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ffmpeg / libavcodec / aac.c @ 039821a8

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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
/**
24
 * @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
/*
31
 * supported tools
32
 *
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 * Support?             Name
34
 * N (code in SoC repo) gain control
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 * Y                    block switching
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 * Y                    window shapes - standard
37
 * N                    window shapes - Low Delay
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 * Y                    filterbank - standard
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 * N (code in SoC repo) filterbank - Scalable Sample Rate
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 * Y                    Temporal Noise Shaping
41
 * 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
64
 * N                    Text-To-Speech Interface
65
 * 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.
76
 */
77

    
78

    
79
#include "avcodec.h"
80
#include "internal.h"
81
#include "bitstream.h"
82
#include "dsputil.h"
83
#include "lpc.h"
84

    
85
#include "aac.h"
86
#include "aactab.h"
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#include "aacdectab.h"
88
#include "mpeg4audio.h"
89
#include "aac_parser.h"
90

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

    
96
static VLC vlc_scalefactors;
97
static VLC vlc_spectral[11];
98

    
99

    
100
/**
101
 * Configure output channel order based on the current program configuration element.
102
 *
103
 * @param   che_pos current channel position configuration
104
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
105
 *
106
 * @return  Returns error status. 0 - OK, !0 - error
107
 */
108
static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID],
109
        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) {
110
    AVCodecContext *avctx = ac->avccontext;
111
    int i, type, channels = 0;
112

    
113
    if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])))
114
        return 0; /* no change */
115

    
116
    memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
117

    
118
    /* Allocate or free elements depending on if they are in the
119
     * current program configuration.
120
     *
121
     * Set up default 1:1 output mapping.
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     *
123
     * For a 5.1 stream the output order will be:
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     *    [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
125
     */
126

    
127
    for(i = 0; i < MAX_ELEM_ID; i++) {
128
        for(type = 0; type < 4; type++) {
129
            if(che_pos[type][i]) {
130
                if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement))))
131
                    return AVERROR(ENOMEM);
132
                if(type != TYPE_CCE) {
133
                    ac->output_data[channels++] = ac->che[type][i]->ch[0].ret;
134
                    if(type == TYPE_CPE) {
135
                        ac->output_data[channels++] = ac->che[type][i]->ch[1].ret;
136
                    }
137
                }
138
            } else
139
                av_freep(&ac->che[type][i]);
140
        }
141
    }
142

    
143
    avctx->channels = channels;
144
    return 0;
145
}
146

    
147
/**
148
 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
149
 *
150
 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
151
 * @param sce_map mono (Single Channel Element) map
152
 * @param type speaker type/position for these channels
153
 */
154
static void decode_channel_map(enum ChannelPosition *cpe_map,
155
        enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
156
    while(n--) {
157
        enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
158
        map[get_bits(gb, 4)] = type;
159
    }
160
}
161

    
162
/**
163
 * Decode program configuration element; reference: table 4.2.
164
 *
165
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
166
 *
167
 * @return  Returns error status. 0 - OK, !0 - error
168
 */
169
static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
170
        GetBitContext * gb) {
171
    int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
172

    
173
    skip_bits(gb, 2);  // object_type
174

    
175
    sampling_index = get_bits(gb, 4);
176
    if(sampling_index > 12) {
177
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
178
        return -1;
179
    }
180
    ac->m4ac.sampling_index = sampling_index;
181
    ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
182
    num_front       = get_bits(gb, 4);
183
    num_side        = get_bits(gb, 4);
184
    num_back        = get_bits(gb, 4);
185
    num_lfe         = get_bits(gb, 2);
186
    num_assoc_data  = get_bits(gb, 3);
187
    num_cc          = get_bits(gb, 4);
188

    
189
    if (get_bits1(gb))
190
        skip_bits(gb, 4); // mono_mixdown_tag
191
    if (get_bits1(gb))
192
        skip_bits(gb, 4); // stereo_mixdown_tag
193

    
194
    if (get_bits1(gb))
195
        skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
196

    
197
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
198
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE,  gb, num_side );
199
    decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK,  gb, num_back );
200
    decode_channel_map(NULL,                  new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE,   gb, num_lfe  );
201

    
202
    skip_bits_long(gb, 4 * num_assoc_data);
203

    
204
    decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC,    gb, num_cc   );
205

    
206
    align_get_bits(gb);
207

    
208
    /* comment field, first byte is length */
209
    skip_bits_long(gb, 8 * get_bits(gb, 8));
210
    return 0;
211
}
212

    
213
/**
214
 * Set up channel positions based on a default channel configuration
215
 * as specified in table 1.17.
216
 *
217
 * @param   new_che_pos New channel position configuration - we only do something if it differs from the current one.
218
 *
219
 * @return  Returns error status. 0 - OK, !0 - error
220
 */
221
static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
222
        int channel_config)
223
{
224
    if(channel_config < 1 || channel_config > 7) {
225
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
226
               channel_config);
227
        return -1;
228
    }
229

    
230
    /* default channel configurations:
231
     *
232
     * 1ch : front center (mono)
233
     * 2ch : L + R (stereo)
234
     * 3ch : front center + L + R
235
     * 4ch : front center + L + R + back center
236
     * 5ch : front center + L + R + back stereo
237
     * 6ch : front center + L + R + back stereo + LFE
238
     * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
239
     */
240

    
241
    if(channel_config != 2)
242
        new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
243
    if(channel_config > 1)
244
        new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
245
    if(channel_config == 4)
246
        new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK;  // back center
247
    if(channel_config > 4)
248
        new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
249
                                 = AAC_CHANNEL_BACK;  // back stereo
250
    if(channel_config > 5)
251
        new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE;   // LFE
252
    if(channel_config == 7)
253
        new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
254

    
255
    return 0;
256
}
257

    
258
/**
259
 * Decode GA "General Audio" specific configuration; reference: table 4.1.
260
 *
261
 * @return  Returns error status. 0 - OK, !0 - error
262
 */
263
static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) {
264
    enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
265
    int extension_flag, ret;
266

    
267
    if(get_bits1(gb)) {  // frameLengthFlag
268
        ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
269
        return -1;
270
    }
271

    
272
    if (get_bits1(gb))       // dependsOnCoreCoder
273
        skip_bits(gb, 14);   // coreCoderDelay
274
    extension_flag = get_bits1(gb);
275

    
276
    if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
277
       ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
278
        skip_bits(gb, 3);     // layerNr
279

    
280
    memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
281
    if (channel_config == 0) {
282
        skip_bits(gb, 4);  // element_instance_tag
283
        if((ret = decode_pce(ac, new_che_pos, gb)))
284
            return ret;
285
    } else {
286
        if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
287
            return ret;
288
    }
289
    if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
290
        return ret;
291

