Lynne
5ffea4c6f1
Better to be safer than usual, since the signal contains literally no audible data, and is so expensive to code, the coder itself is very reluctant to do so except at very high rates.
1694 lines
66 KiB
C
1694 lines
66 KiB
C
/*
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* AAC encoder
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* Copyright (C) 2008 Konstantin Shishkov
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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
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*/
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/***********************************
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* TODOs:
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* add sane pulse detection
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***********************************/
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#include <float.h>
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#include "libavutil/channel_layout.h"
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#include "libavutil/libm.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/mem.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "codec_internal.h"
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#include "encode.h"
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#include "put_bits.h"
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#include "mpeg4audio.h"
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#include "sinewin.h"
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#include "profiles.h"
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#include "version.h"
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#include "aac.h"
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#include "aactab.h"
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#include "aacenc.h"
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#include "aacenctab.h"
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#include "aacenc_utils.h"
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#include "psymodel.h"
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/**
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* List of PCE (Program Configuration Element) for the channel layouts listed
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* in channel_layout.h
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*
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* For those wishing in the future to add other layouts:
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*
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* - num_ele: number of elements in each group of front, side, back, lfe channels
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* (an element is of type SCE (single channel), CPE (channel pair) for
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* the first 3 groups; and is LFE for LFE group).
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*
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* - pairing: 0 for an SCE element or 1 for a CPE; does not apply to LFE group
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*
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* - index: there are three independent indices for SCE, CPE and LFE;
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* they are incremented irrespective of the group to which the element belongs;
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* they are not reset when going from one group to another
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*
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* Example: for 7.0 channel layout,
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* .pairing = { { 1, 0 }, { 1 }, { 1 }, }, (3 CPE and 1 SCE in front group)
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* .index = { { 0, 0 }, { 1 }, { 2 }, },
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* (index is 0 for the single SCE but goes from 0 to 2 for the CPEs)
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*
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* The index order impacts the channel ordering. But is otherwise arbitrary
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* (the sequence could have been 2, 0, 1 instead of 0, 1, 2).
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*
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* Spec allows for discontinuous indices, e.g. if one has a total of two SCE,
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* SCE.0 SCE.15 is OK per spec; BUT it won't be decoded by our AAC decoder
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* which at this time requires that indices fully cover some range starting
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* from 0 (SCE.1 SCE.0 is OK but not SCE.0 SCE.15).
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*
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* - config_map: total number of elements and their types. Beware, the way the
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* types are ordered impacts the final channel ordering.
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*
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* - reorder_map: reorders the channels.
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*
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*/
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static const AACPCEInfo aac_pce_configs[] = {
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{
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.layout = AV_CHANNEL_LAYOUT_MONO,
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.num_ele = { 1, 0, 0, 0 },
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.pairing = { { 0 }, },
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.index = { { 0 }, },
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.config_map = { 1, TYPE_SCE, },
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.reorder_map = { 0 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_STEREO,
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.num_ele = { 1, 0, 0, 0 },
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.pairing = { { 1 }, },
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.index = { { 0 }, },
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.config_map = { 1, TYPE_CPE, },
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.reorder_map = { 0, 1 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_2POINT1,
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.num_ele = { 1, 0, 0, 1 },
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.pairing = { { 1 }, },
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.index = { { 0 },{ 0 },{ 0 },{ 0 } },
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.config_map = { 2, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 0, 1, 2 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_2_1,
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.num_ele = { 1, 0, 1, 0 },
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.pairing = { { 1 },{ 0 },{ 0 } },
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.index = { { 0 },{ 0 },{ 0 }, },
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.config_map = { 2, TYPE_CPE, TYPE_SCE },
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.reorder_map = { 0, 1, 2 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_SURROUND,
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.num_ele = { 2, 0, 0, 0 },
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.pairing = { { 0, 1 }, },
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.index = { { 0, 0 }, },
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.config_map = { 2, TYPE_SCE, TYPE_CPE },
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.reorder_map = { 2, 0, 1 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_3POINT1,
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.num_ele = { 2, 0, 0, 1 },
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.pairing = { { 0, 1 }, },
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.index = { { 0, 0 }, { 0 }, { 0 }, { 0 }, },
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.config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_4POINT0,
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.num_ele = { 2, 0, 1, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 0 }, },
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.index = { { 0, 0 }, { 0 }, { 1 } },
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.config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_SCE },
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.reorder_map = { 2, 0, 1, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_4POINT1,
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.num_ele = { 2, 0, 1, 1 },
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.pairing = { { 0, 1 }, { 0 }, { 0 }, },
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.index = { { 0, 0 }, { 0 }, { 1 }, { 0 } },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 4, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_2_2,
