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ffmpeg / libavcodec / aacdec.c @ 6fd00e9d

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1
/*
2
 * AAC decoder
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 * 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
13
 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * 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
30
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
31
 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
32
 */
33

    
34
/*
35
 * supported tools
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 *
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 * Support?             Name
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 * N (code in SoC repo) gain control
39
 * Y                    block switching
40
 * Y                    window shapes - standard
41
 * N                    window shapes - Low Delay
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 * Y                    filterbank - standard
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 * N (code in SoC repo) filterbank - Scalable Sample Rate
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 * Y                    Temporal Noise Shaping
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 * Y                    Long Term Prediction
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 * Y                    intensity stereo
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 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * Y                    Perceptual Noise Substitution
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 * Y                    Mid/Side stereo
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 * N                    Scalable Inverse AAC Quantization
52
 * N                    Frequency Selective Switch
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 * N                    upsampling filter
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 * Y                    quantization & coding - AAC
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 * N                    quantization & coding - TwinVQ
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 * N                    quantization & coding - BSAC
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 * N                    AAC Error Resilience tools
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 * N                    Error Resilience payload syntax
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 * N                    Error Protection tool
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 * N                    CELP
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 * N                    Silence Compression
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 * N                    HVXC
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 * N                    HVXC 4kbits/s VR
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 * N                    Structured Audio tools
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 * N                    Structured Audio Sample Bank Format
66
 * N                    MIDI
67
 * N                    Harmonic and Individual Lines plus Noise
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 * N                    Text-To-Speech Interface
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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
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           Parametric Stereo.
80
 */
81

    
82

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

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

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

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

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

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

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

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

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

    
205
/**
206
 * Configure output channel order based on the current program configuration element.
207
 *
208
 * @param   che_pos current channel position configuration
209
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
210
 *
211
 * @return  Returns error status. 0 - OK, !0 - error
212
 */
213
static av_cold int output_configure(AACContext *ac,
214
                                    enum ChannelPosition che_pos[4][MAX_ELEM_ID],
215
                                    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
216
                                    int channel_config, enum OCStatus oc_type)
217
{
218
    AVCodecContext *avctx = ac->avctx;
219
    int i, type, channels = 0, ret;
220

    
221
    if (new_che_pos != che_pos)
222
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
223

    
224
    if (channel_config) {
225
        for (i = 0; i < tags_per_config[channel_config]; i++) {
226
            if ((ret = che_configure(ac, che_pos,
227
                                     aac_channel_layout_map[channel_config - 1][i][0],
228
                                     aac_channel_layout_map[channel_config - 1][i][1],
229
                                     &channels)))
230
                return ret;
231
        }
232

    
233
        memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
234

    
235
        avctx->channel_layout = aac_channel_layout[channel_config - 1];
236
    } else {
237
        /* Allocate or free elements depending on if they are in the
238
         * current program configuration.
239
         *
240
         * Set up default 1:1 output mapping.
241
         *
242
         * For a 5.1 stream the output order will be:
243
         *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
244
         */
245

    
246
        for (i = 0; i < MAX_ELEM_ID; i++) {
247
            for (type = 0; type < 4; type++) {
248
                if ((ret = che_configure(ac, che_pos, type, i, &channels)))
249
                    return ret;
250
            }
251
        }
252

    
253
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
254

    
255
        avctx->channel_layout = 0;
256
    }
257

    
258
    avctx->channels = channels;
259

    
260
    ac->output_configured = oc_type;
261

    
262
    return 0;
263
}
264

    
265
/**
266
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
267
 *
268
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
269
 * @param sce_map mono (Single Channel Element) map
270
 * @param type speaker type/position for these channels
271
 */
272
static void decode_channel_map(enum ChannelPosition *cpe_map,
273
                               enum ChannelPosition *sce_map,
274
                               enum ChannelPosition type,
275
                               GetBitContext *gb, int n)
276
{
277
    while (n--) {
278
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
279
        map[get_bits(gb, 4)] = type;
280
    }
281
}
282

    
283
/**
284
 * Decode program configuration element; reference: table 4.2.
285
 *
286
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
287
 *
288
 * @return  Returns error status. 0 - OK, !0 - error
289
 */
290
static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
291
                      enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
292
                      GetBitContext *gb)
293
{
294
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
295
    int comment_len;
296

    
297
    skip_bits(gb, 2);  // object_type
298

    
299
    sampling_index = get_bits(gb, 4);
300
    if (m4ac->sampling_index != sampling_index)
301
        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");
302

    
303
    num_front       = get_bits(gb, 4);
304
    num_side        = get_bits(gb, 4);
305
    num_back        = get_bits(gb, 4);
306
    num_lfe         = get_bits(gb, 2);
307
    num_assoc_data  = get_bits(gb, 3);
308
    num_cc          = get_bits(gb, 4);
309

    
310
    if (get_bits1(gb))
311
        skip_bits(gb, 4); // mono_mixdown_tag
312
    if (get_bits1(gb))
313
        skip_bits(gb, 4); // stereo_mixdown_tag
314

    
315
    if (get_bits1(gb))
316
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
317

    
318
    if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
319
        av_log(avctx, AV_LOG_ERROR, overread_err);
320
        return -1;
321
    }
322
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
323
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
324
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
325
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
326

    
327
    skip_bits_long(gb, 4 * num_assoc_data);
328

    
329
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
330

    
331
    align_get_bits(gb);
332

    
333
    /* comment field, first byte is length */
334
    comment_len = get_bits(gb, 8) * 8;
335
    if (get_bits_left(gb) < comment_len) {
336
        av_log(avctx, AV_LOG_ERROR, overread_err);
337
        return -1;
338
    }
339
    skip_bits_long(gb, comment_len);
340
    return 0;
341
}
342

    
343
/**
344
 * Set up channel positions based on a default channel configuration
345
 * as specified in table 1.17.
346
 *
347
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
348
 *
349
 * @return  Returns error status. 0 - OK, !0 - error
350
 */
351
static av_cold int set_default_channel_config(AVCodecContext *avctx,
352
                                              enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
353
                                              int channel_config)
354
{
355
    if (channel_config < 1 || channel_config > 7) {
356
        av_log(avctx, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
357
               channel_config);
358
        return -1;
359
    }
360

    
361
    /* default channel configurations:
362
     *
363
     * 1ch : front center (mono)
364
     * 2ch : L + R (stereo)
365
     * 3ch : front center + L + R
366
     * 4ch : front center + L + R + back center
367
     * 5ch : front center + L + R + back stereo
368
     * 6ch : front center + L + R + back stereo + LFE
369
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
370
     */
371

    
372
    if (channel_config != 2)
373
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
374
    if (channel_config > 1)
375
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
376
    if (channel_config == 4)
377
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
378
    if (channel_config > 4)
379
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
380
        = AAC_CHANNEL_BACK;  // back stereo
381
    if (channel_config > 5)
382
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
383
    if (channel_config == 7)
384
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
385

    
386
    return 0;
387
}
388

    
389
/**
390
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
391
 *
392
 * @param   ac          pointer to AACContext, may be null
393
 * @param   avctx       pointer to AVCCodecContext, used for logging
394
 *
395
 * @return  Returns error status. 0 - OK, !0 - error
396
 */
397
static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
398
                                     GetBitContext *gb,
399
                                     MPEG4AudioConfig *m4ac,
400
                                     int channel_config)
401
{
402
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
403
    int extension_flag, ret;
404

    
405
    if (get_bits1(gb)) { // frameLengthFlag
406
        av_log_missing_feature(avctx, "960/120 MDCT window is", 1);
407
        return -1;
408
    }
409

    
410
    if (get_bits1(gb))       // dependsOnCoreCoder
411
        skip_bits(gb, 14);   // coreCoderDelay
412
    extension_flag = get_bits1(gb);
413

    
414
    if (m4ac->object_type == AOT_AAC_SCALABLE ||
415
        m4ac->object_type == AOT_ER_AAC_SCALABLE)
416
        skip_bits(gb, 3);     // layerNr
417

    
418
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
419
    if (channel_config == 0) {
420
        skip_bits(gb, 4);  // element_instance_tag
421
        if ((ret = decode_pce(avctx, m4ac, new_che_pos, gb)))
422
            return ret;
423
    } else {
424
        if ((ret = set_default_channel_config(avctx, new_che_pos, channel_config)))
425
            return ret;
426
    }
427
    if (ac && (ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
428
        return ret;
429

    
430
    if (extension_flag) {
431
        switch (m4ac->object_type) {
432
        case AOT_ER_BSAC:
433
            skip_bits(gb, 5);    // numOfSubFrame
434
            skip_bits(gb, 11);   // layer_length
435
            break;
436
        case AOT_ER_AAC_LC:
437
        case AOT_ER_AAC_LTP:
438
        case AOT_ER_AAC_SCALABLE:
439
        case AOT_ER_AAC_LD:
440
            skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
441
                                    * aacScalefactorDataResilienceFlag
442
                                    * aacSpectralDataResilienceFlag
443
                                    */
444
            break;
445
        }
446
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
447
    }
448
    return 0;
449
}
450