    
292
    if (extension_flag) {
293
        switch (ac->m4ac.object_type) {
294
            case AOT_ER_BSAC:
295
                skip_bits(gb, 5);    // numOfSubFrame
296
                skip_bits(gb, 11);   // layer_length
297
                break;
298
            case AOT_ER_AAC_LC:
299
            case AOT_ER_AAC_LTP:
300
            case AOT_ER_AAC_SCALABLE:
301
            case AOT_ER_AAC_LD:
302
                skip_bits(gb, 3);  /* aacSectionDataResilienceFlag
303
                                    * aacScalefactorDataResilienceFlag
304
                                    * aacSpectralDataResilienceFlag
305
                                    */
306
                break;
307
        }
308
        skip_bits1(gb);    // extensionFlag3 (TBD in version 3)
309
    }
310
    return 0;
311
}
312

    
313
/**
314
 * Decode audio specific configuration; reference: table 1.13.
315
 *
316
 * @param   data        pointer to AVCodecContext extradata
317
 * @param   data_size   size of AVCCodecContext extradata
318
 *
319
 * @return  Returns error status. 0 - OK, !0 - error
320
 */
321
static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
322
    GetBitContext gb;
323
    int i;
324

    
325
    init_get_bits(&gb, data, data_size * 8);
326

    
327
    if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
328
        return -1;
329
    if(ac->m4ac.sampling_index > 12) {
330
        av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
331
        return -1;
332
    }
333

    
334
    skip_bits_long(&gb, i);
335

    
336
    switch (ac->m4ac.object_type) {
337
    case AOT_AAC_MAIN:
338
    case AOT_AAC_LC:
339
        if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
340
            return -1;
341
        break;
342
    default:
343
        av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
344
               ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
345
        return -1;
346
    }
347
    return 0;
348
}
349

    
350
/**
351
 * linear congruential pseudorandom number generator
352
 *
353
 * @param   previous_val    pointer to the current state of the generator
354
 *
355
 * @return  Returns a 32-bit pseudorandom integer
356
 */
357
static av_always_inline int lcg_random(int previous_val) {
358
    return previous_val * 1664525 + 1013904223;
359
}
360

    
361
static void reset_predict_state(PredictorState * ps) {
362
    ps->r0 = 0.0f;
363
    ps->r1 = 0.0f;
364
    ps->cor0 = 0.0f;
365
    ps->cor1 = 0.0f;
366
    ps->var0 = 1.0f;
367
    ps->var1 = 1.0f;
368
}
369

    
370
static void reset_all_predictors(PredictorState * ps) {
371
    int i;
372
    for (i = 0; i < MAX_PREDICTORS; i++)
373
        reset_predict_state(&ps[i]);
374
}
375

    
376
static void reset_predictor_group(PredictorState * ps, int group_num) {
377
    int i;
378
    for (i = group_num-1; i < MAX_PREDICTORS; i+=30)
379
        reset_predict_state(&ps[i]);
380
}
381

    
382
static av_cold int aac_decode_init(AVCodecContext * avccontext) {
383
    AACContext * ac = avccontext->priv_data;
384
    int i;
385

    
386
    ac->avccontext = avccontext;
387

    
388
    if (avccontext->extradata_size > 0) {
389
        if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
390
            return -1;
391
        avccontext->sample_rate = ac->m4ac.sample_rate;
392
    } else if (avccontext->channels > 0) {
393
        enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
394
        memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
395
        if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8)))
396
            return -1;
397
        if(output_configure(ac, ac->che_pos, new_che_pos))
398
            return -1;
399
        ac->m4ac.sample_rate = avccontext->sample_rate;
400
    } else {
401
        ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0);
402
        return -1;
403
    }
404

    
405
    avccontext->sample_fmt  = SAMPLE_FMT_S16;
406
    avccontext->frame_size  = 1024;
407

    
408
    AAC_INIT_VLC_STATIC( 0, 144);
409
    AAC_INIT_VLC_STATIC( 1, 114);
410
    AAC_INIT_VLC_STATIC( 2, 188);
411
    AAC_INIT_VLC_STATIC( 3, 180);
412
    AAC_INIT_VLC_STATIC( 4, 172);
413
    AAC_INIT_VLC_STATIC( 5, 140);
414
    AAC_INIT_VLC_STATIC( 6, 168);
415
    AAC_INIT_VLC_STATIC( 7, 114);
416
    AAC_INIT_VLC_STATIC( 8, 262);
417
    AAC_INIT_VLC_STATIC( 9, 248);
418
    AAC_INIT_VLC_STATIC(10, 384);
419

    
420
    dsputil_init(&ac->dsp, avccontext);
421

    
422
    ac->random_state = 0x1f2e3d4c;
423

    
424
    // -1024 - Compensate wrong IMDCT method.
425
    // 32768 - Required to scale values to the correct range for the bias method
426
    //         for float to int16 conversion.
427

    
428
    if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
429
        ac->add_bias = 385.0f;
430
        ac->sf_scale = 1. / (-1024. * 32768.);
431
        ac->sf_offset = 0;
432
    } else {
433
        ac->add_bias = 0.0f;
434
        ac->sf_scale = 1. / -1024.;
435
        ac->sf_offset = 60;
436
    }
437

    
438
#if !CONFIG_HARDCODED_TABLES
439
    for (i = 0; i < 428; i++)
440
        ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
441
#endif /* CONFIG_HARDCODED_TABLES */
442

    
443
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
444
        ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
445
        ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
446
        352);
447

    
448
    ff_mdct_init(&ac->mdct, 11, 1);
449
    ff_mdct_init(&ac->mdct_small, 8, 1);
450
    // window initialization
451
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
452
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
453
    ff_sine_window_init(ff_sine_1024, 1024);
454
    ff_sine_window_init(ff_sine_128, 128);
455

    
456
    return 0;
457
}
458

    
459
/**
460
 * Skip data_stream_element; reference: table 4.10.
461
 */
462
static void skip_data_stream_element(GetBitContext * gb) {
463
    int byte_align = get_bits1(gb);
464
    int count = get_bits(gb, 8);
465
    if (count == 255)
466
        count += get_bits(gb, 8);
467
    if (byte_align)
468
        align_get_bits(gb);
469
    skip_bits_long(gb, 8 * count);
470
}
471

    
472
static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) {
473
    int sfb;
474
    if (get_bits1(gb)) {
475
        ics->predictor_reset_group = get_bits(gb, 5);
476
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
477
            av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
478
            return -1;
479
        }
480
    }
481
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
482
        ics->prediction_used[sfb] = get_bits1(gb);
483
    }
484
    return 0;
485
}
486