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.num_ele = { 1, 0, 1, 0 },
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.pairing = { { 1 }, { 0 }, { 1 }, },
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.index = { { 0 }, { 0 }, { 1 } },
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.config_map = { 2, TYPE_CPE, TYPE_CPE },
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.reorder_map = { 0, 1, 2, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_QUAD,
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.num_ele = { 1, 0, 1, 0 },
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.pairing = { { 1 }, { 0 }, { 1 }, },
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.index = { { 0 }, { 0 }, { 1 } },
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.config_map = { 2, TYPE_CPE, TYPE_CPE },
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.reorder_map = { 0, 1, 2, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_5POINT0,
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.num_ele = { 2, 0, 1, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 1 } },
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.index = { { 0, 0 }, { 0 }, { 1 } },
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.config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_CPE },
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.reorder_map = { 2, 0, 1, 3, 4 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_5POINT1,
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.num_ele = { 2, 0, 1, 1 },
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.pairing = { { 0, 1 }, { 0 }, { 1 }, },
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.index = { { 0, 0 }, { 0 }, { 1 }, { 0 } },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 4, 5, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_5POINT0_BACK,
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.num_ele = { 2, 0, 1, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 1 } },
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.index = { { 0, 0 }, { 0 }, { 1 } },
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.config_map = { 3, TYPE_SCE, TYPE_CPE, TYPE_CPE },
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.reorder_map = { 2, 0, 1, 3, 4 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_5POINT1_BACK,
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.num_ele = { 2, 0, 1, 1 },
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.pairing = { { 0, 1 }, { 0 }, { 1 }, },
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.index = { { 0, 0 }, { 0 }, { 1 }, { 0 } },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 4, 5, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_6POINT0,
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.num_ele = { 2, 0, 2, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 1, 0 } },
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.index = { { 0, 0 }, { 0 }, { 1, 1 } },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
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.reorder_map = { 2, 0, 1, 4, 5, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_6POINT0_FRONT,
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.num_ele = { 2, 0, 1, 0 },
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.pairing = { { 1, 1 }, { 0 }, { 1 } },
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.index = { { 0, 1 }, { 0 }, { 2 }, },
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.config_map = { 3, TYPE_CPE, TYPE_CPE, TYPE_CPE, },
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.reorder_map = { 2, 3, 0, 1, 4, 5 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_HEXAGONAL,
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.num_ele = { 2, 0, 2, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 1, 0 } },
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.index = { { 0, 0 }, { 0 }, { 1, 1 } },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
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.reorder_map = { 2, 0, 1, 3, 4, 5 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_6POINT1,
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.num_ele = { 2, 0, 2, 1 },
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.pairing = { { 0, 1 }, { 0 }, { 1, 0 }, },
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.index = { { 0, 0 }, { 0 }, { 1, 1 }, { 0 } },
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.config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 5, 6, 4, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_6POINT1_BACK,
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.num_ele = { 2, 0, 2, 1 },
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.pairing = { { 0, 1 },{ 0 },{ 1, 0 }, },
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.index = { { 0, 0 },{ 0 },{ 1, 1 },{ 0 } },
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.config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 4, 5, 6, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_6POINT1_FRONT,
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.num_ele = { 2, 0, 1, 1 },
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.pairing = { { 1, 1 }, { 0 }, { 1 }, },
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.index = { { 0, 1 }, { 0 }, { 2 }, { 0 }, },
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.config_map = { 4, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE, },
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.reorder_map = { 3, 4, 0, 1, 5, 6, 2 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_7POINT0,
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.num_ele = { 2, 0, 2, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 1, 1 }, },
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.index = { { 0, 0 }, { 0 }, { 2, 1 }, },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE },
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.reorder_map = { 2, 0, 1, 3, 4, 5, 6 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_7POINT0_FRONT,
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.num_ele = { 3, 0, 1, 0 },
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.pairing = { { 0, 1, 1 }, { 0 }, { 1 }, },
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.index = { { 0, 0, 1 }, { 0 }, { 2 }, },
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.config_map = { 4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE },
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.reorder_map = { 2, 3, 4, 0, 1, 5, 6 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_7POINT1,
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.num_ele = { 2, 0, 2, 1 },
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.pairing = { { 0, 1 }, { 0 }, { 1, 1 }, },
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.index = { { 0, 0 }, { 0 }, { 2, 1 }, { 0 } },
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.config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 2, 0, 1, 4, 5, 6, 7, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_7POINT1_WIDE,
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.num_ele = { 3, 0, 1, 1 },
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.pairing = { { 0, 1, 1 }, { 0 }, { 1 }, },
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.index = { { 0, 0, 1 }, { 0 }, { 2 }, { 0 }, },
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.config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 2, 4, 5, 0, 1, 6, 7, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_7POINT1_WIDE_BACK,
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.num_ele = { 3, 0, 1, 1 },
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.pairing = { { 0, 1, 1 }, { 0 }, { 1 } },
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.index = { { 0, 0, 1 }, { 0 }, { 2 }, { 0 } },
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.config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_LFE },
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.reorder_map = { 2, 6, 7, 0, 1, 4, 5, 3 },
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},
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{
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.layout = AV_CHANNEL_LAYOUT_OCTAGONAL,
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.num_ele = { 2, 0, 3, 0 },
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.pairing = { { 0, 1 }, { 0 }, { 1, 1, 0 }, },
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.index = { { 0, 0 }, { 0 }, { 1, 2, 1 }, },
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.config_map = { 5, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE },
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.reorder_map = { 2, 0, 1, 6, 7, 3, 4, 5 },
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},
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};
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static void put_pce(PutBitContext *pb, AVCodecContext *avctx)
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{
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int i, j;
