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ffmpeg / libavcodec / aacdec.c @ e22910b2

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1
/*
2
 * AAC decoder
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 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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 *
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 * AAC LATM decoder
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 * Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
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 * Copyright (c) 2010      Janne Grunau <janne-ffmpeg@jannau.net>
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 *
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 * This file is part of FFmpeg.
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 *
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 * FFmpeg is free software; you can redistribute it and/or
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 * modify it under the terms of the GNU Lesser General Public
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 * License as published by the Free Software Foundation; either
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 * version 2.1 of the License, or (at your option) any later version.
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 *
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 * FFmpeg is distributed in the hope that it will be useful,
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 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
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 * Lesser General Public License for more details.
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 *
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 * You should have received a copy of the GNU Lesser General Public
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 * License along with FFmpeg; if not, write to the Free Software
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 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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 */
26

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

    
34
/*
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 * supported tools
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 *
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 * Support?             Name
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 * N (code in SoC repo) gain control
39
 * 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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 * Y                    Long Term Prediction
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 * Y                    intensity stereo
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 * Y                    channel coupling
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 * Y                    frequency domain prediction
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 * Y                    Perceptual Noise Substitution
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 * Y                    Mid/Side stereo
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 * N                    Scalable Inverse AAC Quantization
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 * N                    Frequency Selective Switch
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 * N                    upsampling filter
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 * Y                    quantization & coding - AAC
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 * N                    quantization & coding - TwinVQ
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 * N                    quantization & coding - BSAC
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 * N                    AAC Error Resilience tools
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 * N                    Error Resilience payload syntax
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 * N                    Error Protection tool
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 * N                    CELP
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 * N                    Silence Compression
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 * N                    HVXC
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 * N                    HVXC 4kbits/s VR
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 * N                    Structured Audio tools
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 * N                    Structured Audio Sample Bank Format
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 * N                    MIDI
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 * N                    Harmonic and Individual Lines plus Noise
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 * N                    Text-To-Speech Interface
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 * Y                    Spectral Band Replication
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 * Y (not in this code) Layer-1
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 * Y (not in this code) Layer-2
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 * Y (not in this code) Layer-3
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 * N                    SinuSoidal Coding (Transient, Sinusoid, Noise)
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 * Y                    Parametric Stereo
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 * N                    Direct Stream Transfer
76
 *
77
 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
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 *       - HE AAC v2 comprises LC AAC with Spectral Band Replication and
79
           Parametric Stereo.
80
 */
81

    
82

    
83
#include "avcodec.h"
84
#include "internal.h"
85
#include "get_bits.h"
86
#include "dsputil.h"
87
#include "fft.h"
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#include "fmtconvert.h"
89
#include "lpc.h"
90

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

    
254
        avctx->channel_layout = 0;
255
    }
256

    
257
    avctx->channels = channels;
258

    
259
    ac->output_configured = oc_type;
260

    
261
    return 0;
262
}
263

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

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

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

    
298
    sampling_index = get_bits(gb, 4);
299
    if (m4ac->sampling_index != sampling_index)
300
        av_log(avctx, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
301

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

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

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

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

    
322
    skip_bits_long(gb, 4 * num_assoc_data);
323

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

    
326
    align_get_bits(gb);
327

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

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

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

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

    
381
    return 0;
382
}
383

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

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

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

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

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

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

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

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

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

    
476
    skip_bits_long(&gb, i);
477

    
478
    switch (m4ac->object_type) {
479
    case AOT_AAC_MAIN:
480
    case AOT_AAC_LC:
481
    case AOT_AAC_LTP:
482
        if (decode_ga_specific_config(ac, avctx, &gb, m4ac, m4ac->chan_config))
483
            return -1;
484
        break;
485
    default:
486
        av_log(avctx, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
487
               m4ac->sbr == 1? "SBR+" : "", m4ac->object_type);
488
        return -1;
489
    }
490

    
491
    return get_bits_count(&gb);
492
}
493

    
494
/**
495
 * linear congruential pseudorandom number generator
496
 *
497
 * @param   previous_val    pointer to the current state of the generator
498
 *
499
 * @return  Returns a 32-bit pseudorandom integer
500
 */
501
static av_always_inline int lcg_random(int previous_val)
502
{
503
    return previous_val * 1664525 + 1013904223;
504
}
505

    
506
static av_always_inline void reset_predict_state(PredictorState *ps)
507
{
508
    ps->r0   = 0.0f;
509
    ps->r1   = 0.0f;
510
    ps->cor0 = 0.0f;
511
    ps->cor1 = 0.0f;
512
    ps->var0 = 1.0f;
513
    ps->var1 = 1.0f;
514
}
515

    
516
static void reset_all_predictors(PredictorState *ps)
517
{
518
    int i;
519
    for (i = 0; i < MAX_PREDICTORS; i++)
520
        reset_predict_state(&ps[i]);
521
}
522

    
523
static void reset_predictor_group(PredictorState *ps, int group_num)
524
{
525
    int i;
526
    for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
527
        reset_predict_state(&ps[i]);
528
}
529

    
530
#define AAC_INIT_VLC_STATIC(num, size) \
531
    INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
532
         ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
533
        ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
534
        size);
535

    
536
static av_cold int aac_decode_init(AVCodecContext *avctx)
537
{
538
    AACContext *ac = avctx->priv_data;
539

    
540
    ac->avctx = avctx;
541
    ac->m4ac.sample_rate = avctx->sample_rate;
542

    
543
    if (avctx->extradata_size > 0) {
544
        if (decode_audio_specific_config(ac, ac->avctx, &ac->m4ac,
545
                                         avctx->extradata,
546
                                         avctx->extradata_size) < 0)
547
            return -1;
548
    }
549

    
550
    avctx->sample_fmt = AV_SAMPLE_FMT_S16;
551

    
552
    AAC_INIT_VLC_STATIC( 0, 304);
553
    AAC_INIT_VLC_STATIC( 1, 270);
554
    AAC_INIT_VLC_STATIC( 2, 550);
555
    AAC_INIT_VLC_STATIC( 3, 300);
556
    AAC_INIT_VLC_STATIC( 4, 328);
557
    AAC_INIT_VLC_STATIC( 5, 294);
558
    AAC_INIT_VLC_STATIC( 6, 306);
559
    AAC_INIT_VLC_STATIC( 7, 268);
560
    AAC_INIT_VLC_STATIC( 8, 510);
561
    AAC_INIT_VLC_STATIC( 9, 366);
562
    AAC_INIT_VLC_STATIC(10, 462);
563

    
564
    ff_aac_sbr_init();
565

    
566
    dsputil_init(&ac->dsp, avctx);
567
    ff_fmt_convert_init(&ac->fmt_conv, avctx);
568

    
569
    ac->random_state = 0x1f2e3d4c;
570

    
571
    // -1024 - Compensate wrong IMDCT method.
572
    // 60    - Required to scale values to the correct range [-32768,32767]
573
    //         for float to int16 conversion. (1 << (60 / 4)) == 32768
574
    ac->sf_scale  = 1. / -1024.;
575
    ac->sf_offset = 60;
576

    
577
    ff_aac_tableinit();
578

    
579
    INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
580
                    ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
581
                    ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
582
                    352);
583

    
584
    ff_mdct_init(&ac->mdct,       11, 1, 1.0);
585
    ff_mdct_init(&ac->mdct_small,  8, 1, 1.0);
586
    ff_mdct_init(&ac->mdct_ltp,   11, 0, 1.0);
587
    // window initialization
588
    ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
589
    ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
590
    ff_init_ff_sine_windows(10);
591
    ff_init_ff_sine_windows( 7);
592

    
593
    cbrt_tableinit();
594

    
595
    return 0;
596
}
597

    
598
/**
599
 * Skip data_stream_element; reference: table 4.10.
600
 */
601
static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
602
{
603
    int byte_align = get_bits1(gb);
604
    int count = get_bits(gb, 8);
605
    if (count == 255)
606
        count += get_bits(gb, 8);
607
    if (byte_align)
608
        align_get_bits(gb);
609

    
610
    if (get_bits_left(gb) < 8 * count) {
611
        av_log(ac->avctx, AV_LOG_ERROR, overread_err);
612
        return -1;
613
    }
614
    skip_bits_long(gb, 8 * count);
615
    return 0;
616
}
617

    
618
static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
619
                             GetBitContext *gb)
620
{
621
    int sfb;
622
    if (get_bits1(gb)) {
623
        ics->predictor_reset_group = get_bits(gb, 5);
624
        if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
625
            av_log(ac->avctx, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
626
            return -1;
627
        }
628
    }
629
    for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
630
        ics->prediction_used[sfb] = get_bits1(gb);
631
    }
632
    return 0;
633
}
634

    
635
/**
636
 * Decode Long Term Prediction data; reference: table 4.xx.
637
 */
638
static void decode_ltp(AACContext *ac, LongTermPrediction *ltp,
639
                       GetBitContext *gb, uint8_t max_sfb)
640
{
641
    int sfb;
642

    
643
    ltp->lag  = get_bits(gb, 11);
644
    ltp->coef = ltp_coef[get_bits(gb, 3)] * ac->sf_scale;
645
    for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
646
        ltp->used[sfb] = get_bits1(gb);
647
}
648

