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9383533770
The LTP profile of AAC is... terrible. It was an early 90's attempt at bridging the gap between speech codecs and general purpose codecs. It did so by trying to exploit the fact that most speech patterns are regular. Unfortunately, it went about it the same way as AAC Main, by taking the previous frame's samples, modifying them through an LPC filter, transforming them back using a forward MDCT, putting the output coefficients back into the current frame, and using delta coding. But once again, they ignored basic mathematics and MDCT leakage. Thankfully, because AAC LTP is meant to operate at very low bitrates, the extreme quantization results in most leakage being irrelevant. Unfortunately, the result is that the output sounds pretty much terrible regardless of whether LTP is enabled or not. This was the first attempt at trying to couple speech coding into AAC. No, the second attempt did not succeed either. Nnnneither did the third. Or fourth. For the fifth one, they literally just jammed a speech codec into AAC with USAC once they saw Opus do it. Just drop support for encoding AAC LTP. It was always experimental to begin with.
133 lines
4.7 KiB
C
133 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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#include "defs.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 AVChannelLayout aac_normal_chan_layouts[7] = {
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AV_CHANNEL_LAYOUT_MONO,
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AV_CHANNEL_LAYOUT_STEREO,
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AV_CHANNEL_LAYOUT_SURROUND,
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AV_CHANNEL_LAYOUT_4POINT0,
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AV_CHANNEL_LAYOUT_5POINT0_BACK,
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AV_CHANNEL_LAYOUT_5POINT1_BACK,
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AV_CHANNEL_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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AV_PROFILE_AAC_LOW,
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AV_PROFILE_MPEG2_AAC_LOW,
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};
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#endif /* AVCODEC_AACENCTAB_H */
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