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AACEncContext *s = avctx->priv_data;
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AACPCEInfo *pce = &s->pce;
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const int bitexact = avctx->flags & AV_CODEC_FLAG_BITEXACT;
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const char *aux_data = bitexact ? "Lavc" : LIBAVCODEC_IDENT;
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put_bits(pb, 4, 0);
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put_bits(pb, 2, avctx->profile);
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put_bits(pb, 4, s->samplerate_index);
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put_bits(pb, 4, pce->num_ele[0]); /* Front */
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put_bits(pb, 4, pce->num_ele[1]); /* Side */
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put_bits(pb, 4, pce->num_ele[2]); /* Back */
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put_bits(pb, 2, pce->num_ele[3]); /* LFE */
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put_bits(pb, 3, 0); /* Assoc data */
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put_bits(pb, 4, 0); /* CCs */
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put_bits(pb, 1, 0); /* Stereo mixdown */
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put_bits(pb, 1, 0); /* Mono mixdown */
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put_bits(pb, 1, 0); /* Something else */
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for (i = 0; i < 4; i++) {
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for (j = 0; j < pce->num_ele[i]; j++) {
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if (i < 3)
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put_bits(pb, 1, pce->pairing[i][j]);
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put_bits(pb, 4, pce->index[i][j]);
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}
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}
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align_put_bits(pb);
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put_bits(pb, 8, strlen(aux_data));
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ff_put_string(pb, aux_data, 0);
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}
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/**
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* Make AAC audio config object.
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* @see 1.6.2.1 "Syntax - AudioSpecificConfig"
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*/
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static int put_audio_specific_config(AVCodecContext *avctx, int chcfg)
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{
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PutBitContext pb;
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AACEncContext *s = avctx->priv_data;
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const int max_size = 32;
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avctx->extradata = av_mallocz(max_size);
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if (!avctx->extradata)
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return AVERROR(ENOMEM);
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init_put_bits(&pb, avctx->extradata, max_size);
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put_bits(&pb, 5, s->profile+1); //profile
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put_bits(&pb, 4, s->samplerate_index); //sample rate index
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put_bits(&pb, 4, chcfg);
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//GASpecificConfig
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put_bits(&pb, 1, 0); //frame length - 1024 samples
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put_bits(&pb, 1, 0); //does not depend on core coder
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put_bits(&pb, 1, 0); //is not extension
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if (s->needs_pce)
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put_pce(&pb, avctx);
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//Explicitly Mark SBR absent
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put_bits(&pb, 11, 0x2b7); //sync extension
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put_bits(&pb, 5, AOT_SBR);
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put_bits(&pb, 1, 0);
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flush_put_bits(&pb);
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avctx->extradata_size = put_bytes_output(&pb);
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return 0;
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}
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void ff_quantize_band_cost_cache_init(struct AACEncContext *s)
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{
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++s->quantize_band_cost_cache_generation;
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if (s->quantize_band_cost_cache_generation == 0) {
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memset(s->quantize_band_cost_cache, 0, sizeof(s->quantize_band_cost_cache));
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s->quantize_band_cost_cache_generation = 1;
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}
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}
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#define WINDOW_FUNC(type) \
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static void apply_ ##type ##_window(AVFloatDSPContext *fdsp, \
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SingleChannelElement *sce, \
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const float *audio)
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WINDOW_FUNC(only_long)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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float *out = sce->ret_buf;
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fdsp->vector_fmul (out, audio, lwindow, 1024);
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fdsp->vector_fmul_reverse(out + 1024, audio + 1024, pwindow, 1024);
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}
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WINDOW_FUNC(long_start)
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{
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const float *lwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
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const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret_buf;
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fdsp->vector_fmul(out, audio, lwindow, 1024);
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memcpy(out + 1024, audio + 1024, sizeof(out[0]) * 448);
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fdsp->vector_fmul_reverse(out + 1024 + 448, audio + 1024 + 448, swindow, 128);
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memset(out + 1024 + 576, 0, sizeof(out[0]) * 448);
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}
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WINDOW_FUNC(long_stop)
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{
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const float *lwindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024;
|
|
const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
float *out = sce->ret_buf;
|
|
|
|
memset(out, 0, sizeof(out[0]) * 448);
|
|
fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
|
|
memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
|
|
fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
|
|
}
|
|
|
|
WINDOW_FUNC(eight_short)
|
|
{
|
|
const float *swindow = sce->ics.use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
|
|
const float *in = audio + 448;
|
|
float *out = sce->ret_buf;
|
|
int w;
|
|
|
|
for (w = 0; w < 8; w++) {
|
|
fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
|
|
out += 128;
|
|
in += 128;
|
|
fdsp->vector_fmul_reverse(out, in, swindow, 128);
|
|
out += 128;
|
|
}
|
|
}
|
|
|
|
static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
|
|
SingleChannelElement *sce,
|
|
const float *audio) = {
|
|
[ONLY_LONG_SEQUENCE] = apply_only_long_window,
|
|
[LONG_START_SEQUENCE] = apply_long_start_window,
|
|
[EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
|
|
[LONG_STOP_SEQUENCE] = apply_long_stop_window
|
|
};
|
|
|
|
static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
|
|
float *audio)
|
|
{
|
|
int i;
|
|
float *output = sce->ret_buf;
|
|
|
|
apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
|
|
|
|
if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
|
|
s->mdct1024_fn(s->mdct1024, sce->coeffs, output, sizeof(float));
|
|
else
|
|
for (i = 0; i < 1024; i += 128)
|
|
s->mdct128_fn(s->mdct128, &sce->coeffs[i], output + i*2, sizeof(float));
|
|
memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
|
|
memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
|
|
}
|
|
|
|
/**
|
|
* Encode ics_info element.