    
649
/**
650
 * Decode Individual Channel Stream info; reference: table 4.6.
651
 *
652
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
653
 */
654
static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
655
                           GetBitContext *gb, int common_window)
656
{
657
    if (get_bits1(gb)) {
658
        av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
659
        memset(ics, 0, sizeof(IndividualChannelStream));
660
        return -1;
661
    }
662
    ics->window_sequence[1] = ics->window_sequence[0];
663
    ics->window_sequence[0] = get_bits(gb, 2);
664
    ics->use_kb_window[1]   = ics->use_kb_window[0];
665
    ics->use_kb_window[0]   = get_bits1(gb);
666
    ics->num_window_groups  = 1;
667
    ics->group_len[0]       = 1;
668
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
669
        int i;
670
        ics->max_sfb = get_bits(gb, 4);
671
        for (i = 0; i < 7; i++) {
672
            if (get_bits1(gb)) {
673
                ics->group_len[ics->num_window_groups - 1]++;
674
            } else {
675
                ics->num_window_groups++;
676
                ics->group_len[ics->num_window_groups - 1] = 1;
677
            }
678
        }
679
        ics->num_windows       = 8;
680
        ics->swb_offset        =    ff_swb_offset_128[ac->m4ac.sampling_index];
681
        ics->num_swb           =   ff_aac_num_swb_128[ac->m4ac.sampling_index];
682
        ics->tns_max_bands     = ff_tns_max_bands_128[ac->m4ac.sampling_index];
683
        ics->predictor_present = 0;
684
    } else {
685
        ics->max_sfb               = get_bits(gb, 6);
686
        ics->num_windows           = 1;
687
        ics->swb_offset            =    ff_swb_offset_1024[ac->m4ac.sampling_index];
688
        ics->num_swb               =   ff_aac_num_swb_1024[ac->m4ac.sampling_index];
689
        ics->tns_max_bands         = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
690
        ics->predictor_present     = get_bits1(gb);
691
        ics->predictor_reset_group = 0;
692
        if (ics->predictor_present) {
693
            if (ac->m4ac.object_type == AOT_AAC_MAIN) {
694
                if (decode_prediction(ac, ics, gb)) {
695
                    memset(ics, 0, sizeof(IndividualChannelStream));
696
                    return -1;
697
                }
698
            } else if (ac->m4ac.object_type == AOT_AAC_LC) {
699
                av_log(ac->avctx, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
700
                memset(ics, 0, sizeof(IndividualChannelStream));
701
                return -1;
702
            } else {
703
                if ((ics->ltp.present = get_bits(gb, 1)))
704
                    decode_ltp(ac, &ics->ltp, gb, ics->max_sfb);
705
            }
706
        }
707
    }
708

    
709
    if (ics->max_sfb > ics->num_swb) {
710
        av_log(ac->avctx, AV_LOG_ERROR,
711
               "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
712
               ics->max_sfb, ics->num_swb);
713
        memset(ics, 0, sizeof(IndividualChannelStream));
714
        return -1;
715
    }
716

    
717
    return 0;
718
}
719

    
720
/**
721
 * Decode band types (section_data payload); reference: table 4.46.
722
 *
723
 * @param   band_type           array of the used band type
724
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
725
 *
726
 * @return  Returns error status. 0 - OK, !0 - error
727
 */
728
static int decode_band_types(AACContext *ac, enum BandType band_type[120],
729
                             int band_type_run_end[120], GetBitContext *gb,
730
                             IndividualChannelStream *ics)
731
{
732
    int g, idx = 0;
733
    const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
734
    for (g = 0; g < ics->num_window_groups; g++) {
735
        int k = 0;
736
        while (k < ics->max_sfb) {
737
            uint8_t sect_end = k;
738
            int sect_len_incr;
739
            int sect_band_type = get_bits(gb, 4);
740
            if (sect_band_type == 12) {
741
                av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
742
                return -1;
743
            }
744
            while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
745
                sect_end += sect_len_incr;
746
            sect_end += sect_len_incr;
747
            if (get_bits_left(gb) < 0) {
748
                av_log(ac->avctx, AV_LOG_ERROR, overread_err);
749
                return -1;
750
            }
751
            if (sect_end > ics->max_sfb) {
752
                av_log(ac->avctx, AV_LOG_ERROR,
753
                       "Number of bands (%d) exceeds limit (%d).\n",
754
                       sect_end, ics->max_sfb);
755
                return -1;
756
            }
757
            for (; k < sect_end; k++) {
758
                band_type        [idx]   = sect_band_type;
759
                band_type_run_end[idx++] = sect_end;
760
            }
761
        }
762
    }
763
    return 0;
764
}
765

    
766
/**
767
 * Decode scalefactors; reference: table 4.47.
768
 *
769
 * @param   global_gain         first scalefactor value as scalefactors are differentially coded
770
 * @param   band_type           array of the used band type
771
 * @param   band_type_run_end   array of the last scalefactor band of a band type run
772
 * @param   sf                  array of scalefactors or intensity stereo positions
773
 *
774
 * @return  Returns error status. 0 - OK, !0 - error
775
 */
776
static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
777
                               unsigned int global_gain,
778
                               IndividualChannelStream *ics,
779
                               enum BandType band_type[120],
780
                               int band_type_run_end[120])
781
{
782
    const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
783
    int g, i, idx = 0;
784
    int offset[3] = { global_gain, global_gain - 90, 100 };
785
    int noise_flag = 1;
786
    static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
787
    for (g = 0; g < ics->num_window_groups; g++) {
788
        for (i = 0; i < ics->max_sfb;) {
789
            int run_end = band_type_run_end[idx];
790
            if (band_type[idx] == ZERO_BT) {
791
                for (; i < run_end; i++, idx++)
792
                    sf[idx] = 0.;
793
            } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
794
                for (; i < run_end; i++, idx++) {
795
                    offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
796
                    if (offset[2] > 255U) {
797
                        av_log(ac->avctx, AV_LOG_ERROR,
798
                               "%s (%d) out of range.\n", sf_str[2], offset[2]);
799
                        return -1;
800
                    }
801
                    sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
802
                }
803
            } else if (band_type[idx] == NOISE_BT) {
804
                for (; i < run_end; i++, idx++) {
805
                    if (noise_flag-- > 0)
806
                        offset[1] += get_bits(gb, 9) - 256;
807
                    else
808
                        offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
809
                    if (offset[1] > 255U) {
810
                        av_log(ac->avctx, AV_LOG_ERROR,
811
                               "%s (%d) out of range.\n", sf_str[1], offset[1]);
812
                        return -1;
813
                    }
814
                    sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
815
                }
816
            } else {
817
                for (; i < run_end; i++, idx++) {
818
                    offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
819
                    if (offset[0] > 255U) {
820
                        av_log(ac->avctx, AV_LOG_ERROR,
821
                               "%s (%d) out of range.\n", sf_str[0], offset[0]);
822
                        return -1;
823
                    }
824
                    sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
825
                }
826
            }
827
        }
828
    }
829
    return 0;
830
}
831

    
832
/**
833
 * Decode pulse data; reference: table 4.7.
834
 */
835
static int decode_pulses(Pulse *pulse, GetBitContext *gb,
836
                         const uint16_t *swb_offset, int num_swb)
837
{
838
    int i, pulse_swb;
839
    pulse->num_pulse = get_bits(gb, 2) + 1;
840
    pulse_swb        = get_bits(gb, 6);
841
    if (pulse_swb >= num_swb)
842
        return -1;
843
    pulse->pos[0]    = swb_offset[pulse_swb];
844
    pulse->pos[0]   += get_bits(gb, 5);
845
    if (pulse->pos[0] > 1023)
846
        return -1;
847
    pulse->amp[0]    = get_bits(gb, 4);
848
    for (i = 1; i < pulse->num_pulse; i++) {
849
        pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
850
        if (pulse->pos[i] > 1023)
851
            return -1;
852
        pulse->amp[i] = get_bits(gb, 4);
853
    }
854
    return 0;
855
}
856

    
857
/**
858
 * Decode Temporal Noise Shaping data; reference: table 4.48.
859
 *
860
 * @return  Returns error status. 0 - OK, !0 - error
861
 */
862
static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
863
                      GetBitContext *gb, const IndividualChannelStream *ics)
864
{
865
    int w, filt, i, coef_len, coef_res, coef_compress;
866
    const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
867
    const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
868
    for (w = 0; w < ics->num_windows; w++) {
869
        if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
870
            coef_res = get_bits1(gb);
871

    
872
            for (filt = 0; filt < tns->n_filt[w]; filt++) {
873
                int tmp2_idx;
874
                tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
875

    
876
                if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
877
                    av_log(ac->avctx, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.\n",
878
                           tns->order[w][filt], tns_max_order);
879
                    tns->order[w][filt] = 0;
880
                    return -1;
881
                }
882
                if (tns->order[w][filt]) {
883
                    tns->direction[w][filt] = get_bits1(gb);
884
                    coef_compress = get_bits1(gb);
885
                    coef_len = coef_res + 3 - coef_compress;
886
                    tmp2_idx = 2 * coef_compress + coef_res;
887

    
888
                    for (i = 0; i < tns->order[w][filt]; i++)
889
                        tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
890
                }
891
            }
892
        }
893
    }
894
    return 0;
895
}
896

    
897
/**
898
 * Decode Mid/Side data; reference: table 4.54.
899
 *
900
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
901
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
902
 *                      [3] reserved for scalable AAC
903
 */
904
static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
905
                                   int ms_present)
906
{
907
    int idx;
908
    if (ms_present == 1) {
909
        for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
910
            cpe->ms_mask[idx] = get_bits1(gb);
911
    } else if (ms_present == 2) {
912
        memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
913
    }
914
}
915

    
916
#ifndef VMUL2
917
static inline float *VMUL2(float *dst, const float *v, unsigned idx,
918
                           const float *scale)
919
{
920
    float s = *scale;
921
    *dst++ = v[idx    & 15] * s;
922
    *dst++ = v[idx>>4 & 15] * s;
923
    return dst;
924
}
925
#endif
926

    
927
#ifndef VMUL4
928
static inline float *VMUL4(float *dst, const float *v, unsigned idx,
929
                           const float *scale)
930
{
931
    float s = *scale;
932
    *dst++ = v[idx    & 3] * s;
933
    *dst++ = v[idx>>2 & 3] * s;
934
    *dst++ = v[idx>>4 & 3] * s;
935
    *dst++ = v[idx>>6 & 3] * s;
936
    return dst;
937
}
938
#endif
939

