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1
/*
2
 * AAC decoder
3
 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
9
 * modify it under the terms of the GNU Lesser General Public
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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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 */
22

    
23
/**
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 * @file libavcodec/aac.c
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 * AAC decoder
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 * @author Oded Shimon  ( ods15 ods15 dyndns org )
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 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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 */
29

    
30
/*
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 * supported tools
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 *
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 * Support?             Name
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 * N (code in SoC repo) gain control
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 * Y                    block switching
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 * Y                    window shapes - standard
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 * N                    window shapes - Low Delay
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 * Y                    filterbank - standard
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 * N (code in SoC repo) filterbank - Scalable Sample Rate
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 * Y                    Temporal Noise Shaping
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 * N (code in SoC repo) Long Term Prediction
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 * Y                    intensity stereo
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 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * Y                    Perceptual Noise Substitution
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 * Y                    Mid/Side stereo
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 * N                    Scalable Inverse AAC Quantization
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 * N                    Frequency Selective Switch
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 * N                    upsampling filter
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 * Y                    quantization & coding - AAC
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 * N                    quantization & coding - TwinVQ
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 * N                    quantization & coding - BSAC
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 * N                    AAC Error Resilience tools
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 * N                    Error Resilience payload syntax
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 * N                    Error Protection tool
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 * N                    CELP
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 * N                    Silence Compression
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 * N                    HVXC
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 * N                    HVXC 4kbits/s VR
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 * N                    Structured Audio tools
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 * N                    Structured Audio Sample Bank Format
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 * N                    MIDI
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 * N                    Harmonic and Individual Lines plus Noise
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 * N                    Text-To-Speech Interface
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 * N (in progress)      Spectral Band Replication
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 * Y (not in this code) Layer-1
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 * Y (not in this code) Layer-2
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 * Y (not in this code) Layer-3
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 * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
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 * N (planned)          Parametric Stereo
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 * N                    Direct Stream Transfer
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 *
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 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
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 *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
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           Parametric Stereo.
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 */
77

    
78

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

    
85
#include "aac.h"
86
#include "aactab.h"
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#include "aacdectab.h"
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#include "mpeg4audio.h"
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#include "aac_parser.h"
90

    
91
#include <assert.h>
92
#include <errno.h>
93
#include <math.h>
94
#include <string.h>
95

    
96
union float754 { float f; uint32_t i; };
97

    
98
static VLC vlc_scalefactors;
99
static VLC vlc_spectral[11];
100

    
101

    
102
static ChannelElement* get_che(AACContext *ac, int type, int elem_id) {
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    static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 };
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    if (ac->tag_che_map[type][elem_id]) {
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        return ac->tag_che_map[type][elem_id];
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    }
107
    if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
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        return NULL;
109
    }
110
    switch (ac->m4ac.chan_config) {
111
        case 7:
112
            if (ac->tags_mapped == 3 && type == TYPE_CPE) {
113
                ac->tags_mapped++;
114
                return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
115
            }
116
        case 6:
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            /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
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               instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
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               encountered such a stream, transfer the LFE[0] element to SCE[1] */
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            if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
121
                ac->tags_mapped++;
122
                return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
123
            }
124
        case 5:
125
            if (ac->tags_mapped == 2 && type == TYPE_CPE) {
126
                ac->tags_mapped++;
127
                return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
128
            }
129
        case 4:
130
            if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
131
                ac->tags_mapped++;
132
                return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
133
            }
134
        case 3:
135
        case 2:
136
            if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
137
                ac->tags_mapped++;
138
                return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
139
            } else if (ac->m4ac.chan_config == 2) {
140
                return NULL;
141
            }
142
        case 1:
143
            if (!ac->tags_mapped && type == TYPE_SCE) {
144
                ac->tags_mapped++;
145
                return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
146
            }
147
        default:
148
            return NULL;
149
    }
150
}
151

    
152
/**
153
 * Configure output channel order based on the current program configuration element.
154
 *
155
 * @param   che_pos current channel position configuration
156
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
157
 *
158
 * @return  Returns error status. 0 - OK, !0 - error
159
 */
160
static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID],
161
        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) {
162
    AVCodecContext *avctx = ac->avccontext;
163
    int i, type, channels = 0;
164

    
165
    if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])))
166
        return 0; /* no change */
167

    
168
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
169

    
170
    /* Allocate or free elements depending on if they are in the
171
     * current program configuration.
172
     *
173
     * Set up default 1:1 output mapping.
174
     *
175
     * For a 5.1 stream the output order will be:
176
     *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
177
     */
178

    
179
    for(i = 0; i < MAX_ELEM_ID; i++) {
180
        for(type = 0; type < 4; type++) {
181
            if(che_pos[type][i]) {
182
                if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement))))
183
                    return AVERROR(ENOMEM);
184
                if(type != TYPE_CCE) {
185
                    ac->output_data[channels++] = ac->che[type][i]->ch[0].ret;
186
                    if(type == TYPE_CPE) {
187
                        ac->output_data[channels++] = ac->che[type][i]->ch[1].ret;
188
                    }
189
                }
190
            } else
191
                av_freep(&ac->che[type][i]);
192
        }
193
    }
194

    
195
    if (channel_config) {
196
        memset(ac->tag_che_map, 0,       4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
197
        ac->tags_mapped = 0;
198
    } else {
199
        memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
200
        ac->tags_mapped = 4*MAX_ELEM_ID;
201
    }
202

    
203
    avctx->channels = channels;
204

    
205
    return 0;
206
}
207

    
208
/**
209
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
210
 *
211
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
212
 * @param sce_map mono (Single Channel Element) map
213
 * @param type speaker type/position for these channels
214
 */
215
static void decode_channel_map(enum ChannelPosition *cpe_map,
216
        enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
217
    while(n--) {
218
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
219
        map[get_bits(gb, 4)] = type;
220
    }
221
}
222

    
223
/**
224
 * Decode program configuration element; reference: table 4.2.
225
 *
226
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
227
 *
228
 * @return  Returns error status. 0 - OK, !0 - error
229
 */
230
static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
231
        GetBitContext * gb) {
232
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
233

    
234
    skip_bits(gb, 2);  // object_type
235

    
236
    sampling_index = get_bits(gb, 4);
237
    if(sampling_index > 12) {
238
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
239
        return -1;
240
    }
241
    ac->m4ac.sampling_index = sampling_index;
242
    ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
243
    num_front       = get_bits(gb, 4);
244
    num_side        = get_bits(gb, 4);
245
    num_back        = get_bits(gb, 4);
246
    num_lfe         = get_bits(gb, 2);
247
    num_assoc_data  = get_bits(gb, 3);
248
    num_cc          = get_bits(gb, 4);
249

    
250
    if (get_bits1(gb))
251
        skip_bits(gb, 4); // mono_mixdown_tag
252
    if (get_bits1(gb))
253
        skip_bits(gb, 4); // stereo_mixdown_tag
254