|
|
* @see Table 4.6 (syntax of ics_info)
|
|
*/
|
|
static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
|
|
{
|
|
int w;
|
|
|
|
put_bits(&s->pb, 1, 0); // ics_reserved bit
|
|
put_bits(&s->pb, 2, info->window_sequence[0]);
|
|
put_bits(&s->pb, 1, info->use_kb_window[0]);
|
|
if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
|
|
put_bits(&s->pb, 6, info->max_sfb);
|
|
put_bits(&s->pb, 1, 0); /* No predictor present */
|
|
} else {
|
|
put_bits(&s->pb, 4, info->max_sfb);
|
|
for (w = 1; w < 8; w++)
|
|
put_bits(&s->pb, 1, !info->group_len[w]);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode MS data.
|
|
* @see 4.6.8.1 "Joint Coding - M/S Stereo"
|
|
*/
|
|
static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
|
|
{
|
|
int i, w;
|
|
|
|
put_bits(pb, 2, cpe->ms_mode);
|
|
if (cpe->ms_mode == 1)
|
|
for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
|
|
for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
|
|
put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
|
|
}
|
|
|
|
/**
|
|
* Produce integer coefficients from scalefactors provided by the model.
|
|
*/
|
|
static void adjust_frame_information(ChannelElement *cpe, int chans)
|
|
{
|
|
int i, w, w2, g, ch;
|
|
int maxsfb, cmaxsfb;
|
|
|
|
for (ch = 0; ch < chans; ch++) {
|
|
IndividualChannelStream *ics = &cpe->ch[ch].ics;
|
|
maxsfb = 0;
|
|
cpe->ch[ch].pulse.num_pulse = 0;
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
|
|
for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
|
|
;
|
|
maxsfb = FFMAX(maxsfb, cmaxsfb);
|
|
}
|
|
ics->max_sfb = maxsfb;
|
|
|
|
//adjust zero bands for window groups
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
|
|
for (g = 0; g < ics->max_sfb; g++) {
|
|
i = 1;
|
|
for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
|
|
if (!cpe->ch[ch].zeroes[w2*16 + g]) {
|
|
i = 0;
|
|
break;
|
|
}
|
|
}
|
|
cpe->ch[ch].zeroes[w*16 + g] = i;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (chans > 1 && cpe->common_window) {
|
|
IndividualChannelStream *ics0 = &cpe->ch[0].ics;
|
|
IndividualChannelStream *ics1 = &cpe->ch[1].ics;
|
|
int msc = 0;
|
|
ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
|
|
ics1->max_sfb = ics0->max_sfb;
|
|
for (w = 0; w < ics0->num_windows*16; w += 16)
|
|
for (i = 0; i < ics0->max_sfb; i++)
|
|
if (cpe->ms_mask[w+i])
|
|
msc++;
|
|
if (msc == 0 || ics0->max_sfb == 0)
|
|
cpe->ms_mode = 0;
|
|
else
|
|
cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
|
|
}
|
|
}
|
|
|
|
static void apply_intensity_stereo(ChannelElement *cpe)
|
|
{
|
|
int w, w2, g, i;
|
|
IndividualChannelStream *ics = &cpe->ch[0].ics;
|
|
if (!cpe->common_window)
|
|
return;
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
|
|
for (w2 = 0; w2 < ics->group_len[w]; w2++) {
|
|
int start = (w+w2) * 128;
|
|
for (g = 0; g < ics->num_swb; g++) {
|
|
int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
|
|
float scale = cpe->ch[0].is_ener[w*16+g];
|
|
if (!cpe->is_mask[w*16 + g]) {
|
|
start += ics->swb_sizes[g];
|
|
continue;
|
|
}
|
|
if (cpe->ms_mask[w*16 + g])
|
|
p *= -1;
|
|
for (i = 0; i < ics->swb_sizes[g]; i++) {
|
|
float sum = (cpe->ch[0].coeffs[start+i] + p*cpe->ch[1].coeffs[start+i])*scale;
|
|
cpe->ch[0].coeffs[start+i] = sum;
|
|
cpe->ch[1].coeffs[start+i] = 0.0f;
|
|
}
|
|
