GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavcodec/aacdec_fixed.c Lines: 88 203 43.3 %
Date: 2019-11-18 18:00:01 Branches: 32 88 36.4 %

Line Branch Exec Source
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/*
2
 * Copyright (c) 2013
3
 *      MIPS Technologies, Inc., California.
4
 *
5
 * Redistribution and use in source and binary forms, with or without
6
 * modification, are permitted provided that the following conditions
7
 * are met:
8
 * 1. Redistributions of source code must retain the above copyright
9
 *    notice, this list of conditions and the following disclaimer.
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 * 2. Redistributions in binary form must reproduce the above copyright
11
 *    notice, this list of conditions and the following disclaimer in the
12
 *    documentation and/or other materials provided with the distribution.
13
 * 3. Neither the name of the MIPS Technologies, Inc., nor the names of its
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 *    contributors may be used to endorse or promote products derived from
15
 *    this software without specific prior written permission.
16
 *
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 * THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND
18
 * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
19
 * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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 * ARE DISCLAIMED.  IN NO EVENT SHALL THE MIPS TECHNOLOGIES, INC. BE LIABLE
21
 * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
22
 * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS
23
 * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION)
24
 * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT
25
 * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY
26
 * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF
27
 * SUCH DAMAGE.
28
 *
29
 * AAC decoder fixed-point implementation
30
 *
31
 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
32
 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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 *
34
 * This file is part of FFmpeg.
35
 *
36
 * FFmpeg is free software; you can redistribute it and/or
37
 * modify it under the terms of the GNU Lesser General Public
38
 * License as published by the Free Software Foundation; either
39
 * version 2.1 of the License, or (at your option) any later version.
40
 *
41
 * FFmpeg is distributed in the hope that it will be useful,
42
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
43
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
44
 * Lesser General Public License for more details.
45
 *
46
 * You should have received a copy of the GNU Lesser General Public
47
 * License along with FFmpeg; if not, write to the Free Software
48
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
49
 */
50
51
/**
52
 * @file
53
 * AAC decoder
54
 * @author Oded Shimon  ( ods15 ods15 dyndns org )
55
 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
56
 *
57
 * Fixed point implementation
58
 * @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
59
 */
60
61
#define FFT_FLOAT 0
62
#define FFT_FIXED_32 1
63
#define USE_FIXED 1
64
65
#include "libavutil/fixed_dsp.h"
66
#include "libavutil/opt.h"
67
#include "avcodec.h"
68
#include "internal.h"
69
#include "get_bits.h"
70
#include "fft.h"
71
#include "lpc.h"
72
#include "kbdwin.h"
73
#include "sinewin.h"
74
75
#include "aac.h"
76
#include "aactab.h"
77
#include "aacdectab.h"
78
#include "adts_header.h"
79
#include "cbrt_data.h"
80
#include "sbr.h"
81
#include "aacsbr.h"
82
#include "mpeg4audio.h"
83
#include "profiles.h"
84
#include "libavutil/intfloat.h"
85
86
#include <math.h>
87
#include <string.h>
88
89
static av_always_inline void reset_predict_state(PredictorState *ps)
90
{
91
    ps->r0.mant   = 0;
92
    ps->r0.exp   = 0;
93
    ps->r1.mant   = 0;
94
    ps->r1.exp   = 0;
95
    ps->cor0.mant = 0;
96
    ps->cor0.exp = 0;
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    ps->cor1.mant = 0;
98
    ps->cor1.exp = 0;
99
