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/*
* AAC decoder
* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
*
* AAC LATM decoder
* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
*
* AAC decoder fixed-point implementation
* Copyright (c) 2013
* MIPS Technologies, Inc., California.
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
#define AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
#include "aacdec_tab.h"
static void inline vector_pow43(int *coefs, int len)
{
int i, coef;
for (i=0; i<len; i++) {
coef = coefs[i];
if (coef < 0)
coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
else
coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
coefs[i] = coef;
}
}
/* 2^0, 2^0.25, 2^0.5, 2^0.75 */
static const int exp2tab[4] = {
Q31(1.0000000000/2), Q31(1.1892071150/2),
Q31(1.4142135624/2), Q31(1.6817928305/2)
};
static void inline subband_scale(int *dst, int *src, int scale,
int offset, int len, void *log_context)
{
int ssign = scale < 0 ? -1 : 1;
int s = FFABS(scale);
unsigned int round;
int i, out, c = exp2tab[s & 3];
s = offset - (s >> 2);
if (s > 31) {
for (i=0; i<len; i++) {
dst[i] = 0;
}
} else if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)(((int64_t)src[i] * c) >> 32);
dst[i] = ((int)(out+round) >> s) * ssign;
}
} else if (s > -32) {
s = s + 32;
round = 1U << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
dst[i] = out * (unsigned)ssign;
}
} else {
av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
}
}
static void noise_scale(int *coefs, int scale, int band_energy, int len)
{
int s = -scale;
unsigned int round;
int i, out, c = exp2tab[s & 3];
int nlz = 0;
av_assert0(s >= 0);
while (band_energy > 0x7fff) {
band_energy >>= 1;
nlz++;
}
c /= band_energy;
s = 21 + nlz - (s >> 2);
if (s > 31) {
for (i=0; i<len; i++) {
coefs[i] = 0;
}
} else if (s >= 0) {
round = s ? 1 << (s-1) : 0;
for (i=0; i<len; i++) {
out = (int)(((int64_t)coefs[i] * c) >> 32);
coefs[i] = -((int)(out+round) >> s);
}
}
else {
s = s + 32;
if (s > 0) {
round = 1 << (s-1);
for (i=0; i<len; i++) {
out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
coefs[i] = -out;
}
} else {
for (i=0; i<len; i++)
coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
}
}
}
static inline int *DEC_SPAIR(int *dst, unsigned idx)
{
dst[0] = (idx & 15) - 4;
dst[1] = (idx >> 4 & 15) - 4;
return dst + 2;
}
static inline int *DEC_SQUAD(int *dst, unsigned idx)
{
dst[0] = (idx & 3) - 1;
dst[1] = (idx >> 2 & 3) - 1;
dst[2] = (idx >> 4 & 3) - 1;
dst[3] = (idx >> 6 & 3) - 1;
return dst + 4;
}
static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
{
dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
return dst + 2;
}
static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
{
unsigned nz = idx >> 12;
dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
sign <<= nz & 1;
nz >>= 1;
dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
return dst + 4;
}
#endif /* AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H */
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