    
451
/**
452
 * Decode audio specific configuration; reference: table 1.13.
453
 *
454
 * @param   ac          pointer to AACContext, may be null
455
 * @param   avctx       pointer to AVCCodecContext, used for logging
456
 * @param   m4ac        pointer to MPEG4AudioConfig, used for parsing
457
 * @param   data        pointer to AVCodecContext extradata
458
 * @param   data_size   size of AVCCodecContext extradata
459
 *
460
 * @return  Returns error status or number of consumed bits. <0 - error
461
 */
462
static int decode_audio_specific_config(AACContext *ac,
463
                                        AVCodecContext *avctx,
464
                                        MPEG4AudioConfig *m4ac,
465
                                        const uint8_t *data, int data_size)
466
{
467
    GetBitContext gb;
468
    int i;
469

    
470
    av_dlog(avctx, "extradata size %d\n", avctx->extradata_size);
471
    for (i = 0; i < avctx->extradata_size; i++)
472
         av_dlog(avctx, "%02x ", avctx->extradata[i]);
473
    av_dlog(avctx, "\n");
474

    
475
    init_get_bits(&gb, data, data_size * 8);
476

    
477
    if ((i = ff_mpeg4audio_get_config(m4ac, data, data_size)) < 0)
478
        return -1;
479
    if (m4ac->sampling_index > 12) {
480
        av_log(avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", m4ac->sampling_index);
481
        return -1;
482
    }
483
    if (m4ac->sbr == 1 && m4ac->ps == -1)
484
        m4ac->ps = 1;
485

    
486
    skip_bits_long(&gb, i);
487

    
488
    switch (m4ac->object_type) {
489
    case AOT_AAC_MAIN:
490
    case AOT_AAC_LC:
491
    case AOT_AAC_LTP:
492
        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
493
            return -1;
494
        break;
495
    default:
496
        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
497
               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
498
        return -1;
499
    }
500

    
501
    av_dlog(avctx, "AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
502
            m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
503
            m4ac->sample_rate, m4ac->sbr, m4ac->ps);
504

    
505
    return get_bits_count(&gb);
506
}
507

    
508
/**
509
 * linear congruential pseudorandom number generator
510
 *
511
 * @param   previous_val    pointer to the current state of the generator
512
 *
513
 * @return  Returns a 32-bit pseudorandom integer
514
 */
515
static av_always_inline int lcg_random(int previous_val)
516
{
517
    return previous_val * 1664525 + 1013904223;
518
}
519

    
520
static av_always_inline void reset_predict_state(PredictorState *ps)
521
{
522
    ps->r0   = 0.0f;
523
    ps->r1   = 0.0f;
524
    ps->cor0 = 0.0f;
525
    ps->cor1 = 0.0f;
526
    ps->var0 = 1.0f;
527
    ps->var1 = 1.0f;
528
}
529

    
530
static void reset_all_predictors(PredictorState *ps)
531
{
532
    int i;
533
    for (i = 0; i < MAX_PREDICTORS; i++)
534
        reset_predict_state(&ps[i]);
535
}
536

    
537
static void reset_predictor_group(PredictorState *ps, int group_num)
538
{
539
    int i;
540
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
541
        reset_predict_state(&ps[i]);
542
}
543

    
544
#define AAC_INIT_VLC_STATIC(num, size) \
545
    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
546
         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
547
        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
548
        size);
549

    
550
static av_cold int aac_decode_init(AVCodecContext *avctx)
551
{
552
    AACContext *ac = avctx->priv_data;
553

    
554
    ac->avctx = avctx;
555
    ac->m4ac.sample_rate = avctx->sample_rate;
556

    
557
    if (avctx->extradata_size > 0) {
558
        if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
559
                                         avctx->extradata,
560
                                         avctx->extradata_size) < 0)
561
            return -1;
562
    }
563

    
564
    avctx->sample_fmt = avctx->request_sample_fmt == AV_SAMPLE_FMT_FLT ?
565
                        AV_SAMPLE_FMT_FLT : AV_SAMPLE_FMT_S16;
566

    
567
    AAC_INIT_VLC_STATIC( 0, 304);
568
    AAC_INIT_VLC_STATIC( 1, 270);
569
    AAC_INIT_VLC_STATIC( 2, 550);
570
    AAC_INIT_VLC_STATIC( 3, 300);
571
    AAC_INIT_VLC_STATIC( 4, 328);
572
    AAC_INIT_VLC_STATIC( 5, 294);
573
    AAC_INIT_VLC_STATIC( 6, 306);
574
    AAC_INIT_VLC_STATIC( 7, 268);
575
    AAC_INIT_VLC_STATIC( 8, 510);
576
    AAC_INIT_VLC_STATIC( 9, 366);
577
    AAC_INIT_VLC_STATIC(10, 462);
578

    
579
    ff_aac_sbr_init();
580

    
581
    dsputil_init(&ac->dsp, avctx);
582
    ff_fmt_convert_init(&ac->fmt_conv, avctx);
583

    
584
    ac->random_state = 0x1f2e3d4c;
585

    
586
    ff_aac_tableinit();
587

    
588
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
589
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
590
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
591
                    352);
592

    
593
    ff_mdct_init(&ac->mdct,       11, 1, 1.0/1024.0);
594
    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0/128.0);
595
    ff_mdct_init(&ac->mdct_ltp,   11, 0, -2.0);
596
    // window initialization
597
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
598
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
599
    ff_init_ff_sine_windows(10);
600
    ff_init_ff_sine_windows( 7);
601

    
602
    cbrt_tableinit();
603

    
604
    return 0;
605
}
606

    
607
/**
608
 * Skip data_stream_element; reference: table 4.10.
609
 */
610
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
611
{
612
    int byte_align = get_bits1(gb);
613
    int count = get_bits(gb, 8);
614
    if (count == 255)
615
        count += get_bits(gb, 8);
616
    if (byte_align)
617
        align_get_bits(gb);
618

    
619
    if (get_bits_left(gb) < 8 * count) {
620
        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
621
        return -1;
622
    }
623
    skip_bits_long(gb, 8 * count);
624
    return 0;
625
}
626

    
627
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
628
                             GetBitContext *gb)
629
{
630
    int sfb;
631
    if (get_bits1(gb)) {
632
        ics->predictor_reset_group = get_bits(gb, 5);
633
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
634
            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
635
            return -1;
636
        }
637
    }
638
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
639
        ics->prediction_used[sfb] = get_bits1(gb);
640
    }
641
    return 0;
642
}
643

    
644
/**
645
 * Decode Long Term Prediction data; reference: table 4.xx.
646
 */
647
static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
648
                       GetBitContext *gb, uint8_t max_sfb)
649
{
650
    int sfb;
651

    
652
    ltp->lag  = get_bits(gb, 11);
653
    ltp->coef = ltp_coef[get_bits(gb, 3)];
654
    for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
655
        ltp->used[sfb] = get_bits1(gb);
656
}
657

    
658
/**
659
 * Decode Individual Channel Stream info; reference: table 4.6.
660
 *
661
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
662
 */
663
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
664
                           GetBitContext *gb, int common_window)
665
{
666
    if (get_bits1(gb)) {
667
        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
668
        memset(ics, 0, sizeof(IndividualChannelStream));
669
        return -1;
670
    }
671
    ics->window_sequence[1] = ics->window_sequence[0];
672
    ics->window_sequence[0] = get_bits(gb, 2);
673
    ics->use_kb_window[1]   = ics->use_kb_window[0];
674
    ics->use_kb_window[0]   = get_bits1(gb);
675
    ics->num_window_groups  = 1;
676
    ics->group_len[0]       = 1;
677
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
678
        int i;
679
        ics->max_sfb = get_bits(gb, 4);
680
        for (i = 0; i < 7; i++) {
681
            if (get_bits1(gb)) {
682
                ics->group_len[ics->num_window_groups - 1]++;
683
            } else {
684
                ics->num_window_groups++;
685
                ics->group_len[ics->num_window_groups - 1] = 1;
686
            }
687
        }
688
        ics->num_windows       = 8;
689
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
690
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
691
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
692
        ics->predictor_present = 0;
693
    } else {
694
        ics->max_sfb               = get_bits(gb, 6);
695
        ics->num_windows           = 1;
696
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
697
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
698
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
699
        ics->predictor_present     = get_bits1(gb);
700
        ics->predictor_reset_group = 0;
701
        if (ics->predictor_present) {
702
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
703
                if (decode_prediction(ac, ics, gb)) {
704
                    memset(ics, 0, sizeof(IndividualChannelStream));
705
                    return -1;
706
                }
707
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
708
                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
709
                memset(ics, 0, sizeof(IndividualChannelStream));
710
                return -1;
711
            } else {
712
                if ((ics->ltp.present = get_bits(gb, 1)))
713
                    decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
714
            }
715
        }
716
    }
717

    
718
    if (ics->max_sfb > ics->num_swb) {
719
        av_log(ac->avctx, AV_LOG_ERROR,
720
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
721
               ics->max_sfb, ics->num_swb);
722
        memset(ics, 0, sizeof(IndividualChannelStream));
723
        return -1;
724
    }
725