    
487
/**
488
 * Decode Individual Channel Stream info; reference: table 4.6.
489
 *
490
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
491
 */
492
static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
493
    if (get_bits1(gb)) {
494
        av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
495
        memset(ics, 0, sizeof(IndividualChannelStream));
496
        return -1;
497
    }
498
    ics->window_sequence[1] = ics->window_sequence[0];
499
    ics->window_sequence[0] = get_bits(gb, 2);
500
    ics->use_kb_window[1] = ics->use_kb_window[0];
501
    ics->use_kb_window[0] = get_bits1(gb);
502
    ics->num_window_groups = 1;
503
    ics->group_len[0] = 1;
504
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
505
        int i;
506
        ics->max_sfb = get_bits(gb, 4);
507
        for (i = 0; i < 7; i++) {
508
            if (get_bits1(gb)) {
509
                ics->group_len[ics->num_window_groups-1]++;
510
            } else {
511
                ics->num_window_groups++;
512
                ics->group_len[ics->num_window_groups-1] = 1;
513
            }
514
        }
515
        ics->num_windows   = 8;
516
        ics->swb_offset    =      swb_offset_128[ac->m4ac.sampling_index];
517
        ics->num_swb       =  ff_aac_num_swb_128[ac->m4ac.sampling_index];
518
        ics->tns_max_bands =   tns_max_bands_128[ac->m4ac.sampling_index];
519
        ics->predictor_present = 0;
520
    } else {
521
        ics->max_sfb       = get_bits(gb, 6);
522
        ics->num_windows   = 1;
523
        ics->swb_offset    =     swb_offset_1024[ac->m4ac.sampling_index];
524
        ics->num_swb       = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
525
        ics->tns_max_bands =  tns_max_bands_1024[ac->m4ac.sampling_index];
526
        ics->predictor_present = get_bits1(gb);
527
        ics->predictor_reset_group = 0;
528
        if (ics->predictor_present) {
529
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
530
                if (decode_prediction(ac, ics, gb)) {
531
                    memset(ics, 0, sizeof(IndividualChannelStream));
532
                    return -1;
533
                }
534
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
535
                av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
536
                memset(ics, 0, sizeof(IndividualChannelStream));
537
                return -1;
538
            } else {
539
                ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
540
                memset(ics, 0, sizeof(IndividualChannelStream));
541
                return -1;
542
            }
543
        }
544
    }
545

    
546
    if(ics->max_sfb > ics->num_swb) {
547
        av_log(ac->avccontext, AV_LOG_ERROR,
548
            "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
549
            ics->max_sfb, ics->num_swb);
550
        memset(ics, 0, sizeof(IndividualChannelStream));
551
        return -1;
552
    }
553

    
554
    return 0;
555
}
556

    
557
/**
558
 * Decode band types (section_data payload); reference: table 4.46.
559
 *
560
 * @param   band_type           array of the used band type
561
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
562
 *
563
 * @return  Returns error status. 0 - OK, !0 - error
564
 */
565
static int decode_band_types(AACContext * ac, enum BandType band_type[120],
566
        int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
567
    int g, idx = 0;
568
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
569
    for (g = 0; g < ics->num_window_groups; g++) {
570
        int k = 0;
571
        while (k < ics->max_sfb) {
572
            uint8_t sect_len = k;
573
            int sect_len_incr;
574
            int sect_band_type = get_bits(gb, 4);
575
            if (sect_band_type == 12) {
576
                av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
577
                return -1;
578
            }
579
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
580
                sect_len += sect_len_incr;
581
            sect_len += sect_len_incr;
582
            if (sect_len > ics->max_sfb) {
583
                av_log(ac->avccontext, AV_LOG_ERROR,
584
                    "Number of bands (%d) exceeds limit (%d).\n",
585
                    sect_len, ics->max_sfb);
586
                return -1;
587
            }
588
            for (; k < sect_len; k++) {
589
                band_type        [idx]   = sect_band_type;
590
                band_type_run_end[idx++] = sect_len;
591
            }
592
        }
593
    }
594
    return 0;
595
}
596

    
597
/**
598
 * Decode scalefactors; reference: table 4.47.
599
 *
600
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
601
 * @param   band_type           array of the used band type
602
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
603
 * @param   sf                  array of scalefactors or intensity stereo positions
604
 *
605
 * @return  Returns error status. 0 - OK, !0 - error
606
 */
607
static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
608
        unsigned int global_gain, IndividualChannelStream * ics,
609
        enum BandType band_type[120], int band_type_run_end[120]) {
610
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
611
    int g, i, idx = 0;
612
    int offset[3] = { global_gain, global_gain - 90, 100 };
613
    int noise_flag = 1;
614
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
615
    for (g = 0; g < ics->num_window_groups; g++) {
616
        for (i = 0; i < ics->max_sfb;) {
617
            int run_end = band_type_run_end[idx];
618
            if (band_type[idx] == ZERO_BT) {
619
                for(; i < run_end; i++, idx++)
620
                    sf[idx] = 0.;
621
            }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
622
                for(; i < run_end; i++, idx++) {
623
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
624
                    if(offset[2] > 255U) {
625
                        av_log(ac->avccontext, AV_LOG_ERROR,
626
                            "%s (%d) out of range.\n", sf_str[2], offset[2]);
627
                        return -1;
628
                    }
629
                    sf[idx]  = ff_aac_pow2sf_tab[-offset[2] + 300];
630
                }
631
            }else if(band_type[idx] == NOISE_BT) {
632
                for(; i < run_end; i++, idx++) {
633
                    if(noise_flag-- > 0)
634
                        offset[1] += get_bits(gb, 9) - 256;
635
                    else
636
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
637
                    if(offset[1] > 255U) {
638
                        av_log(ac->avccontext, AV_LOG_ERROR,
639
                            "%s (%d) out of range.\n", sf_str[1], offset[1]);
640
                        return -1;
641
                    }
642
                    sf[idx]  = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100];
643
                }
644
            }else {
645
                for(; i < run_end; i++, idx++) {
646
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
647
                    if(offset[0] > 255U) {
648
                        av_log(ac->avccontext, AV_LOG_ERROR,
649
                            "%s (%d) out of range.\n", sf_str[0], offset[0]);
650
                        return -1;
651
                    }
652
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
653
                }
654
            }
655
        }
656
    }
657
    return 0;
658
}
659

