view mpegaudiotab.h @ 10874:bcfe2acbf190 libavcodec

AAC: Compress codebook tables and optimise sign bit handling The codebooks each consist of small number of values repeated in groups of 2 or 4. Storing the codebooks as a packed list of 2- or 4-bit indexes into a table reduces their size substantially (from 7.5k to 1.5k), resulting in less cache pressure. For the band types with sign bits in the bitstream, storing the number and position of non-zero codebook values using a few bits avoids multiple get_bits() calls and floating-point comparisons which gcc handles miserably. Some float/int type punning also avoids gcc brain damage. Overall speedup 20-35% on Cortex-A8, 20% on Core i7.

/*
 * mpeg audio layer 2 tables. Most of them come from the mpeg audio
 * specification.
 *
 * Copyright (c) 2000, 2001 Fabrice Bellard
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file libavcodec/mpegaudiotab.h
 * mpeg audio layer 2 tables.
 * Most of them come from the mpeg audio specification.
 */

#ifndef AVCODEC_MPEGAUDIOTAB_H
#define AVCODEC_MPEGAUDIOTAB_H

#include <stdint.h>
#include "mpegaudio.h"

#define SQRT2 1.41421356237309514547

static const int costab32[30] = {
    FIX(0.54119610014619701222),
    FIX(1.3065629648763763537),

    FIX(0.50979557910415917998),
    FIX(2.5629154477415054814),
    FIX(0.89997622313641556513),
    FIX(0.60134488693504528634),

    FIX(0.5024192861881556782),
    FIX(5.1011486186891552563),
    FIX(0.78815462345125020249),
    FIX(0.64682178335999007679),
    FIX(0.56694403481635768927),
    FIX(1.0606776859903470633),
    FIX(1.7224470982383341955),
    FIX(0.52249861493968885462),

    FIX(10.19000812354803287),
    FIX(0.674808341455005678),
    FIX(1.1694399334328846596),
    FIX(0.53104259108978413284),
    FIX(2.0577810099534108446),
    FIX(0.58293496820613388554),
    FIX(0.83934964541552681272),
    FIX(0.50547095989754364798),
    FIX(3.4076084184687189804),
    FIX(0.62250412303566482475),
    FIX(0.97256823786196078263),
    FIX(0.51544730992262455249),
    FIX(1.4841646163141661852),
    FIX(0.5531038960344445421),
    FIX(0.74453627100229857749),
    FIX(0.5006029982351962726),
};

static const int bitinv32[32] = {
    0,  16,  8, 24,  4,  20,  12,  28,
    2,  18, 10, 26,  6,  22,  14,  30,
    1,  17,  9, 25,  5,  21,  13,  29,
    3,  19, 11, 27,  7,  23,  15,  31
};


static int16_t filter_bank[512];

static int scale_factor_table[64];
#ifdef USE_FLOATS
static float scale_factor_inv_table[64];
#else
static int8_t scale_factor_shift[64];
static unsigned short scale_factor_mult[64];
#endif
static unsigned char scale_diff_table[128];

/* total number of bits per allocation group */
static unsigned short total_quant_bits[17];

/* signal to noise ratio of each quantification step (could be
   computed from quant_steps[]). The values are dB multiplied by 10
*/
static const unsigned short quant_snr[17] = {
     70, 110, 160, 208,
    253, 316, 378, 439,
    499, 559, 620, 680,
    740, 800, 861, 920,
    980
};

/* fixed psycho acoustic model. Values of SNR taken from the 'toolame'
   project */
static const float fixed_smr[SBLIMIT] =  {
    30, 17, 16, 10, 3, 12, 8, 2.5,
    5, 5, 6, 6, 5, 6, 10, 6,
    -4, -10, -21, -30, -42, -55, -68, -75,
    -75, -75, -75, -75, -91, -107, -110, -108
};

static const unsigned char nb_scale_factors[4] = { 3, 2, 1, 2 };

#endif /* AVCODEC_MPEGAUDIOTAB_H */