    
726
    return 0;
727
}
728

    
729
/**
730
 * Decode band types (section_data payload); reference: table 4.46.
731
 *
732
 * @param   band_type           array of the used band type
733
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
734
 *
735
 * @return  Returns error status. 0 - OK, !0 - error
736
 */
737
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
738
                             int band_type_run_end[120], GetBitContext *gb,
739
                             IndividualChannelStream *ics)
740
{
741
    int g, idx = 0;
742
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
743
    for (g = 0; g < ics->num_window_groups; g++) {
744
        int k = 0;
745
        while (k < ics->max_sfb) {
746
            uint8_t sect_end = k;
747
            int sect_len_incr;
748
            int sect_band_type = get_bits(gb, 4);
749
            if (sect_band_type == 12) {
750
                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
751
                return -1;
752
            }
753
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
754
                sect_end += sect_len_incr;
755
            sect_end += sect_len_incr;
756
            if (get_bits_left(gb) < 0) {
757
                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
758
                return -1;
759
            }
760
            if (sect_end > ics->max_sfb) {
761
                av_log(ac->avctx, AV_LOG_ERROR,
762
                       "Number of bands (%d) exceeds limit (%d).\n",
763
                       sect_end, ics->max_sfb);
764
                return -1;
765
            }
766
            for (; k < sect_end; k++) {
767
                band_type        [idx]   = sect_band_type;
768
                band_type_run_end[idx++] = sect_end;
769
            }
770
        }
771
    }
772
    return 0;
773
}
774

    
775
/**
776
 * Decode scalefactors; reference: table 4.47.
777
 *
778
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
779
 * @param   band_type           array of the used band type
780
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
781
 * @param   sf                  array of scalefactors or intensity stereo positions
782
 *
783
 * @return  Returns error status. 0 - OK, !0 - error
784
 */
785
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
786
                               unsigned int global_gain,
787
                               IndividualChannelStream *ics,
788
                               enum BandType band_type[120],
789
                               int band_type_run_end[120])
790
{
791
    int g, i, idx = 0;
792
    int offset[3] = { global_gain, global_gain - 90, 0 };
793
    int clipped_offset;
794
    int noise_flag = 1;
795
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
796
    for (g = 0; g < ics->num_window_groups; g++) {
797
        for (i = 0; i < ics->max_sfb;) {
798
            int run_end = band_type_run_end[idx];
799
            if (band_type[idx] == ZERO_BT) {
800
                for (; i < run_end; i++, idx++)
801
                    sf[idx] = 0.;
802
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
803
                for (; i < run_end; i++, idx++) {
804
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
805
                    clipped_offset = av_clip(offset[2], -155, 100);
806
                    if (offset[2] != clipped_offset) {
807
                        av_log_ask_for_sample(ac->avctx, "Intensity stereo "
808
                                "position clipped (%d -> %d).\nIf you heard an "
809
                                "audible artifact, there may be a bug in the "
810
                                "decoder. ", offset[2], clipped_offset);
811
                    }
812
                    sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
813
                }
814
            } else if (band_type[idx] == NOISE_BT) {
815
                for (; i < run_end; i++, idx++) {
816
                    if (noise_flag-- > 0)
817
                        offset[1] += get_bits(gb, 9) - 256;
818
                    else
819
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
820
                    clipped_offset = av_clip(offset[1], -100, 155);
821
                    if (offset[2] != clipped_offset) {
822
                        av_log_ask_for_sample(ac->avctx, "Noise gain clipped "
823
                                "(%d -> %d).\nIf you heard an audible "
824
                                "artifact, there may be a bug in the decoder. ",
825
                                offset[1], clipped_offset);
826
                    }
827
                    sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
828
                }
829
            } else {
830
                for (; i < run_end; i++, idx++) {
831
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
832
                    if (offset[0] > 255U) {
833
                        av_log(ac->avctx, AV_LOG_ERROR,
834
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
835
                        return -1;
836
                    }
837
                    sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
838
                }
839
            }
840
        }
841
    }
842
    return 0;
843
}
844

    
845
/**
846
 * Decode pulse data; reference: table 4.7.
847
 */
848
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
849
                         const uint16_t *swb_offset, int num_swb)
850
{
851
    int i, pulse_swb;
852
    pulse->num_pulse = get_bits(gb, 2) + 1;
853
    pulse_swb        = get_bits(gb, 6);
854
    if (pulse_swb >= num_swb)
855
        return -1;
856
    pulse->pos[0]    = swb_offset[pulse_swb];
857
    pulse->pos[0]   += get_bits(gb, 5);
858
    if (pulse->pos[0] > 1023)
859
        return -1;
860
    pulse->amp[0]    = get_bits(gb, 4);
861
    for (i = 1; i < pulse->num_pulse; i++) {
862
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
863
        if (pulse->pos[i] > 1023)
864
            return -1;
865
        pulse->amp[i] = get_bits(gb, 4);
866
    }
867
    return 0;
868
}
869

    
870
/**
871
 * Decode Temporal Noise Shaping data; reference: table 4.48.
872
 *
873
 * @return  Returns error status. 0 - OK, !0 - error
874
 */
875
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
876
                      GetBitContext *gb, const IndividualChannelStream *ics)
877
{
878
    int w, filt, i, coef_len, coef_res, coef_compress;
879
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
880
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
881
    for (w = 0; w < ics->num_windows; w++) {
882
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
883
            coef_res = get_bits1(gb);
884

    
885
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
886
                int tmp2_idx;
887
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
888

    
889
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
890
                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
891
                           tns->order[w][filt], tns_max_order);
892
                    tns->order[w][filt] = 0;
893
                    return -1;
894
                }
895
                if (tns->order[w][filt]) {
896
                    tns->direction[w][filt] = get_bits1(gb);
897
                    coef_compress = get_bits1(gb);
898
                    coef_len = coef_res + 3 - coef_compress;
899
                    tmp2_idx = 2 * coef_compress + coef_res;
900

    
901
                    for (i = 0; i < tns->order[w][filt]; i++)
902
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
903
                }
904
            }
905
        }
906
    }
907
    return 0;
908
}
909

    
910
/**
911
 * Decode Mid/Side data; reference: table 4.54.
912
 *
913
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
914
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
915
 *                      [3] reserved for scalable AAC
916
 */
917
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
918
                                   int ms_present)
919
{
920
    int idx;
921
    if (ms_present == 1) {
922
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
923
            cpe->ms_mask[idx] = get_bits1(gb);
924
    } else if (ms_present == 2) {
925
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
926
    }
927
}
928

    
929
#ifndef VMUL2
930
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
931
                           const float *scale)
932
{
933
    float s = *scale;
934
    *dst++ = v[idx    & 15] * s;
935
    *dst++ = v[idx>>4 & 15] * s;
936
    return dst;
937
}
938
#endif
939

    
940
#ifndef VMUL4
941
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
942
                           const float *scale)
943
{
944
    float s = *scale;
945
    *dst++ = v[idx    & 3] * s;
946
    *dst++ = v[idx>>2 & 3] * s;
947
    *dst++ = v[idx>>4 & 3] * s;
948
    *dst++ = v[idx>>6 & 3] * s;
949
    return dst;
950
}
951
#endif
952

    
953
#ifndef VMUL2S
954
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
955
                            unsigned sign, const float *scale)
956
{
957
    union float754 s0, s1;
958

    
959
    s0.f = s1.f = *scale;
960
    s0.i ^= sign >> 1 << 31;
961
    s1.i ^= sign      << 31;
962

    
963
    *dst++ = v[idx    & 15] * s0.f;
964
    *dst++ = v[idx>>4 & 15] * s1.f;
965

    
966
    return dst;
967
}
968
#endif
969

    
970
#ifndef VMUL4S
971
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
972
                            unsigned sign, const float *scale)
973
{
974
    unsigned nz = idx >> 12;
975
    union float754 s = { .f = *scale };
976
    union float754 t;
977

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

    
981
    sign <<= nz & 1; nz >>= 1;
982
    t.i = s.i ^ (sign & 1U<<31);
983
    *dst++ = v[idx>>2 & 3] * t.f;
984

    
985
    sign <<= nz & 1; nz >>= 1;
986
    t.i = s.i ^ (sign & 1U<<31);
987
    *dst++ = v[idx>>4 & 3] * t.f;
988

    
989
    sign <<= nz & 1; nz >>= 1;
990
    t.i = s.i ^ (sign & 1U<<31);
991
    *dst++ = v[idx>>6 & 3] * t.f;
992

    
993
    return dst;
994
}
995
#endif
996

    
997
/**
998
 * Decode spectral data; reference: table 4.50.
999
 * Dequantize and scale spectral data; reference: 4.6.3.3.
1000
 *
1001
 * @param   coef            array of dequantized, scaled spectral data
1002
 * @param   sf              array of scalefactors or intensity stereo positions
1003
 * @param   pulse_present   set if pulses are present
1004
 * @param   pulse           pointer to pulse data struct
1005
 * @param   band_type       array of the used band type
1006
 *
1007
 * @return  Returns error status. 0 - OK, !0 - error
1008
 */
1009
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
1010
                                       GetBitContext *gb, const float sf[120],
1011
                                       int pulse_present, const Pulse *pulse,
1012
                                       const IndividualChannelStream *ics,
1013
                                       enum BandType band_type[120])
1014
{
1015
    int i, k, g, idx = 0;
1016
    const int c = 1024 / ics->num_windows;
1017
    const uint16_t *offsets = ics->swb_offset;
1018
    float *coef_base = coef;
1019

    
1020
    for (g = 0; g < ics->num_windows; g++)
1021
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
1022

    
1023
    for (g = 0; g < ics->num_window_groups; g++) {
1024
        unsigned g_len = ics->group_len[g];
1025

    
1026
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1027
            const unsigned cbt_m1 = band_type[idx] - 1;
1028
            float *cfo = coef + offsets[i];
1029
            int off_len = offsets[i + 1] - offsets[i];
1030
            int group;
1031