    
940
#ifndef VMUL2S
941
static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
942
                            unsigned sign, const float *scale)
943
{
944
    union float754 s0, s1;
945

    
946
    s0.f = s1.f = *scale;
947
    s0.i ^= sign >> 1 << 31;
948
    s1.i ^= sign      << 31;
949

    
950
    *dst++ = v[idx    & 15] * s0.f;
951
    *dst++ = v[idx>>4 & 15] * s1.f;
952

    
953
    return dst;
954
}
955
#endif
956

    
957
#ifndef VMUL4S
958
static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
959
                            unsigned sign, const float *scale)
960
{
961
    unsigned nz = idx >> 12;
962
    union float754 s = { .f = *scale };
963
    union float754 t;
964

    
965
    t.i = s.i ^ (sign & 1<<31);
966
    *dst++ = v[idx    & 3] * t.f;
967

    
968
    sign <<= nz & 1; nz >>= 1;
969
    t.i = s.i ^ (sign & 1<<31);
970
    *dst++ = v[idx>>2 & 3] * t.f;
971

    
972
    sign <<= nz & 1; nz >>= 1;
973
    t.i = s.i ^ (sign & 1<<31);
974
    *dst++ = v[idx>>4 & 3] * t.f;
975

    
976
    sign <<= nz & 1; nz >>= 1;
977
    t.i = s.i ^ (sign & 1<<31);
978
    *dst++ = v[idx>>6 & 3] * t.f;
979

    
980
    return dst;
981
}
982
#endif
983

    
984
/**
985
 * Decode spectral data; reference: table 4.50.
986
 * Dequantize and scale spectral data; reference: 4.6.3.3.
987
 *
988
 * @param   coef            array of dequantized, scaled spectral data
989
 * @param   sf              array of scalefactors or intensity stereo positions
990
 * @param   pulse_present   set if pulses are present
991
 * @param   pulse           pointer to pulse data struct
992
 * @param   band_type       array of the used band type
993
 *
994
 * @return  Returns error status. 0 - OK, !0 - error
995
 */
996
static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
997
                                       GetBitContext *gb, const float sf[120],
998
                                       int pulse_present, const Pulse *pulse,
999
                                       const IndividualChannelStream *ics,
1000
                                       enum BandType band_type[120])
1001
{
1002
    int i, k, g, idx = 0;
1003
    const int c = 1024 / ics->num_windows;
1004
    const uint16_t *offsets = ics->swb_offset;
1005
    float *coef_base = coef;
1006

    
1007
    for (g = 0; g < ics->num_windows; g++)
1008
        memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
1009

    
1010
    for (g = 0; g < ics->num_window_groups; g++) {
1011
        unsigned g_len = ics->group_len[g];
1012

    
1013
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1014
            const unsigned cbt_m1 = band_type[idx] - 1;
1015
            float *cfo = coef + offsets[i];
1016
            int off_len = offsets[i + 1] - offsets[i];
1017
            int group;
1018

    
1019
            if (cbt_m1 >= INTENSITY_BT2 - 1) {
1020
                for (group = 0; group < g_len; group++, cfo+=128) {
1021
                    memset(cfo, 0, off_len * sizeof(float));
1022
                }
1023
            } else if (cbt_m1 == NOISE_BT - 1) {
1024
                for (group = 0; group < g_len; group++, cfo+=128) {
1025
                    float scale;
1026
                    float band_energy;
1027

    
1028
                    for (k = 0; k < off_len; k++) {
1029
                        ac->random_state  = lcg_random(ac->random_state);
1030
                        cfo[k] = ac->random_state;
1031
                    }
1032

    
1033
                    band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1034
                    scale = sf[idx] / sqrtf(band_energy);
1035
                    ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1036
                }
1037
            } else {
1038
                const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1039
                const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1040
                VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1041
                OPEN_READER(re, gb);
1042

    
1043
                switch (cbt_m1 >> 1) {
1044
                case 0:
1045
                    for (group = 0; group < g_len; group++, cfo+=128) {
1046
                        float *cf = cfo;
1047
                        int len = off_len;
1048

    
1049
                        do {
1050
                            int code;
1051
                            unsigned cb_idx;
1052

    
1053
                            UPDATE_CACHE(re, gb);
1054
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1055
                            cb_idx = cb_vector_idx[code];
1056
                            cf = VMUL4(cf, vq, cb_idx, sf + idx);
1057
                        } while (len -= 4);
1058
                    }
1059
                    break;
1060

    
1061
                case 1:
1062
                    for (group = 0; group < g_len; group++, cfo+=128) {
1063
                        float *cf = cfo;
1064
                        int len = off_len;
1065

    
1066
                        do {
1067
                            int code;
1068
                            unsigned nnz;
1069
                            unsigned cb_idx;
1070
                            uint32_t bits;
1071

    
1072
                            UPDATE_CACHE(re, gb);
1073
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1074
                            cb_idx = cb_vector_idx[code];
1075
                            nnz = cb_idx >> 8 & 15;
1076
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1077
                            LAST_SKIP_BITS(re, gb, nnz);
1078
                            cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1079
                        } while (len -= 4);
1080
                    }
1081
                    break;
1082

    
1083
                case 2:
1084
                    for (group = 0; group < g_len; group++, cfo+=128) {
1085
                        float *cf = cfo;
1086
                        int len = off_len;
1087

    
1088
                        do {
1089
                            int code;
1090
                            unsigned cb_idx;
1091

    
1092
                            UPDATE_CACHE(re, gb);
1093
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1094
                            cb_idx = cb_vector_idx[code];
1095
                            cf = VMUL2(cf, vq, cb_idx, sf + idx);
1096
                        } while (len -= 2);
1097
                    }
1098
                    break;
1099

    
1100
                case 3:
1101
                case 4:
1102
                    for (group = 0; group < g_len; group++, cfo+=128) {
1103
                        float *cf = cfo;
1104
                        int len = off_len;
1105

    
1106
                        do {
1107
                            int code;
1108
                            unsigned nnz;
1109
                            unsigned cb_idx;
1110
                            unsigned sign;
1111

    
1112
                            UPDATE_CACHE(re, gb);
1113
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1114
                            cb_idx = cb_vector_idx[code];
1115
                            nnz = cb_idx >> 8 & 15;
1116
                            sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1117
                            LAST_SKIP_BITS(re, gb, nnz);
1118
                            cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1119
                        } while (len -= 2);
1120
                    }
1121
                    break;
1122

    
1123
                default:
1124
                    for (group = 0; group < g_len; group++, cfo+=128) {
1125
                        float *cf = cfo;
1126
                        uint32_t *icf = (uint32_t *) cf;
1127
                        int len = off_len;
1128

    
1129
                        do {
1130
                            int code;
1131
                            unsigned nzt, nnz;
1132
                            unsigned cb_idx;
1133
                            uint32_t bits;
1134
                            int j;
1135

    
1136
                            UPDATE_CACHE(re, gb);
1137
                            GET_VLC(code, re, gb, vlc_tab, 8, 2);
1138

    
1139
                            if (!code) {
1140
                                *icf++ = 0;
1141
                                *icf++ = 0;
1142
                                continue;
1143
                            }
1144

    
1145
                            cb_idx = cb_vector_idx[code];
1146
                            nnz = cb_idx >> 12;
1147
                            nzt = cb_idx >> 8;
1148
                            bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1149
                            LAST_SKIP_BITS(re, gb, nnz);
1150

    
1151
                            for (j = 0; j < 2; j++) {
1152
                                if (nzt & 1<<j) {
1153
                                    uint32_t b;
1154
                                    int n;
1155
                                    /* The total length of escape_sequence must be < 22 bits according
1156
                                       to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1157
                                    UPDATE_CACHE(re, gb);
1158
                                    b = GET_CACHE(re, gb);
1159
                                    b = 31 - av_log2(~b);
1160

    
1161
                                    if (b > 8) {
1162
                                        av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1163
                                        return -1;
1164
                                    }
1165

    
1166
                                    SKIP_BITS(re, gb, b + 1);
1167
                                    b += 4;
1168
                                    n = (1 << b) + SHOW_UBITS(re, gb, b);
1169
                                    LAST_SKIP_BITS(re, gb, b);
1170
                                    *icf++ = cbrt_tab[n] | (bits & 1<<31);
1171
                                    bits <<= 1;
1172
                                } else {
1173
                                    unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1174
                                    *icf++ = (bits & 1<<31) | v;
1175
                                    bits <<= !!v;
1176
                                }
1177
                                cb_idx >>= 4;
1178
                            }
1179
                        } while (len -= 2);
1180

    
1181
                        ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1182
                    }
1183
                }
1184

    
1185
                CLOSE_READER(re, gb);
1186
            }
1187
        }
1188
        coef += g_len << 7;
1189
    }
1190

    
1191
    if (pulse_present) {
1192
        idx = 0;
1193
        for (i = 0; i < pulse->num_pulse; i++) {
1194
            float co = coef_base[ pulse->pos[i] ];
1195
            while (offsets[idx + 1] <= pulse->pos[i])
1196
                idx++;
1197
            if (band_type[idx] != NOISE_BT && sf[idx]) {
1198
                float ico = -pulse->amp[i];
1199
                if (co) {
1200
                    co /= sf[idx];
1201
                    ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1202
                }
1203
                coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1204
            }
1205
        }
1206
    }
1207
    return 0;
1208
}
1209

    
1210
static av_always_inline float flt16_round(float pf)
1211
{
1212
    union float754 tmp;
1213
    tmp.f = pf;
1214
    tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1215
    return tmp.f;
1216
}
1217