start += ics->swb_sizes[g];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Intensity stereo is only allowed when its irreducible image error */
|
|
#define NMR_IS_IMG_GATE 0.5f
|
|
|
|
/* Frequency in Hz for the lower limit of intensity stereo */
|
|
#define NMR_IS_LOW_LIMIT 6100
|
|
|
|
/* Rate ceiling (bits/sample/channel) above which intensity is skipped, ~145kbps */
|
|
#define NMR_IS_MAXBPS 1.52f
|
|
|
|
/* The rate ceiling is lifted on hard-to-code frames. The signal is the bit
|
|
* reservoir going into deficit: a negative fill means the trellis is spending
|
|
* more than the nominal rate to hold quality (operating lambda has climbed). */
|
|
#define NMR_IS_FILLGAIN 0.27f
|
|
#define NMR_IS_FILLMAX 0.40f
|
|
|
|
/* M/S thresholds: a band is recoded as mid+side when the side is negligible */
|
|
#define NMR_MS_EQUIV 0.01f
|
|
#define NMR_MS_MASK 0.0f
|
|
|
|
/* PNS-stereo decorrelation gate: a band may be noise-substituted in a CPE only if its
|
|
* side energy is at least this fraction of its mid energy, i.e. the image is genuinely
|
|
* wide (channels decorrelated). PNS renders uncorrelated noise per channel, so it only
|
|
* preserves the image on already-wide bands; a much stricter bar than I/S (which can
|
|
* collapse correlated bands). Lower = more PNS / more imaging risk. */
|
|
#define NMR_PNS_STEREO_DECORR 0.6f
|
|
|
|
/* Recode one band's window group as mid+side in place, updating the psy band
|
|
* energies/thresholds to the M/S spectra. The threshold is halved as a coarse guard
|
|
* against L/R unmasking of the independently-quantized M/S noise (M/S is a lossless
|
|
* rotation but lossy coding). Used for the M/S decision and the intensity fallback. */
|
|
static void nmr_apply_ms_band(AACEncContext *s, ChannelElement *cpe,
|
|
int w, int g, int start, int len, int gl)
|
|
{
|
|
SingleChannelElement *sce0 = &cpe->ch[0];
|
|
SingleChannelElement *sce1 = &cpe->ch[1];
|
|
cpe->ms_mask[w*16+g] = 1;
|
|
for (int w2 = 0; w2 < gl; w2++) {
|
|
FFPsyBand *b0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
|
|
FFPsyBand *b1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
|
|
float *L = sce0->coeffs + start + (w+w2)*128;
|
|
float *R = sce1->coeffs + start + (w+w2)*128;
|
|
float em = 0.0f, es = 0.0f;
|
|
for (int i = 0; i < len; i++) {
|
|
float m = (L[i] + R[i]) * 0.5f;
|
|
R[i] = m - R[i]; L[i] = m;
|
|
em += L[i]*L[i]; es += R[i]*R[i];
|
|
}
|
|
b0->threshold = b1->threshold = FFMIN(b0->threshold, b1->threshold) * 0.5f;
|
|
b0->energy = em; b1->energy = es;
|
|
}
|
|
}
|
|
|
|
/* Intensity-stereo perceptual test for one band's window group: collapse the pair
|
|
* to a single carrier (L + p*R)*scale that the decoder rescales per channel, and
|
|
* check that the irreducible image error, which no bit budget can reduce, is
|
|
* masked in both channels. On success returns 1 and fills the carrier scale, the
|
|
* decoder's R/carrier ratio sr_, and the phase p. The caller restricts this to HF
|
|
* bands with energy in both channels. */
|
|
static int nmr_is_image_masked(AACEncContext *s, ChannelElement *cpe,
|
|
int w, int g, int start, int len, int gl,
|
|
float ener0, float ener1, float dot,
|
|
float minthr0, float minthr1,
|
|
float *scale_out, float *sr_out, int *p_out)
|
|
{
|
|
int p = dot >= 0.0f ? 1 : -1;
|
|
float ener01 = ener0 + ener1 + 2*p*dot; /* energy of L + p*R */
|
|