    
1032
            if (cbt_m1 >= INTENSITY_BT2 - 1) {
1033
                for (group = 0; group < g_len; group++, cfo+=128) {
1034
                    memset(cfo, 0, off_len * sizeof(float));
1035
                }
1036
            } else if (cbt_m1 == NOISE_BT - 1) {
1037
                for (group = 0; group < g_len; group++, cfo+=128) {
1038
                    float scale;
1039
                    float band_energy;
1040

    
1041
                    for (k = 0; k < off_len; k++) {
1042
                        ac->random_state  = lcg_random(ac->random_state);
1043
                        cfo[k] = ac->random_state;
1044
                    }
1045

    
1046
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1047
                    scale = sf[idx] / sqrtf(band_energy);
1048
                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1049
                }
1050
            } else {
1051
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1052
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1053
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1054
                OPEN_READER(re, gb);
1055

    
1056
                switch (cbt_m1 >> 1) {
1057
                case 0:
1058
                    for (group = 0; group < g_len; group++, cfo+=128) {
1059
                        float *cf = cfo;
1060
                        int len = off_len;
1061

    
1062
                        do {
1063
                            int code;
1064
                            unsigned cb_idx;
1065

    
1066
                            UPDATE_CACHE(re, gb);
1067
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1068
                            cb_idx = cb_vector_idx[code];
1069
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1070
                        } while (len -= 4);
1071
                    }
1072
                    break;
1073

    
1074
                case 1:
1075
                    for (group = 0; group < g_len; group++, cfo+=128) {
1076
                        float *cf = cfo;
1077
                        int len = off_len;
1078

    
1079
                        do {
1080
                            int code;
1081
                            unsigned nnz;
1082
                            unsigned cb_idx;
1083
                            uint32_t bits;
1084

    
1085
                            UPDATE_CACHE(re, gb);
1086
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1087
                            cb_idx = cb_vector_idx[code];
1088
                            nnz = cb_idx >> 8 & 15;
1089
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1090
                            LAST_SKIP_BITS(re, gb, nnz);
1091
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1092
                        } while (len -= 4);
1093
                    }
1094
                    break;
1095

    
1096
                case 2:
1097
                    for (group = 0; group < g_len; group++, cfo+=128) {
1098
                        float *cf = cfo;
1099
                        int len = off_len;
1100

    
1101
                        do {
1102
                            int code;
1103
                            unsigned cb_idx;
1104

    
1105
                            UPDATE_CACHE(re, gb);
1106
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1107
                            cb_idx = cb_vector_idx[code];
1108
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1109
                        } while (len -= 2);
1110
                    }
1111
                    break;
1112

    
1113
                case 3:
1114
                case 4:
1115
                    for (group = 0; group < g_len; group++, cfo+=128) {
1116
                        float *cf = cfo;
1117
                        int len = off_len;
1118

    
1119
                        do {
1120
                            int code;
1121
                            unsigned nnz;
1122
                            unsigned cb_idx;
1123
                            unsigned sign;
1124

    
1125
                            UPDATE_CACHE(re, gb);
1126
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1127
                            cb_idx = cb_vector_idx[code];
1128
                            nnz = cb_idx >> 8 & 15;
1129
                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1130
                            LAST_SKIP_BITS(re, gb, nnz);
1131
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1132
                        } while (len -= 2);
1133
                    }
1134
                    break;
1135

    
1136
                default:
1137
                    for (group = 0; group < g_len; group++, cfo+=128) {
1138
                        float *cf = cfo;
1139
                        uint32_t *icf = (uint32_t *) cf;
1140
                        int len = off_len;
1141

    
1142
                        do {
1143
                            int code;
1144
                            unsigned nzt, nnz;
1145
                            unsigned cb_idx;
1146
                            uint32_t bits;
1147
                            int j;
1148

    
1149
                            UPDATE_CACHE(re, gb);
1150
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1151

    
1152
                            if (!code) {
1153
                                *icf++ = 0;
1154
                                *icf++ = 0;
1155
                                continue;
1156
                            }
1157

    
1158
                            cb_idx = cb_vector_idx[code];
1159
                            nnz = cb_idx >> 12;
1160
                            nzt = cb_idx >> 8;
1161
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1162
                            LAST_SKIP_BITS(re, gb, nnz);
1163

    
1164
                            for (j = 0; j < 2; j++) {
1165
                                if (nzt & 1<<j) {
1166
                                    uint32_t b;
1167
                                    int n;
1168
                                    /* The total length of escape_sequence must be < 22 bits according
1169
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1170
                                    UPDATE_CACHE(re, gb);
1171
                                    b = GET_CACHE(re, gb);
1172
                                    b = 31 - av_log2(~b);
1173

    
1174
                                    if (b > 8) {
1175
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1176
                                        return -1;
1177
                                    }
1178

    
1179
                                    SKIP_BITS(re, gb, b + 1);
1180
                                    b += 4;
1181
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
1182
                                    LAST_SKIP_BITS(re, gb, b);
1183
                                    *icf++ = cbrt_tab[n] | (bits & 1U<<31);
1184
                                    bits <<= 1;
1185
                                } else {
1186
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1187
                                    *icf++ = (bits & 1U<<31) | v;
1188
                                    bits <<= !!v;
1189
                                }
1190
                                cb_idx >>= 4;
1191
                            }
1192
                        } while (len -= 2);
1193

    
1194
                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1195
                    }
1196
                }
1197

    
1198
                CLOSE_READER(re, gb);
1199
            }
1200
        }
1201
        coef += g_len << 7;
1202
    }
1203

    
1204
    if (pulse_present) {
1205
        idx = 0;
1206
        for (i = 0; i < pulse->num_pulse; i++) {
1207
            float co = coef_base[ pulse->pos[i] ];
1208
            while (offsets[idx + 1] <= pulse->pos[i])
1209
                idx++;
1210
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1211
                float ico = -pulse->amp[i];
1212
                if (co) {
1213
                    co /= sf[idx];
1214
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1215
                }
1216
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1217
            }
1218
        }
1219
    }
1220
    return 0;
1221
}
1222

    
1223
static av_always_inline float flt16_round(float pf)
1224
{
1225
    union float754 tmp;
1226
    tmp.f = pf;
1227
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1228
    return tmp.f;
1229
}
1230

    
1231
static av_always_inline float flt16_even(float pf)
1232
{
1233
    union float754 tmp;
1234
    tmp.f = pf;
1235
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1236
    return tmp.f;
1237
}
1238

    
1239
static av_always_inline float flt16_trunc(float pf)
1240
{
1241
    union float754 pun;
1242
    pun.f = pf;
1243
    pun.i &= 0xFFFF0000U;
1244
    return pun.f;
1245
}
1246

    
1247
static av_always_inline void predict(PredictorState *ps, float *coef,
1248
                                     int output_enable)
1249
{
1250
    const float a     = 0.953125; // 61.0 / 64
1251
    const float alpha = 0.90625;  // 29.0 / 32
1252
    float e0, e1;
1253
    float pv;
1254
    float k1, k2;
1255
    float   r0 = ps->r0,     r1 = ps->r1;
1256
    float cor0 = ps->cor0, cor1 = ps->cor1;
1257
    float var0 = ps->var0, var1 = ps->var1;
1258

    
1259
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
1260
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1261

    
1262
    pv = flt16_round(k1 * r0 + k2 * r1);
1263
    if (output_enable)
1264
        *coef += pv;
1265

    
1266
    e0 = *coef;
1267
    e1 = e0 - k1 * r0;
1268

    
1269
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
1270
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
1271
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
1272
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1273

    
1274
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1275
    ps->r0 = flt16_trunc(a * e0);
1276
}
1277

    
1278
/**
1279
 * Apply AAC-Main style frequency domain prediction.
1280
 */
1281
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1282
{
1283
    int sfb, k;
1284

    
1285
    if (!sce->ics.predictor_initialized) {
1286
        reset_all_predictors(sce->predictor_state);
1287
        sce->ics.predictor_initialized = 1;
1288
    }
1289

    
1290
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1291
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1292
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1293
                predict(&sce->predictor_state[k], &sce->coeffs[k],
1294
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1295
            }
1296
        }
1297
        if (sce->ics.predictor_reset_group)
1298
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1299
    } else
1300
        reset_all_predictors(sce->predictor_state);
1301
}
1302

    
1303
/**
1304
 * Decode an individual_channel_stream payload; reference: table 4.44.
1305
 *
1306
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1307
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1308
 *
1309
 * @return  Returns error status. 0 - OK, !0 - error
1310
 */
1311
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1312
                      GetBitContext *gb, int common_window, int scale_flag)
1313
{
1314
    Pulse pulse;
1315
    TemporalNoiseShaping    *tns = &sce->tns;
1316
    IndividualChannelStream *ics = &sce->ics;
1317
    float *out = sce->coeffs;
1318
    int global_gain, pulse_present = 0;
1319

    
1320
    /* This assignment is to silence a GCC warning about the variable being used
1321
     * uninitialized when in fact it always is.
1322
     */
1323
    pulse.num_pulse = 0;
1324

    
1325
    global_gain = get_bits(gb, 8);
1326

    
1327
    if (!common_window && !scale_flag) {
1328
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1329
            return -1;
1330
    }
1331

    
1332
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1333
        return -1;
1334
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1335
        return -1;
1336