    
255
    if (get_bits1(gb))
256
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
257

    
258
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
259
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
260
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
261
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
262

    
263
    skip_bits_long(gb, 4 * num_assoc_data);
264

    
265
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
266

    
267
    align_get_bits(gb);
268

    
269
    /* comment field, first byte is length */
270
    skip_bits_long(gb, 8 * get_bits(gb, 8));
271
    return 0;
272
}
273

    
274
/**
275
 * Set up channel positions based on a default channel configuration
276
 * as specified in table 1.17.
277
 *
278
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
279
 *
280
 * @return  Returns error status. 0 - OK, !0 - error
281
 */
282
static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
283
        int channel_config)
284
{
285
    if(channel_config < 1 || channel_config > 7) {
286
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
287
               channel_config);
288
        return -1;
289
    }
290

    
291
    /* default channel configurations:
292
     *
293
     * 1ch : front center (mono)
294
     * 2ch : L + R (stereo)
295
     * 3ch : front center + L + R
296
     * 4ch : front center + L + R + back center
297
     * 5ch : front center + L + R + back stereo
298
     * 6ch : front center + L + R + back stereo + LFE
299
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
300
     */
301

    
302
    if(channel_config != 2)
303
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
304
    if(channel_config > 1)
305
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
306
    if(channel_config == 4)
307
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
308
    if(channel_config > 4)
309
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
310
                                 = AAC_CHANNEL_BACK;  // back stereo
311
    if(channel_config > 5)
312
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
313
    if(channel_config == 7)
314
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
315

    
316
    return 0;
317
}
318

    
319
/**
320
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
321
 *
322
 * @return  Returns error status. 0 - OK, !0 - error
323
 */
324
static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) {
325
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
326
    int extension_flag, ret;
327

    
328
    if(get_bits1(gb)) {  // frameLengthFlag
329
        ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
330
        return -1;
331
    }
332

    
333
    if (get_bits1(gb))       // dependsOnCoreCoder
334
        skip_bits(gb, 14);   // coreCoderDelay
335
    extension_flag = get_bits1(gb);
336

    
337
    if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
338
       ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
339
        skip_bits(gb, 3);     // layerNr
340

    
341
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
342
    if (channel_config == 0) {
343
        skip_bits(gb, 4);  // element_instance_tag
344
        if((ret = decode_pce(ac, new_che_pos, gb)))
345
            return ret;
346
    } else {
347
        if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
348
            return ret;
349
    }
350
    if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config)))
351
        return ret;
352

    
353
    if (extension_flag) {
354
        switch (ac->m4ac.object_type) {
355
            case AOT_ER_BSAC:
356
                skip_bits(gb, 5);    // numOfSubFrame
357
                skip_bits(gb, 11);   // layer_length
358
                break;
359
            case AOT_ER_AAC_LC:
360
            case AOT_ER_AAC_LTP:
361
            case AOT_ER_AAC_SCALABLE:
362
            case AOT_ER_AAC_LD:
363
                skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
364
                                    * aacScalefactorDataResilienceFlag
365
                                    * aacSpectralDataResilienceFlag
366
                                    */
367
                break;
368
        }
369
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
370
    }
371
    return 0;
372
}
373

    
374
/**
375
 * Decode audio specific configuration; reference: table 1.13.
376
 *
377
 * @param   data        pointer to AVCodecContext extradata
378
 * @param   data_size   size of AVCCodecContext extradata
379
 *
380
 * @return  Returns error status. 0 - OK, !0 - error
381
 */
382
static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
383
    GetBitContext gb;
384
    int i;
385

    
386
    init_get_bits(&gb, data, data_size * 8);
387

    
388
    if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
389
        return -1;
390
    if(ac->m4ac.sampling_index > 12) {
391
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
392
        return -1;
393
    }
394

    
395
    skip_bits_long(&gb, i);
396

    
397
    switch (ac->m4ac.object_type) {
398
    case AOT_AAC_MAIN:
399
    case AOT_AAC_LC:
400
        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
401
            return -1;
402
        break;
403
    default:
404
        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
405
               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
406
        return -1;
407
    }
408
    return 0;
409
}
410

    
411
/**
412
 * linear congruential pseudorandom number generator
413
 *
414
 * @param   previous_val    pointer to the current state of the generator
415
 *
416
 * @return  Returns a 32-bit pseudorandom integer
417
 */
418
static av_always_inline int lcg_random(int previous_val) {
419
    return previous_val * 1664525 + 1013904223;
420
}
421

    
422
static void reset_predict_state(PredictorState * ps) {
423
    ps->r0 = 0.0f;
424
    ps->r1 = 0.0f;
425
    ps->cor0 = 0.0f;
426
    ps->cor1 = 0.0f;
427
    ps->var0 = 1.0f;
428
    ps->var1 = 1.0f;
429
}
430

    
431
static void reset_all_predictors(PredictorState * ps) {
432
    int i;
433
    for (i = 0; i < MAX_PREDICTORS; i++)
434
        reset_predict_state(&ps[i]);
435
}
436

    
437
static void reset_predictor_group(PredictorState * ps, int group_num) {
438
    int i;
439
    for (i = group_num-1; i < MAX_PREDICTORS; i+=30)
440
        reset_predict_state(&ps[i]);
441
}
442

    
443
static av_cold int aac_decode_init(AVCodecContext * avccontext) {
444
    AACContext * ac = avccontext->priv_data;
445
    int i;
446

    
447
    ac->avccontext = avccontext;
448

    
449
    if (avccontext->extradata_size > 0) {
450
        if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
451
            return -1;
452
        avccontext->sample_rate = ac->m4ac.sample_rate;
453
    } else if (avccontext->channels > 0) {
454
        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
455
        memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
456
        if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8)))
457
            return -1;
458
        if(output_configure(ac, ac->che_pos, new_che_pos, 1))
459
            return -1;
460
        ac->m4ac.sample_rate = avccontext->sample_rate;
461
    } else {
462
        ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0);
463
        return -1;
464
    }
465

    
466
    avccontext->sample_fmt  = SAMPLE_FMT_S16;
467
    avccontext->frame_size  = 1024;
468

    
469
    AAC_INIT_VLC_STATIC( 0, 144);
470
    AAC_INIT_VLC_STATIC( 1, 114);
471
    AAC_INIT_VLC_STATIC( 2, 188);
472
    AAC_INIT_VLC_STATIC( 3, 180);
473
    AAC_INIT_VLC_STATIC( 4, 172);
474
    AAC_INIT_VLC_STATIC( 5, 140);
475
    AAC_INIT_VLC_STATIC( 6, 168);
476
    AAC_INIT_VLC_STATIC( 7, 114);
477
    AAC_INIT_VLC_STATIC( 8, 262);
478
    AAC_INIT_VLC_STATIC( 9, 248);
479
    AAC_INIT_VLC_STATIC(10, 384);
480