    
660
/**
661
 * Decode pulse data; reference: table 4.7.
662
 */
663
static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) {
664
    int i, pulse_swb;
665
    pulse->num_pulse = get_bits(gb, 2) + 1;
666
    pulse_swb        = get_bits(gb, 6);
667
    if (pulse_swb >= num_swb)
668
        return -1;
669
    pulse->pos[0]    = swb_offset[pulse_swb];
670
    pulse->pos[0]   += get_bits(gb, 5);
671
    if (pulse->pos[0] > 1023)
672
        return -1;
673
    pulse->amp[0]    = get_bits(gb, 4);
674
    for (i = 1; i < pulse->num_pulse; i++) {
675
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1];
676
        if (pulse->pos[i] > 1023)
677
            return -1;
678
        pulse->amp[i] = get_bits(gb, 4);
679
    }
680
    return 0;
681
}
682

    
683
/**
684
 * Decode Temporal Noise Shaping data; reference: table 4.48.
685
 *
686
 * @return  Returns error status. 0 - OK, !0 - error
687
 */
688
static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns,
689
        GetBitContext * gb, const IndividualChannelStream * ics) {
690
    int w, filt, i, coef_len, coef_res, coef_compress;
691
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
692
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
693
    for (w = 0; w < ics->num_windows; w++) {
694
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
695
            coef_res = get_bits1(gb);
696

    
697
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
698
                int tmp2_idx;
699
                tns->length[w][filt] = get_bits(gb, 6 - 2*is8);
700

    
701
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) {
702
                    av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
703
                           tns->order[w][filt], tns_max_order);
704
                    tns->order[w][filt] = 0;
705
                    return -1;
706
                }
707
                if (tns->order[w][filt]) {
708
                    tns->direction[w][filt] = get_bits1(gb);
709
                    coef_compress = get_bits1(gb);
710
                    coef_len = coef_res + 3 - coef_compress;
711
                    tmp2_idx = 2*coef_compress + coef_res;
712

    
713
                    for (i = 0; i < tns->order[w][filt]; i++)
714
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
715
                }
716
            }
717
        }
718
    }
719
    return 0;
720
}
721

    
722
/**
723
 * Decode Mid/Side data; reference: table 4.54.
724
 *
725
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
726
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
727
 *                      [3] reserved for scalable AAC
728
 */
729
static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
730
        int ms_present) {
731
    int idx;
732
    if (ms_present == 1) {
733
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
734
            cpe->ms_mask[idx] = get_bits1(gb);
735
    } else if (ms_present == 2) {
736
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
737
    }
738
}
739

    
740
/**
741
 * Decode spectral data; reference: table 4.50.
742
 * Dequantize and scale spectral data; reference: 4.6.3.3.
743
 *
744
 * @param   coef            array of dequantized, scaled spectral data
745
 * @param   sf              array of scalefactors or intensity stereo positions
746
 * @param   pulse_present   set if pulses are present
747
 * @param   pulse           pointer to pulse data struct
748
 * @param   band_type       array of the used band type
749
 *
750
 * @return  Returns error status. 0 - OK, !0 - error
751
 */
752
static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120],
753
        int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) {
754
    int i, k, g, idx = 0;
755
    const int c = 1024/ics->num_windows;
756
    const uint16_t * offsets = ics->swb_offset;
757
    float *coef_base = coef;
758
    static const float sign_lookup[] = { 1.0f, -1.0f };
759

    
760
    for (g = 0; g < ics->num_windows; g++)
761
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb]));
762

    
763
    for (g = 0; g < ics->num_window_groups; g++) {
764
        for (i = 0; i < ics->max_sfb; i++, idx++) {
765
            const int cur_band_type = band_type[idx];
766
            const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4;
767
            const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type);
768
            int group;
769
            if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) {
770
                for (group = 0; group < ics->group_len[g]; group++) {
771
                    memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float));
772
                }
773
            }else if (cur_band_type == NOISE_BT) {
774
                for (group = 0; group < ics->group_len[g]; group++) {
775
                    float scale;
776
                    float band_energy = 0;
777
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
778
                        ac->random_state  = lcg_random(ac->random_state);
779
                        coef[group*128+k] = ac->random_state;
780
                        band_energy += coef[group*128+k]*coef[group*128+k];
781
                    }
782
                    scale = sf[idx] / sqrtf(band_energy);
783
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
784
                        coef[group*128+k] *= scale;
785
                    }
786
                }
787
            }else {
788
                for (group = 0; group < ics->group_len[g]; group++) {
789
                    for (k = offsets[i]; k < offsets[i+1]; k += dim) {
790
                        const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3);
791
                        const int coef_tmp_idx = (group << 7) + k;
792
                        const float *vq_ptr;
793
                        int j;
794
                        if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) {
795
                            av_log(ac->avccontext, AV_LOG_ERROR,
796
                                "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
797
                                cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]);
798
                            return -1;
799
                        }
800
                        vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim];
801
                        if (is_cb_unsigned) {
802
                            if (vq_ptr[0]) coef[coef_tmp_idx    ] = sign_lookup[get_bits1(gb)];
803
                            if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)];
804
                            if (dim == 4) {
805
                                if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)];
806
                                if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)];
807
                            }
808
                        }else {
809
                            coef[coef_tmp_idx    ] = 1.0f;
810
                            coef[coef_tmp_idx + 1] = 1.0f;
811
                            if (dim == 4) {
812
                                coef[coef_tmp_idx + 2] = 1.0f;
813
                                coef[coef_tmp_idx + 3] = 1.0f;
814
                            }
815
                        }
816
                        if (cur_band_type == ESC_BT) {
817
                            for (j = 0; j < 2; j++) {
818
                                if (vq_ptr[j] == 64.0f) {
819
                                    int n = 4;
820
                                    /* The total length of escape_sequence must be < 22 bits according
821
                                       to the specification (i.e. max is 11111111110xxxxxxxxxx). */
822
                                    while (get_bits1(gb) && n < 15) n++;
823
                                    if(n == 15) {
824
                                        av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
825
                                        return -1;
826
                                    }
827
                                    n = (1<<n) + get_bits(gb, n);
828
                                    coef[coef_tmp_idx + j] *= cbrtf(n) * n;
829
                                }else
830
                                    coef[coef_tmp_idx + j] *= vq_ptr[j];
831
                            }
832
                        }else
833
                        {
834
                            coef[coef_tmp_idx    ] *= vq_ptr[0];
835
                            coef[coef_tmp_idx + 1] *= vq_ptr[1];
836
                            if (dim == 4) {
837
                                coef[coef_tmp_idx + 2] *= vq_ptr[2];
838
                                coef[coef_tmp_idx + 3] *= vq_ptr[3];
839
                            }
840
                        }
841
                        coef[coef_tmp_idx    ] *= sf[idx];
842
                        coef[coef_tmp_idx + 1] *= sf[idx];
843
                        if (dim == 4) {
844
                            coef[coef_tmp_idx + 2] *= sf[idx];
845
                            coef[coef_tmp_idx + 3] *= sf[idx];
846
                        }
847
                    }
848
                }
849
            }
850
        }
851
        coef += ics->group_len[g]<<7;
852
    }
853