    
1337
    pulse_present = 0;
1338
    if (!scale_flag) {
1339
        if ((pulse_present = get_bits1(gb))) {
1340
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1341
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1342
                return -1;
1343
            }
1344
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1345
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1346
                return -1;
1347
            }
1348
        }
1349
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1350
            return -1;
1351
        if (get_bits1(gb)) {
1352
            av_log_missing_feature(ac->avctx, "SSR", 1);
1353
            return -1;
1354
        }
1355
    }
1356

    
1357
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1358
        return -1;
1359

    
1360
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1361
        apply_prediction(ac, sce);
1362

    
1363
    return 0;
1364
}
1365

    
1366
/**
1367
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1368
 */
1369
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1370
{
1371
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1372
    float *ch0 = cpe->ch[0].coeffs;
1373
    float *ch1 = cpe->ch[1].coeffs;
1374
    int g, i, group, idx = 0;
1375
    const uint16_t *offsets = ics->swb_offset;
1376
    for (g = 0; g < ics->num_window_groups; g++) {
1377
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1378
            if (cpe->ms_mask[idx] &&
1379
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1380
                for (group = 0; group < ics->group_len[g]; group++) {
1381
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1382
                                              ch1 + group * 128 + offsets[i],
1383
                                              offsets[i+1] - offsets[i]);
1384
                }
1385
            }
1386
        }
1387
        ch0 += ics->group_len[g] * 128;
1388
        ch1 += ics->group_len[g] * 128;
1389
    }
1390
}
1391

    
1392
/**
1393
 * intensity stereo decoding; reference: 4.6.8.2.3
1394
 *
1395
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1396
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1397
 *                      [3] reserved for scalable AAC
1398
 */
1399
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
1400
{
1401
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1402
    SingleChannelElement         *sce1 = &cpe->ch[1];
1403
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1404
    const uint16_t *offsets = ics->swb_offset;
1405
    int g, group, i, idx = 0;
1406
    int c;
1407
    float scale;
1408
    for (g = 0; g < ics->num_window_groups; g++) {
1409
        for (i = 0; i < ics->max_sfb;) {
1410
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1411
                const int bt_run_end = sce1->band_type_run_end[idx];
1412
                for (; i < bt_run_end; i++, idx++) {
1413
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1414
                    if (ms_present)
1415
                        c *= 1 - 2 * cpe->ms_mask[idx];
1416
                    scale = c * sce1->sf[idx];
1417
                    for (group = 0; group < ics->group_len[g]; group++)
1418
                        ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
1419
                                                   coef0 + group * 128 + offsets[i],
1420
                                                   scale,
1421
                                                   offsets[i + 1] - offsets[i]);
1422
                }
1423
            } else {
1424
                int bt_run_end = sce1->band_type_run_end[idx];
1425
                idx += bt_run_end - i;
1426
                i    = bt_run_end;
1427
            }
1428
        }
1429
        coef0 += ics->group_len[g] * 128;
1430
        coef1 += ics->group_len[g] * 128;
1431
    }
1432
}
1433

    
1434
/**
1435
 * Decode a channel_pair_element; reference: table 4.4.
1436
 *
1437
 * @return  Returns error status. 0 - OK, !0 - error
1438
 */
1439
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1440
{
1441
    int i, ret, common_window, ms_present = 0;
1442

    
1443
    common_window = get_bits1(gb);
1444
    if (common_window) {
1445
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1446
            return -1;
1447
        i = cpe->ch[1].ics.use_kb_window[0];
1448
        cpe->ch[1].ics = cpe->ch[0].ics;
1449
        cpe->ch[1].ics.use_kb_window[1] = i;
1450
        if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
1451
            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
1452
                decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
1453
        ms_present = get_bits(gb, 2);
1454
        if (ms_present == 3) {
1455
            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1456
            return -1;
1457
        } else if (ms_present)
1458
            decode_mid_side_stereo(cpe, gb, ms_present);
1459
    }
1460
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1461
        return ret;
1462
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1463
        return ret;
1464

    
1465
    if (common_window) {
1466
        if (ms_present)
1467
            apply_mid_side_stereo(ac, cpe);
1468
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1469
            apply_prediction(ac, &cpe->ch[0]);
1470
            apply_prediction(ac, &cpe->ch[1]);
1471
        }
1472
    }
1473

    
1474
    apply_intensity_stereo(ac, cpe, ms_present);
1475
    return 0;
1476
}
1477

    
1478
static const float cce_scale[] = {
1479
    1.09050773266525765921, //2^(1/8)
1480
    1.18920711500272106672, //2^(1/4)
1481
    M_SQRT2,
1482
    2,
1483
};
1484

    
1485
/**
1486
 * Decode coupling_channel_element; reference: table 4.8.
1487
 *
1488
 * @return  Returns error status. 0 - OK, !0 - error
1489
 */
1490
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1491
{
1492
    int num_gain = 0;
1493
    int c, g, sfb, ret;
1494
    int sign;
1495
    float scale;
1496
    SingleChannelElement *sce = &che->ch[0];
1497
    ChannelCoupling     *coup = &che->coup;
1498

    
1499
    coup->coupling_point = 2 * get_bits1(gb);
1500
    coup->num_coupled = get_bits(gb, 3);
1501
    for (c = 0; c <= coup->num_coupled; c++) {
1502
        num_gain++;
1503
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1504
        coup->id_select[c] = get_bits(gb, 4);
1505
        if (coup->type[c] == TYPE_CPE) {
1506
            coup->ch_select[c] = get_bits(gb, 2);
1507
            if (coup->ch_select[c] == 3)
1508
                num_gain++;
1509
        } else
1510
            coup->ch_select[c] = 2;
1511
    }
1512
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1513

    
1514
    sign  = get_bits(gb, 1);
1515
    scale = cce_scale[get_bits(gb, 2)];
1516

    
1517
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1518
        return ret;
1519

    
1520
    for (c = 0; c < num_gain; c++) {
1521
        int idx  = 0;
1522
        int cge  = 1;
1523
        int gain = 0;
1524
        float gain_cache = 1.;
1525
        if (c) {
1526
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1527
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1528
            gain_cache = powf(scale, -gain);
1529
        }
1530
        if (coup->coupling_point == AFTER_IMDCT) {
1531
            coup->gain[c][0] = gain_cache;
1532
        } else {
1533
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1534
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1535
                    if (sce->band_type[idx] != ZERO_BT) {
1536
                        if (!cge) {
1537
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1538
                            if (t) {
1539
                                int s = 1;
1540
                                t = gain += t;
1541
                                if (sign) {
1542
                                    s  -= 2 * (t & 0x1);
1543
                                    t >>= 1;
1544
                                }
1545
                                gain_cache = powf(scale, -t) * s;
1546
                            }
1547
                        }
1548
                        coup->gain[c][idx] = gain_cache;
1549
                    }
1550
                }
1551
            }
1552
        }
1553
    }
1554
    return 0;
1555
}
1556

    
1557
/**
1558
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1559
 *
1560
 * @return  Returns number of bytes consumed.
1561
 */
1562
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1563
                                         GetBitContext *gb)
1564
{
1565
    int i;
1566
    int num_excl_chan = 0;
1567

    
1568
    do {
1569
        for (i = 0; i < 7; i++)
1570
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1571
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1572

    
1573
    return num_excl_chan / 7;
1574
}
1575

    
1576
/**
1577
 * Decode dynamic range information; reference: table 4.52.
1578
 *
1579
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1580
 *
1581
 * @return  Returns number of bytes consumed.
1582
 */
1583
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1584
                                GetBitContext *gb, int cnt)
1585
{
1586
    int n             = 1;
1587
    int drc_num_bands = 1;
1588
    int i;
1589

    
1590
    /* pce_tag_present? */
1591
    if (get_bits1(gb)) {
1592
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1593
        skip_bits(gb, 4); // tag_reserved_bits
1594
        n++;
1595
    }
1596

    
1597
    /* excluded_chns_present? */
1598
    if (get_bits1(gb)) {
1599
        n += decode_drc_channel_exclusions(che_drc, gb);
1600
    }
1601

    
1602
    /* drc_bands_present? */
1603
    if (get_bits1(gb)) {
1604
        che_drc->band_incr            = get_bits(gb, 4);
1605
        che_drc->interpolation_scheme = get_bits(gb, 4);
1606
        n++;
1607
        drc_num_bands += che_drc->band_incr;
1608
        for (i = 0; i < drc_num_bands; i++) {
1609
            che_drc->band_top[i] = get_bits(gb, 8);
1610
            n++;
1611
        }
1612
    }
1613

    
1614
    /* prog_ref_level_present? */
1615
    if (get_bits1(gb)) {
1616
        che_drc->prog_ref_level = get_bits(gb, 7);
1617
        skip_bits1(gb); // prog_ref_level_reserved_bits
1618
        n++;
1619
    }
1620

    
1621
    for (i = 0; i < drc_num_bands; i++) {
1622
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1623
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1624
        n++;
1625
    }
1626

