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https://github.com/FFmpeg/FFmpeg.git
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49bf94536f
avpriv_mpeg4audio_sample_rates has a size of 64B and it is currently avpriv; a clone of it exists in aacenctab.h and from there it is inlined in aacenc.c (which also uses the avpriv version) and in the FLV muxer. This means that despite it being avpriv both libavformat as well as libavcodec have copies already. This situation is clearly suboptimal. Given the overhead of exporting symbols (for x64 Elf/Linux/GNU: 2x2B version, 2x24B .dynsym, 24B .rela.dyn, 8B .got, 4B hash + twice the size of the name (here 31B)) the object is unavprived, i.e. duplicated into libavformat when creating a shared build; but the duplicates in the AAC encoder and FLV muxer are removed. This involves splitting of the sample rate table into a file of its own; this allowed to break some spurious dependencies (e.g. both the AAC encoder as well as the Matroska demuxer actually don't need the mpeg4audio_get_config stuff). Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
134 lines
4.7 KiB
C
134 lines
4.7 KiB
C
/*
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* AAC encoder data
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* Copyright (c) 2015 Rostislav Pehlivanov ( atomnuker 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 encoder data
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* @author Rostislav Pehlivanov ( atomnuker gmail com )
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*/
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#ifndef AVCODEC_AACENCTAB_H
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#define AVCODEC_AACENCTAB_H
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#include "libavutil/channel_layout.h"
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#include "aac.h"
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/** Total number of usable codebooks **/
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#define CB_TOT 12
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/** Total number of codebooks, including special ones **/
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#define CB_TOT_ALL 15
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#define AAC_MAX_CHANNELS 16
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extern const uint8_t *const ff_aac_swb_size_1024[];
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extern const int ff_aac_swb_size_1024_len;
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extern const uint8_t *const ff_aac_swb_size_128[];
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extern const int ff_aac_swb_size_128_len;
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/* Supported layouts without using a PCE */
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static const int64_t aac_normal_chan_layouts[7] = {
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AV_CH_LAYOUT_MONO,
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AV_CH_LAYOUT_STEREO,
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AV_CH_LAYOUT_SURROUND,
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AV_CH_LAYOUT_4POINT0,
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AV_CH_LAYOUT_5POINT0_BACK,
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AV_CH_LAYOUT_5POINT1_BACK,
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AV_CH_LAYOUT_7POINT1,
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};
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/** default channel configurations */
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static const uint8_t aac_chan_configs[AAC_MAX_CHANNELS][6] = {
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{1, TYPE_SCE}, // 1 channel - single channel element
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{1, TYPE_CPE}, // 2 channels - channel pair
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{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
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{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
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{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
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{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
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{0}, // 7 channels - invalid without PCE
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{5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 8 channels - front center + front stereo + side stereo + back stereo + LFE
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};
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/**
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* Table to remap channels from libavcodec's default order to AAC order.
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*/
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static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
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{ 0 },
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{ 0, 1 },
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{ 2, 0, 1 },
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{ 2, 0, 1, 3 },
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{ 2, 0, 1, 3, 4 },
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{ 2, 0, 1, 4, 5, 3 },
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{ 0 },
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{ 2, 0, 1, 6, 7, 4, 5, 3 },
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};
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/** bits needed to code codebook run value for long windows */
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static const uint8_t run_value_bits_long[64] = {
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5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
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5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
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10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
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10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
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};
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/** bits needed to code codebook run value for short windows */
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static const uint8_t run_value_bits_short[16] = {
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3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
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};
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/* TNS starting SFBs for long and short windows */
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static const uint8_t tns_min_sfb_short[16] = {
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2, 2, 2, 3, 3, 4, 6, 6, 8, 10, 10, 12, 12, 12, 12, 12
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};
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static const uint8_t tns_min_sfb_long[16] = {
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12, 13, 15, 16, 17, 20, 25, 26, 24, 28, 30, 31, 31, 31, 31, 31
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};
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static const uint8_t * const tns_min_sfb[2] = {
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tns_min_sfb_long, tns_min_sfb_short
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};
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static const uint8_t * const run_value_bits[2] = {
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run_value_bits_long, run_value_bits_short
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};
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/** Map to convert values from BandCodingPath index to a codebook index **/
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static const uint8_t aac_cb_out_map[CB_TOT_ALL] = {0,1,2,3,4,5,6,7,8,9,10,11,13,14,15};
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/** Inverse map to convert from codebooks to BandCodingPath indices **/
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static const uint8_t aac_cb_in_map[CB_TOT_ALL+1] = {0,1,2,3,4,5,6,7,8,9,10,11,0,12,13,14};
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static const uint8_t aac_cb_range [12] = {0, 3, 3, 3, 3, 9, 9, 8, 8, 13, 13, 17};
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static const uint8_t aac_cb_maxval[12] = {0, 1, 1, 2, 2, 4, 4, 7, 7, 12, 12, 16};
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static const unsigned char aac_maxval_cb[] = {
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0, 1, 3, 5, 5, 7, 7, 7, 9, 9, 9, 9, 9, 11
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};
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static const int aacenc_profiles[] = {
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FF_PROFILE_AAC_MAIN,
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FF_PROFILE_AAC_LOW,
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FF_PROFILE_AAC_LTP,
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FF_PROFILE_MPEG2_AAC_LOW,
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};
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#endif /* AVCODEC_AACENCTAB_H */
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