    
854
    if (pulse_present) {
855
        idx = 0;
856
        for(i = 0; i < pulse->num_pulse; i++){
857
            float co  = coef_base[ pulse->pos[i] ];
858
            while(offsets[idx + 1] <= pulse->pos[i])
859
                idx++;
860
            if (band_type[idx] != NOISE_BT && sf[idx]) {
861
                float ico = -pulse->amp[i];
862
                if (co) {
863
                    co /= sf[idx];
864
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
865
                }
866
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
867
            }
868
        }
869
    }
870
    return 0;
871
}
872

    
873
static av_always_inline float flt16_round(float pf) {
874
    int exp;
875
    pf = frexpf(pf, &exp);
876
    pf = ldexpf(roundf(ldexpf(pf, 8)), exp-8);
877
    return pf;
878
}
879

    
880
static av_always_inline float flt16_even(float pf) {
881
    int exp;
882
    pf = frexpf(pf, &exp);
883
    pf = ldexpf(rintf(ldexpf(pf, 8)), exp-8);
884
    return pf;
885
}
886

    
887
static av_always_inline float flt16_trunc(float pf) {
888
    int exp;
889
    pf = frexpf(pf, &exp);
890
    pf = ldexpf(truncf(ldexpf(pf, 8)), exp-8);
891
    return pf;
892
}
893

    
894
static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) {
895
    const float a     = 0.953125; // 61.0/64
896
    const float alpha = 0.90625;  // 29.0/32
897
    float e0, e1;
898
    float pv;
899
    float k1, k2;
900

    
901
    k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
902
    k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
903

    
904
    pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
905
    if (output_enable)
906
        *coef += pv * ac->sf_scale;
907

    
908
    e0 = *coef / ac->sf_scale;
909
    e1 = e0 - k1 * ps->r0;
910

    
911
    ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
912
    ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
913
    ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
914
    ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
915

    
916
    ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
917
    ps->r0 = flt16_trunc(a * e0);
918
}
919

    
920
/**
921
 * Apply AAC-Main style frequency domain prediction.
922
 */
923
static void apply_prediction(AACContext * ac, SingleChannelElement * sce) {
924
    int sfb, k;
925

    
926
    if (!sce->ics.predictor_initialized) {
927
        reset_all_predictors(sce->predictor_state);
928
        sce->ics.predictor_initialized = 1;
929
    }
930

    
931
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
932
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
933
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
934
                predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
935
                    sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
936
            }
937
        }
938
        if (sce->ics.predictor_reset_group)
939
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
940
    } else
941
        reset_all_predictors(sce->predictor_state);
942
}
943

    
944
/**
945
 * Decode an individual_channel_stream payload; reference: table 4.44.
946
 *
947
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
948
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
949
 *
950
 * @return  Returns error status. 0 - OK, !0 - error
951
 */
952
static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
953
    Pulse pulse;
954
    TemporalNoiseShaping * tns = &sce->tns;
955
    IndividualChannelStream * ics = &sce->ics;
956
    float * out = sce->coeffs;
957
    int global_gain, pulse_present = 0;
958

    
959
    /* This assignment is to silence a GCC warning about the variable being used
960
     * uninitialized when in fact it always is.
961
     */
962
    pulse.num_pulse = 0;
963

    
964
    global_gain = get_bits(gb, 8);
965

    
966
    if (!common_window && !scale_flag) {
967
        if (decode_ics_info(ac, ics, gb, 0) < 0)
968
            return -1;
969
    }
970

    
971
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
972
        return -1;
973
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
974
        return -1;
975

    
976
    pulse_present = 0;
977
    if (!scale_flag) {
978
        if ((pulse_present = get_bits1(gb))) {
979
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
980
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
981
                return -1;
982
            }
983
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
984
                av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
985
                return -1;
986
            }
987
        }
988
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
989
            return -1;
990
        if (get_bits1(gb)) {
991
            ff_log_missing_feature(ac->avccontext, "SSR", 1);
992
            return -1;
993
        }
994
    }
995

    
996
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
997
        return -1;
998

    
999
    if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1000
        apply_prediction(ac, sce);
1001

    
1002
    return 0;
1003
}
1004

    
1005
/**
1006
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1007
 */
1008
static void apply_mid_side_stereo(ChannelElement * cpe) {
1009
    const IndividualChannelStream * ics = &cpe->ch[0].ics;
1010
    float *ch0 = cpe->ch[0].coeffs;
1011
    float *ch1 = cpe->ch[1].coeffs;
1012
    int g, i, k, group, idx = 0;
1013
    const uint16_t * offsets = ics->swb_offset;
1014
    for (g = 0; g < ics->num_window_groups; g++) {
1015
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1016
            if (cpe->ms_mask[idx] &&
1017
                cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1018
                for (group = 0; group < ics->group_len[g]; group++) {
1019
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
1020
                        float tmp = ch0[group*128 + k] - ch1[group*128 + k];
1021
                        ch0[group*128 + k] += ch1[group*128 + k];
1022
                        ch1[group*128 + k] = tmp;
1023
                    }
1024
                }
1025
            }
1026
        }
1027
        ch0 += ics->group_len[g]*128;
1028
        ch1 += ics->group_len[g]*128;
1029
    }
1030
}
1031

    
1032
/**
1033
 * intensity stereo decoding; reference: 4.6.8.2.3
1034
 *
1035
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1036
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1037
 *                      [3] reserved for scalable AAC
1038
 */
1039
static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) {
1040
    const IndividualChannelStream * ics = &cpe->ch[1].ics;
1041
    SingleChannelElement * sce1 = &cpe->ch[1];
1042
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1043
    const uint16_t * offsets = ics->swb_offset;
1044
    int g, group, i, k, idx = 0;
1045
    int c;
1046
    float scale;
1047
    for (g = 0; g < ics->num_window_groups; g++) {
1048
        for (i = 0; i < ics->max_sfb;) {
1049
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1050
                const int bt_run_end = sce1->band_type_run_end[idx];
1051
                for (; i < bt_run_end; i++, idx++) {
1052
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1053
                    if (ms_present)
1054
                        c *= 1 - 2 * cpe->ms_mask[idx];
1055
                    scale = c * sce1->sf[idx];
1056
                    for (group = 0; group < ics->group_len[g]; group++)
1057
                        for (k = offsets[i]; k < offsets[i+1]; k++)
1058
                            coef1[group*128 + k] = scale * coef0[group*128 + k];
1059
                }
1060
            } else {
1061
                int bt_run_end = sce1->band_type_run_end[idx];
1062
                idx += bt_run_end - i;
1063
                i    = bt_run_end;
1064
            }
1065
        }
1066
        coef0 += ics->group_len[g]*128;
1067
        coef1 += ics->group_len[g]*128;
1068
    }
1069
}
1070