    
1218
static av_always_inline float flt16_even(float pf)
1219
{
1220
    union float754 tmp;
1221
    tmp.f = pf;
1222
    tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1223
    return tmp.f;
1224
}
1225

    
1226
static av_always_inline float flt16_trunc(float pf)
1227
{
1228
    union float754 pun;
1229
    pun.f = pf;
1230
    pun.i &= 0xFFFF0000U;
1231
    return pun.f;
1232
}
1233

    
1234
static av_always_inline void predict(PredictorState *ps, float *coef,
1235
                                     float sf_scale, float inv_sf_scale,
1236
                    int output_enable)
1237
{
1238
    const float a     = 0.953125; // 61.0 / 64
1239
    const float alpha = 0.90625;  // 29.0 / 32
1240
    float e0, e1;
1241
    float pv;
1242
    float k1, k2;
1243
    float   r0 = ps->r0,     r1 = ps->r1;
1244
    float cor0 = ps->cor0, cor1 = ps->cor1;
1245
    float var0 = ps->var0, var1 = ps->var1;
1246

    
1247
    k1 = var0 > 1 ? cor0 * flt16_even(a / var0) : 0;
1248
    k2 = var1 > 1 ? cor1 * flt16_even(a / var1) : 0;
1249

    
1250
    pv = flt16_round(k1 * r0 + k2 * r1);
1251
    if (output_enable)
1252
        *coef += pv * sf_scale;
1253

    
1254
    e0 = *coef * inv_sf_scale;
1255
    e1 = e0 - k1 * r0;
1256

    
1257
    ps->cor1 = flt16_trunc(alpha * cor1 + r1 * e1);
1258
    ps->var1 = flt16_trunc(alpha * var1 + 0.5f * (r1 * r1 + e1 * e1));
1259
    ps->cor0 = flt16_trunc(alpha * cor0 + r0 * e0);
1260
    ps->var0 = flt16_trunc(alpha * var0 + 0.5f * (r0 * r0 + e0 * e0));
1261

    
1262
    ps->r1 = flt16_trunc(a * (r0 - k1 * e0));
1263
    ps->r0 = flt16_trunc(a * e0);
1264
}
1265

    
1266
/**
1267
 * Apply AAC-Main style frequency domain prediction.
1268
 */
1269
static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1270
{
1271
    int sfb, k;
1272
    float sf_scale = ac->sf_scale, inv_sf_scale = 1 / ac->sf_scale;
1273

    
1274
    if (!sce->ics.predictor_initialized) {
1275
        reset_all_predictors(sce->predictor_state);
1276
        sce->ics.predictor_initialized = 1;
1277
    }
1278

    
1279
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1280
        for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1281
            for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1282
                predict(&sce->predictor_state[k], &sce->coeffs[k],
1283
                        sf_scale, inv_sf_scale,
1284
                        sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1285
            }
1286
        }
1287
        if (sce->ics.predictor_reset_group)
1288
            reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1289
    } else
1290
        reset_all_predictors(sce->predictor_state);
1291
}
1292

    
1293
/**
1294
 * Decode an individual_channel_stream payload; reference: table 4.44.
1295
 *
1296
 * @param   common_window   Channels have independent [0], or shared [1], Individual Channel Stream information.
1297
 * @param   scale_flag      scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1298
 *
1299
 * @return  Returns error status. 0 - OK, !0 - error
1300
 */
1301
static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1302
                      GetBitContext *gb, int common_window, int scale_flag)
1303
{
1304
    Pulse pulse;
1305
    TemporalNoiseShaping    *tns = &sce->tns;
1306
    IndividualChannelStream *ics = &sce->ics;
1307
    float *out = sce->coeffs;
1308
    int global_gain, pulse_present = 0;
1309

    
1310
    /* This assignment is to silence a GCC warning about the variable being used
1311
     * uninitialized when in fact it always is.
1312
     */
1313
    pulse.num_pulse = 0;
1314

    
1315
    global_gain = get_bits(gb, 8);
1316

    
1317
    if (!common_window && !scale_flag) {
1318
        if (decode_ics_info(ac, ics, gb, 0) < 0)
1319
            return -1;
1320
    }
1321

    
1322
    if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1323
        return -1;
1324
    if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1325
        return -1;
1326

    
1327
    pulse_present = 0;
1328
    if (!scale_flag) {
1329
        if ((pulse_present = get_bits1(gb))) {
1330
            if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1331
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1332
                return -1;
1333
            }
1334
            if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1335
                av_log(ac->avctx, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1336
                return -1;
1337
            }
1338
        }
1339
        if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1340
            return -1;
1341
        if (get_bits1(gb)) {
1342
            av_log_missing_feature(ac->avctx, "SSR", 1);
1343
            return -1;
1344
        }
1345
    }
1346

    
1347
    if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1348
        return -1;
1349

    
1350
    if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1351
        apply_prediction(ac, sce);
1352

    
1353
    return 0;
1354
}
1355

    
1356
/**
1357
 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1358
 */
1359
static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1360
{
1361
    const IndividualChannelStream *ics = &cpe->ch[0].ics;
1362
    float *ch0 = cpe->ch[0].coeffs;
1363
    float *ch1 = cpe->ch[1].coeffs;
1364
    int g, i, group, idx = 0;
1365
    const uint16_t *offsets = ics->swb_offset;
1366
    for (g = 0; g < ics->num_window_groups; g++) {
1367
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1368
            if (cpe->ms_mask[idx] &&
1369
                    cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1370
                for (group = 0; group < ics->group_len[g]; group++) {
1371
                    ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1372
                                              ch1 + group * 128 + offsets[i],
1373
                                              offsets[i+1] - offsets[i]);
1374
                }
1375
            }
1376
        }
1377
        ch0 += ics->group_len[g] * 128;
1378
        ch1 += ics->group_len[g] * 128;
1379
    }
1380
}
1381

    
1382
/**
1383
 * intensity stereo decoding; reference: 4.6.8.2.3
1384
 *
1385
 * @param   ms_present  Indicates mid/side stereo presence. [0] mask is all 0s;
1386
 *                      [1] mask is decoded from bitstream; [2] mask is all 1s;
1387
 *                      [3] reserved for scalable AAC
1388
 */
1389
static void apply_intensity_stereo(AACContext *ac, ChannelElement *cpe, int ms_present)
1390
{
1391
    const IndividualChannelStream *ics = &cpe->ch[1].ics;
1392
    SingleChannelElement         *sce1 = &cpe->ch[1];
1393
    float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1394
    const uint16_t *offsets = ics->swb_offset;
1395
    int g, group, i, idx = 0;
1396
    int c;
1397
    float scale;
1398
    for (g = 0; g < ics->num_window_groups; g++) {
1399
        for (i = 0; i < ics->max_sfb;) {
1400
            if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1401
                const int bt_run_end = sce1->band_type_run_end[idx];
1402
                for (; i < bt_run_end; i++, idx++) {
1403
                    c = -1 + 2 * (sce1->band_type[idx] - 14);
1404
                    if (ms_present)
1405
                        c *= 1 - 2 * cpe->ms_mask[idx];
1406
                    scale = c * sce1->sf[idx];
1407
                    for (group = 0; group < ics->group_len[g]; group++)
1408
                        ac->dsp.vector_fmul_scalar(coef1 + group * 128 + offsets[i],
1409
                                                   coef0 + group * 128 + offsets[i],
1410
                                                   scale,
1411
                                                   offsets[i + 1] - offsets[i]);
1412
                }
1413
            } else {
1414
                int bt_run_end = sce1->band_type_run_end[idx];
1415
                idx += bt_run_end - i;
1416
                i    = bt_run_end;
1417
            }
1418
        }
1419
        coef0 += ics->group_len[g] * 128;
1420
        coef1 += ics->group_len[g] * 128;
1421
    }
1422
}
1423

    
1424
/**
1425
 * Decode a channel_pair_element; reference: table 4.4.
1426
 *
1427
 * @return  Returns error status. 0 - OK, !0 - error
1428
 */
1429
static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1430
{
1431
    int i, ret, common_window, ms_present = 0;
1432

    
1433
    common_window = get_bits1(gb);
1434
    if (common_window) {
1435
        if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1436
            return -1;
1437
        i = cpe->ch[1].ics.use_kb_window[0];
1438
        cpe->ch[1].ics = cpe->ch[0].ics;
1439
        cpe->ch[1].ics.use_kb_window[1] = i;
1440
        if (cpe->ch[1].ics.predictor_present && (ac->m4ac.object_type != AOT_AAC_MAIN))
1441
            if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
1442
                decode_ltp(ac, &cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
1443
        ms_present = get_bits(gb, 2);
1444
        if (ms_present == 3) {
1445
            av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1446
            return -1;
1447
        } else if (ms_present)
1448
            decode_mid_side_stereo(cpe, gb, ms_present);
1449
    }
1450
    if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1451
        return ret;
1452
    if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1453
        return ret;
1454

    
1455
    if (common_window) {
1456
        if (ms_present)
1457
            apply_mid_side_stereo(ac, cpe);
1458
        if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1459
            apply_prediction(ac, &cpe->ch[0]);
1460
            apply_prediction(ac, &cpe->ch[1]);
1461
        }
1462
    }
1463

    
1464
    apply_intensity_stereo(ac, cpe, ms_present);
1465
    return 0;
1466
}
1467

    
1468
static const float cce_scale[] = {
1469
    1.09050773266525765921, //2^(1/8)
1470
    1.18920711500272106672, //2^(1/4)
1471
    M_SQRT2,
1472
    2,
1473
};
1474

    
1475
/**
1476
 * Decode coupling_channel_element; reference: table 4.8.
1477
 *
1478
 * @return  Returns error status. 0 - OK, !0 - error
1479
 */
1480
static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1481
{
1482
    int num_gain = 0;
1483
    int c, g, sfb, ret;
1484
    int sign;
1485
    float scale;
1486
    SingleChannelElement *sce = &che->ch[0];
1487
    ChannelCoupling     *coup = &che->coup;
1488