if (ener01 <= FLT_MIN)
|
|
return 0;
|
|
float scale = sqrtf(ener0 / ener01); /* carrier = (L + p*R)*scale */
|
|
float sr_ = sqrtf(ener1 / ener0); /* decoder: R = p*sr_*carrier */
|
|
float img0 = 0.0f, img1 = 0.0f;
|
|
for (int w2 = 0; w2 < gl; w2++) {
|
|
const float *L = cpe->ch[0].coeffs + start + (w+w2)*128;
|
|
const float *R = cpe->ch[1].coeffs + start + (w+w2)*128;
|
|
for (int i = 0; i < len; i++) {
|
|
float c = (L[i] + p*R[i]) * scale;
|
|
float dl = L[i] - c, dr = R[i] - p*sr_*c;
|
|
img0 += dl*dl; img1 += dr*dr;
|
|
}
|
|
}
|
|
if (img0 >= NMR_IS_IMG_GATE * minthr0 * gl ||
|
|
img1 >= NMR_IS_IMG_GATE * minthr1 * gl)
|
|
return 0;
|
|
*scale_out = scale; *sr_out = sr_; *p_out = p;
|
|
return 1;
|
|
}
|
|
|
|
/* Recode one band's window group as intensity stereo in place: replace L with the
|
|
* carrier, zero R, signal the phase via the side channel's band type, and fold the
|
|
* pair's masking into the surviving (carrier) channel. */
|
|
static void nmr_apply_is_band(AACEncContext *s, ChannelElement *cpe,
|
|
int w, int g, int start, int len, int gl,
|
|
float scale, float sr_, int p,
|
|
float ener0, float ener1)
|
|
{
|
|
cpe->is_mask[w*16+g] = 1;
|
|
cpe->ch[0].is_ener[w*16+g] = scale;
|
|
cpe->ch[1].is_ener[w*16+g] = ener0 / ener1;
|
|
cpe->ch[1].band_type[w*16+g] = p > 0 ? INTENSITY_BT : INTENSITY_BT2;
|
|
for (int w2 = 0; w2 < gl; w2++) {
|
|
FFPsyBand *b0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
|
|
FFPsyBand *b1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
|
|
float *L = cpe->ch[0].coeffs + start + (w+w2)*128;
|
|
float *R = cpe->ch[1].coeffs + start + (w+w2)*128;
|
|
float ec = 0.0f;
|
|
for (int i = 0; i < len; i++) {
|
|
L[i] = (L[i] + p*R[i]) * scale;
|
|
R[i] = 0.0f;
|
|
ec += L[i]*L[i];
|
|
}
|
|
b0->threshold = FFMIN(b0->threshold, b1->threshold / FFMAX(sr_*sr_, 1e-9f));
|
|
b0->energy = ec; b1->energy = 0.0f;
|
|
}
|
|
}
|
|
|
|
/*
|
|
* Per-band stereo-mode decision (L/R vs M/S vs intensity) for the NMR coder,
|
|
* made before quantization from the psychoacoustic model alone, so the
|
|
* quantizer search allocates natively on the spectra that are actually coded.
|
|
*/
|
|
static void nmr_decide_stereo(AACEncContext *s, ChannelElement *cpe)
|
|
{
|
|
SingleChannelElement *sce0 = &cpe->ch[0];
|
|
SingleChannelElement *sce1 = &cpe->ch[1];
|
|
IndividualChannelStream *ics = &sce0->ics;
|
|
const AVCodecContext *avctx = s->psy.avctx;
|
|
const float freq_mult = avctx->sample_rate / (1024.0f / ics->num_windows) / 2.0f;
|
|
const float bps = avctx->bit_rate > 0 ?
|
|
(float)avctx->bit_rate / avctx->sample_rate / avctx->ch_layout.nb_channels : 0.0f;
|
|
int is_count = 0;
|
|
|
|
/* Stereo decision, with no bitrate dependence. Start from full L/R and depart from
|
|
* it only where the change is inaudible. M/S and I/S differ in what they trade:
|
|
* M/S recodes the pair as mid+side -- an invertible rotation, but the M and S
|
|
* are quantized independently, so it is lossy coding whose noise un-mixes
|
|
* back to L/R. Used where it barely changes the result (the side is
|
|
* negligible vs the mid, so it is ~equivalent to L/R at the same rate) --
|
|
* OR where the doubled side energy is masked.
|
|
* I/S drops the side phase and keeps its energy, where the residual image error
|
|
* is masked. Used for the decorrelated HF that M/S cannot help.
