mirror of https://github.com/FFmpeg/FFmpeg.git
Add fixed point implementation Signed-off-by: Nedeljko Babic <nedeljko.babic@imgtec.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>pull/140/head
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/*
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* Copyright (c) 2013 |
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* MIPS Technologies, Inc., California. |
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* |
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* Redistribution and use in source and binary forms, with or without |
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* modification, are permitted provided that the following conditions |
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* are met: |
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* 1. Redistributions of source code must retain the above copyright |
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* notice, this list of conditions and the following disclaimer. |
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* 2. Redistributions in binary form must reproduce the above copyright |
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* notice, this list of conditions and the following disclaimer in the |
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* documentation and/or other materials provided with the distribution. |
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* 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 |
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* this software without specific prior written permission. |
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* |
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* THIS SOFTWARE IS PROVIDED BY THE MIPS TECHNOLOGIES, INC. ``AS IS'' AND |
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* ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE |
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* 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 |
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* FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL |
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* DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS |
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* OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) |
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* HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT |
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* LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY |
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* OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF |
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* SUCH DAMAGE. |
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* |
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* AAC decoder fixed-point implementation |
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* |
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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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* 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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/**
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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 ) |
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* |
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* Fixed point implementation |
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* @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com ) |
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*/ |
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#define FFT_FLOAT 0 |
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#define FFT_FIXED_32 1 |
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#define USE_FIXED 1 |
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#define CONFIG_FIXED 1 |
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#include "libavutil/fixed_dsp.h" |
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#include "libavutil/opt.h" |
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#include "avcodec.h" |
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#include "internal.h" |
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#include "get_bits.h" |
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#include "fft.h" |
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#include "lpc.h" |
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#include "kbdwin.h" |
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#include "sinewin.h" |
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#include "aac.h" |
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#include "aactab.h" |
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#include "aacdectab.h" |
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#include "cbrt_tablegen.h" |
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#include "sbr.h" |
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#include "aacsbr.h" |
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#include "mpeg4audio.h" |
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#include "aacadtsdec.h" |
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#include "libavutil/intfloat.h" |
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#include <math.h> |
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#include <string.h> |
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static av_always_inline void reset_predict_state(PredictorState *ps) |
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{ |
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ps->r0.mant = 0; |
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ps->r0.exp = 0; |
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ps->r1.mant = 0; |
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ps->r1.exp = 0; |
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ps->cor0.mant = 0; |
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ps->cor0.exp = 0; |
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ps->cor1.mant = 0; |
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ps->cor1.exp = 0; |
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ps->var0.mant = 0x20000000; |
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ps->var0.exp = 1; |
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ps->var1.mant = 0x20000000; |
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ps->var1.exp = 1; |
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} |
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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
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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) << 1)); |
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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) << 1)); |
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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) << 1)); |
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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) << 1)); |
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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) << 1)); |
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return dst + 4; |
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} |
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static void 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)cbrt_tab[-coef]; |
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else |
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coef = (int)cbrt_tab[coef]; |
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coefs[i] = coef; |
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} |
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} |
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static void subband_scale(int *dst, int *src, int scale, int offset, int len) |
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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 > 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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} |
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else { |
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s = s + 32; |
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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)src[i] * c + round) >> s); |