    
1071
/**
1072
 * Decode a channel_pair_element; reference: table 4.4.
1073
 *
1074
 * @param   elem_id Identifies the instance of a syntax element.
1075
 *
1076
 * @return  Returns error status. 0 - OK, !0 - error
1077
 */
1078
static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
1079
    int i, ret, common_window, ms_present = 0;
1080
    ChannelElement * cpe;
1081

    
1082
    cpe = ac->che[TYPE_CPE][elem_id];
1083
    common_window = get_bits1(gb);
1084
    if (common_window) {
1085
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1086
            return -1;
1087
        i = cpe->ch[1].ics.use_kb_window[0];
1088
        cpe->ch[1].ics = cpe->ch[0].ics;
1089
        cpe->ch[1].ics.use_kb_window[1] = i;
1090
        ms_present = get_bits(gb, 2);
1091
        if(ms_present == 3) {
1092
            av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1093
            return -1;
1094
        } else if(ms_present)
1095
            decode_mid_side_stereo(cpe, gb, ms_present);
1096
    }
1097
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1098
        return ret;
1099
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1100
        return ret;
1101

    
1102
    if (common_window) {
1103
        if (ms_present)
1104
            apply_mid_side_stereo(cpe);
1105
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1106
            apply_prediction(ac, &cpe->ch[0]);
1107
            apply_prediction(ac, &cpe->ch[1]);
1108
        }
1109
    }
1110

    
1111
    apply_intensity_stereo(cpe, ms_present);
1112
    return 0;
1113
}
1114

    
1115
/**
1116
 * Decode coupling_channel_element; reference: table 4.8.
1117
 *
1118
 * @param   elem_id Identifies the instance of a syntax element.
1119
 *
1120
 * @return  Returns error status. 0 - OK, !0 - error
1121
 */
1122
static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) {
1123
    int num_gain = 0;
1124
    int c, g, sfb, ret;
1125
    int sign;
1126
    float scale;
1127
    SingleChannelElement * sce = &che->ch[0];
1128
    ChannelCoupling * coup     = &che->coup;
1129

    
1130
    coup->coupling_point = 2*get_bits1(gb);
1131
    coup->num_coupled = get_bits(gb, 3);
1132
    for (c = 0; c <= coup->num_coupled; c++) {
1133
        num_gain++;
1134
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1135
        coup->id_select[c] = get_bits(gb, 4);
1136
        if (coup->type[c] == TYPE_CPE) {
1137
            coup->ch_select[c] = get_bits(gb, 2);
1138
            if (coup->ch_select[c] == 3)
1139
                num_gain++;
1140
        } else
1141
            coup->ch_select[c] = 2;
1142
    }
1143
    coup->coupling_point += get_bits1(gb);
1144

    
1145
    if (coup->coupling_point == 2) {
1146
        av_log(ac->avccontext, AV_LOG_ERROR,
1147
            "Independently switched CCE with 'invalid' domain signalled.\n");
1148
        memset(coup, 0, sizeof(ChannelCoupling));
1149
        return -1;
1150
    }
1151

    
1152
    sign = get_bits(gb, 1);
1153
    scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1154

    
1155
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1156
        return ret;
1157

    
1158
    for (c = 0; c < num_gain; c++) {
1159
        int idx = 0;
1160
        int cge = 1;
1161
        int gain = 0;
1162
        float gain_cache = 1.;
1163
        if (c) {
1164
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1165
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1166
            gain_cache = pow(scale, -gain);
1167
        }
1168
        if (coup->coupling_point == AFTER_IMDCT) {
1169
            coup->gain[c][0] = gain_cache;
1170
        } else {
1171
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1172
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1173
                    if (sce->band_type[idx] != ZERO_BT) {
1174
                        if (!cge) {
1175
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1176
                                if (t) {
1177
                                int s = 1;
1178
                                t = gain += t;
1179
                                if (sign) {
1180
                                    s  -= 2 * (t & 0x1);
1181
                                    t >>= 1;
1182
                                }
1183
                                gain_cache = pow(scale, -t) * s;
1184
                            }
1185
                        }
1186
                        coup->gain[c][idx] = gain_cache;
1187
                    }
1188
                }
1189
            }
1190
        }
1191
    }
1192
    return 0;
1193
}
1194

    
1195
/**
1196
 * Decode Spectral Band Replication extension data; reference: table 4.55.
1197
 *
1198
 * @param   crc flag indicating the presence of CRC checksum
1199
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1200
 *
1201
 * @return  Returns number of bytes consumed from the TYPE_FIL element.
1202
 */
1203
static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
1204
    // TODO : sbr_extension implementation
1205
    ff_log_missing_feature(ac->avccontext, "SBR", 0);
1206
    skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
1207
    return cnt;
1208
}
1209

    
1210
/**
1211
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1212
 *
1213
 * @return  Returns number of bytes consumed.
1214
 */
1215
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) {
1216
    int i;
1217
    int num_excl_chan = 0;
1218

    
1219
    do {
1220
        for (i = 0; i < 7; i++)
1221
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1222
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1223

    
1224
    return num_excl_chan / 7;
1225
}
1226

    
1227
/**
1228
 * Decode dynamic range information; reference: table 4.52.
1229
 *
1230
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1231
 *
1232
 * @return  Returns number of bytes consumed.
1233
 */
1234
static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
1235
    int n = 1;
1236
    int drc_num_bands = 1;
1237
    int i;
1238

    
1239
    /* pce_tag_present? */
1240
    if(get_bits1(gb)) {
1241
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1242
        skip_bits(gb, 4); // tag_reserved_bits
1243
        n++;
1244
    }
1245

    
1246
    /* excluded_chns_present? */
1247
    if(get_bits1(gb)) {
1248
        n += decode_drc_channel_exclusions(che_drc, gb);
1249
    }
1250

    
1251
    /* drc_bands_present? */
1252
    if (get_bits1(gb)) {
1253
        che_drc->band_incr            = get_bits(gb, 4);
1254
        che_drc->interpolation_scheme = get_bits(gb, 4);
1255
        n++;
1256
        drc_num_bands += che_drc->band_incr;
1257
        for (i = 0; i < drc_num_bands; i++) {
1258
            che_drc->band_top[i] = get_bits(gb, 8);
1259
            n++;
1260
        }
1261
    }
1262