    
1627
    return n;
1628
}
1629

    
1630
/**
1631
 * Decode extension data (incomplete); reference: table 4.51.
1632
 *
1633
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1634
 *
1635
 * @return Returns number of bytes consumed
1636
 */
1637
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
1638
                                    ChannelElement *che, enum RawDataBlockType elem_type)
1639
{
1640
    int crc_flag = 0;
1641
    int res = cnt;
1642
    switch (get_bits(gb, 4)) { // extension type
1643
    case EXT_SBR_DATA_CRC:
1644
        crc_flag++;
1645
    case EXT_SBR_DATA:
1646
        if (!che) {
1647
            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
1648
            return res;
1649
        } else if (!ac->m4ac.sbr) {
1650
            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
1651
            skip_bits_long(gb, 8 * cnt - 4);
1652
            return res;
1653
        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
1654
            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
1655
            skip_bits_long(gb, 8 * cnt - 4);
1656
            return res;
1657
        } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
1658
            ac->m4ac.sbr = 1;
1659
            ac->m4ac.ps = 1;
1660
            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
1661
        } else {
1662
            ac->m4ac.sbr = 1;
1663
        }
1664
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
1665
        break;
1666
    case EXT_DYNAMIC_RANGE:
1667
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1668
        break;
1669
    case EXT_FILL:
1670
    case EXT_FILL_DATA:
1671
    case EXT_DATA_ELEMENT:
1672
    default:
1673
        skip_bits_long(gb, 8 * cnt - 4);
1674
        break;
1675
    };
1676
    return res;
1677
}
1678

    
1679
/**
1680
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1681
 *
1682
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1683
 * @param   coef    spectral coefficients
1684
 */
1685
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1686
                      IndividualChannelStream *ics, int decode)
1687
{
1688
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1689
    int w, filt, m, i;
1690
    int bottom, top, order, start, end, size, inc;
1691
    float lpc[TNS_MAX_ORDER];
1692
    float tmp[TNS_MAX_ORDER];
1693

    
1694
    for (w = 0; w < ics->num_windows; w++) {
1695
        bottom = ics->num_swb;
1696
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1697
            top    = bottom;
1698
            bottom = FFMAX(0, top - tns->length[w][filt]);
1699
            order  = tns->order[w][filt];
1700
            if (order == 0)
1701
                continue;
1702

    
1703
            // tns_decode_coef
1704
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1705

    
1706
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1707
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1708
            if ((size = end - start) <= 0)
1709
                continue;
1710
            if (tns->direction[w][filt]) {
1711
                inc = -1;
1712
                start = end - 1;
1713
            } else {
1714
                inc = 1;
1715
            }
1716
            start += w * 128;
1717

    
1718
            if (decode) {
1719
                // ar filter
1720
                for (m = 0; m < size; m++, start += inc)
1721
                    for (i = 1; i <= FFMIN(m, order); i++)
1722
                        coef[start] -= coef[start - i * inc] * lpc[i - 1];
1723
            } else {
1724
                // ma filter
1725
                for (m = 0; m < size; m++, start += inc) {
1726
                    tmp[0] = coef[start];
1727
                    for (i = 1; i <= FFMIN(m, order); i++)
1728
                        coef[start] += tmp[i] * lpc[i - 1];
1729
                    for (i = order; i > 0; i--)
1730
                        tmp[i] = tmp[i - 1];
1731
                }
1732
            }
1733
        }
1734
    }
1735
}
1736

    
1737
/**
1738
 *  Apply windowing and MDCT to obtain the spectral
1739
 *  coefficient from the predicted sample by LTP.
1740
 */
1741
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
1742
                                   float *in, IndividualChannelStream *ics)
1743
{
1744
    const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1745
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1746
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1747
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1748

    
1749
    if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
1750
        ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
1751
    } else {
1752
        memset(in, 0, 448 * sizeof(float));
1753
        ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
1754
        memcpy(in + 576, in + 576, 448 * sizeof(float));
1755
    }
1756
    if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
1757
        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
1758
    } else {
1759
        memcpy(in + 1024, in + 1024, 448 * sizeof(float));
1760
        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
1761
        memset(in + 1024 + 576, 0, 448 * sizeof(float));
1762
    }
1763
    ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
1764
}
1765

    
1766
/**
1767
 * Apply the long term prediction
1768
 */
1769
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
1770
{
1771
    const LongTermPrediction *ltp = &sce->ics.ltp;
1772
    const uint16_t *offsets = sce->ics.swb_offset;
1773
    int i, sfb;
1774

    
1775
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1776
        float *predTime = sce->ret;
1777
        float *predFreq = ac->buf_mdct;
1778
        int16_t num_samples = 2048;
1779

    
1780
        if (ltp->lag < 1024)
1781
            num_samples = ltp->lag + 1024;
1782
        for (i = 0; i < num_samples; i++)
1783
            predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
1784
        memset(&predTime[i], 0, (2048 - i) * sizeof(float));
1785

    
1786
        windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
1787

    
1788
        if (sce->tns.present)
1789
            apply_tns(predFreq, &sce->tns, &sce->ics, 0);
1790

    
1791
        for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
1792
            if (ltp->used[sfb])
1793
                for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
1794
                    sce->coeffs[i] += predFreq[i];
1795
    }
1796
}
1797

    
1798
/**
1799
 * Update the LTP buffer for next frame
1800
 */
1801
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
1802
{
1803
    IndividualChannelStream *ics = &sce->ics;
1804
    float *saved     = sce->saved;
1805
    float *saved_ltp = sce->coeffs;
1806
    const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1807
    const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1808
    int i;
1809

    
1810
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1811
        memcpy(saved_ltp,       saved, 512 * sizeof(float));
1812
        memset(saved_ltp + 576, 0,     448 * sizeof(float));
1813
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
1814
        for (i = 0; i < 64; i++)
1815
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
1816
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1817
        memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
1818
        memset(saved_ltp + 576, 0,                  448 * sizeof(float));
1819
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
1820
        for (i = 0; i < 64; i++)
1821
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
1822
    } else { // LONG_STOP or ONLY_LONG
1823
        ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
1824
        for (i = 0; i < 512; i++)
1825
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
1826
    }
1827

    
1828
    memcpy(sce->ltp_state, &sce->ltp_state[1024], 1024 * sizeof(int16_t));
1829
    ac->fmt_conv.float_to_int16(&(sce->ltp_state[1024]), sce->ret,  1024);
1830
    ac->fmt_conv.float_to_int16(&(sce->ltp_state[2048]), saved_ltp, 1024);
1831
}
1832

    
1833
/**
1834
 * Conduct IMDCT and windowing.
1835
 */
1836
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1837
{
1838
    IndividualChannelStream *ics = &sce->ics;
1839
    float *in    = sce->coeffs;
1840
    float *out   = sce->ret;
1841
    float *saved = sce->saved;
1842
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1843
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1844
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1845
    float *buf  = ac->buf_mdct;
1846
    float *temp = ac->temp;
1847
    int i;
1848

    
1849
    // imdct
1850
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1851
        for (i = 0; i < 1024; i += 128)
1852
            ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
1853
    } else
1854
        ac->mdct.imdct_half(&ac->mdct, buf, in);
1855

    
1856
    /* window overlapping
1857
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1858
     * and long to short transitions are considered to be short to short
1859
     * transitions. This leaves just two cases (long to long and short to short)
1860
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1861
     */
1862
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1863
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1864
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
1865
    } else {
1866
        memcpy(                        out,               saved,            448 * sizeof(float));
1867

    
1868
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1869
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
1870
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
1871
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
1872
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
1873
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
1874
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1875
        } else {
1876
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
1877
            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
1878
        }
1879
    }
1880

    
1881
    // buffer update
1882
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1883
        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
1884
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
1885
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
1886
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
1887
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1888
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1889
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1890
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1891
    } else { // LONG_STOP or ONLY_LONG
1892
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1893
    }
1894
}
1895

    
1896
/**
1897
 * Apply dependent channel coupling (applied before IMDCT).
1898
 *
1899
 * @param   index   index into coupling gain array
1900
 */
1901
static void apply_dependent_coupling(AACContext *ac,
1902
                                     SingleChannelElement *target,
1903
                                     ChannelElement *cce, int index)
1904
{
1905
    IndividualChannelStream *ics = &cce->ch[0].ics;
1906
    const uint16_t *offsets = ics->swb_offset;
1907
    float *dest = target->coeffs;
1908
    const float *src = cce->ch[0].coeffs;
1909
    int g, i, group, k, idx = 0;
1910
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1911
        av_log(ac->avctx, AV_LOG_ERROR,
1912
               "Dependent coupling is not supported together with LTP\n");
1913
        return;
1914
    }
1915
    for (g = 0; g < ics->num_window_groups; g++) {
1916
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1917
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1918
                const float gain = cce->coup.gain[index][idx];
1919
                for (group = 0; group < ics->group_len[g]; group++) {
1920
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1921
                        // XXX dsputil-ize
1922
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1923
                    }
1924
                }
1925
            }
1926
        }
1927
        dest += ics->group_len[g] * 128;
1928
        src  += ics->group_len[g] * 128;
1929
    }
1930
}
1931

    
1932
/**
1933
 * Apply independent channel coupling (applied after IMDCT).
1934
 *
1935
 * @param   index   index into coupling gain array
1936
 */
1937
static void apply_independent_coupling(AACContext *ac,
1938
                                       SingleChannelElement *target,
1939
                                       ChannelElement *cce, int index)
1940
{
1941
    int i;
1942
    const float gain = cce->coup.gain[index][0];
1943
    const float *src = cce->ch[0].ret;
1944
    float *dest = target->ret;
1945
    const int len = 1024 << (ac->m4ac.sbr == 1);
1946

    
1947
    for (i = 0; i < len; i++)
1948
        dest[i] += gain * src[i];
1949
}
1950

