mirror of
https://github.com/FFmpeg/FFmpeg.git
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175 lines
4.7 KiB
C
175 lines
4.7 KiB
C
/*
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* 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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* Copyright (c) 2008-2013 Alex Converse <alex.converse@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-libav@jannau.net>
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*
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* AAC decoder fixed-point implementation
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* Copyright (c) 2013
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* MIPS Technologies, Inc., California.
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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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*/
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#ifndef AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
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#define AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H
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#include "aacdec_tab.h"
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static void inline vector_pow43(int *coefs, int len)
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{
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int i, coef;
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for (i=0; i<len; i++) {
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coef = coefs[i];
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if (coef < 0)
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coef = -(int)ff_cbrt_tab_fixed[(-coef) & 8191];
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else
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coef = (int)ff_cbrt_tab_fixed[ coef & 8191];
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coefs[i] = coef;
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}
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}
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/* 2^0, 2^0.25, 2^0.5, 2^0.75 */
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static const int exp2tab[4] = {
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Q31(1.0000000000/2), Q31(1.1892071150/2),
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Q31(1.4142135624/2), Q31(1.6817928305/2)
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};
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static void inline subband_scale(int *dst, int *src, int scale,
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int offset, int len, void *log_context)
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{
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int ssign = scale < 0 ? -1 : 1;
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int s = FFABS(scale);
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unsigned int round;
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int i, out, c = exp2tab[s & 3];
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s = offset - (s >> 2);
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if (s > 31) {
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for (i=0; i<len; i++) {
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dst[i] = 0;
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}
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} else if (s > 0) {
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round = 1 << (s-1);
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for (i=0; i<len; i++) {
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out = (int)(((int64_t)src[i] * c) >> 32);
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dst[i] = ((int)(out+round) >> s) * ssign;
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}
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} else if (s > -32) {
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s = s + 32;
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round = 1U << (s-1);
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for (i=0; i<len; i++) {
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out = (int)((int64_t)((int64_t)src[i] * c + round) >> s);
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dst[i] = out * (unsigned)ssign;
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}
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} else {
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av_log(log_context, AV_LOG_ERROR, "Overflow in subband_scale()\n");
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}
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}
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static void noise_scale(int *coefs, int scale, int band_energy, int len)
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{
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int s = -scale;
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unsigned int round;
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int i, out, c = exp2tab[s & 3];
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int nlz = 0;
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av_assert0(s >= 0);
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while (band_energy > 0x7fff) {
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band_energy >>= 1;
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nlz++;
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}
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c /= band_energy;
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s = 21 + nlz - (s >> 2);
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if (s > 31) {
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for (i=0; i<len; i++) {
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coefs[i] = 0;
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}
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} else if (s >= 0) {
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round = s ? 1 << (s-1) : 0;
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for (i=0; i<len; i++) {
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out = (int)(((int64_t)coefs[i] * c) >> 32);
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coefs[i] = -((int)(out+round) >> s);
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}
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}
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else {
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s = s + 32;
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if (s > 0) {
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round = 1 << (s-1);
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for (i=0; i<len; i++) {
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out = (int)((int64_t)((int64_t)coefs[i] * c + round) >> s);
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coefs[i] = -out;
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}
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} else {
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for (i=0; i<len; i++)
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coefs[i] = -(int64_t)coefs[i] * c * (1 << -s);
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}
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}
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}
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static inline int *DEC_SPAIR(int *dst, unsigned idx)
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{
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dst[0] = (idx & 15) - 4;
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dst[1] = (idx >> 4 & 15) - 4;
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return dst + 2;
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}
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static inline int *DEC_SQUAD(int *dst, unsigned idx)
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{
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dst[0] = (idx & 3) - 1;
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dst[1] = (idx >> 2 & 3) - 1;
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dst[2] = (idx >> 4 & 3) - 1;
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dst[3] = (idx >> 6 & 3) - 1;
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return dst + 4;
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}
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static inline int *DEC_UPAIR(int *dst, unsigned idx, unsigned sign)
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{
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dst[0] = (idx & 15) * (1 - (sign & 0xFFFFFFFE));
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dst[1] = (idx >> 4 & 15) * (1 - ((sign & 1) * 2));
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return dst + 2;
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}
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static inline int *DEC_UQUAD(int *dst, unsigned idx, unsigned sign)
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{
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unsigned nz = idx >> 12;
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dst[0] = (idx & 3) * (1 + (((int)sign >> 31) * 2));
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sign <<= nz & 1;
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nz >>= 1;
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dst[1] = (idx >> 2 & 3) * (1 + (((int)sign >> 31) * 2));
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sign <<= nz & 1;
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nz >>= 1;
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dst[2] = (idx >> 4 & 3) * (1 + (((int)sign >> 31) * 2));
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sign <<= nz & 1;
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nz >>= 1;
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dst[3] = (idx >> 6 & 3) * (1 + (((int)sign >> 31) * 2));
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return dst + 4;
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}
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#endif /* AVCODEC_AAC_AACDEC_FIXED_DEQUANT_H */
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