    
1489
    coup->coupling_point = 2 * get_bits1(gb);
1490
    coup->num_coupled = get_bits(gb, 3);
1491
    for (c = 0; c <= coup->num_coupled; c++) {
1492
        num_gain++;
1493
        coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1494
        coup->id_select[c] = get_bits(gb, 4);
1495
        if (coup->type[c] == TYPE_CPE) {
1496
            coup->ch_select[c] = get_bits(gb, 2);
1497
            if (coup->ch_select[c] == 3)
1498
                num_gain++;
1499
        } else
1500
            coup->ch_select[c] = 2;
1501
    }
1502
    coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1503

    
1504
    sign  = get_bits(gb, 1);
1505
    scale = cce_scale[get_bits(gb, 2)];
1506

    
1507
    if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1508
        return ret;
1509

    
1510
    for (c = 0; c < num_gain; c++) {
1511
        int idx  = 0;
1512
        int cge  = 1;
1513
        int gain = 0;
1514
        float gain_cache = 1.;
1515
        if (c) {
1516
            cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1517
            gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1518
            gain_cache = powf(scale, -gain);
1519
        }
1520
        if (coup->coupling_point == AFTER_IMDCT) {
1521
            coup->gain[c][0] = gain_cache;
1522
        } else {
1523
            for (g = 0; g < sce->ics.num_window_groups; g++) {
1524
                for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1525
                    if (sce->band_type[idx] != ZERO_BT) {
1526
                        if (!cge) {
1527
                            int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1528
                            if (t) {
1529
                                int s = 1;
1530
                                t = gain += t;
1531
                                if (sign) {
1532
                                    s  -= 2 * (t & 0x1);
1533
                                    t >>= 1;
1534
                                }
1535
                                gain_cache = powf(scale, -t) * s;
1536
                            }
1537
                        }
1538
                        coup->gain[c][idx] = gain_cache;
1539
                    }
1540
                }
1541
            }
1542
        }
1543
    }
1544
    return 0;
1545
}
1546

    
1547
/**
1548
 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1549
 *
1550
 * @return  Returns number of bytes consumed.
1551
 */
1552
static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1553
                                         GetBitContext *gb)
1554
{
1555
    int i;
1556
    int num_excl_chan = 0;
1557

    
1558
    do {
1559
        for (i = 0; i < 7; i++)
1560
            che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1561
    } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1562

    
1563
    return num_excl_chan / 7;
1564
}
1565

    
1566
/**
1567
 * Decode dynamic range information; reference: table 4.52.
1568
 *
1569
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1570
 *
1571
 * @return  Returns number of bytes consumed.
1572
 */
1573
static int decode_dynamic_range(DynamicRangeControl *che_drc,
1574
                                GetBitContext *gb, int cnt)
1575
{
1576
    int n             = 1;
1577
    int drc_num_bands = 1;
1578
    int i;
1579

    
1580
    /* pce_tag_present? */
1581
    if (get_bits1(gb)) {
1582
        che_drc->pce_instance_tag  = get_bits(gb, 4);
1583
        skip_bits(gb, 4); // tag_reserved_bits
1584
        n++;
1585
    }
1586

    
1587
    /* excluded_chns_present? */
1588
    if (get_bits1(gb)) {
1589
        n += decode_drc_channel_exclusions(che_drc, gb);
1590
    }
1591

    
1592
    /* drc_bands_present? */
1593
    if (get_bits1(gb)) {
1594
        che_drc->band_incr            = get_bits(gb, 4);
1595
        che_drc->interpolation_scheme = get_bits(gb, 4);
1596
        n++;
1597
        drc_num_bands += che_drc->band_incr;
1598
        for (i = 0; i < drc_num_bands; i++) {
1599
            che_drc->band_top[i] = get_bits(gb, 8);
1600
            n++;
1601
        }
1602
    }
1603

    
1604
    /* prog_ref_level_present? */
1605
    if (get_bits1(gb)) {
1606
        che_drc->prog_ref_level = get_bits(gb, 7);
1607
        skip_bits1(gb); // prog_ref_level_reserved_bits
1608
        n++;
1609
    }
1610

    
1611
    for (i = 0; i < drc_num_bands; i++) {
1612
        che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1613
        che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1614
        n++;
1615
    }
1616

    
1617
    return n;
1618
}
1619

    
1620
/**
1621
 * Decode extension data (incomplete); reference: table 4.51.
1622
 *
1623
 * @param   cnt length of TYPE_FIL syntactic element in bytes
1624
 *
1625
 * @return Returns number of bytes consumed
1626
 */
1627
static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
1628
                                    ChannelElement *che, enum RawDataBlockType elem_type)
1629
{
1630
    int crc_flag = 0;
1631
    int res = cnt;
1632
    switch (get_bits(gb, 4)) { // extension type
1633
    case EXT_SBR_DATA_CRC:
1634
        crc_flag++;
1635
    case EXT_SBR_DATA:
1636
        if (!che) {
1637
            av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
1638
            return res;
1639
        } else if (!ac->m4ac.sbr) {
1640
            av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
1641
            skip_bits_long(gb, 8 * cnt - 4);
1642
            return res;
1643
        } else if (ac->m4ac.sbr == -1 && ac->output_configured == OC_LOCKED) {
1644
            av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
1645
            skip_bits_long(gb, 8 * cnt - 4);
1646
            return res;
1647
        } else if (ac->m4ac.ps == -1 && ac->output_configured < OC_LOCKED && ac->avctx->channels == 1) {
1648
            ac->m4ac.sbr = 1;
1649
            ac->m4ac.ps = 1;
1650
            output_configure(ac, ac->che_pos, ac->che_pos, ac->m4ac.chan_config, ac->output_configured);
1651
        } else {
1652
            ac->m4ac.sbr = 1;
1653
        }
1654
        res = ff_decode_sbr_extension(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
1655
        break;
1656
    case EXT_DYNAMIC_RANGE:
1657
        res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1658
        break;
1659
    case EXT_FILL:
1660
    case EXT_FILL_DATA:
1661
    case EXT_DATA_ELEMENT:
1662
    default:
1663
        skip_bits_long(gb, 8 * cnt - 4);
1664
        break;
1665
    };
1666
    return res;
1667
}
1668

    
1669
/**
1670
 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1671
 *
1672
 * @param   decode  1 if tool is used normally, 0 if tool is used in LTP.
1673
 * @param   coef    spectral coefficients
1674
 */
1675
static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1676
                      IndividualChannelStream *ics, int decode)
1677
{
1678
    const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1679
    int w, filt, m, i;
1680
    int bottom, top, order, start, end, size, inc;
1681
    float lpc[TNS_MAX_ORDER];
1682
    float tmp[TNS_MAX_ORDER];
1683

    
1684
    for (w = 0; w < ics->num_windows; w++) {
1685
        bottom = ics->num_swb;
1686
        for (filt = 0; filt < tns->n_filt[w]; filt++) {
1687
            top    = bottom;
1688
            bottom = FFMAX(0, top - tns->length[w][filt]);
1689
            order  = tns->order[w][filt];
1690
            if (order == 0)
1691
                continue;
1692

    
1693
            // tns_decode_coef
1694
            compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1695

    
1696
            start = ics->swb_offset[FFMIN(bottom, mmm)];
1697
            end   = ics->swb_offset[FFMIN(   top, mmm)];
1698
            if ((size = end - start) <= 0)
1699
                continue;
1700
            if (tns->direction[w][filt]) {
1701
                inc = -1;
1702
                start = end - 1;
1703
            } else {
1704
                inc = 1;
1705
            }
1706
            start += w * 128;
1707

    
1708
            if (decode) {
1709
                // ar filter
1710
                for (m = 0; m < size; m++, start += inc)
1711
                    for (i = 1; i <= FFMIN(m, order); i++)
1712
                        coef[start] -= coef[start - i * inc] * lpc[i - 1];
1713
            } else {
1714
                // ma filter
1715
                for (m = 0; m < size; m++, start += inc) {
1716
                    tmp[0] = coef[start];
1717
                    for (i = 1; i <= FFMIN(m, order); i++)
1718
                        coef[start] += tmp[i] * lpc[i - 1];
1719
                    for (i = order; i > 0; i--)
1720
                        tmp[i] = tmp[i - 1];
1721
                }
1722
            }
1723
        }
1724
    }
1725
}
1726

    
1727
/**
1728
 *  Apply windowing and MDCT to obtain the spectral
1729
 *  coefficient from the predicted sample by LTP.
1730
 */
1731
static void windowing_and_mdct_ltp(AACContext *ac, float *out,
1732
                                   float *in, IndividualChannelStream *ics)
1733
{
1734
    const float *lwindow      = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1735
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1736
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1737
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1738

    
1739
    if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
1740
        ac->dsp.vector_fmul(in, in, lwindow_prev, 1024);
1741
    } else {
1742
        memset(in, 0, 448 * sizeof(float));
1743
        ac->dsp.vector_fmul(in + 448, in + 448, swindow_prev, 128);
1744
        memcpy(in + 576, in + 576, 448 * sizeof(float));
1745
    }
1746
    if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
1747
        ac->dsp.vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
1748
    } else {
1749
        memcpy(in + 1024, in + 1024, 448 * sizeof(float));
1750
        ac->dsp.vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
1751
        memset(in + 1024 + 576, 0, 448 * sizeof(float));
1752
    }
1753
    ff_mdct_calc(&ac->mdct_ltp, out, in);
1754
}
1755

    
1756
/**
1757
 * Apply the long term prediction
1758
 */
1759
static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
1760
{
1761
    const LongTermPrediction *ltp = &sce->ics.ltp;
1762
    const uint16_t *offsets = sce->ics.swb_offset;
1763
    int i, sfb;
1764