    
481
    dsputil_init(&ac->dsp, avccontext);
482

    
483
    ac->random_state = 0x1f2e3d4c;
484

    
485
    // -1024 - Compensate wrong IMDCT method.
486
    // 32768 - Required to scale values to the correct range for the bias method
487
    //         for float to int16 conversion.
488

    
489
    if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
490
        ac->add_bias = 385.0f;
491
        ac->sf_scale = 1. / (-1024. * 32768.);
492
        ac->sf_offset = 0;
493
    } else {
494
        ac->add_bias = 0.0f;
495
        ac->sf_scale = 1. / -1024.;
496
        ac->sf_offset = 60;
497
    }
498

    
499
#if !CONFIG_HARDCODED_TABLES
500
    for (i = 0; i < 428; i++)
501
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
502
#endif /* CONFIG_HARDCODED_TABLES */
503

    
504
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
505
        ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
506
        ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
507
        352);
508

    
509
    ff_mdct_init(&ac->mdct, 11, 1);
510
    ff_mdct_init(&ac->mdct_small, 8, 1);
511
    // window initialization
512
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
513
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
514
    ff_sine_window_init(ff_sine_1024, 1024);
515
    ff_sine_window_init(ff_sine_128, 128);
516

    
517
    return 0;
518
}
519

    
520
/**
521
 * Skip data_stream_element; reference: table 4.10.
522
 */
523
static void skip_data_stream_element(GetBitContext * gb) {
524
    int byte_align = get_bits1(gb);
525
    int count = get_bits(gb, 8);
526
    if (count == 255)
527
        count += get_bits(gb, 8);
528
    if (byte_align)
529
        align_get_bits(gb);
530
    skip_bits_long(gb, 8 * count);
531
}
532

    
533
static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) {
534
    int sfb;
535
    if (get_bits1(gb)) {
536
        ics->predictor_reset_group = get_bits(gb, 5);
537
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
538
            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
539
            return -1;
540
        }
541
    }
542
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
543
        ics->prediction_used[sfb] = get_bits1(gb);
544
    }
545
    return 0;
546
}
547

    
548
/**
549
 * Decode Individual Channel Stream info; reference: table 4.6.
550
 *
551
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
552
 */
553
static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
554
    if (get_bits1(gb)) {
555
        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
556
        memset(ics, 0, sizeof(IndividualChannelStream));
557
        return -1;
558
    }
559
    ics->window_sequence[1] = ics->window_sequence[0];
560
    ics->window_sequence[0] = get_bits(gb, 2);
561
    ics->use_kb_window[1] = ics->use_kb_window[0];
562
    ics->use_kb_window[0] = get_bits1(gb);
563
    ics->num_window_groups = 1;
564
    ics->group_len[0] = 1;
565
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
566
        int i;
567
        ics->max_sfb = get_bits(gb, 4);
568
        for (i = 0; i < 7; i++) {
569
            if (get_bits1(gb)) {
570
                ics->group_len[ics->num_window_groups-1]++;
571
            } else {
572
                ics->num_window_groups++;
573
                ics->group_len[ics->num_window_groups-1] = 1;
574
            }
575
        }
576
        ics->num_windows   = 8;
577
        ics->swb_offset    =      swb_offset_128[ac->m4ac.sampling_index];
578
        ics->num_swb       =  ff_aac_num_swb_128[ac->m4ac.sampling_index];
579
        ics->tns_max_bands =   tns_max_bands_128[ac->m4ac.sampling_index];
580
        ics->predictor_present = 0;
581
    } else {
582
        ics->max_sfb       = get_bits(gb, 6);
583
        ics->num_windows   = 1;
584
        ics->swb_offset    =     swb_offset_1024[ac->m4ac.sampling_index];
585
        ics->num_swb       = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
586
        ics->tns_max_bands =  tns_max_bands_1024[ac->m4ac.sampling_index];
587
        ics->predictor_present = get_bits1(gb);
588
        ics->predictor_reset_group = 0;
589
        if (ics->predictor_present) {
590
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
591
                if (decode_prediction(ac, ics, gb)) {
592
                    memset(ics, 0, sizeof(IndividualChannelStream));
593
                    return -1;
594
                }
595
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
596
                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
597
                memset(ics, 0, sizeof(IndividualChannelStream));
598
                return -1;
599
            } else {
600
                ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
601
                memset(ics, 0, sizeof(IndividualChannelStream));
602
                return -1;
603
            }
604
        }
605
    }
606

    
607
    if(ics->max_sfb > ics->num_swb) {
608
        av_log(ac->avccontext, AV_LOG_ERROR,
609
            "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
610
            ics->max_sfb, ics->num_swb);
611
        memset(ics, 0, sizeof(IndividualChannelStream));
612
        return -1;
613
    }
614

    
615
    return 0;
616
}
617

    
618
/**
619
 * Decode band types (section_data payload); reference: table 4.46.
620
 *
621
 * @param   band_type           array of the used band type
622
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
623
 *
624
 * @return  Returns error status. 0 - OK, !0 - error
625
 */
626
static int decode_band_types(AACContext * ac, enum BandType band_type[120],
627
        int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
628
    int g, idx = 0;
629
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
630
    for (g = 0; g < ics->num_window_groups; g++) {
631
        int k = 0;
632
        while (k < ics->max_sfb) {
633
            uint8_t sect_len = k;
634
            int sect_len_incr;
635
            int sect_band_type = get_bits(gb, 4);
636
            if (sect_band_type == 12) {
637
                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
638
                return -1;
639
            }
640
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
641
                sect_len += sect_len_incr;
642
            sect_len += sect_len_incr;
643
            if (sect_len > ics->max_sfb) {
644
                av_log(ac->avccontext, AV_LOG_ERROR,
645
                    "Number of bands (%d) exceeds limit (%d).\n",
646
                    sect_len, ics->max_sfb);
647
                return -1;
648
            }
649
            for (; k < sect_len; k++) {
650
                band_type        [idx]   = sect_band_type;
651
                band_type_run_end[idx++] = sect_len;
652
            }
653
        }
654
    }
655
    return 0;
656
}
657

