ac->output_configured < OC_LOCKED) {
2023
        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
2024
        avctx->frame_size = samples;
2025
    }
2026

    
2027
    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
2028
    if (*data_size < data_size_tmp) {
2029
        av_log(avctx, AV_LOG_ERROR,
2030
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2031
               *data_size, data_size_tmp);
2032
        return -1;
2033
    }
2034
    *data_size = data_size_tmp;
2035

    
2036
    if (samples)
2037
        ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
2038

    
2039
    if (ac->output_configured)
2040
        ac->output_configured = OC_LOCKED;
2041

    
2042
    return 0;
2043
}
2044

    
2045
static int aac_decode_frame(AVCodecContext *avctx, void *data,
2046
                            int *data_size, AVPacket *avpkt)
2047
{
2048
    const uint8_t *buf = avpkt->data;
2049
    int buf_size = avpkt->size;
2050
    GetBitContext gb;
2051
    int buf_consumed;
2052
    int buf_offset;
2053
    int err;
2054

    
2055
    init_get_bits(&gb, buf, buf_size * 8);
2056

    
2057
    if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
2058
        return err;
2059

    
2060
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2061
    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
2062
        if (buf[buf_offset])
2063
            break;
2064

    
2065
    return buf_size > buf_offset ? buf_consumed : buf_size;
2066
}
2067

    
2068
static av_cold int aac_decode_close(AVCodecContext *avctx)
2069
{
2070
    AACContext *ac = avctx->priv_data;
2071
    int i, type;
2072

    
2073
    for (i = 0; i < MAX_ELEM_ID; i++) {
2074
        for (type = 0; type < 4; type++) {
2075
            if (ac->che[type][i])
2076
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
2077
            av_freep(&ac->che[type][i]);
2078
        }
2079
    }
2080

    
2081
    ff_mdct_end(&ac->mdct);
2082
    ff_mdct_end(&ac->mdct_small);
2083
    return 0;
2084
}
2085

    
2086

    
2087
#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
2088

    
2089
struct LATMContext {
2090
    AACContext      aac_ctx;             ///< containing AACContext
2091
    int             initialized;         ///< initilized after a valid extradata was seen
2092

    
2093
    // parser data
2094
    int             audio_mux_version_A; ///< LATM syntax version
2095
    int             frame_length_type;   ///< 0/1 variable/fixed frame length
2096
    int             frame_length;        ///< frame length for fixed frame length
2097
};
2098

    
2099
static inline uint32_t latm_get_value(GetBitContext *b)
2100
{
2101
    int length = get_bits(b, 2);
2102

    
2103
    return get_bits_long(b, (length+1)*8);
2104
}
2105

    
2106
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
2107
                                             GetBitContext *gb)
2108
{
2109
    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
2110
    MPEG4AudioConfig m4ac;
2111
    int  config_start_bit = get_bits_count(gb);
2112
    int     bits_consumed, esize;
2113

    
2114
    if (config_start_bit % 8) {
2115
        av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
2116
                               "config not byte aligned.\n", 1);
2117
        return AVERROR_INVALIDDATA;
2118
    } else {
2119
        bits_consumed =
2120
            decode_audio_specific_config(NULL, avctx, &m4ac,
2121
                                         gb->buffer + (config_start_bit / 8),
2122
                                         get_bits_left(gb) / 8);
2123

    
2124
        if (bits_consumed < 0)
2125
            return AVERROR_INVALIDDATA;
2126

    
2127
        esize = (bits_consumed+7) / 8;
2128

    
2129
        if (avctx->extradata_size <= esize) {
2130
            av_free(avctx->extradata);
2131
            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
2132
            if (!avctx->extradata)
2133
                return AVERROR(ENOMEM);
2134
        }
2135

    
2136
        avctx->extradata_size = esize;
2137
        memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
2138
        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
2139