    
1765
    if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1766
        float *predTime = sce->ret;
1767
        float *predFreq = ac->buf_mdct;
1768
        int16_t num_samples = 2048;
1769

    
1770
        if (ltp->lag < 1024)
1771
            num_samples = ltp->lag + 1024;
1772
        for (i = 0; i < num_samples; i++)
1773
            predTime[i] = sce->ltp_state[i + 2048 - ltp->lag] * ltp->coef;
1774
        memset(&predTime[i], 0, (2048 - i) * sizeof(float));
1775

    
1776
        windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
1777

    
1778
        if (sce->tns.present)
1779
            apply_tns(predFreq, &sce->tns, &sce->ics, 0);
1780

    
1781
        for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
1782
            if (ltp->used[sfb])
1783
                for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
1784
                    sce->coeffs[i] += predFreq[i];
1785
    }
1786
}
1787

    
1788
/**
1789
 * Update the LTP buffer for next frame
1790
 */
1791
static void update_ltp(AACContext *ac, SingleChannelElement *sce)
1792
{
1793
    IndividualChannelStream *ics = &sce->ics;
1794
    float *saved     = sce->saved;
1795
    float *saved_ltp = sce->coeffs;
1796
    const float *lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1797
    const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1798
    int i;
1799

    
1800
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1801
        memcpy(saved_ltp,       saved, 512 * sizeof(float));
1802
        memset(saved_ltp + 576, 0,     448 * sizeof(float));
1803
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
1804
        for (i = 0; i < 64; i++)
1805
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
1806
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1807
        memcpy(saved_ltp,       ac->buf_mdct + 512, 448 * sizeof(float));
1808
        memset(saved_ltp + 576, 0,                  448 * sizeof(float));
1809
        ac->dsp.vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960,     &swindow[64],      64);
1810
        for (i = 0; i < 64; i++)
1811
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * swindow[63 - i];
1812
    } else { // LONG_STOP or ONLY_LONG
1813
        ac->dsp.vector_fmul_reverse(saved_ltp,       ac->buf_mdct + 512,     &lwindow[512],     512);
1814
        for (i = 0; i < 512; i++)
1815
            saved_ltp[i + 512] = ac->buf_mdct[1023 - i] * lwindow[511 - i];
1816
    }
1817

    
1818
    memcpy(sce->ltp_state, &sce->ltp_state[1024], 1024 * sizeof(int16_t));
1819
    ac->fmt_conv.float_to_int16(&(sce->ltp_state[1024]), sce->ret,  1024);
1820
    ac->fmt_conv.float_to_int16(&(sce->ltp_state[2048]), saved_ltp, 1024);
1821
}
1822

    
1823
/**
1824
 * Conduct IMDCT and windowing.
1825
 */
1826
static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1827
{
1828
    IndividualChannelStream *ics = &sce->ics;
1829
    float *in    = sce->coeffs;
1830
    float *out   = sce->ret;
1831
    float *saved = sce->saved;
1832
    const float *swindow      = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1833
    const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1834
    const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1835
    float *buf  = ac->buf_mdct;
1836
    float *temp = ac->temp;
1837
    int i;
1838

    
1839
    // imdct
1840
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1841
        for (i = 0; i < 1024; i += 128)
1842
            ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1843
    } else
1844
        ff_imdct_half(&ac->mdct, buf, in);
1845

    
1846
    /* window overlapping
1847
     * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1848
     * and long to short transitions are considered to be short to short
1849
     * transitions. This leaves just two cases (long to long and short to short)
1850
     * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1851
     */
1852
    if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1853
            (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1854
        ac->dsp.vector_fmul_window(    out,               saved,            buf,         lwindow_prev, 512);
1855
    } else {
1856
        memcpy(                        out,               saved,            448 * sizeof(float));
1857

    
1858
        if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1859
            ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448,      buf + 0*128, swindow_prev, 64);
1860
            ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow,      64);
1861
            ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow,      64);
1862
            ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow,      64);
1863
            ac->dsp.vector_fmul_window(temp,              buf + 3*128 + 64, buf + 4*128, swindow,      64);
1864
            memcpy(                    out + 448 + 4*128, temp, 64 * sizeof(float));
1865
        } else {
1866
            ac->dsp.vector_fmul_window(out + 448,         saved + 448,      buf,         swindow_prev, 64);
1867
            memcpy(                    out + 576,         buf + 64,         448 * sizeof(float));
1868
        }
1869
    }
1870

    
1871
    // buffer update
1872
    if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1873
        memcpy(                    saved,       temp + 64,         64 * sizeof(float));
1874
        ac->dsp.vector_fmul_window(saved + 64,  buf + 4*128 + 64, buf + 5*128, swindow, 64);
1875
        ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
1876
        ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
1877
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1878
    } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1879
        memcpy(                    saved,       buf + 512,        448 * sizeof(float));
1880
        memcpy(                    saved + 448, buf + 7*128 + 64,  64 * sizeof(float));
1881
    } else { // LONG_STOP or ONLY_LONG
1882
        memcpy(                    saved,       buf + 512,        512 * sizeof(float));
1883
    }
1884
}
1885

    
1886
/**
1887
 * Apply dependent channel coupling (applied before IMDCT).
1888
 *
1889
 * @param   index   index into coupling gain array
1890
 */
1891
static void apply_dependent_coupling(AACContext *ac,
1892
                                     SingleChannelElement *target,
1893
                                     ChannelElement *cce, int index)
1894
{
1895
    IndividualChannelStream *ics = &cce->ch[0].ics;
1896
    const uint16_t *offsets = ics->swb_offset;
1897
    float *dest = target->coeffs;
1898
    const float *src = cce->ch[0].coeffs;
1899
    int g, i, group, k, idx = 0;
1900
    if (ac->m4ac.object_type == AOT_AAC_LTP) {
1901
        av_log(ac->avctx, AV_LOG_ERROR,
1902
               "Dependent coupling is not supported together with LTP\n");
1903
        return;
1904
    }
1905
    for (g = 0; g < ics->num_window_groups; g++) {
1906
        for (i = 0; i < ics->max_sfb; i++, idx++) {
1907
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
1908
                const float gain = cce->coup.gain[index][idx];
1909
                for (group = 0; group < ics->group_len[g]; group++) {
1910
                    for (k = offsets[i]; k < offsets[i + 1]; k++) {
1911
                        // XXX dsputil-ize
1912
                        dest[group * 128 + k] += gain * src[group * 128 + k];
1913
                    }
1914
                }
1915
            }
1916
        }
1917
        dest += ics->group_len[g] * 128;
1918
        src  += ics->group_len[g] * 128;
1919
    }
1920
}
1921

    
1922
/**
1923
 * Apply independent channel coupling (applied after IMDCT).
1924
 *
1925
 * @param   index   index into coupling gain array
1926
 */
1927
static void apply_independent_coupling(AACContext *ac,
1928
                                       SingleChannelElement *target,
1929
                                       ChannelElement *cce, int index)
1930
{
1931
    int i;
1932
    const float gain = cce->coup.gain[index][0];
1933
    const float *src = cce->ch[0].ret;
1934
    float *dest = target->ret;
1935
    const int len = 1024 << (ac->m4ac.sbr == 1);
1936

    
1937
    for (i = 0; i < len; i++)
1938
        dest[i] += gain * src[i];
1939
}
1940

    
1941
/**
1942
 * channel coupling transformation interface
1943
 *
1944
 * @param   apply_coupling_method   pointer to (in)dependent coupling function
1945
 */
1946
static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1947
                                   enum RawDataBlockType type, int elem_id,
1948
                                   enum CouplingPoint coupling_point,
1949
                                   void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1950
{
1951
    int i, c;
1952

    
1953
    for (i = 0; i < MAX_ELEM_ID; i++) {
1954
        ChannelElement *cce = ac->che[TYPE_CCE][i];
1955
        int index = 0;
1956

    
1957
        if (cce && cce->coup.coupling_point == coupling_point) {
1958
            ChannelCoupling *coup = &cce->coup;
1959

    
1960
            for (c = 0; c <= coup->num_coupled; c++) {
1961
                if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1962
                    if (coup->ch_select[c] != 1) {
1963
                        apply_coupling_method(ac, &cc->ch[0], cce, index);
1964
                        if (coup->ch_select[c] != 0)
1965
                            index++;
1966
                    }
1967
                    if (coup->ch_select[c] != 2)
1968
                        apply_coupling_method(ac, &cc->ch[1], cce, index++);
1969
                } else
1970
                    index += 1 + (coup->ch_select[c] == 3);
1971
            }
1972
        }
1973
    }
1974
}
1975

    
1976
/**
1977
 * Convert spectral data to float samples, applying all supported tools as appropriate.
1978
 */
1979
static void spectral_to_sample(AACContext *ac)
1980
{
1981
    int i, type;
1982
    for (type = 3; type >= 0; type--) {
1983
        for (i = 0; i < MAX_ELEM_ID; i++) {
1984
            ChannelElement *che = ac->che[type][i];
1985
            if (che) {
1986
                if (type <= TYPE_CPE)
1987
                    apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1988
                if (ac->m4ac.object_type == AOT_AAC_LTP) {
1989
                    if (che->ch[0].ics.predictor_present) {
1990
                        if (che->ch[0].ics.ltp.present)
1991
                            apply_ltp(ac, &che->ch[0]);
1992
                        if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
1993
                            apply_ltp(ac, &che->ch[1]);
1994
                    }
1995
                }
1996
                if (che->ch[0].tns.present)
1997
                    apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1998
                if (che->ch[1].tns.present)
1999
                    apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2000
                if (type <= TYPE_CPE)
2001
                    apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
2002
                if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2003
                    imdct_and_windowing(ac, &che->ch[0]);
2004
                    if (ac->m4ac.object_type == AOT_AAC_LTP)
2005
                        update_ltp(ac, &che->ch[0]);
2006
                    if (type == TYPE_CPE) {
2007
                        imdct_and_windowing(ac, &che->ch[1]);
2008
                        if (ac->m4ac.object_type == AOT_AAC_LTP)
2009
                            update_ltp(ac, &che->ch[1]);
2010
                    }
2011
                    if (ac->m4ac.sbr > 0) {
2012
                        ff_sbr_apply(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2013
                    }
2014
                }
2015
                if (type <= TYPE_CCE)
2016
                    apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
2017
            }
2018
        }
2019
    }
2020
}
2021