    
658
/**
659
 * Decode scalefactors; reference: table 4.47.
660
 *
661
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
662
 * @param   band_type           array of the used band type
663
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
664
 * @param   sf                  array of scalefactors or intensity stereo positions
665
 *
666
 * @return  Returns error status. 0 - OK, !0 - error
667
 */
668
static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
669
        unsigned int global_gain, IndividualChannelStream * ics,
670
        enum BandType band_type[120], int band_type_run_end[120]) {
671
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
672
    int g, i, idx = 0;
673
    int offset[3] = { global_gain, global_gain - 90, 100 };
674
    int noise_flag = 1;
675
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
676
    for (g = 0; g < ics->num_window_groups; g++) {
677
        for (i = 0; i < ics->max_sfb;) {
678
            int run_end = band_type_run_end[idx];
679
            if (band_type[idx] == ZERO_BT) {
680
                for(; i < run_end; i++, idx++)
681
                    sf[idx] = 0.;
682
            }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
683
                for(; i < run_end; i++, idx++) {
684
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
685
                    if(offset[2] > 255U) {
686
                        av_log(ac->avccontext, AV_LOG_ERROR,
687
                            "%s (%d) out of range.\n", sf_str[2], offset[2]);
688
                        return -1;
689
                    }
690
                    sf[idx]  = ff_aac_pow2sf_tab[-offset[2] + 300];
691
                }
692
            }else if(band_type[idx] == NOISE_BT) {
693
                for(; i < run_end; i++, idx++) {
694
                    if(noise_flag-- > 0)
695
                        offset[1] += get_bits(gb, 9) - 256;
696
                    else
697
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
698
                    if(offset[1] > 255U) {
699
                        av_log(ac->avccontext, AV_LOG_ERROR,
700
                            "%s (%d) out of range.\n", sf_str[1], offset[1]);
701
                        return -1;
702
                    }
703
                    sf[idx]  = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100];
704
                }
705
            }else {
706
                for(; i < run_end; i++, idx++) {
707
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
708
                    if(offset[0] > 255U) {
709
                        av_log(ac->avccontext, AV_LOG_ERROR,
710
                            "%s (%d) out of range.\n", sf_str[0], offset[0]);
711
                        return -1;
712
                    }
713
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
714
                }
715
            }
716
        }
717
    }
718
    return 0;
719
}
720

    
721
/**
722
 * Decode pulse data; reference: table 4.7.
723
 */
724
static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) {
725
    int i, pulse_swb;
726
    pulse->num_pulse = get_bits(gb, 2) + 1;
727
    pulse_swb        = get_bits(gb, 6);
728
    if (pulse_swb >= num_swb)
729
        return -1;
730
    pulse->pos[0]    = swb_offset[pulse_swb];
731
    pulse->pos[0]   += get_bits(gb, 5);
732
    if (pulse->pos[0] > 1023)
733
        return -1;
734
    pulse->amp[0]    = get_bits(gb, 4);
735
    for (i = 1; i < pulse->num_pulse; i++) {
736
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1];
737
        if (pulse->pos[i] > 1023)
738
            return -1;
739
        pulse->amp[i] = get_bits(gb, 4);
740
    }
741
    return 0;
742
}
743

    
744
/**
745
 * Decode Temporal Noise Shaping data; reference: table 4.48.
746
 *
747
 * @return  Returns error status. 0 - OK, !0 - error
748
 */
749
static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns,
750
        GetBitContext * gb, const IndividualChannelStream * ics) {
751
    int w, filt, i, coef_len, coef_res, coef_compress;
752
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
753
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
754
    for (w = 0; w < ics->num_windows; w++) {
755
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
756
            coef_res = get_bits1(gb);
757

    
758
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
759
                int tmp2_idx;
760
                tns->length[w][filt] = get_bits(gb, 6 - 2*is8);
761

    
762
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) {
763
                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
764
                           tns->order[w][filt], tns_max_order);
765
                    tns->order[w][filt] = 0;
766
                    return -1;
767
                }
768
                if (tns->order[w][filt]) {
769
                    tns->direction[w][filt] = get_bits1(gb);
770
                    coef_compress = get_bits1(gb);
771
                    coef_len = coef_res + 3 - coef_compress;
772
                    tmp2_idx = 2*coef_compress + coef_res;
773

    
774
                    for (i = 0; i < tns->order[w][filt]; i++)
775
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
776
                }
777
            }
778
        }
779
    }
780
    return 0;
781
}
782

    
783
/**
784
 * Decode Mid/Side data; reference: table 4.54.
785
 *
786
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
787
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
788
 *                      [3] reserved for scalable AAC
789
 */
790
static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
791
        int ms_present) {
792
    int idx;
793
    if (ms_present == 1) {
794
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
795
            cpe->ms_mask[idx] = get_bits1(gb);
796
    } else if (ms_present == 2) {
797
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
798
    }
799
}
800

    
801
/**
802
 * Decode spectral data; reference: table 4.50.
803
 * Dequantize and scale spectral data; reference: 4.6.3.3.
804
 *
805
 * @param   coef            array of dequantized, scaled spectral data
806
 * @param   sf              array of scalefactors or intensity stereo positions
807
 * @param   pulse_present   set if pulses are present
808
 * @param   pulse           pointer to pulse data struct
809
 * @param   band_type       array of the used band type
810
 *
811
 * @return  Returns error status. 0 - OK, !0 - error
812
 */
813
static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120],
814
        int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) {
815
    int i, k, g, idx = 0;
816
    const int c = 1024/ics->num_windows;
817
    const uint16_t * offsets = ics->swb_offset;
818
    float *coef_base = coef;
819
    static const float sign_lookup[] = { 1.0f, -1.0f };
820

    
821
    for (g = 0; g < ics->num_windows; g++)
822
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb]));
823

    
824
    for (g = 0; g < ics->num_window_groups; g++) {
825
        for (i = 0; i < ics->max_sfb; i++, idx++) {
826
            const int cur_band_type = band_type[idx];
827
            const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4;
828
            const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type);
829
            int group;
830
            if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) {
831
                for (group = 0; group < ics->group_len[g]; group++) {
832
                    memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float));
833
                }
834
            }else if (cur_band_type == NOISE_BT) {
835
                for (group = 0; group < ics->group_len[g]; group++) {
836
                    float scale;
837
                    float band_energy = 0;
838
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
839
                        ac->random_state  = lcg_random(ac->random_state);
840
                        coef[group*128+k] = ac->random_state;
841
                        band_energy += coef[group*128+k]*coef[group*128+k];
842
                    }
843
                    scale = sf[idx] / sqrtf(band_energy);
844
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
845
                        coef[group*128+k] *= scale;
846
                    }
847
                }
848
            }else {
849
                for (group = 0; group < ics->group_len[g]; group++) {
850
                    for (k = offsets[i]; k < offsets[i+1]; k += dim) {
851
                        const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3);
852
                        const int coef_tmp_idx = (group << 7) + k;
853
                        const float *vq_ptr;
854
                        int j;
855
                        if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) {
856
                            av_log(ac->avccontext, AV_LOG_ERROR,
857
                                "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
858
                                cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]);
859
                            return -1;
860
                        }
861
                        vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
862
                        if (is_cb_unsigned) {
863
                            if (vq_ptr[0]) coef[coef_tmp_idx    ] = sign_lookup[get_bits1(gb)];
864
                            if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)];
865
                            if (dim == 4) {
866
                                if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)];
867
                                if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)];
868
                            }
869
                            if (cur_band_type == ESC_BT) {
870
                                for (j = 0; j < 2; j++) {
871
                                    if (vq_ptr[j] == 64.0f) {
872
                                        int n = 4;
873
                                        /* The total length of escape_sequence must be < 22 bits according
874
                                           to the specification (i.e. max is 11111111110xxxxxxxxxx). */
875
                                        while (get_bits1(gb) && n < 15) n++;
876
                                        if(n == 15) {
877
                                            av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
878
                                            return -1;
879
                                        }
880
                                        n = (1<<n) + get_bits(gb, n);
881
                                        coef[coef_tmp_idx + j] *= cbrtf(n) * n;
882
                                    }else
883
                                        coef[coef_tmp_idx + j] *= vq_ptr[j];
884
                                }
885
                            }else
886
                            {
887
                                coef[coef_tmp_idx    ] *= vq_ptr[0];
888
                                coef[coef_tmp_idx + 1] *= vq_ptr[1];
889
                                if (dim == 4) {
890
                                    coef[coef_tmp_idx + 2] *= vq_ptr[2];
891
                                    coef[coef_tmp_idx + 3] *= vq_ptr[3];
892
                                }
893
                            }
894
                        }else {
895
                            coef[coef_tmp_idx    ] = vq_ptr[0];
896
                            coef[coef_tmp_idx + 1] = vq_ptr[1];
897
                            if (dim == 4) {
898
                                coef[coef_tmp_idx + 2] = vq_ptr[2];
899
                                coef[coef_tmp_idx + 3] = vq_ptr[3];
900
                            }
901
                        }
902
                        coef[coef_tmp_idx    ] *= sf[idx];
903
                        coef[coef_tmp_idx + 1] *= sf[idx];
904
                        if (dim == 4) {
905
                            coef[coef_tmp_idx + 2] *= sf[idx];
906
                            coef[coef_tmp_idx + 3] *= sf[idx];
907
                        }
908
                    }
909
                }
910
            }
911
        }
912
        coef += ics->group_len[g]<<7;
913
    }
914