|
|
* Both tests are content/perceptual and frame-stable, so the image holds. */
|
|
|
|
/* I/S rate gate: eligible at/below ~128 kbps, with the ceiling lifted on hard
|
|
* frames (bit reservoir in deficit) so a starved high-rate passage can still
|
|
* call on intensity. Where an I/S candidate is found but IS is not eligible, fall
|
|
* back to M/S: not free, but ~equivalent to L/R there and it lets the energy
|
|
* compact into the mid. */
|
|
const float rate_frame = avctx->bit_rate * 1024.0f / FFMAX(avctx->sample_rate, 1);
|
|
const float deficit = (s->nmr && rate_frame > 0.0f)
|
|
? FFMAX(0.0f, -(float)s->nmr->rc_fill / rate_frame) : 0.0f;
|
|
const float is_bonus = FFMIN(NMR_IS_FILLMAX, NMR_IS_FILLGAIN * deficit);
|
|
const int allow_is = s->options.intensity_stereo && bps < NMR_IS_MAXBPS + is_bonus;
|
|
|
|
for (int w = 0; w < ics->num_windows; w += ics->group_len[w]) {
|
|
int start = 0;
|
|
for (int g = 0; g < ics->num_swb; start += ics->swb_sizes[g++]) {
|
|
int len = ics->swb_sizes[g], gl = ics->group_len[w];
|
|
float ener0 = 0.0f, ener1 = 0.0f, dot = 0.0f, es_tot = 0.0f, em_tot = 0.0f;
|
|
float minthr0 = FLT_MAX, minthr1 = FLT_MAX;
|
|
|
|
cpe->is_mask[w*16+g] = 0;
|
|
cpe->ms_mask[w*16+g] = 0;
|
|
|
|
for (int w2 = 0; w2 < gl; w2++) {
|
|
FFPsyBand *b0 = &s->psy.ch[s->cur_channel+0].psy_bands[(w+w2)*16+g];
|
|
FFPsyBand *b1 = &s->psy.ch[s->cur_channel+1].psy_bands[(w+w2)*16+g];
|
|
const float *L = sce0->coeffs + start + (w+w2)*128;
|
|
const float *R = sce1->coeffs + start + (w+w2)*128;
|
|
float el = 0.0f, er = 0.0f, em = 0.0f, es = 0.0f, d = 0.0f;
|
|
for (int i = 0; i < len; i++) {
|
|
float m = (L[i] + R[i]) * 0.5f;
|
|
float sv = m - R[i];
|
|
el += L[i]*L[i]; er += R[i]*R[i];
|
|
em += m*m; es += sv*sv; d += L[i]*R[i];
|
|
}
|
|
ener0 += el; ener1 += er; dot += d; es_tot += es; em_tot += em;
|
|
minthr0 = FFMIN(minthr0, b0->threshold);
|
|
minthr1 = FFMIN(minthr1, b1->threshold);
|
|
}
|
|
float thr_g = FFMIN(minthr0, minthr1) * gl; /* group masking budget */
|
|
|
|
/* PNS-stereo reservation. Reserve a band for noise substitution only if it
|
|
* is noise-like in both channels (intersected can_pns) and clearly
|
|
* decorrelated (wide image). */
|
|
if (cpe->ch[0].can_pns[w*16+g] && cpe->ch[1].can_pns[w*16+g] &&
|
|
es_tot > NMR_PNS_STEREO_DECORR * em_tot)
|
|
continue;
|
|
cpe->ch[0].can_pns[w*16+g] = cpe->ch[1].can_pns[w*16+g] = 0;
|
|
|
|
int ms_ok = s->options.mid_side &&
|
|
(s->options.mid_side == 1 ||
|
|
es_tot < NMR_MS_EQUIV * em_tot ||
|
|
es_tot < NMR_MS_MASK * thr_g);
|
|
float scale, sr_; int p;
|
|
int is_ok = !ms_ok &&
|
|
start * freq_mult > NMR_IS_LOW_LIMIT &&
|
|
ener0 > FLT_MIN && ener1 > FLT_MIN &&
|
|
nmr_is_image_masked(s, cpe, w, g, start, len, gl,
|
|
ener0, ener1, dot, minthr0, minthr1,
|
|
&scale, &sr_, &p);
|
|
|
|
if (ms_ok) {
|
|
nmr_apply_ms_band(s, cpe, w, g, start, len, gl);
|
|
} else if (is_ok && allow_is) {
|
|
nmr_apply_is_band(s, cpe, w, g, start, len, gl,
|
|
scale, sr_, p, ener0, ener1);
|
|
is_count++;
|
|
} else if (is_ok && s->options.mid_side) {
|
|
nmr_apply_ms_band(s, cpe, w, g, start, len, gl);
|
|
}
|
|
/* else: keep full L/R stereo */
|
|
}
|
|
}
|
|
cpe->is_mode = !!is_count;
|
|
}
|
|
|
|
static void apply_mid_side_stereo(ChannelElement *cpe)
|
|
{
|
|
int w, w2, g, i;
|
|
IndividualChannelStream *ics = &cpe->ch[0].ics;
|
|
if (!cpe->common_window)
|
|
return;
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
|
|
for (w2 = 0; w2 < ics->group_len[w]; w2++) {
|
|
int start = (w+w2) * 128;
|
|
for (g = 0; g < ics->num_swb; g++) {
|
|
/* ms_mask can be used for other purposes in PNS and I/S,
|
|
* so must not apply M/S if any band uses either, even if
|
|
* ms_mask is set.
|
|
*/
|
|
if (!cpe->ms_mask[w*16 + g] || cpe->is_mask[w*16 + g]
|
|
|| cpe->ch[0].band_type[w*16 + g] >= NOISE_BT
|
|
|| cpe->ch[1].band_type[w*16 + g] >= NOISE_BT) {
|
|
start += ics->swb_sizes[g];
|
|
continue;
|
|
}
|
|
for (i = 0; i < ics->swb_sizes[g]; i++) {
|
|
float L = (cpe->ch[0].coeffs[start+i] + cpe->ch[1].coeffs[start+i]) * 0.5f;
|
|
float R = L - cpe->ch[1].coeffs[start+i];
|
|
cpe->ch[0].coeffs[start+i] = L;
|
|
cpe->ch[1].coeffs[start+i] = R;
|
|
}
|
|
start += ics->swb_sizes[g];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode scalefactor band coding type.