    
2022
static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2023
{
2024
    int size;
2025
    AACADTSHeaderInfo hdr_info;
2026

    
2027
    size = ff_aac_parse_header(gb, &hdr_info);
2028
    if (size > 0) {
2029
        if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
2030
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2031
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2032
            ac->m4ac.chan_config = hdr_info.chan_config;
2033
            if (set_default_channel_config(ac->avctx, new_che_pos, hdr_info.chan_config))
2034
                return -7;
2035
            if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
2036
                return -7;
2037
        } else if (ac->output_configured != OC_LOCKED) {
2038
            ac->output_configured = OC_NONE;
2039
        }
2040
        if (ac->output_configured != OC_LOCKED) {
2041
            ac->m4ac.sbr = -1;
2042
            ac->m4ac.ps  = -1;
2043
        }
2044
        ac->m4ac.sample_rate     = hdr_info.sample_rate;
2045
        ac->m4ac.sampling_index  = hdr_info.sampling_index;
2046
        ac->m4ac.object_type     = hdr_info.object_type;
2047
        if (!ac->avctx->sample_rate)
2048
            ac->avctx->sample_rate = hdr_info.sample_rate;
2049
        if (hdr_info.num_aac_frames == 1) {
2050
            if (!hdr_info.crc_absent)
2051
                skip_bits(gb, 16);
2052
        } else {
2053
            av_log_missing_feature(ac->avctx, "More than one AAC RDB per ADTS frame is", 0);
2054
            return -1;
2055
        }
2056
    }
2057
    return size;
2058
}
2059

    
2060
static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2061
                                int *data_size, GetBitContext *gb)
2062
{
2063
    AACContext *ac = avctx->priv_data;
2064
    ChannelElement *che = NULL, *che_prev = NULL;
2065
    enum RawDataBlockType elem_type, elem_type_prev = TYPE_END;
2066
    int err, elem_id, data_size_tmp;
2067
    int samples = 0, multiplier;
2068

    
2069
    if (show_bits(gb, 12) == 0xfff) {
2070
        if (parse_adts_frame_header(ac, gb) < 0) {
2071
            av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2072
            return -1;
2073
        }
2074
        if (ac->m4ac.sampling_index > 12) {
2075
            av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
2076
            return -1;
2077
        }
2078
    }
2079

    
2080
    ac->tags_mapped = 0;
2081
    // parse
2082
    while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2083
        elem_id = get_bits(gb, 4);
2084

    
2085
        if (elem_type < TYPE_DSE) {
2086
            if (!(che=get_che(ac, elem_type, elem_id))) {
2087
                av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2088
                       elem_type, elem_id);
2089
                return -1;
2090
            }
2091
            samples = 1024;
2092
        }
2093

    
2094
        switch (elem_type) {
2095

    
2096
        case TYPE_SCE:
2097
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2098
            break;
2099

    
2100
        case TYPE_CPE:
2101
            err = decode_cpe(ac, gb, che);
2102
            break;
2103

    
2104
        case TYPE_CCE:
2105
            err = decode_cce(ac, gb, che);
2106
            break;
2107

    
2108
        case TYPE_LFE:
2109
            err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2110
            break;
2111

    
2112
        case TYPE_DSE:
2113
            err = skip_data_stream_element(ac, gb);
2114
            break;
2115

    
2116
        case TYPE_PCE: {
2117
            enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
2118
            memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
2119
            if ((err = decode_pce(avctx, &ac->m4ac, new_che_pos, gb)))
2120
                break;
2121
            if (ac->output_configured > OC_TRIAL_PCE)
2122
                av_log(avctx, AV_LOG_ERROR,
2123
                       "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2124
            else
2125
                err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
2126
            break;
2127
        }
2128

    
2129
        case TYPE_FIL:
2130
            if (elem_id == 15)
2131
                elem_id += get_bits(gb, 8) - 1;
2132
            if (get_bits_left(gb) < 8 * elem_id) {
2133
                    av_log(avctx, AV_LOG_ERROR, overread_err);
2134
                    return -1;
2135
            }
2136
            while (elem_id > 0)
2137
                elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, elem_type_prev);
2138
            err = 0; /* FIXME */
2139
            break;
2140

    
2141
        default:
2142
            err = -1; /* should not happen, but keeps compiler happy */
2143
            break;
2144
        }
2145

    
2146
        che_prev       = che;
2147
        elem_type_prev = elem_type;
2148

    
2149
        if (err)
2150
            return err;
2151

    
2152
        if (get_bits_left(gb) < 3) {
2153
            av_log(avctx, AV_LOG_ERROR, overread_err);
2154
            return -1;
2155
        }
2156
    }
2157

    
2158
    spectral_to_sample(ac);
2159

    
2160
    multiplier = (ac->m4ac.sbr == 1) ? ac->m4ac.ext_sample_rate > ac->m4ac.sample_rate : 0;
2161
    samples <<= multiplier;
2162
    if (ac->output_configured < OC_LOCKED) {
2163
        avctx->sample_rate = ac->m4ac.sample_rate << multiplier;
2164
        avctx->frame_size = samples;
2165
    }
2166

    
2167
    data_size_tmp = samples * avctx->channels * sizeof(int16_t);
2168
    if (*data_size < data_size_tmp) {
2169
        av_log(avctx, AV_LOG_ERROR,
2170
               "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2171
               *data_size, data_size_tmp);
2172
        return -1;
2173
    }
2174
    *data_size = data_size_tmp;
2175

    
2176
    if (samples)
2177
        ac->fmt_conv.float_to_int16_interleave(data, (const float **)ac->output_data, samples, avctx->channels);
2178

    
2179
    if (ac->output_configured)
2180
        ac->output_configured = OC_LOCKED;
2181

    
2182
    return 0;
2183
}
2184

    
2185
static int aac_decode_frame(AVCodecContext *avctx, void *data,
2186
                            int *data_size, AVPacket *avpkt)
2187
{
2188
    const uint8_t *buf = avpkt->data;
2189
    int buf_size = avpkt->size;
2190
    GetBitContext gb;
2191
    int buf_consumed;
2192
    int buf_offset;
2193
    int err;
2194

    
2195
    init_get_bits(&gb, buf, buf_size * 8);
2196

    
2197
    if ((err = aac_decode_frame_int(avctx, data, data_size, &gb)) < 0)
2198
        return err;
2199

    
2200
    buf_consumed = (get_bits_count(&gb) + 7) >> 3;
2201
    for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
2202
        if (buf[buf_offset])
2203
            break;
2204

    
2205
    return buf_size > buf_offset ? buf_consumed : buf_size;
2206
}
2207

    
2208
static av_cold int aac_decode_close(AVCodecContext *avctx)
2209
{
2210
    AACContext *ac = avctx->priv_data;
2211
    int i, type;
2212

    
2213
    for (i = 0; i < MAX_ELEM_ID; i++) {
2214
        for (type = 0; type < 4; type++) {
2215
            if (ac->che[type][i])
2216
                ff_aac_sbr_ctx_close(&ac->che[type][i]->sbr);
2217
            av_freep(&ac->che[type][i]);
2218
        }
2219
    }
2220

    
2221
    ff_mdct_end(&ac->mdct);
2222
    ff_mdct_end(&ac->mdct_small);
2223
    ff_mdct_end(&ac->mdct_ltp);
2224
    return 0;
2225
}
2226

    
2227

    
2228
#define LOAS_SYNC_WORD   0x2b7       ///< 11 bits LOAS sync word
2229

    
2230
struct LATMContext {
2231
    AACContext      aac_ctx;             ///< containing AACContext
2232
    int             initialized;         ///< initilized after a valid extradata was seen
2233

    
2234
    // parser data
2235
    int             audio_mux_version_A; ///< LATM syntax version
2236
    int             frame_length_type;   ///< 0/1 variable/fixed frame length
2237
    int             frame_length;        ///< frame length for fixed frame length
2238
};
2239

    
2240
static inline uint32_t latm_get_value(GetBitContext *b)
2241
{
2242
    int length = get_bits(b, 2);
2243

    
2244
    return get_bits_long(b, (length+1)*8);
2245
}
2246

    
2247
static int latm_decode_audio_specific_config(struct LATMContext *latmctx,
2248
                                             GetBitContext *gb)
2249
{
2250
    AVCodecContext *avctx = latmctx->aac_ctx.avctx;
2251
    MPEG4AudioConfig m4ac;
2252
    int  config_start_bit = get_bits_count(gb);
2253
    int     bits_consumed, esize;
2254

    
2255
    if (config_start_bit % 8) {
2256
        av_log_missing_feature(latmctx->aac_ctx.avctx, "audio specific "
2257
                               "config not byte aligned.\n", 1);
2258
        return AVERROR_INVALIDDATA;
2259
    } else {
2260
        bits_consumed =
2261
            decode_audio_specific_config(NULL, avctx, &m4ac,
2262
                                         gb->buffer + (config_start_bit / 8),
2263
                                         get_bits_left(gb) / 8);
2264

    
2265
        if (bits_consumed < 0)
2266
            return AVERROR_INVALIDDATA;
2267