    ps->var0.mant = 0x20000000;
100
    ps->var0.exp = 1;
101
    ps->var1.mant = 0x20000000;
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    ps->var1.exp = 1;
103
}
104
105
static const int exp2tab[4] = { Q31(1.0000000000/2), Q31(1.1892071150/2), Q31(1.4142135624/2), Q31(1.6817928305/2) };  // 2^0, 2^0.25, 2^0.5, 2^0.75
106
107
672272
static inline int *DEC_SPAIR(int *dst, unsigned idx)
108
{
109
672272
    dst[0] = (idx & 15) - 4;
110
672272
    dst[1] = (idx >> 4 & 15) - 4;
111
112
672272
    return dst + 2;
113
}
114
115
978157
static inline int *DEC_SQUAD(int *dst, unsigned idx)
116
{
117
978157
    dst[0] = (idx & 3) - 1;
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978157
    dst[1] = (idx >> 2 & 3) - 1;
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978157
    dst[2] = (idx >> 4 & 3) - 1;
120
978157
    dst[3] = (idx >> 6 & 3) - 1;
121
122
978157
    return dst + 4;
123
}
124
125
703742
static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
126
{
127
703742
    dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
128
703742
    dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
129
130
703742
    return dst + 2;
131
}
132
133
816691
static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
134
{
135
816691
    unsigned nz = idx >> 12;
136
137
816691
    dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
138
816691
    sign <<= nz & 1;
139
816691
    nz >>= 1;
140
816691
    dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
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816691
    sign <<= nz & 1;
142
816691
    nz >>= 1;
143
816691
    dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
144
816691
    sign <<= nz & 1;
145
816691
    nz >>= 1;
146
816691
    dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
147
148
816691
    return dst + 4;
149
}
150
151
787858
static void vector_pow43(int *coefs, int len)
152
{
153
    int i, coef;
154
155
11368194
    for (i=0; i<len; i++) {
156
10580336
        coef = coefs[i];
157
10580336
        if (coef < 0)
158
1834713
            coef = -(int)ff_cbrt_tab_fixed[-coef];
159
        else
160
8745623
            coef = (int)ff_cbrt_tab_fixed[coef];
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10580336
        coefs[i] = coef;
162
    }
163
787858
}
164
165
805813
static void subband_scale(int *dst, int *src, int scale, int offset, int len, void *log_context)
166
{
167
805813
    int ssign = scale < 0 ? -1 : 1;
168
805813
    int s = FFABS(scale);
169
    unsigned int round;
170
805813
    int i, out, c = exp2tab[s & 3];
171
172
805813
    s = offset - (s >> 2);
173
174
805813
    if (s > 31) {
175
        for (i=0; i<len; i++) {
176
            dst[i] = 0;
177
        }
178
805813
    } else if (s > 0) {
179
94338
        round = 1 << (s-1);
180
1385238
        for (i=0; i<len; i++) {
181
1290900
            out = (int)(((int64_t)src[i] * c) >> 32);
182
1290900
            dst[i] = ((int)(out+round) >> s) * ssign;
183
        }
184
711475
    } else if (s > -32) {
185
711475
        s = s + 32;
186
711475
        round = 1U << (s-1);
187
10536111
        for (i=0; i<len; i++) {
188
9824636
            out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
189
9824636
            dst[i] = out * (unsigned)ssign;
190
        }
191
    } else {
192
        av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
193
    }
194
805813
}
195
196
137931
static void noise_scale(int *coefs, int scale, int band_energy, int len)
197
{
198
137931
    int s = -scale;
199
    unsigned int round;
200
137931
    int i, out, c = exp2tab[s & 3];
201
137931
    int nlz = 0;
202
203
137931
    av_assert0(s >= 0);
204
2147210
    while (band_energy > 0x7fff) {
205
2009279
        band_energy >>= 1;
206
2009279
        nlz++;
207
    }
208
137931
    c /= band_energy;
209
137931
    s = 21 + nlz - (s >> 2);
210
211
137931
    if (s > 31) {
212
        for (i=0; i<len; i++) {
213
            coefs[i] = 0;
214
        }
215
137931
    } else if (s >= 0) {
216
73177
        round = s ? 1 << (s-1) : 0;
217
2460461
        for (i=0; i<len; i++) {
218