    
1951
/**
1952
 * channel coupling transformation interface
1953
 *
1954
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1955
 */
1956
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1957
                                   enum RawDataBlockType type, int elem_id,
1958
                                   enum CouplingPoint coupling_point,
1959
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1960
{
1961
    int i, c;
1962

    
1963
    for (i = 0; i < MAX_ELEM_ID; i++) {
1964
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1965
        int index = 0;
1966

    
1967
        if (cce && cce->coup.coupling_point == coupling_point) {
1968
            ChannelCoupling *coup = &cce->coup;
1969

    
1970
            for (c = 0; c <= coup->num_coupled; c++) {
1971
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1972
                    if (coup->ch_select[c] != 1) {
1973
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1974
                        if (coup->ch_select[c] != 0)
1975
                            index++;
1976
                    }
1977
                    if (coup->ch_select[c] != 2)
1978
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1979
                } else
1980
                    index += 1 + (coup->ch_select[c] == 3);
1981
            }
1982
        }
1983
    }
1984
}
1985

    
1986
/**
1987
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1988
 */
1989
static void spectral_to_sample(AACContext *ac)
1990
{
1991
    int i, type;
1992
    for (type = 3; type >= 0; type--) {
1993
        for (i = 0; i < MAX_ELEM_ID; i++) {
1994
            ChannelElement *che = ac->che[type][i];
1995
            if (che) {
1996
                if (type <= TYPE_CPE)
1997
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1998
                if (ac->m4ac.object_type == AOT_AAC_LTP) {
1999
                    if (che->ch[0].ics.predictor_present) {
2000
                        if (che->ch[0].ics.ltp.present)
2001
                            apply_ltp(ac, &che->ch[0]);
2002
                        if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2003
                            apply_ltp(ac, &che->ch[1]);
2004
                    }
2005
                }
2006
                if (che->ch[0].tns.present)
2007
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2008
                if (che->ch[1].tns.present)
2009
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2010
                if (type <= TYPE_CPE)
2011
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
2012
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2013
                    imdct_and_windowing(ac, &che->ch[0]);
2014
                    if (ac->m4ac.object_type == AOT_AAC_LTP)
2015
                        update_ltp(ac, &che->ch[0]);
2016
                    if (type == TYPE_CPE) {
2017
                        imdct_and_windowing(ac, &che->ch[1]);
2018
                        if (ac->m4ac.object_type == AOT_AAC_LTP)
2019
                            update_ltp(ac, &che->ch[1]);
2020
                    }
2021
                    if (ac->m4ac.sbr > 0) {
2022
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2023
                    }
2024
                }
2025
                if (type <= TYPE_CCE)
2026
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
2027
            }
2028
        }
2029
    }
2030
}
2031

    
2032
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2033
{
2034
    int size;
2035
    AACADTSHeaderInfo hdr_info;
2036

    
2037
    size = ff_aac_parse_header(gb, &hdr_info);
2038
    if (size > 0) {
2039
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
2040
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2041
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2042
            ac->m4ac.chan_config = hdr_info.chan_config;
2043
            if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
2044
                return -7;
2045
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
2046
                return -7;
2047
        } else if (ac->output_configured != OC_LOCKED) {
2048
            ac->output_configured = OC_NONE;
2049
        }
2050
        if (ac->output_configured != OC_LOCKED) {
2051
            ac->m4ac.sbr = -1;
2052
            ac->m4ac.ps  = -1;
2053
        }
2054
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
2055
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
2056
        ac->m4ac.object_type     = hdr_info.object_type;
2057
        if (!ac->avctx->sample_rate)
2058
            ac->avctx->sample_rate = hdr_info.sample_rate;
2059
        if (hdr_info.num_aac_frames == 1) {
2060
            if (!hdr_info.crc_absent)
2061
                skip_bits(gb, 16);
2062
        } else {
2063
            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
2064
            return -1;
2065
        }
2066
    }
2067
    return size;
2068
}
2069

    
2070
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2071
                                int *data_size, GetBitContext *gb)
2072
{
2073
    AACContext *ac = avctx->priv_data;
2074
    ChannelElement *che = NULL, *che_prev = NULL;
2075
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2076
    int err, elem_id, data_size_tmp;
2077
    int samples = 0, multiplier;
2078

    
2079
    if (show_bits(gb, 12) == 0xfff) {
2080
        if (parse_adts_frame_header(ac, gb) < 0) {
2081
            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2082
            return -1;
2083
        }
2084
        if (ac->m4ac.sampling_index > 12) {
2085
            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
2086
            return -1;
2087
        }
2088
    }
2089

    
2090
    ac->tags_mapped = 0;
2091
    // parse
2092
    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2093
        elem_id = get_bits(gb, 4);
2094

    
2095
        if (elem_type < TYPE_DSE) {
2096
            if (!(che=get_che(ac, elem_type, elem_id))) {
2097
                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2098
                       elem_type, elem_id);
2099
                return -1;
2100
            }
2101
            samples = 1024;
2102
        }
2103

    
2104
        switch (elem_type) {
2105

    
2106
        case TYPE_SCE:
2107
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2108
            break;
2109

    
2110
        case TYPE_CPE:
2111
            err = decode_cpe(ac, gb, che);
2112
            break;
2113

    
2114
        case TYPE_CCE:
2115
            err = decode_cce(ac, gb, che);
2116
            break;
2117

    
2118
        case TYPE_LFE:
2119
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2120
            break;
2121

    
2122
        case TYPE_DSE:
2123
            err = skip_data_stream_element(ac, gb);
2124
            break;
2125

    
2126
        case TYPE_PCE: {
2127
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2128
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2129
            if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
2130
                break;
2131
            if (ac->output_configured > OC_TRIAL_PCE)
2132
                av_log(avctx, AV_LOG_ERROR,
2133
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2134
            else
2135
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
2136
            break;
2137
        }
2138

    
2139
        case TYPE_FIL:
2140
            if (elem_id == 15)
2141
                elem_id += get_bits(gb, 8) - 1;
2142
            if (get_bits_left(gb) < 8 * elem_id) {
2143
                    av_log(avctx, AV_LOG_ERROR, overread_err);
2144
                    return -1;
2145
            }
2146
            while (elem_id > 0)
2147
                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
2148
            err = 0; /* FIXME */
2149
            break;
2150

    
2151
        default:
2152
            err = -1; /* should not happen, but keeps compiler happy */
2153
            break;
2154
        }
2155

    
2156
        che_prev       = che;
2157
        elem_type_prev = elem_type;
2158

    
2159
        if (err)
2160
            return err;
2161

    
2162
        if (get_bits_left(gb) < 3) {
2163
            av_log(avctx, AV_LOG_ERROR, overread_err);
2164
            return -1;
2165
        }
2166
    }
2167

    
2168
    spectral_to_sample(ac);
2169

    
2170
    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
2171
    samples <<= multiplier;
2172
    if (ac->output_configured < OC_LOCKED) {
2173
        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
2174
        avctx->frame_size = samples;
2175
    }
2176

    
2177
    data_size_tmp = samples * avctx->channels;
2178
    data_size_tmp *= avctx->sample_fmt == AV_SAMPLE_FMT_FLT ? sizeof(float) : sizeof(int16_t);
2179
    if (*data_size < data_size_tmp) {
2180
        av_log(avctx, AV_LOG_ERROR,
2181
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2182
               *data_size, data_size_tmp);
2183
        return -1;
2184
    }
2185
    *data_size = data_size_tmp;
2186

    
2187
    if (samples) {
2188
        if (avctx->sample_fmt == AV_SAMPLE_FMT_FLT) {
2189
            float_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
2190
        } else
2191
            ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
2192
    }
2193

    
2194
    if (ac->output_configured)
2195
        ac->output_configured = OC_LOCKED;
2196

    
2197
    return 0;
2198
}
2199

    
2200
static int aac_decode_frame(AVCodecContext *avctx, void *data,
2201
                            int *data_size, AVPacket *avpkt)
2202
{
2203
    const uint8_t *buf = avpkt->data;
2204
    int buf_size = avpkt->size;
2205
    GetBitContext gb;
2206
    int buf_consumed;
2207
    int buf_offset;
2208
    int err;
2209

    
2210
    init_get_bits(&gb, buf, buf_size * 8);
2211

    
2212
    if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
2213
        return err;
2214

    
2215
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2216
    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
2217
        if (buf[buf_offset])
2218
            break;
2219

    
2220
    return buf_size > buf_offset ? buf_consumed : buf_size;
2221
}
2222

    
2223
static av_cold int aac_decode_close(AVCodecContext *avctx)
2224
{
2225
    AACContext *ac = avctx->priv_data;
2226
    int i, type;
2227

    
2228
    for (i = 0; i < MAX_ELEM_ID; i++) {
2229
        for (type = 0; type < 4; type++) {
2230
            if (ac->che[type][i])
2231
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
2232
            av_freep(&ac->che[type][i]);
2233
        }
2234
    }
2235

    
2236
    ff_mdct_end(&ac->mdct);
2237
    ff_mdct_end(&ac->mdct_small);
2238
    ff_mdct_end(&ac->mdct_ltp);
2239
    return 0;
2240
}
2241

    
2242

    
2243
#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
2244

    
2245
struct LATMContext {
2246
    AACContext      aac_ctx;             ///< containing AACContext
2247
    int             initialized;         ///< initilized after a valid extradata was seen
2248

    
2249
    // parser data
2250
    int             audio_mux_version_A; ///< LATM syntax version
2251
    int             frame_length_type;   ///< 0/1 variable/fixed frame length
2252
    int             frame_length;        ///< frame length for fixed frame length
2253
};
2254