    
2268
        esize = (bits_consumed+7) / 8;
2269

    
2270
        if (avctx->extradata_size <= esize) {
2271
            av_free(avctx->extradata);
2272
            avctx->extradata = av_malloc(esize + FF_INPUT_BUFFER_PADDING_SIZE);
2273
            if (!avctx->extradata)
2274
                return AVERROR(ENOMEM);
2275
        }
2276

    
2277
        avctx->extradata_size = esize;
2278
        memcpy(avctx->extradata, gb->buffer + (config_start_bit/8), esize);
2279
        memset(avctx->extradata+esize, 0, FF_INPUT_BUFFER_PADDING_SIZE);
2280

    
2281
        skip_bits_long(gb, bits_consumed);
2282
    }
2283

    
2284
    return bits_consumed;
2285
}
2286

    
2287
static int read_stream_mux_config(struct LATMContext *latmctx,
2288
                                  GetBitContext *gb)
2289
{
2290
    int ret, audio_mux_version = get_bits(gb, 1);
2291

    
2292
    latmctx->audio_mux_version_A = 0;
2293
    if (audio_mux_version)
2294
        latmctx->audio_mux_version_A = get_bits(gb, 1);
2295

    
2296
    if (!latmctx->audio_mux_version_A) {
2297

    
2298
        if (audio_mux_version)
2299
            latm_get_value(gb);                 // taraFullness
2300

    
2301
        skip_bits(gb, 1);                       // allStreamSameTimeFraming
2302
        skip_bits(gb, 6);                       // numSubFrames
2303
        // numPrograms
2304
        if (get_bits(gb, 4)) {                  // numPrograms
2305
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2306
                                   "multiple programs are not supported\n", 1);
2307
            return AVERROR_PATCHWELCOME;
2308
        }
2309

    
2310
        // for each program (which there is only on in DVB)
2311

    
2312
        // for each layer (which there is only on in DVB)
2313
        if (get_bits(gb, 3)) {                   // numLayer
2314
            av_log_missing_feature(latmctx->aac_ctx.avctx,
2315
                                   "multiple layers are not supported\n", 1);
2316
            return AVERROR_PATCHWELCOME;
2317
        }
2318

    
2319
        // for all but first stream: use_same_config = get_bits(gb, 1);
2320
        if (!audio_mux_version) {
2321
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2322
                return ret;
2323
        } else {
2324
            int ascLen = latm_get_value(gb);
2325
            if ((ret = latm_decode_audio_specific_config(latmctx, gb)) < 0)
2326
                return ret;
2327
            ascLen -= ret;
2328
            skip_bits_long(gb, ascLen);
2329
        }
2330

    
2331
        latmctx->frame_length_type = get_bits(gb, 3);
2332
        switch (latmctx->frame_length_type) {
2333
        case 0:
2334
            skip_bits(gb, 8);       // latmBufferFullness
2335
            break;
2336
        case 1:
2337
            latmctx->frame_length = get_bits(gb, 9);
2338
            break;
2339
        case 3:
2340
        case 4:
2341
        case 5:
2342
            skip_bits(gb, 6);       // CELP frame length table index
2343
            break;
2344
        case 6:
2345
        case 7:
2346
            skip_bits(gb, 1);       // HVXC frame length table index
2347
            break;
2348
        }
2349

    
2350
        if (get_bits(gb, 1)) {                  // other data
2351
            if (audio_mux_version) {
2352
                latm_get_value(gb);             // other_data_bits
2353
            } else {
2354
                int esc;
2355
                do {
2356
                    esc = get_bits(gb, 1);
2357
                    skip_bits(gb, 8);
2358
                } while (esc);
2359
            }
2360
        }
2361

    
2362
        if (get_bits(gb, 1))                     // crc present
2363
            skip_bits(gb, 8);                    // config_crc
2364
    }
2365

    
2366
    return 0;
2367
}
2368

    
2369
static int read_payload_length_info(struct LATMContext *ctx, GetBitContext *gb)
2370
{
2371
    uint8_t tmp;
2372

    
2373
    if (ctx->frame_length_type == 0) {
2374
        int mux_slot_length = 0;
2375
        do {
2376
            tmp = get_bits(gb, 8);
2377
            mux_slot_length += tmp;
2378
        } while (tmp == 255);
2379
        return mux_slot_length;
2380
    } else if (ctx->frame_length_type == 1) {
2381
        return ctx->frame_length;
2382
    } else if (ctx->frame_length_type == 3 ||
2383
               ctx->frame_length_type == 5 ||
2384
               ctx->frame_length_type == 7) {
2385
        skip_bits(gb, 2);          // mux_slot_length_coded
2386
    }
2387
    return 0;
2388
}
2389

    
2390
static int read_audio_mux_element(struct LATMContext *latmctx,
2391
                                  GetBitContext *gb)
2392
{
2393
    int err;
2394
    uint8_t use_same_mux = get_bits(gb, 1);
2395
    if (!use_same_mux) {
2396
        if ((err = read_stream_mux_config(latmctx, gb)) < 0)
2397
            return err;
2398
    } else if (!latmctx->aac_ctx.avctx->extradata) {
2399
        av_log(latmctx->aac_ctx.avctx, AV_LOG_DEBUG,
2400
               "no decoder config found\n");
2401
        return AVERROR(EAGAIN);
2402
    }
2403
    if (latmctx->audio_mux_version_A == 0) {
2404
        int mux_slot_length_bytes = read_payload_length_info(latmctx, gb);
2405
        if (mux_slot_length_bytes * 8 > get_bits_left(gb)) {
2406
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR, "incomplete frame\n");
2407
            return AVERROR_INVALIDDATA;
2408
        } else if (mux_slot_length_bytes * 8 + 256 < get_bits_left(gb)) {
2409
            av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2410
                   "frame length mismatch %d << %d\n",
2411
                   mux_slot_length_bytes * 8, get_bits_left(gb));
2412
            return AVERROR_INVALIDDATA;
2413
        }
2414
    }
2415
    return 0;
2416
}
2417

    
2418

    
2419
static int latm_decode_frame(AVCodecContext *avctx, void *out, int *out_size,
2420
                             AVPacket *avpkt)
2421
{
2422
    struct LATMContext *latmctx = avctx->priv_data;
2423
    int                 muxlength, err;
2424
    GetBitContext       gb;
2425

    
2426
    if (avpkt->size == 0)
2427
        return 0;
2428

    
2429
    init_get_bits(&gb, avpkt->data, avpkt->size * 8);
2430

    
2431
    // check for LOAS sync word
2432
    if (get_bits(&gb, 11) != LOAS_SYNC_WORD)
2433
        return AVERROR_INVALIDDATA;
2434

    
2435
    muxlength = get_bits(&gb, 13) + 3;
2436
    // not enough data, the parser should have sorted this
2437
    if (muxlength > avpkt->size)
2438
        return AVERROR_INVALIDDATA;
2439

    
2440
    if ((err = read_audio_mux_element(latmctx, &gb)) < 0)
2441
        return err;
2442

    
2443
    if (!latmctx->initialized) {
2444
        if (!avctx->extradata) {
2445
            *out_size = 0;
2446
            return avpkt->size;
2447
        } else {
2448
            if ((err = aac_decode_init(avctx)) < 0)
2449
                return err;
2450
            latmctx->initialized = 1;
2451
        }
2452
    }
2453

    
2454
    if (show_bits(&gb, 12) == 0xfff) {
2455
        av_log(latmctx->aac_ctx.avctx, AV_LOG_ERROR,
2456
               "ADTS header detected, probably as result of configuration "
2457
               "misparsing\n");
2458
        return AVERROR_INVALIDDATA;
2459
    }
2460

    
2461
    if ((err = aac_decode_frame_int(avctx, out, out_size, &gb)) < 0)
2462
        return err;
2463

    
2464
    return muxlength;
2465
}
2466

    
2467
av_cold static int latm_decode_init(AVCodecContext *avctx)
2468
{
2469
    struct LATMContext *latmctx = avctx->priv_data;
2470
    int ret;
2471

    
2472
    ret = aac_decode_init(avctx);
2473

    
2474
    if (avctx->extradata_size > 0) {
2475
        latmctx->initialized = !ret;
2476
    } else {
2477
        latmctx->initialized = 0;
2478
    }
2479

    
2480
    return ret;
2481
}
2482

    
2483

    
2484
AVCodec ff_aac_decoder = {
2485
    "aac",
2486
    AVMEDIA_TYPE_AUDIO,
2487
    CODEC_ID_AAC,
2488
    sizeof(AACContext),
2489
    aac_decode_init,
2490
    NULL,
2491
    aac_decode_close,
2492
    aac_decode_frame,
2493
    .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2494
    .sample_fmts = (const enum AVSampleFormat[]) {
2495
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
2496
    },
2497
    .channel_layouts = aac_channel_layout,
2498
};
2499

    
2500
/*
2501
    Note: This decoder filter is intended to decode LATM streams transferred
2502
    in MPEG transport streams which only contain one program.
2503
    To do a more complex LATM demuxing a separate LATM demuxer should be used.
2504
*/
2505
AVCodec ff_aac_latm_decoder = {
2506
    .name = "aac_latm",
2507
    .type = AVMEDIA_TYPE_AUDIO,
2508
    .id   = CODEC_ID_AAC_LATM,
2509
    .priv_data_size = sizeof(struct LATMContext),
2510
    .init   = latm_decode_init,
2511
    .close  = aac_decode_close,
2512
    .decode = latm_decode_frame,
2513
    .long_name = NULL_IF_CONFIG_SMALL("AAC LATM (Advanced Audio Codec LATM syntax)"),
2514
    .sample_fmts = (const enum AVSampleFormat[]) {
2515
        AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE
2516
    },
2517
    .channel_layouts = aac_channel_layout,
2518
};