view mpegaudiotab.h @ 4166:eced83504436 libavcodec

mp3 header (de)compression bitstream filter this will make mp3 frames 4 bytes smaller, it will not give you binary identical mp3 files, but it will give you mp3 files which decode to binary identical output this will only work in containers providing at least packet size, sample_rate and number of channels bugreports about mp3 files for which this fails are welcome and this is experimental (dont expect compatibility and dont even expect to be able to decompress what you compressed, hell dont even expect this to work without editing the source a little)

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

#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 };