    
2140
        skip_bits_long(gb, bits_consumed);
2141
    }
2142

    
2143
    return bits_consumed;
2144
}
2145

    
2146
static int read_stream_mux_config(struct LATMContext *latmctx,
2147
                                  GetBitContext *gb)
2148
{
2149
    int ret, audio_mux_version = get_bits(gb, 1);
2150

    
2151
    latmctx->audio_mux_version_A = 0;
2152
    if (audio_mux_version)
2153
        latmctx->audio_mux_version_A = get_bits(gb, 1);
2154

    
2155
    if (!latmctx->audio_mux_version_A) {
2156

    
2157
        if (audio_mux_version)
2158
            latm_get_value(gb);                 // taraFullness
2159

    
2160
        skip_bits(gb, 1);                       // allStreamSameTimeFraming
2161
        skip_bits(gb, 6);                       // numSubFrames
2162
        // numPrograms
2163
        if (get_bits(gb, 4)) {                  // numPrograms
2164
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2165
                                   "multiple programs are not supported\n", 1);
2166
            return AVERROR_PATCHWELCOME;
2167
        }
2168

    
2169
        // for each program (which there is only on in DVB)
2170

    
2171
        // for each layer (which there is only on in DVB)
2172
        if (get_bits(gb, 3)) {                   // numLayer
2173
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2174
                                   "multiple layers are not supported\n", 1);
2175
            return AVERROR_PATCHWELCOME;
2176
        }
2177

    
2178
        // for all but first stream: use_same_config = get_bits(gb, 1);
2179
        if (!audio_mux_version) {
2180
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2181
                return ret;
2182
        } else {
2183
            int ascLen = latm_get_value(gb);
2184
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2185
                return ret;
2186
            ascLen -= ret;
2187
            skip_bits_long(gb, ascLen);
2188
        }
2189

    
2190
        latmctx->frame_length_type = get_bits(gb, 3);
2191
        switch (latmctx->frame_length_type) {
2192
        case 0:
2193
            skip_bits(gb, 8);       // latmBufferFullness
2194
            break;
2195
        case 1:
2196
            latmctx->frame_length = get_bits(gb, 9);
2197
            break;
2198
        case 3:
2199
        case 4:
2200
        case 5:
2201
            skip_bits(gb, 6);       // CELP frame length table index
2202
            break;
2203
        case 6:
2204
        case 7:
2205
            skip_bits(gb, 1);       // HVXC frame length table index
2206
            break;
2207
        }
2208

    
2209
        if (get_bits(gb, 1)) {                  // other data
2210
            if (audio_mux_version) {
2211
                latm_get_value(gb);             // other_data_bits
2212
            } else {
2213
                int esc;
2214
                do {
2215
                    esc = get_bits(gb, 1);
2216
                    skip_bits(gb, 8);
2217
                } while (esc);
2218
            }
2219
        }
2220

    
2221
        if (get_bits(gb, 1))                     // crc present
2222
            skip_bits(gb, 8);                    // config_crc
2223
    }
2224

    
2225
    return 0;
2226
}
2227

    
2228
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
2229
{
2230
    uint8_t tmp;
2231

    
2232
    if (ctx->frame_length_type == 0) {
2233
        int mux_slot_length = 0;
2234
        do {
2235
            tmp = get_bits(gb, 8);
2236
            mux_slot_length += tmp;
2237
        } while (tmp == 255);
2238
        return mux_slot_length;
2239
    } else if (ctx->frame_length_type == 1) {
2240
        return ctx->frame_length;
2241
    } else if (ctx->frame_length_type == 3 ||
2242
               ctx->frame_length_type == 5 ||
2243
               ctx->frame_length_type == 7) {
2244
        skip_bits(gb, 2);          // mux_slot_length_coded
2245
    }
2246
    return 0;
2247
}
2248