2387284
            out = (int)(((int64_t)coefs[i] * c) >> 32);
219
2387284
            coefs[i] = -((int)(out+round) >> s);
220
        }
221
    }
222
    else {
223
64754
        s = s + 32;
224
64754
        if (s > 0) {
225
64754
            round = 1 << (s-1);
226
560946
            for (i=0; i<len; i++) {
227
496192
                out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
228
496192
                coefs[i] = -out;
229
            }
230
        } else {
231
            for (i=0; i<len; i++)
232
                coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
233
        }
234
    }
235
137931
}
236
237
static av_always_inline SoftFloat flt16_round(SoftFloat pf)
238
{
239
    SoftFloat tmp;
240
    int s;
241
242
    tmp.exp = pf.exp;
243
    s = pf.mant >> 31;
244
    tmp.mant = (pf.mant ^ s) - s;
245
    tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U;
246
    tmp.mant = (tmp.mant ^ s) - s;
247
248
    return tmp;
249
}
250
251
static av_always_inline SoftFloat flt16_even(SoftFloat pf)
252
{
253
    SoftFloat tmp;
254
    int s;
255
256
    tmp.exp = pf.exp;
257
    s = pf.mant >> 31;
258
    tmp.mant = (pf.mant ^ s) - s;
259
    tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U;
260
    tmp.mant = (tmp.mant ^ s) - s;
261
262
    return tmp;
263
}
264
265
static av_always_inline SoftFloat flt16_trunc(SoftFloat pf)
266
{
267
    SoftFloat pun;
268
    int s;
269
270
    pun.exp = pf.exp;
271
    s = pf.mant >> 31;
272
    pun.mant = (pf.mant ^ s) - s;
273
    pun.mant = pun.mant & 0xFFC00000U;
274
    pun.mant = (pun.mant ^ s) - s;
275
276
    return pun;
277
}
278
279
static av_always_inline void predict(PredictorState *ps, int *coef,
280
                                     int output_enable)
281
{
282
    const SoftFloat a     = { 1023410176, 0 };  // 61.0 / 64
283
    const SoftFloat alpha = {  973078528, 0 };  // 29.0 / 32
284
    SoftFloat e0, e1;
285
    SoftFloat pv;
286
    SoftFloat k1, k2;
287
    SoftFloat   r0 = ps->r0,     r1 = ps->r1;
288
    SoftFloat cor0 = ps->cor0, cor1 = ps->cor1;
289
    SoftFloat var0 = ps->var0, var1 = ps->var1;
290
    SoftFloat tmp;
291
292
    if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) {
293
        k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0)));
294
    }
295
    else {
296
        k1.mant = 0;
297
        k1.exp = 0;
298
    }
299
300
    if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) {
301
        k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1)));
302
    }
303
    else {
304
        k2.mant = 0;
305
        k2.exp = 0;
306
    }
307
308
    tmp = av_mul_sf(k1, r0);
309
    pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1)));
310
    if (output_enable) {
311
        int shift = 28 - pv.exp;
312
313
        if (shift < 31) {
314
            if (shift > 0) {
315
                *coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
316
            } else
317
                *coef += (unsigned)pv.mant << -shift;
318
        }
319
    }
320
321
    e0 = av_int2sf(*coef, 2);
322
    e1 = av_sub_sf(e0, tmp);
323
324
    ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1)));
325
    tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1));
326
    tmp.exp--;
327
    ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp));
328
    ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0)));
329
    tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0));
330
    tmp.exp--;
331
    ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp));
332
333
    ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0))));
334
    ps->r0 = flt16_trunc(av_mul_sf(a, e0));
335
}
336
337
338
static const int cce_scale_fixed[8] = {
339
    Q30(1.0),          //2^(0/8)
340
    Q30(1.0905077327), //2^(1/8)
341
    Q30(1.1892071150), //2^(2/8)
342
    Q30(1.2968395547), //2^(3/8)
343
    Q30(1.4142135624), //2^(4/8)
344
    Q30(1.5422108254), //2^(5/8)
345
    Q30(1.6817928305), //2^(6/8)
346
    Q30(1.8340080864), //2^(7/8)
347
};
348
349
/**
350
 * Apply dependent channel coupling (applied before IMDCT).