    
915
    if (pulse_present) {
916
        idx = 0;
917
        for(i = 0; i < pulse->num_pulse; i++){
918
            float co  = coef_base[ pulse->pos[i] ];
919
            while(offsets[idx + 1] <= pulse->pos[i])
920
                idx++;
921
            if (band_type[idx] != NOISE_BT && sf[idx]) {
922
                float ico = -pulse->amp[i];
923
                if (co) {
924
                    co /= sf[idx];
925
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
926
                }
927
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
928
            }
929
        }
930
    }
931
    return 0;
932
}
933

    
934
static av_always_inline float flt16_round(float pf) {
935
    union float754 tmp;
936
    tmp.f = pf;
937
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
938
    return tmp.f;
939
}
940

    
941
static av_always_inline float flt16_even(float pf) {
942
    union float754 tmp;
943
    tmp.f = pf;
944
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U>>16)) & 0xFFFF0000U;
945
    return tmp.f;
946
}
947

    
948
static av_always_inline float flt16_trunc(float pf) {
949
    union float754 pun;
950
    pun.f = pf;
951
    pun.i &= 0xFFFF0000U;
952
    return pun.f;
953
}
954

    
955
static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) {
956
    const float a     = 0.953125; // 61.0/64
957
    const float alpha = 0.90625;  // 29.0/32
958
    float e0, e1;
959
    float pv;
960
    float k1, k2;
961

    
962
    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
963
    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
964

    
965
    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
966
    if (output_enable)
967
        *coef += pv * ac->sf_scale;
968

    
969
    e0 = *coef / ac->sf_scale;
970
    e1 = e0 - k1 * ps->r0;
971

    
972
    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
973
    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
974
    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
975
    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
976

    
977
    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
978
    ps->r0 = flt16_trunc(a * e0);
979
}
980

    
981
/**
982
 * Apply AAC-Main style frequency domain prediction.
983
 */
984
static void apply_prediction(AACContext * ac, SingleChannelElement * sce) {
985
    int sfb, k;
986

    
987
    if (!sce->ics.predictor_initialized) {
988
        reset_all_predictors(sce->predictor_state);
989
        sce->ics.predictor_initialized = 1;
990
    }
991

    
992
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
993
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
994
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
995
                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
996
                    sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
997
            }
998
        }
999
        if (sce->ics.predictor_reset_group)
1000
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1001
    } else
1002
        reset_all_predictors(sce->predictor_state);
1003
}
1004

    
1005
/**
1006
 * Decode an individual_channel_stream payload; reference: table 4.44.
1007
 *
1008
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1009
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1010
 *
1011
 * @return  Returns error status. 0 - OK, !0 - error
1012
 */
1013
static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
1014
    Pulse pulse;
1015
    TemporalNoiseShaping * tns = &sce->tns;
1016
    IndividualChannelStream * ics = &sce->ics;
1017
    float * out = sce->coeffs;
1018
    int global_gain, pulse_present = 0;
1019

    
1020
    /* This assignment is to silence a GCC warning about the variable being used
1021
     * uninitialized when in fact it always is.
1022
     */
1023
    pulse.num_pulse = 0;
1024

    
1025
    global_gain = get_bits(gb, 8);
1026

    
1027
    if (!common_window && !scale_flag) {
1028
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1029
            return -1;
1030
    }
1031

    
1032
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1033
        return -1;
1034
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1035
        return -1;
1036

    
1037
    pulse_present = 0;
1038
    if (!scale_flag) {
1039
        if ((pulse_present = get_bits1(gb))) {
1040
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1041
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1042
                return -1;
1043
            }
1044
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1045
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1046
                return -1;
1047
            }
1048
        }
1049
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1050
            return -1;
1051
        if (get_bits1(gb)) {
1052
            ff_log_missing_feature(ac->avccontext, "SSR", 1);
1053
            return -1;
1054
        }
1055
    }
1056

    
1057
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1058
        return -1;
1059

    
1060
    if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1061
        apply_prediction(ac, sce);
1062

    
1063
    return 0;
1064
}
1065

    
1066
/**
1067
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1068
 */
1069
static void apply_mid_side_stereo(ChannelElement * cpe) {
1070
    const IndividualChannelStream * ics = &cpe->ch[0].ics;
1071
    float *ch0 = cpe->ch[0].coeffs;
1072
    float *ch1 = cpe->ch[1].coeffs;
1073
    int g, i, k, group, idx = 0;
1074
    const uint16_t * offsets = ics->swb_offset;
1075
    for (g = 0; g < ics->num_window_groups; g++) {
1076
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1077
            if (cpe->ms_mask[idx] &&
1078
                cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1079
                for (group = 0; group < ics->group_len[g]; group++) {
1080
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
1081
                        float tmp = ch0[group*128 + k] - ch1[group*128 + k];
1082
                        ch0[group*128 + k] += ch1[group*128 + k];
1083
                        ch1[group*128 + k] = tmp;
1084
                    }
1085
                }
1086
            }
1087
        }
1088
        ch0 += ics->group_len[g]*128;
1089
        ch1 += ics->group_len[g]*128;
1090
    }
1091
}
1092

