469 lines
14 KiB
C
469 lines
14 KiB
C
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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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#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 "adts_header.h"
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#include "cbrt_data.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 "profiles.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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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
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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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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)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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static void subband_scale(int *dst, int *src, int scale, 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 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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if (shift > 0) {
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*coef += (unsigned)((pv.mant + (1 << (shift - 1))) >> shift);
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} else
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*coef += (unsigned)pv.mant << -shift;
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}
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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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|
|
*
|
||
|
|
* @param index index into coupling gain array
|
||
|
|
*/
|
||
|
|
static void apply_dependent_coupling_fixed(AACContext *ac,
|
||
|
|
SingleChannelElement *target,
|
||
|
|
ChannelElement *cce, int index)
|
||
|
|
{
|
||
|
|
IndividualChannelStream *ics = &cce->ch[0].ics;
|
||
|
|
const uint16_t *offsets = ics->swb_offset;
|
||
|
|
int *dest = target->coeffs;
|
||
|
|
const int *src = cce->ch[0].coeffs;
|
||
|
|
int g, i, group, k, idx = 0;
|
||
|
|
if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
|
||
|
|
av_log(ac->avctx, AV_LOG_ERROR,
|
||
|
|
"Dependent coupling is not supported together with LTP\n");
|
||
|
|
return;
|
||
|
|
}
|
||
|
|
for (g = 0; g < ics->num_window_groups; g++) {
|
||
|
|
for (i = 0; i < ics->max_sfb; i++, idx++) {
|
||
|
|
if (cce->ch[0].band_type[idx] != ZERO_BT) {
|
||
|
|
const int gain = cce->coup.gain[index][idx];
|
||
|
|
int shift, round, c, tmp;
|
||
|
|
|
||
|
|
if (gain < 0) {
|
||
|
|
c = -cce_scale_fixed[-gain & 7];
|
||
|
|
shift = (-gain-1024) >> 3;
|
||
|
|
}
|
||
|
|
else {
|
||
|
|
c = cce_scale_fixed[gain & 7];
|
||
|
|
shift = (gain-1024) >> 3;
|
||
|
|
}
|
||
|
|
|
||
|
|
if (shift < -31) {
|
||
|
|
// Nothing to do
|
||
|
|
} else if (shift < 0) {
|
||
|
|
shift = -shift;
|
||
|
|
round = 1 << (shift - 1);
|
||
|
|
|
||
|
|
for (group = 0; group < ics->group_len[g]; group++) {
|
||
|
|
for (k = offsets[i]; k < offsets[i + 1]; k++) {
|
||
|
|
tmp = (int)(((int64_t)src[group * 128 + k] * c + \
|
||
|
|
(int64_t)0x1000000000) >> 37);
|
||
|
|
dest[group * 128 + k] += (tmp + (int64_t)round) >> shift;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
else {
|
||
|
|
for (group = 0; group < ics->group_len[g]; group++) {
|
||
|
|
for (k = offsets[i]; k < offsets[i + 1]; k++) {
|
||
|
|
tmp = (int)(((int64_t)src[group * 128 + k] * c + \
|
||
|
|
(int64_t)0x1000000000) >> 37);
|
||
|
|
dest[group * 128 + k] += tmp * (1U << shift);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
dest += ics->group_len[g] * 128;
|
||
|
|
src += ics->group_len[g] * 128;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
/**
|
||
|
|
* Apply independent channel coupling (applied after IMDCT).
|
||
|
|
*
|
||
|
|
* @param index index into coupling gain array
|
||
|
|
*/
|
||
|
|
static void apply_independent_coupling_fixed(AACContext *ac,
|
||
|
|
SingleChannelElement *target,
|
||
|
|
ChannelElement *cce, int index)
|
||
|
|
{
|
||
|
|
int i, c, shift, round, tmp;
|
||
|
|
const int gain = cce->coup.gain[index][0];
|
||
|
|
const int *src = cce->ch[0].ret;
|
||
|
|
unsigned int *dest = target->ret;
|
||
|
|
const int len = 1024 << (ac->oc[1].m4ac.sbr == 1);
|
||
|
|
|
||
|
|
c = cce_scale_fixed[gain & 7];
|
||
|
|
shift = (gain-1024) >> 3;
|
||
|
|
if (shift < -31) {
|
||
|
|
return;
|
||
|
|
} else if (shift < 0) {
|
||
|
|
shift = -shift;
|
||
|
|
round = 1 << (shift - 1);
|
||
|
|
|
||
|
|
for (i = 0; i < len; i++) {
|
||
|
|
tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
|
||
|
|
dest[i] += (tmp + round) >> shift;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
else {
|
||
|
|
for (i = 0; i < len; i++) {
|
||
|
|
tmp = (int)(((int64_t)src[i] * c + (int64_t)0x1000000000) >> 37);
|
||
|
|
dest[i] += tmp * (1U << shift);
|
||
|
|
}
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
#include "aacdec_template.c"
|
||
|
|
|
||
|
|
AVCodec ff_aac_fixed_decoder = {
|
||
|
|
.name = "aac_fixed",
|
||
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
|
||
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
||
|
|
.id = AV_CODEC_ID_AAC,
|
||
|
|
.priv_data_size = sizeof(AACContext),
|
||
|
|
.init = aac_decode_init,
|
||
|
|
.close = aac_decode_close,
|
||
|
|
.decode = aac_decode_frame,
|
||
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
|
||
|
|
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
|
||
|
|
},
|
||
|
|
.capabilities = AV_CODEC_CAP_CHANNEL_CONF | AV_CODEC_CAP_DR1,
|
||
|
|
.caps_internal = FF_CODEC_CAP_INIT_THREADSAFE | FF_CODEC_CAP_INIT_CLEANUP,
|
||
|
|
.channel_layouts = aac_channel_layout,
|
||
|
|
.profiles = NULL_IF_CONFIG_SMALL(ff_aac_profiles),
|
||
|
|
.flush = flush,
|
||
|
|
};
|