    
1263
    /* prog_ref_level_present? */
1264
    if (get_bits1(gb)) {
1265
        che_drc->prog_ref_level = get_bits(gb, 7);
1266
        skip_bits1(gb); // prog_ref_level_reserved_bits
1267
        n++;
1268
    }
1269

    
1270
    for (i = 0; i < drc_num_bands; i++) {
1271
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1272
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1273
        n++;
1274
    }
1275

    
1276
    return n;
1277
}
1278

    
1279
/**
1280
 * Decode extension data (incomplete); reference: table 4.51.
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_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
1287
    int crc_flag = 0;
1288
    int res = cnt;
1289
    switch (get_bits(gb, 4)) { // extension type
1290
        case EXT_SBR_DATA_CRC:
1291
            crc_flag++;
1292
        case EXT_SBR_DATA:
1293
            res = decode_sbr_extension(ac, gb, crc_flag, cnt);
1294
            break;
1295
        case EXT_DYNAMIC_RANGE:
1296
            res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1297
            break;
1298
        case EXT_FILL:
1299
        case EXT_FILL_DATA:
1300
        case EXT_DATA_ELEMENT:
1301
        default:
1302
            skip_bits_long(gb, 8*cnt - 4);
1303
            break;
1304
    };
1305
    return res;
1306
}
1307

    
1308
/**
1309
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1310
 *
1311
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1312
 * @param   coef    spectral coefficients
1313
 */
1314
static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) {
1315
    const int mmm = FFMIN(ics->tns_max_bands,  ics->max_sfb);
1316
    int w, filt, m, i;
1317
    int bottom, top, order, start, end, size, inc;
1318
    float lpc[TNS_MAX_ORDER];
1319

    
1320
    for (w = 0; w < ics->num_windows; w++) {
1321
        bottom = ics->num_swb;
1322
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1323
            top    = bottom;
1324
            bottom = FFMAX(0, top - tns->length[w][filt]);
1325
            order  = tns->order[w][filt];
1326
            if (order == 0)
1327
                continue;
1328

    
1329
            // tns_decode_coef
1330
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1331

    
1332
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1333
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1334
            if ((size = end - start) <= 0)
1335
                continue;
1336
            if (tns->direction[w][filt]) {
1337
                inc = -1; start = end - 1;
1338
            } else {
1339
                inc = 1;
1340
            }
1341
            start += w * 128;
1342

    
1343
            // ar filter
1344
            for (m = 0; m < size; m++, start += inc)
1345
                for (i = 1; i <= FFMIN(m, order); i++)
1346
                    coef[start] -= coef[start - i*inc] * lpc[i-1];
1347
        }
1348
    }
1349
}
1350

    
1351
/**
1352
 * Conduct IMDCT and windowing.
1353
 */
1354
static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
1355
    IndividualChannelStream * ics = &sce->ics;
1356
    float * in = sce->coeffs;
1357
    float * out = sce->ret;
1358
    float * saved = sce->saved;
1359
    const float * swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1360
    const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1361
    const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1362
    float * buf = ac->buf_mdct;
1363
    float * temp = ac->temp;
1364
    int i;
1365

    
1366
    // imdct
1367
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1368
        if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1369
            av_log(ac->avccontext, AV_LOG_WARNING,
1370
                   "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1371
                   "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1372
        for (i = 0; i < 1024; i += 128)
1373
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1374
    } else
1375
        ff_imdct_half(&ac->mdct, buf, in);
1376

    
1377
    /* window overlapping
1378
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1379
     * and long to short transitions are considered to be short to short
1380
     * transitions. This leaves just two cases (long to long and short to short)
1381
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1382
     */
1383
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1384
        (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1385
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, ac->add_bias, 512);
1386
    } else {
1387
        for (i = 0; i < 448; i++)
1388
            out[i] = saved[i] + ac->add_bias;
1389

    
1390
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1391
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, ac->add_bias, 64);
1392
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      ac->add_bias, 64);
1393
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      ac->add_bias, 64);
1394
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      ac->add_bias, 64);
1395
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      ac->add_bias, 64);
1396
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1397
        } else {
1398
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, ac->add_bias, 64);
1399
            for (i = 576; i < 1024; i++)
1400
                out[i] = buf[i-512] + ac->add_bias;
1401
        }
1402
    }
1403

    
1404
    // buffer update
1405
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1406
        for (i = 0; i < 64; i++)
1407
            saved[i] = temp[64 + i] - ac->add_bias;
1408
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1409
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1410
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1411
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1412
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1413
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1414
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1415
    } else { // LONG_STOP or ONLY_LONG
1416
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1417
    }
1418
}
1419

    
1420
/**
1421
 * Apply dependent channel coupling (applied before IMDCT).
1422
 *
1423
 * @param   index   index into coupling gain array
1424
 */
1425
static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) {
1426
    IndividualChannelStream * ics = &cce->ch[0].ics;
1427
    const uint16_t * offsets = ics->swb_offset;
1428
    float * dest = target->coeffs;
1429
    const float * src = cce->ch[0].coeffs;
1430
    int g, i, group, k, idx = 0;
1431
    if(ac->m4ac.object_type == AOT_AAC_LTP) {
1432
        av_log(ac->avccontext, AV_LOG_ERROR,
1433
               "Dependent coupling is not supported together with LTP\n");
1434
        return;
1435
    }
1436
    for (g = 0; g < ics->num_window_groups; g++) {
1437
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1438
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1439
                for (group = 0; group < ics->group_len[g]; group++) {
1440
                    for (k = offsets[i]; k < offsets[i+1]; k++) {
1441
                        // XXX dsputil-ize
1442
                        dest[group*128+k] += cce->coup.gain[index][idx] * src[group*128+k];
1443
                    }
1444
                }
1445
            }
1446
        }
1447
        dest += ics->group_len[g]*128;
1448
        src  += ics->group_len[g]*128;
1449
    }
1450
}
1451

    
1452
/**
1453
 * Apply independent channel coupling (applied after IMDCT).
1454
 *
1455
 * @param   index   index into coupling gain array
1456
 */
1457
static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) {
1458
    int i;
1459
    const float gain = cce->coup.gain[index][0];
1460
    const float bias = ac->add_bias;
1461
    const float* src = cce->ch[0].ret;
1462
    float* dest = target->ret;
1463

    
1464
    for (i = 0; i < 1024; i++)
1465
        dest[i] += gain * (src[i] - bias);
1466
}
1467

    
1468
/**
1469
 * channel coupling transformation interface
1470
 *
1471
 * @param   index   index into coupling gain array
1472
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1473
 */
1474
static void apply_channel_coupling(AACContext * ac, ChannelElement * cc,
1475
        enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point,
1476
        void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index))
1477
{
1478
    int i, c;
1479