    
1093
/**
1094
 * intensity stereo decoding; reference: 4.6.8.2.3
1095
 *
1096
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1097
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1098
 *                      [3] reserved for scalable AAC
1099
 */
1100
static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) {
1101
    const IndividualChannelStream * ics = &cpe->ch[1].ics;
1102
    SingleChannelElement * sce1 = &cpe->ch[1];
1103
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1104
    const uint16_t * offsets = ics->swb_offset;
1105
    int g, group, i, k, idx = 0;
1106
    int c;
1107
    float scale;
1108
    for (g = 0; g < ics->num_window_groups; g++) {
1109
        for (i = 0; i < ics->max_sfb;) {
1110
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1111
                const int bt_run_end = sce1->band_type_run_end[idx];
1112
                for (; i < bt_run_end; i++, idx++) {
1113
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1114
                    if (ms_present)
1115
                        c *= 1 - 2 * cpe->ms_mask[idx];
1116
                    scale = c * sce1->sf[idx];
1117
                    for (group = 0; group < ics->group_len[g]; group++)
1118
                        for (k = offsets[i]; k < offsets[i+1]; k++)
1119
                            coef1[group*128 + k] = scale * coef0[group*128 + k];
1120
                }
1121
            } else {
1122
                int bt_run_end = sce1->band_type_run_end[idx];
1123
                idx += bt_run_end - i;
1124
                i    = bt_run_end;
1125
            }
1126
        }
1127
        coef0 += ics->group_len[g]*128;
1128
        coef1 += ics->group_len[g]*128;
1129
    }
1130
}
1131

    
1132
/**
1133
 * Decode a channel_pair_element; reference: table 4.4.
1134
 *
1135
 * @param   elem_id Identifies the instance of a syntax element.
1136
 *
1137
 * @return  Returns error status. 0 - OK, !0 - error
1138
 */
1139
static int decode_cpe(AACContext * ac, GetBitContext * gb, ChannelElement * cpe) {
1140
    int i, ret, common_window, ms_present = 0;
1141

    
1142
    common_window = get_bits1(gb);
1143
    if (common_window) {
1144
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1145
            return -1;
1146
        i = cpe->ch[1].ics.use_kb_window[0];
1147
        cpe->ch[1].ics = cpe->ch[0].ics;
1148
        cpe->ch[1].ics.use_kb_window[1] = i;
1149
        ms_present = get_bits(gb, 2);
1150
        if(ms_present == 3) {
1151
            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1152
            return -1;
1153
        } else if(ms_present)
1154
            decode_mid_side_stereo(cpe, gb, ms_present);
1155
    }
1156
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1157
        return ret;
1158
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1159
        return ret;
1160

    
1161
    if (common_window) {
1162
        if (ms_present)
1163
            apply_mid_side_stereo(cpe);
1164
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1165
            apply_prediction(ac, &cpe->ch[0]);
1166
            apply_prediction(ac, &cpe->ch[1]);
1167
        }
1168
    }
1169

    
1170
    apply_intensity_stereo(cpe, ms_present);
1171
    return 0;
1172
}
1173

    
1174
/**
1175
 * Decode coupling_channel_element; reference: table 4.8.
1176
 *
1177
 * @param   elem_id Identifies the instance of a syntax element.
1178
 *
1179
 * @return  Returns error status. 0 - OK, !0 - error
1180
 */
1181
static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) {
1182
    int num_gain = 0;
1183
    int c, g, sfb, ret;
1184
    int sign;
1185
    float scale;
1186
    SingleChannelElement * sce = &che->ch[0];
1187
    ChannelCoupling * coup     = &che->coup;
1188

    
1189
    coup->coupling_point = 2*get_bits1(gb);
1190
    coup->num_coupled = get_bits(gb, 3);
1191
    for (c = 0; c <= coup->num_coupled; c++) {
1192
        num_gain++;
1193
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1194
        coup->id_select[c] = get_bits(gb, 4);
1195
        if (coup->type[c] == TYPE_CPE) {
1196
            coup->ch_select[c] = get_bits(gb, 2);
1197
            if (coup->ch_select[c] == 3)
1198
                num_gain++;
1199
        } else
1200
            coup->ch_select[c] = 2;
1201
    }
1202
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point>>1);
1203

    
1204
    sign = get_bits(gb, 1);
1205
    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1206

    
1207
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1208
        return ret;
1209

    
1210
    for (c = 0; c < num_gain; c++) {
1211
        int idx = 0;
1212
        int cge = 1;
1213
        int gain = 0;
1214
        float gain_cache = 1.;
1215
        if (c) {
1216
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1217
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1218
            gain_cache = pow(scale, -gain);
1219
        }
1220
        if (coup->coupling_point == AFTER_IMDCT) {
1221
            coup->gain[c][0] = gain_cache;
1222
        } else {
1223
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1224
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1225
                    if (sce->band_type[idx] != ZERO_BT) {
1226
                        if (!cge) {
1227
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1228
                                if (t) {
1229
                                int s = 1;
1230
                                t = gain += t;
1231
                                if (sign) {
1232
                                    s  -= 2 * (t & 0x1);
1233
                                    t >>= 1;
1234
                                }
1235
                                gain_cache = pow(scale, -t) * s;
1236
                            }
1237
                        }
1238
                        coup->gain[c][idx] = gain_cache;
1239
                    }
1240
                }
1241
            }
1242
        }
1243
    }
1244
    return 0;
1245
}
1246

    
1247
/**
1248
 * Decode Spectral Band Replication extension data; reference: table 4.55.
1249
 *
1250
 * @param   crc flag indicating the presence of CRC checksum
1251
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1252
 *
1253
 * @return  Returns number of bytes consumed from the TYPE_FIL element.
1254
 */
1255
static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
1256
    // TODO : sbr_extension implementation
1257
    ff_log_missing_feature(ac->avccontext, "SBR", 0);
1258
    skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
1259
    return cnt;
1260
}
1261

    
1262
/**
1263
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1264
 *
1265
 * @return  Returns number of bytes consumed.
1266
 */
1267
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) {
1268
    int i;
1269
    int num_excl_chan = 0;
1270

    
1271
    do {
1272
        for (i = 0; i < 7; i++)
1273
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1274
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1275

    
1276
    return num_excl_chan / 7;
1277
}
1278

    
1279
/**
1280
 * Decode dynamic range information; reference: table 4.52.
1281
 *
1282
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1283
 *
1284
 * @return  Returns number of bytes consumed.
1285
 */
1286
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
1287
    int n = 1;
1288
    int drc_num_bands = 1;
1289
    int i;
1290

    
1291
    /* pce_tag_present? */
1292
    if(get_bits1(gb)) {
1293
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1294
        skip_bits(gb, 4); // tag_reserved_bits
1295
        n++;
1296
    }
1297