351
 *
352
 * @param   index   index into coupling gain array
353
 */
354
static void apply_dependent_coupling_fixed(AACContext *ac,
355
                                     SingleChannelElement *target,
356
                                     ChannelElement *cce, int index)
357
{
358
    IndividualChannelStream *ics = &cce->ch[0].ics;
359
    const uint16_t *offsets = ics->swb_offset;
360
    int *dest = target->coeffs;
361
    const int *src = cce->ch[0].coeffs;
362
    int g, i, group, k, idx = 0;
363
    if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
364
        av_log(ac->avctx, AV_LOG_ERROR,
365
               "Dependent coupling is not supported together with LTP\n");
366
        return;
367
    }
368
    for (g = 0; g < ics->num_window_groups; g++) {
369
        for (i = 0; i < ics->max_sfb; i++, idx++) {
370
            if (cce->ch[0].band_type[idx] != ZERO_BT) {
371
                const int gain = cce->coup.gain[index][idx];
372
                int shift, round, c, tmp;
373
374
                if (gain < 0) {
375
                    c = -cce_scale_fixed[-gain & 7];
376
                    shift = (-gain-1024) >> 3;
377
                }
378
                else {
379
                    c = cce_scale_fixed[gain & 7];
380
                    shift = (gain-1024) >> 3;
381
                }
382
383
                if (shift < -31) {
384
                    // Nothing to do
385
                } else if (shift < 0) {
386
                    shift = -shift;
387
                    round = 1 << (shift - 1);
388
389
                    for (group = 0; group < ics->group_len[g]; group++) {
390
                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
391
                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
392
                                       (int64_t)0x1000000000) >> 37);
393
                            dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
394
                        }
395
                    }
396
                }
397
                else {
398
                    for (group = 0; group < ics->group_len[g]; group++) {
399
                        for (k = offsets[i]; k < offsets[i + 1]; k++) {
400
                            tmp = (int)(((int64_t)src[group * 128 + k] * c + \
401
                                        (int64_t)0x1000000000) >> 37);
402
                            dest[group * 128 + k] += tmp * (1U << shift);
403
                        }
404
                    }
405
                }
406
            }
407
        }
408
        dest += ics->group_len[g] * 128;
409
        src  += ics->group_len[g] * 128;
410
    }
411
}
412
413
/**
414
 * Apply independent channel coupling (applied after IMDCT).
415
 *
416
 * @param   index   index into coupling gain array
417
 */
418
650
static void apply_independent_coupling_fixed(AACContext *ac,
419
                                       SingleChannelElement *target,
420
                                       ChannelElement *cce, int index)
421
{
422
    int i, c, shift, round, tmp;
423
650
    const int gain = cce->coup.gain[index][0];
424
650
    const int *src = cce->ch[0].ret;
425
650
    unsigned int *dest = target->ret;
426
650
    const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
427
428
650
    c = cce_scale_fixed[gain & 7];
429
650
    shift = (gain-1024) >> 3;
430
650
    if (shift < -31) {
431
        return;
432
650
    } else if (shift < 0) {
433
        shift = -shift;
434
        round = 1 << (shift - 1);
435
436
        for (i = 0; i < len; i++) {
437
            tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
438
            dest[i] += (tmp + round) >> shift;
439
        }
440
    }
441
    else {
442
666250
      for (i = 0; i < len; i++) {
443
665600
          tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
444
665600
          dest[i] += tmp * (1U << shift);
445
      }
446
    }
447
}
448
449
#include "aacdec_template.c"
450
451
AVCodec ff_aac_fixed_decoder = {
452
    .name            = "aac_fixed",
453
    .long_name       = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
454
    .type            = AVMEDIA_TYPE_AUDIO,
455
    .id              = AV_CODEC_ID_AAC,
456
    .priv_data_size  = sizeof(AACContext),
457
    .init            = aac_decode_init,
458
    .close           = aac_decode_close,
459
    .decode          = aac_decode_frame,
460
    .sample_fmts     = (const enum AVSampleFormat[]) {
461
        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
462
    },
463
    .capabilities    = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
464
    .caps_internal   = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
465
    .channel_layouts = aac_channel_layout,
466
    .profiles        = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
467
    .flush = flush,
468
};