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dst[i] = out * ssign; |
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} |
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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 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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int nlz = 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 > 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)coefs[i] * c) >> 32); |
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coefs[i] = ((int)(out+round) >> s) * ssign; |
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} |
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} |
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else { |
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s = s + 32; |
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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 * ssign; |
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} |
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} |
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} |
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static av_always_inline SoftFloat flt16_round(SoftFloat pf) |
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{ |
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SoftFloat tmp; |
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int s; |
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tmp.exp = pf.exp; |
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s = pf.mant >> 31; |
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tmp.mant = (pf.mant ^ s) - s; |
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tmp.mant = (tmp.mant + 0x00200000U) & 0xFFC00000U; |
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tmp.mant = (tmp.mant ^ s) - s; |
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return tmp; |
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} |
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static av_always_inline SoftFloat flt16_even(SoftFloat pf) |
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{ |
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SoftFloat tmp; |
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int s; |
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tmp.exp = pf.exp; |
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s = pf.mant >> 31; |
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tmp.mant = (pf.mant ^ s) - s; |
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tmp.mant = (tmp.mant + 0x001FFFFFU + (tmp.mant & 0x00400000U >> 16)) & 0xFFC00000U; |
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tmp.mant = (tmp.mant ^ s) - s; |
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return tmp; |
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} |
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static av_always_inline SoftFloat flt16_trunc(SoftFloat pf) |
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{ |
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SoftFloat pun; |
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int s; |
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pun.exp = pf.exp; |
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s = pf.mant >> 31; |
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pun.mant = (pf.mant ^ s) - s; |
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pun.mant = pun.mant & 0xFFC00000U; |
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pun.mant = (pun.mant ^ s) - s; |
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return pun; |
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} |
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static av_always_inline void predict(PredictorState *ps, int *coef, |
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int output_enable) |
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{ |
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const SoftFloat a = { 1023410176, 0 }; // 61.0 / 64
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const SoftFloat alpha = { 973078528, 0 }; // 29.0 / 32
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SoftFloat e0, e1; |
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SoftFloat pv; |
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SoftFloat k1, k2; |
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SoftFloat r0 = ps->r0, r1 = ps->r1; |
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SoftFloat cor0 = ps->cor0, cor1 = ps->cor1; |
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SoftFloat var0 = ps->var0, var1 = ps->var1; |
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SoftFloat tmp; |
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if (var0.exp > 1 || (var0.exp == 1 && var0.mant > 0x20000000)) { |
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k1 = av_mul_sf(cor0, flt16_even(av_div_sf(a, var0))); |
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} |
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else { |
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k1.mant = 0; |
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k1.exp = 0; |
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} |
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if (var1.exp > 1 || (var1.exp == 1 && var1.mant > 0x20000000)) { |
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k2 = av_mul_sf(cor1, flt16_even(av_div_sf(a, var1))); |
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} |
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else { |
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k2.mant = 0; |
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k2.exp = 0; |
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} |
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tmp = av_mul_sf(k1, r0); |
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pv = flt16_round(av_add_sf(tmp, av_mul_sf(k2, r1))); |
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if (output_enable) { |
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int shift = 28 - pv.exp; |
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if (shift < 31) |
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*coef += (pv.mant + (1 << (shift - 1))) >> shift; |
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} |
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e0 = av_int2sf(*coef, 2); |
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e1 = av_sub_sf(e0, tmp); |
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ps->cor1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor1), av_mul_sf(r1, e1))); |
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tmp = av_add_sf(av_mul_sf(r1, r1), av_mul_sf(e1, e1)); |
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tmp.exp--; |
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ps->var1 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var1), tmp)); |
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ps->cor0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, cor0), av_mul_sf(r0, e0))); |
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tmp = av_add_sf(av_mul_sf(r0, r0), av_mul_sf(e0, e0)); |
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tmp.exp--; |
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ps->var0 = flt16_trunc(av_add_sf(av_mul_sf(alpha, var0), tmp)); |
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ps->r1 = flt16_trunc(av_mul_sf(a, av_sub_sf(r0, av_mul_sf(k1, e0)))); |
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ps->r0 = flt16_trunc(av_mul_sf(a, e0)); |
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} |
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static const int cce_scale_fixed[8] = { |
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Q30(1.0), //2^(0/8)
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Q30(1.0905077327), //2^(1/8)
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Q30(1.1892071150), //2^(2/8)
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Q30(1.2968395547), //2^(3/8)
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Q30(1.4142135624), //2^(4/8)
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Q30(1.5422108254), //2^(5/8)
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Q30(1.6817928305), //2^(6/8)
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Q30(1.8340080864), //2^(7/8)
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}; |
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/**
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* Apply dependent channel coupling (applied before IMDCT). |