    
1298
    /* excluded_chns_present? */
1299
    if(get_bits1(gb)) {
1300
        n += decode_drc_channel_exclusions(che_drc, gb);
1301
    }
1302

    
1303
    /* drc_bands_present? */
1304
    if (get_bits1(gb)) {
1305
        che_drc->band_incr            = get_bits(gb, 4);
1306
        che_drc->interpolation_scheme = get_bits(gb, 4);
1307
        n++;
1308
        drc_num_bands += che_drc->band_incr;
1309
        for (i = 0; i < drc_num_bands; i++) {
1310
            che_drc->band_top[i] = get_bits(gb, 8);
1311
            n++;
1312
        }
1313
    }
1314

    
1315
    /* prog_ref_level_present? */
1316
    if (get_bits1(gb)) {
1317
        che_drc->prog_ref_level = get_bits(gb, 7);
1318
        skip_bits1(gb); // prog_ref_level_reserved_bits
1319
        n++;
1320
    }
1321

    
1322
    for (i = 0; i < drc_num_bands; i++) {
1323
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1324
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1325
        n++;
1326
    }
1327

    
1328
    return n;
1329
}
1330

    
1331
/**
1332
 * Decode extension data (incomplete); reference: table 4.51.
1333
 *
1334
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1335
 *
1336
 * @return Returns number of bytes consumed
1337
 */
1338
static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
1339
    int crc_flag = 0;
1340
    int res = cnt;
1341
    switch (get_bits(gb, 4)) { // extension type
1342
        case EXT_SBR_DATA_CRC:
1343
            crc_flag++;
1344
        case EXT_SBR_DATA:
1345
            res = decode_sbr_extension(ac, gb, crc_flag, cnt);
1346
            break;
1347
        case EXT_DYNAMIC_RANGE:
1348
            res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1349
            break;
1350
        case EXT_FILL:
1351
        case EXT_FILL_DATA:
1352
        case EXT_DATA_ELEMENT:
1353
        default:
1354
            skip_bits_long(gb, 8*cnt - 4);
1355
            break;
1356
    };
1357
    return res;
1358
}
1359

    
1360
/**
1361
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1362
 *
1363
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1364
 * @param   coef    spectral coefficients
1365
 */
1366
static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) {
1367
    const int mmm = FFMIN(ics->tns_max_bands,  ics->max_sfb);
1368
    int w, filt, m, i;
1369
    int bottom, top, order, start, end, size, inc;
1370
    float lpc[TNS_MAX_ORDER];
1371

    
1372
    for (w = 0; w < ics->num_windows; w++) {
1373
        bottom = ics->num_swb;
1374
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1375
            top    = bottom;
1376
            bottom = FFMAX(0, top - tns->length[w][filt]);
1377
            order  = tns->order[w][filt];
1378
            if (order == 0)
1379
                continue;
1380

    
1381
            // tns_decode_coef
1382
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1383

    
1384
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1385
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1386
            if ((size = end - start) <= 0)
1387
                continue;
1388
            if (tns->direction[w][filt]) {
1389
                inc = -1; start = end - 1;
1390
            } else {
1391
                inc = 1;
1392
            }
1393
            start += w * 128;
1394

    
1395
            // ar filter
1396
            for (m = 0; m < size; m++, start += inc)
1397
                for (i = 1; i <= FFMIN(m, order); i++)
1398
                    coef[start] -= coef[start - i*inc] * lpc[i-1];
1399
        }
1400
    }
1401
}
1402

    
1403
/**
1404
 * Conduct IMDCT and windowing.
1405
 */
1406
static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
1407
    IndividualChannelStream * ics = &sce->ics;
1408
    float * in = sce->coeffs;
1409
    float * out = sce->ret;
1410
    float * saved = sce->saved;
1411
    const float * swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1412
    const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1413
    const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1414
    float * buf = ac->buf_mdct;
1415
    float * temp = ac->temp;
1416
    int i;
1417

    
1418
    // imdct
1419
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1420
        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1421
            av_log(ac->avccontext, AV_LOG_WARNING,
1422
                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1423
                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1424
        for (i = 0; i < 1024; i += 128)
1425
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1426
    } else
1427
        ff_imdct_half(&ac->mdct, buf, in);
1428

    
1429
    /* window overlapping
1430
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1431
     * and long to short transitions are considered to be short to short
1432
     * transitions. This leaves just two cases (long to long and short to short)
1433
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1434
     */
1435
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1436
        (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1437
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, ac->add_bias, 512);
1438
    } else {
1439
        for (i = 0; i < 448; i++)
1440
            out[i] = saved[i] + ac->add_bias;
1441

    
1442
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1443
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, ac->add_bias, 64);
1444
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      ac->add_bias, 64);
1445
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      ac->add_bias, 64);
1446
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      ac->add_bias, 64);
1447
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      ac->add_bias, 64);
1448
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1449
        } else {
1450
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, ac->add_bias, 64);
1451
            for (i = 576; i < 1024; i++)
1452
                out[i] = buf[i-512] + ac->add_bias;
1453
        }
1454
    }
1455

    
1456
    // buffer update
1457
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1458
        for (i = 0; i < 64; i++)
1459
            saved[i] = temp[64 + i] - ac->add_bias;
1460
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1461
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1462
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1463
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1464
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1465
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1466
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1467
    } else { // LONG_STOP or ONLY_LONG
1468
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1469
    }
1470
}
1471

    
1472
/**
1473
 * Apply dependent channel coupling (applied before IMDCT).
1474
 *
1475
 * @param   index   index into coupling gain array
1476
 */
1477
static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) {
1478
    IndividualChannelStream * ics = &cce->ch[0].ics;
1479
    const uint16_t * offsets = ics->swb_offset;
1480
    float * dest = target->coeffs;
1481
    const float * src = cce->ch[0].coeffs;
1482
    int g, i, group, k, idx = 0;
1483
    if(ac->m4ac.object_type == AOT_AAC_LTP) {
1484
        av_log(ac->avccontext, AV_LOG_ERROR,
1485
               "Dependent coupling is not supported together with LTP\n");
1486
        return;
1487
    }
1488
    for (g = 0; g < ics->num_window_groups; g++) {
1489
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1490
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1491
                const float gain = cce->coup.gain[index][idx];
1492
                for (group = 0; group < ics->group_len[g]; group++) {
1493
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
1494
                        // XXX dsputil-ize
1495
                        dest[group*128+k] += gain * src[group*128+k];
1496
                    }
1497
                }
1498
            }
1499
        }
1500
        dest += ics->group_len[g]*128;
1501
        src  += ics->group_len[g]*128;
1502
    }
1503
}
1504

    
1505
/**
1506
 * Apply independent channel coupling (applied after IMDCT).
1507
 *
1508
 * @param   index   index into coupling gain array
1509
 */
1510
static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) {
1511
    int i;
1512
    const float gain = cce->coup.gain[index][0];
1513
    const float bias = ac->add_bias;
1514
    const float* src = cce->ch[0].ret;
1515
    float* dest = target->ret;
1516