    
1480
    for (i = 0; i < MAX_ELEM_ID; i++) {
1481
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1482
        int index = 0;
1483

    
1484
        if (cce && cce->coup.coupling_point == coupling_point) {
1485
            ChannelCoupling * coup = &cce->coup;
1486

    
1487
            for (c = 0; c <= coup->num_coupled; c++) {
1488
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1489
                    if (coup->ch_select[c] != 1) {
1490
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1491
                        if (coup->ch_select[c] != 0)
1492
                            index++;
1493
                    }
1494
                    if (coup->ch_select[c] != 2)
1495
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1496
                } else
1497
                    index += 1 + (coup->ch_select[c] == 3);
1498
            }
1499
        }
1500
    }
1501
}
1502

    
1503
/**
1504
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1505
 */
1506
static void spectral_to_sample(AACContext * ac) {
1507
    int i, type;
1508
    for(type = 3; type >= 0; type--) {
1509
        for (i = 0; i < MAX_ELEM_ID; i++) {
1510
            ChannelElement *che = ac->che[type][i];
1511
            if(che) {
1512
                if(type <= TYPE_CPE)
1513
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1514
                if(che->ch[0].tns.present)
1515
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1516
                if(che->ch[1].tns.present)
1517
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1518
                if(type <= TYPE_CPE)
1519
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1520
                if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
1521
                    imdct_and_windowing(ac, &che->ch[0]);
1522
                if(type == TYPE_CPE)
1523
                    imdct_and_windowing(ac, &che->ch[1]);
1524
                if(type <= TYPE_CCE)
1525
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1526
            }
1527
        }
1528
    }
1529
}
1530

    
1531
static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) {
1532

    
1533
    int size;
1534
    AACADTSHeaderInfo hdr_info;
1535

    
1536
    size = ff_aac_parse_header(gb, &hdr_info);
1537
    if (size > 0) {
1538
        if (hdr_info.chan_config)
1539
            ac->m4ac.chan_config = hdr_info.chan_config;
1540
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
1541
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
1542
        ac->m4ac.object_type     = hdr_info.object_type;
1543
    }
1544
    if (hdr_info.num_aac_frames == 1) {
1545
        if (!hdr_info.crc_absent)
1546
            skip_bits(gb, 16);
1547
    } else {
1548
        ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1549
        return -1;
1550
    }
1551
    return size;
1552
}
1553

    
1554
static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) {
1555
    AACContext * ac = avccontext->priv_data;
1556
    GetBitContext gb;
1557
    enum RawDataBlockType elem_type;
1558
    int err, elem_id, data_size_tmp;
1559

    
1560
    init_get_bits(&gb, buf, buf_size*8);
1561

    
1562
    if (show_bits(&gb, 12) == 0xfff) {
1563
        if ((err = parse_adts_frame_header(ac, &gb)) < 0) {
1564
            av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1565
            return -1;
1566
        }
1567
        if (ac->m4ac.sampling_index > 12) {
1568
            av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1569
            return -1;
1570
        }
1571
    }
1572

    
1573
    // parse
1574
    while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1575
        elem_id = get_bits(&gb, 4);
1576
        err = -1;
1577

    
1578
        if(elem_type == TYPE_SCE && elem_id == 1 &&
1579
                !ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) {
1580
            /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
1581
               instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
1582
               encountered such a stream, transfer the LFE[0] element to SCE[1] */
1583
            ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0];
1584
            ac->che[TYPE_LFE][0] = NULL;
1585
        }
1586
        if(elem_type < TYPE_DSE && !ac->che[elem_type][elem_id]) {
1587
            av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1588
            return -1;
1589
        }
1590

    
1591
        switch (elem_type) {
1592

    
1593
        case TYPE_SCE:
1594
            err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0);
1595
            break;
1596

    
1597
        case TYPE_CPE:
1598
            err = decode_cpe(ac, &gb, elem_id);
1599
            break;
1600

    
1601
        case TYPE_CCE:
1602
            err = decode_cce(ac, &gb, ac->che[TYPE_CCE][elem_id]);
1603
            break;
1604

    
1605
        case TYPE_LFE:
1606
            err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0);
1607
            break;
1608

    
1609
        case TYPE_DSE:
1610
            skip_data_stream_element(&gb);
1611
            err = 0;
1612
            break;
1613

    
1614
        case TYPE_PCE:
1615
        {
1616
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1617
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1618
            if((err = decode_pce(ac, new_che_pos, &gb)))
1619
                break;
1620
            err = output_configure(ac, ac->che_pos, new_che_pos);
1621
            break;
1622
        }
1623

    
1624
        case TYPE_FIL:
1625
            if (elem_id == 15)
1626
                elem_id += get_bits(&gb, 8) - 1;
1627
            while (elem_id > 0)
1628
                elem_id -= decode_extension_payload(ac, &gb, elem_id);
1629
            err = 0; /* FIXME */
1630
            break;
1631

    
1632
        default:
1633
            err = -1; /* should not happen, but keeps compiler happy */
1634
            break;
1635
        }
1636

    
1637
        if(err)
1638
            return err;
1639
    }
1640

    
1641
    spectral_to_sample(ac);
1642

    
1643
    if (!ac->is_saved) {
1644
        ac->is_saved = 1;
1645
        *data_size = 0;
1646
        return buf_size;
1647
    }
1648

    
1649
    data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
1650
    if(*data_size < data_size_tmp) {
1651
        av_log(avccontext, AV_LOG_ERROR,
1652
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
1653
               *data_size, data_size_tmp);
1654
        return -1;
1655
    }
1656
    *data_size = data_size_tmp;
1657

    
1658
    ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
1659

    
1660
    return buf_size;
1661
}
1662

    
1663
static av_cold int aac_decode_close(AVCodecContext * avccontext) {
1664
    AACContext * ac = avccontext->priv_data;
1665
    int i, type;
1666

    
1667
    for (i = 0; i < MAX_ELEM_ID; i++) {
1668
        for(type = 0; type < 4; type++)
1669
            av_freep(&ac->che[type][i]);
1670
    }
1671

    
1672
    ff_mdct_end(&ac->mdct);
1673
    ff_mdct_end(&ac->mdct_small);
1674
    return 0 ;
1675
}
1676

    
1677
AVCodec aac_decoder = {
1678
    "aac",
1679
    CODEC_TYPE_AUDIO,
1680
    CODEC_ID_AAC,
1681
    sizeof(AACContext),
1682
    aac_decode_init,
1683
    NULL,
1684
    aac_decode_close,
1685
    aac_decode_frame,
1686
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
1687
    .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
1688
};