    
2255
static inline uint32_t latm_get_value(GetBitContext *b)
2256
{
2257
    int length = get_bits(b, 2);
2258

    
2259
    return get_bits_long(b, (length+1)*8);
2260
}
2261

    
2262
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
2263
                                             GetBitContext *gb)
2264
{
2265
    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
2266
    MPEG4AudioConfig m4ac;
2267
    int  config_start_bit = get_bits_count(gb);
2268
    int     bits_consumed, esize;
2269

    
2270
    if (config_start_bit % 8) {
2271
        av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
2272
                               "config not byte aligned.\n", 1);
2273
        return AVERROR_INVALIDDATA;
2274
    } else {
2275
        bits_consumed =
2276
            decode_audio_specific_config(NULL, avctx, &m4ac,
2277
                                         gb->buffer + (config_start_bit / 8),
2278
                                         get_bits_left(gb) / 8);
2279

    
2280
        if (bits_consumed < 0)
2281
            return AVERROR_INVALIDDATA;
2282

    
2283
        esize = (bits_consumed+7) / 8;
2284

    
2285
        if (avctx->extradata_size <= esize) {
2286
            av_free(avctx->extradata);
2287
            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
2288
            if (!avctx->extradata)
2289
                return AVERROR(ENOMEM);
2290
        }
2291

    
2292
        avctx->extradata_size = esize;
2293
        memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
2294
        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
2295

    
2296
        skip_bits_long(gb, bits_consumed);
2297
    }
2298

    
2299
    return bits_consumed;
2300
}
2301

    
2302
static int read_stream_mux_config(struct LATMContext *latmctx,
2303
                                  GetBitContext *gb)
2304
{
2305
    int ret, audio_mux_version = get_bits(gb, 1);
2306

    
2307
    latmctx->audio_mux_version_A = 0;
2308
    if (audio_mux_version)
2309
        latmctx->audio_mux_version_A = get_bits(gb, 1);
2310

    
2311
    if (!latmctx->audio_mux_version_A) {
2312

    
2313
        if (audio_mux_version)
2314
            latm_get_value(gb);                 // taraFullness
2315

    
2316
        skip_bits(gb, 1);                       // allStreamSameTimeFraming
2317
        skip_bits(gb, 6);                       // numSubFrames
2318
        // numPrograms
2319
        if (get_bits(gb, 4)) {                  // numPrograms
2320
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2321
                                   "multiple programs are not supported\n", 1);
2322
            return AVERROR_PATCHWELCOME;
2323
        }
2324

    
2325
        // for each program (which there is only on in DVB)
2326

    
2327
        // for each layer (which there is only on in DVB)
2328
        if (get_bits(gb, 3)) {                   // numLayer
2329
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2330
                                   "multiple layers are not supported\n", 1);
2331
            return AVERROR_PATCHWELCOME;
2332
        }
2333

    
2334
        // for all but first stream: use_same_config = get_bits(gb, 1);
2335
        if (!audio_mux_version) {
2336
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2337
                return ret;
2338
        } else {
2339
            int ascLen = latm_get_value(gb);
2340
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2341
                return ret;
2342
            ascLen -= ret;
2343
            skip_bits_long(gb, ascLen);
2344
        }
2345

    
2346
        latmctx->frame_length_type = get_bits(gb, 3);
2347
        switch (latmctx->frame_length_type) {
2348
        case 0:
2349
            skip_bits(gb, 8);       // latmBufferFullness
2350
            break;
2351
        case 1:
2352
            latmctx->frame_length = get_bits(gb, 9);
2353
            break;
2354
        case 3:
2355
        case 4:
2356
        case 5:
2357
            skip_bits(gb, 6);       // CELP frame length table index
2358
            break;
2359
        case 6:
2360
        case 7:
2361
            skip_bits(gb, 1);       // HVXC frame length table index
2362
            break;
2363
        }
2364

    
2365
        if (get_bits(gb, 1)) {                  // other data
2366
            if (audio_mux_version) {
2367
                latm_get_value(gb);             // other_data_bits
2368
            } else {
2369
                int esc;
2370
                do {
2371
                    esc = get_bits(gb, 1);
2372
                    skip_bits(gb, 8);
2373
                } while (esc);
2374
            }
2375
        }
2376

    
2377
        if (get_bits(gb, 1))                     // crc present
2378
            skip_bits(gb, 8);                    // config_crc
2379
    }
2380

    
2381
    return 0;
2382
}
2383

    
2384
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
2385
{
2386
    uint8_t tmp;
2387

    
2388
    if (ctx->frame_length_type == 0) {
2389
        int mux_slot_length = 0;
2390
        do {
2391
            tmp = get_bits(gb, 8);
2392
            mux_slot_length += tmp;
2393
        } while (tmp == 255);
2394
        return mux_slot_length;
2395
    } else if (ctx->frame_length_type == 1) {
2396
        return ctx->frame_length;
2397
    } else if (ctx->frame_length_type == 3 ||
2398
               ctx->frame_length_type == 5 ||
2399
               ctx->frame_length_type == 7) {
2400
        skip_bits(gb, 2);          // mux_slot_length_coded
2401
    }
2402
    return 0;
2403
}
2404

    
2405
static int read_audio_mux_element(struct LATMContext *latmctx,
2406
                                  GetBitContext *gb)
2407
{
2408
    int err;
2409
    uint8_t use_same_mux = get_bits(gb, 1);
2410
    if (!use_same_mux) {
2411
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
2412
            return err;
2413
    } else if (!latmctx->aac_ctx.avctx->extradata) {
2414
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
2415
               "no decoder config found\n");
2416
        return AVERROR(EAGAIN);
2417
    }
2418
    if (latmctx->audio_mux_version_A == 0) {
2419
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
2420
        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
2421
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
2422
            return AVERROR_INVALIDDATA;
2423
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
2424
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2425
                   "frame length mismatch %d << %d\n",
2426
                   mux_slot_length_bytes * 8, get_bits_left(gb));
2427
            return AVERROR_INVALIDDATA;
2428
        }
2429
    }
2430
    return 0;
2431
}
2432

    
2433

    
2434
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
2435
                             AVPacket *avpkt)
2436
{
2437
    struct LATMContext *latmctx = avctx->priv_data;
2438
    int                 muxlength, err;
2439
    GetBitContext       gb;
2440

    
2441
    if (avpkt->size == 0)
2442
        return 0;
2443

    
2444
    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
2445

    
2446
    // check for LOAS sync word
2447
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
2448
        return AVERROR_INVALIDDATA;
2449

    
2450
    muxlength = get_bits(&gb, 13) + 3;
2451
    // not enough data, the parser should have sorted this
2452
    if (muxlength > avpkt->size)
2453
        return AVERROR_INVALIDDATA;
2454

    
2455
    if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
2456
        return err;
2457

    
2458
    if (!latmctx->initialized) {
2459
        if (!avctx->extradata) {
2460
            *out_size = 0;
2461
            return avpkt->size;
2462
        } else {
2463
            if ((err = aac_decode_init(avctx)) < 0)
2464
                return err;
2465
            latmctx->initialized = 1;
2466
        }
2467
    }
2468

    
2469
    if (show_bits(&gb, 12) == 0xfff) {
2470
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2471
               "ADTS header detected, probably as result of configuration "
2472
               "misparsing\n");
2473
        return AVERROR_INVALIDDATA;
2474
    }
2475

    
2476
    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
2477
        return err;
2478

    
2479
    return muxlength;
2480
}
2481

    
2482
av_cold static int latm_decode_init(AVCodecContext *avctx)
2483
{
2484
    struct LATMContext *latmctx = avctx->priv_data;
2485
    int ret;
2486

    
2487
    ret = aac_decode_init(avctx);
2488

    
2489
    if (avctx->extradata_size > 0) {
2490
        latmctx->initialized = !ret;
2491
    } else {
2492
        latmctx->initialized = 0;
2493
    }
2494

    
2495
    return ret;
2496
}
2497

    
2498

    
2499
AVCodec ff_aac_decoder = {
2500
    "aac",
2501
    AVMEDIA_TYPE_AUDIO,
2502
    CODEC_ID_AAC,
2503
    sizeof(AACContext),
2504
    aac_decode_init,
2505
    NULL,
2506
    aac_decode_close,
2507
    aac_decode_frame,
2508
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2509
    .sample_fmts = (const enum AVSampleFormat[]) {
2510
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
2511
    },
2512
    .channel_layouts = aac_channel_layout,
2513
};
2514

    
2515
/*
2516
    Note: This decoder filter is intended to decode LATM streams transferred
2517
    in MPEG transport streams which only contain one program.
2518
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
2519
*/
2520
AVCodec ff_aac_latm_decoder = {
2521
    .name = "aac_latm",
2522
    .type = AVMEDIA_TYPE_AUDIO,
2523
    .id   = CODEC_ID_AAC_LATM,
2524
    .priv_data_size = sizeof(struct LATMContext),
2525
    .init   = latm_decode_init,
2526
    .close  = aac_decode_close,
2527
    .decode = latm_decode_frame,
2528
    .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
2529
    .sample_fmts = (const enum AVSampleFormat[]) {
2530
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_FLT,AV_SAMPLE_FMT_NONE
2531
    },
2532
    .channel_layouts = aac_channel_layout,
2533
};