Mercurial > libavcodec.hg
view mpegaudiotab.h @ 12235:e08d65897115 libavcodec
VP8: clear DCT blocks in iDCT instead of using clear_blocks.
~0.3% faster overall.
author | darkshikari |
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date | Fri, 23 Jul 2010 00:07:16 +0000 |
parents | 7dd2a45249a9 |
children |
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/* * 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 * 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 */