698a4b22d0
The fixed-point AAC decoder is the only user of the fixed-point sinewin
tables from sinewin; and it only uses a few of them (about 10% when
counting by size). This means that guarding initializing these tables by
an AVOnce (as done in 3719122065) is
unnecessary for them. Furthermore the array of pointers to the
individual arrays is also unneeded.
Therefore this commit moves these tables directly into aacdec_fixed.c;
this is done by ridding the original sinewin.h and sinewin_tablegen.h
headers completely of any fixed-point code at the cost of a bit of
duplicated code (the alternative is an ugly ifdef-mess).
This saves about 58KB from the binary when using hardcoded tables (as
these tables are hardcoded in this scenario); when not using hardcoded
tables, most of these savings only affect the .bss segment, but the rest
(< 1KB) contains relocations (i.e. savings in .data.rel.ro).
Reviewed-by: Lynne <dev@lynne.ee>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
472 lines
14 KiB
C
472 lines
14 KiB
C
/*
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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_fixed_tablegen.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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DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_long_1024))[1024];
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DECLARE_ALIGNED(32, static int, AAC_RENAME2(aac_kbd_short_128))[128];
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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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*
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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 < -31) {
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// Nothing to do
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} else 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 + (int64_t)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 * (1U << 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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unsigned 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 < -31) {
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return;
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} else 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 * (1U << 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",
|
|
.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,
|
|
.priv_data_size = sizeof(AACContext),
|
|
.init = aac_decode_init,
|
|
.close = aac_decode_close,
|
|
.decode = aac_decode_frame,
|
|
.sample_fmts = (const enum AVSampleFormat[]) {
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|
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,
|
|
};
|