    
1517
    for (i = 0; i < 1024; i++)
1518
        dest[i] += gain * (src[i] - bias);
1519
}
1520

    
1521
/**
1522
 * channel coupling transformation interface
1523
 *
1524
 * @param   index   index into coupling gain array
1525
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1526
 */
1527
static void apply_channel_coupling(AACContext * ac, ChannelElement * cc,
1528
        enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point,
1529
        void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index))
1530
{
1531
    int i, c;
1532

    
1533
    for (i = 0; i < MAX_ELEM_ID; i++) {
1534
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1535
        int index = 0;
1536

    
1537
        if (cce && cce->coup.coupling_point == coupling_point) {
1538
            ChannelCoupling * coup = &cce->coup;
1539

    
1540
            for (c = 0; c <= coup->num_coupled; c++) {
1541
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1542
                    if (coup->ch_select[c] != 1) {
1543
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1544
                        if (coup->ch_select[c] != 0)
1545
                            index++;
1546
                    }
1547
                    if (coup->ch_select[c] != 2)
1548
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1549
                } else
1550
                    index += 1 + (coup->ch_select[c] == 3);
1551
            }
1552
        }
1553
    }
1554
}
1555

    
1556
/**
1557
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1558
 */
1559
static void spectral_to_sample(AACContext * ac) {
1560
    int i, type;
1561
    for(type = 3; type >= 0; type--) {
1562
        for (i = 0; i < MAX_ELEM_ID; i++) {
1563
            ChannelElement *che = ac->che[type][i];
1564
            if(che) {
1565
                if(type <= TYPE_CPE)
1566
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1567
                if(che->ch[0].tns.present)
1568
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1569
                if(che->ch[1].tns.present)
1570
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1571
                if(type <= TYPE_CPE)
1572
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1573
                if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
1574
                    imdct_and_windowing(ac, &che->ch[0]);
1575
                if(type == TYPE_CPE)
1576
                    imdct_and_windowing(ac, &che->ch[1]);
1577
                if(type <= TYPE_CCE)
1578
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1579
            }
1580
        }
1581
    }
1582
}
1583

    
1584
static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) {
1585

    
1586
    int size;
1587
    AACADTSHeaderInfo hdr_info;
1588

    
1589
    size = ff_aac_parse_header(gb, &hdr_info);
1590
    if (size > 0) {
1591
        if (hdr_info.chan_config)
1592
            ac->m4ac.chan_config = hdr_info.chan_config;
1593
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1594
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1595
        ac->m4ac.object_type     = hdr_info.object_type;
1596
        if (hdr_info.num_aac_frames == 1) {
1597
            if (!hdr_info.crc_absent)
1598
                skip_bits(gb, 16);
1599
        } else {
1600
            ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1601
            return -1;
1602
        }
1603
    }
1604
    return size;
1605
}
1606

    
1607
static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, AVPacket *avpkt) {
1608
    const uint8_t *buf = avpkt->data;
1609
    int buf_size = avpkt->size;
1610
    AACContext * ac = avccontext->priv_data;
1611
    ChannelElement * che = NULL;
1612
    GetBitContext gb;
1613
    enum RawDataBlockType elem_type;
1614
    int err, elem_id, data_size_tmp;
1615

    
1616
    init_get_bits(&gb, buf, buf_size*8);
1617

    
1618
    if (show_bits(&gb, 12) == 0xfff) {
1619
        if ((err = parse_adts_frame_header(ac, &gb)) < 0) {
1620
            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1621
            return -1;
1622
        }
1623
        if (ac->m4ac.sampling_index > 12) {
1624
            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1625
            return -1;
1626
        }
1627
    }
1628

    
1629
    // parse
1630
    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1631
        elem_id = get_bits(&gb, 4);
1632
        err = -1;
1633

    
1634
        if(elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
1635
            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1636
            return -1;
1637
        }
1638

    
1639
        switch (elem_type) {
1640

    
1641
        case TYPE_SCE:
1642
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1643
            break;
1644

    
1645
        case TYPE_CPE:
1646
            err = decode_cpe(ac, &gb, che);
1647
            break;
1648

    
1649
        case TYPE_CCE:
1650
            err = decode_cce(ac, &gb, che);
1651
            break;
1652

    
1653
        case TYPE_LFE:
1654
            err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1655
            break;
1656

    
1657
        case TYPE_DSE:
1658
            skip_data_stream_element(&gb);
1659
            err = 0;
1660
            break;
1661

    
1662
        case TYPE_PCE:
1663
        {
1664
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1665
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1666
            if((err = decode_pce(ac, new_che_pos, &gb)))
1667
                break;
1668
            err = output_configure(ac, ac->che_pos, new_che_pos, 0);
1669
            break;
1670
        }
1671

    
1672
        case TYPE_FIL:
1673
            if (elem_id == 15)
1674
                elem_id += get_bits(&gb, 8) - 1;
1675
            while (elem_id > 0)
1676
                elem_id -= decode_extension_payload(ac, &gb, elem_id);
1677
            err = 0; /* FIXME */
1678
            break;
1679

    
1680
        default:
1681
            err = -1; /* should not happen, but keeps compiler happy */
1682
            break;
1683
        }
1684

    
1685
        if(err)
1686
            return err;
1687
    }
1688

    
1689
    spectral_to_sample(ac);
1690

    
1691
    if (!ac->is_saved) {
1692
        ac->is_saved = 1;
1693
        *data_size = 0;
1694
        return buf_size;
1695
    }
1696

    
1697
    data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
1698
    if(*data_size < data_size_tmp) {
1699
        av_log(avccontext, AV_LOG_ERROR,
1700
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
1701
               *data_size, data_size_tmp);
1702
        return -1;
1703
    }
1704
    *data_size = data_size_tmp;
1705

    
1706
    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
1707

    
1708
    return buf_size;
1709
}
1710

    
1711
static av_cold int aac_decode_close(AVCodecContext * avccontext) {
1712
    AACContext * ac = avccontext->priv_data;
1713
    int i, type;
1714

    
1715
    for (i = 0; i < MAX_ELEM_ID; i++) {
1716
        for(type = 0; type < 4; type++)
1717
            av_freep(&ac->che[type][i]);
1718
    }
1719

    
1720
    ff_mdct_end(&ac->mdct);
1721
    ff_mdct_end(&ac->mdct_small);
1722
    return 0 ;
1723
}
1724

    
1725
AVCodec aac_decoder = {
1726
    "aac",
1727
    CODEC_TYPE_AUDIO,
1728
    CODEC_ID_AAC,
1729
    sizeof(AACContext),
1730
    aac_decode_init,
1731
    NULL,
1732
    aac_decode_close,
1733
    aac_decode_frame,
1734
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
1735
    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
1736
};