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* |
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* @param index index into coupling gain array |
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*/ |
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static void apply_dependent_coupling_fixed(AACContext *ac, |
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SingleChannelElement *target, |
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ChannelElement *cce, int index) |
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{ |
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IndividualChannelStream *ics = &cce->ch[0].ics; |
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const uint16_t *offsets = ics->swb_offset; |
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int *dest = target->coeffs; |
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const int *src = cce->ch[0].coeffs; |
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int g, i, group, k, idx = 0; |
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if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) { |
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av_log(ac->avctx, AV_LOG_ERROR, |
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"Dependent coupling is not supported together with LTP\n"); |
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return; |
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} |
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for (g = 0; g < ics->num_window_groups; g++) { |
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for (i = 0; i < ics->max_sfb; i++, idx++) { |
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if (cce->ch[0].band_type[idx] != ZERO_BT) { |
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const int gain = cce->coup.gain[index][idx]; |
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int shift, round, c, tmp; |
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if (gain < 0) { |
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c = -cce_scale_fixed[-gain & 7]; |
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shift = (-gain-1024) >> 3; |
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} |
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else { |
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c = cce_scale_fixed[gain & 7]; |
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shift = (gain-1024) >> 3; |
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} |
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if (shift < 0) { |
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shift = -shift; |
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round = 1 << (shift - 1); |
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for (group = 0; group < ics->group_len[g]; group++) { |
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for (k = offsets[i]; k < offsets[i + 1]; k++) { |
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tmp = (int)(((int64_t)src[group * 128 + k] * c + \
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(int64_t)0x1000000000) >> 37); |
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dest[group * 128 + k] += (tmp + round) >> shift; |
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} |
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} |
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} |
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else { |
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for (group = 0; group < ics->group_len[g]; group++) { |
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for (k = offsets[i]; k < offsets[i + 1]; k++) { |
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tmp = (int)(((int64_t)src[group * 128 + k] * c + \
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(int64_t)0x1000000000) >> 37); |
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dest[group * 128 + k] += tmp << shift; |
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} |
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} |
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} |
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} |
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} |
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dest += ics->group_len[g] * 128; |
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src += ics->group_len[g] * 128; |
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} |
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} |
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/**
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* Apply independent channel coupling (applied after IMDCT). |
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* |
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* @param index index into coupling gain array |
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*/ |
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static void apply_independent_coupling_fixed(AACContext *ac, |
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SingleChannelElement *target, |
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ChannelElement *cce, int index) |
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{ |
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int i, c, shift, round, tmp; |
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const int gain = cce->coup.gain[index][0]; |
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const int *src = cce->ch[0].ret; |
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int *dest = target->ret; |
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const int len = 1024 << (ac->oc[1].m4ac.sbr == 1); |
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c = cce_scale_fixed[gain & 7]; |
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shift = (gain-1024) >> 3; |
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if (shift < 0) { |
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shift = -shift; |
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round = 1 << (shift - 1); |
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for (i = 0; i < len; i++) { |
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tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37); |
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dest[i] += (tmp + round) >> shift; |
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} |
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} |
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else { |
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for (i = 0; i < len; i++) { |
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tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37); |
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dest[i] += tmp << shift; |
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} |
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} |
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} |
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#include "aacdec_template.c" |
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AVCodec ff_aac_fixed_decoder = { |
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.name = "aac_fixed", |
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.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"), |
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.type = AVMEDIA_TYPE_AUDIO, |
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.id = AV_CODEC_ID_AAC, |
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.priv_data_size = sizeof(AACContext), |
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.init = aac_decode_init, |
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.close = aac_decode_close, |
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.decode = aac_decode_frame, |
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.sample_fmts = (const enum AVSampleFormat[]) { |
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AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE |
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}, |
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.capabilities = CODEC_CAP_CHANNEL_CONF | CODEC_CAP_DR1, |
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.channel_layouts = aac_channel_layout, |
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.flush = flush, |
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}; |
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