    
2249
static int read_audio_mux_element(struct LATMContext *latmctx,
2250
                                  GetBitContext *gb)
2251
{
2252
    int err;
2253
    uint8_t use_same_mux = get_bits(gb, 1);
2254
    if (!use_same_mux) {
2255
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
2256
            return err;
2257
    } else if (!latmctx->aac_ctx.avctx->extradata) {
2258
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
2259
               "no decoder config found\n");
2260
        return AVERROR(EAGAIN);
2261
    }
2262
    if (latmctx->audio_mux_version_A == 0) {
2263
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
2264
        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
2265
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
2266
            return AVERROR_INVALIDDATA;
2267
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
2268
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2269
                   "frame length mismatch %d << %d\n",
2270
                   mux_slot_length_bytes * 8, get_bits_left(gb));
2271
            return AVERROR_INVALIDDATA;
2272
        }
2273
    }
2274
    return 0;
2275
}
2276

    
2277

    
2278
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
2279
                             AVPacket *avpkt)
2280
{
2281
    struct LATMContext *latmctx = avctx->priv_data;
2282
    int                 muxlength, err;
2283
    GetBitContext       gb;
2284

    
2285
    if (avpkt->size == 0)
2286
        return 0;
2287

    
2288
    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
2289

    
2290
    // check for LOAS sync word
2291
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
2292
        return AVERROR_INVALIDDATA;
2293

    
2294
    muxlength = get_bits(&gb, 13) + 3;
2295
    // not enough data, the parser should have sorted this
2296
    if (muxlength > avpkt->size)
2297
        return AVERROR_INVALIDDATA;
2298

    
2299
    if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
2300
        return err;
2301

    
2302
    if (!latmctx->initialized) {
2303
        if (!avctx->extradata) {
2304
            *out_size = 0;
2305
            return avpkt->size;
2306
        } else {
2307
            if ((err = aac_decode_init(avctx)) < 0)
2308
                return err;
2309
            latmctx->initialized = 1;
2310
        }
2311
    }
2312

    
2313
    if (show_bits(&gb, 12) == 0xfff) {
2314
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2315
               "ADTS header detected, probably as result of configuration "
2316
               "misparsing\n");
2317
        return AVERROR_INVALIDDATA;
2318
    }
2319

    
2320
    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
2321
        return err;
2322

    
2323
    return muxlength;
2324
}
2325

    
2326
av_cold static int latm_decode_init(AVCodecContext *avctx)
2327
{
2328
    struct LATMContext *latmctx = avctx->priv_data;
2329
    int ret;
2330

    
2331
    ret = aac_decode_init(avctx);
2332

    
2333
    if (avctx->extradata_size > 0) {
2334
        latmctx->initialized = !ret;
2335
    } else {
2336
        latmctx->initialized = 0;
2337
    }
2338

    
2339
    return ret;
2340
}
2341

    
2342

    
2343
AVCodec ff_aac_decoder = {
2344
    "aac",
2345
    AVMEDIA_TYPE_AUDIO,
2346
    CODEC_ID_AAC,
2347
    sizeof(AACContext),
2348
    aac_decode_init,
2349
    NULL,
2350
    aac_decode_close,
2351
    aac_decode_frame,
2352
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2353
    .sample_fmts = (const enum AVSampleFormat[]) {
2354
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
2355
    },
2356
    .channel_layouts = aac_channel_layout,
2357
};
2358

    
2359
/*
2360
    Note: This decoder filter is intended to decode LATM streams transferred
2361
    in MPEG transport streams which only contain one program.
2362
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
2363
*/
2364
AVCodec ff_aac_latm_decoder = {
2365
    .name = "aac_latm",
2366
    .type = CODEC_TYPE_AUDIO,
2367
    .id   = CODEC_ID_AAC_LATM,
2368
    .priv_data_size = sizeof(struct LATMContext),
2369
    .init   = latm_decode_init,
2370
    .close  = aac_decode_close,
2371
    .decode = latm_decode_frame,
2372
    .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
2373
    .sample_fmts = (const enum AVSampleFormat[]) {
2374
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
2375
    },
2376
    .channel_layouts = aac_channel_layout,
2377
};