|
|
*/
|
|
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
|
|
{
|
|
int w;
|
|
|
|
if (s->coder->set_special_band_scalefactors)
|
|
s->coder->set_special_band_scalefactors(s, sce);
|
|
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w])
|
|
s->coder->encode_window_bands_info(s, sce, w, sce->ics.group_len[w], s->lambda);
|
|
}
|
|
|
|
/**
|
|
* Encode scalefactors.
|
|
*/
|
|
static void encode_scale_factors(AVCodecContext *avctx, AACEncContext *s,
|
|
SingleChannelElement *sce)
|
|
{
|
|
int diff, off_sf = sce->sf_idx[0], off_pns = sce->sf_idx[0] - NOISE_OFFSET;
|
|
int off_is = 0, noise_flag = 1;
|
|
int i, w;
|
|
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
for (i = 0; i < sce->ics.max_sfb; i++) {
|
|
if (!sce->zeroes[w*16 + i]) {
|
|
if (sce->band_type[w*16 + i] == NOISE_BT) {
|
|
diff = sce->sf_idx[w*16 + i] - off_pns;
|
|
off_pns = sce->sf_idx[w*16 + i];
|
|
if (noise_flag-- > 0) {
|
|
put_bits(&s->pb, NOISE_PRE_BITS, diff + NOISE_PRE);
|
|
continue;
|
|
}
|
|
} else if (sce->band_type[w*16 + i] == INTENSITY_BT ||
|
|
sce->band_type[w*16 + i] == INTENSITY_BT2) {
|
|
diff = sce->sf_idx[w*16 + i] - off_is;
|
|
off_is = sce->sf_idx[w*16 + i];
|
|
} else {
|
|
diff = sce->sf_idx[w*16 + i] - off_sf;
|
|
off_sf = sce->sf_idx[w*16 + i];
|
|
}
|
|
diff += SCALE_DIFF_ZERO;
|
|
av_assert0(diff >= 0 && diff <= 120);
|
|
put_bits(&s->pb, ff_aac_scalefactor_bits[diff], ff_aac_scalefactor_code[diff]);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode pulse data.
|
|
*/
|
|
static void encode_pulses(AACEncContext *s, Pulse *pulse)
|
|
{
|
|
int i;
|
|
|
|
put_bits(&s->pb, 1, !!pulse->num_pulse);
|
|
if (!pulse->num_pulse)
|
|
return;
|
|
|
|
put_bits(&s->pb, 2, pulse->num_pulse - 1);
|
|
put_bits(&s->pb, 6, pulse->start);
|
|
for (i = 0; i < pulse->num_pulse; i++) {
|
|
put_bits(&s->pb, 5, pulse->pos[i]);
|
|
put_bits(&s->pb, 4, pulse->amp[i]);
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode spectral coefficients processed by psychoacoustic model.
|
|
*/
|
|
static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
|
|
{
|
|
int start, i, w, w2;
|
|
|
|
for (w = 0; w < sce->ics.num_windows; w += sce->ics.group_len[w]) {
|
|
start = 0;
|
|
for (i = 0; i < sce->ics.max_sfb; i++) {
|
|
if (sce->zeroes[w*16 + i]) {
|
|
start += sce->ics.swb_sizes[i];
|
|
continue;
|
|
}
|
|
for (w2 = w; w2 < w + sce->ics.group_len[w]; w2++) {
|
|
s->coder->quantize_and_encode_band(s, &s->pb,
|
|
&sce->coeffs[start + w2*128],
|
|
NULL, sce->ics.swb_sizes[i],
|
|
sce->sf_idx[w*16 + i],
|
|
sce->band_type[w*16 + i],
|
|
s->lambda,
|
|
sce->ics.window_clipping[w]);
|
|
}
|
|
start += sce->ics.swb_sizes[i];
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Downscale spectral coefficients for near-clipping windows to avoid artifacts
|
|
*/
|
|
static void avoid_clipping(AACEncContext *s, SingleChannelElement *sce)
|
|
{
|
|
int start, i, j, w;
|
|
|
|
if (sce->ics.clip_avoidance_factor < 1.0f) {
|
|
for (w = 0; w < sce->ics.num_windows; w++) {
|
|
start = 0;
|
|
for (i = 0; i < sce->ics.max_sfb; i++) {
|
|
float *swb_coeffs = &sce->coeffs[start + w*128];
|
|
for (j = 0; j < sce->ics.swb_sizes[i]; j++)
|
|
swb_coeffs[j] *= sce->ics.clip_avoidance_factor;
|
|
start += sce->ics.swb_sizes[i];
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode one channel of audio data.
|
|
*/
|
|
static int encode_individual_channel(AVCodecContext *avctx, AACEncContext *s,
|
|
SingleChannelElement *sce,
|
|
int common_window)
|
|
{
|
|
put_bits(&s->pb, 8, sce->sf_idx[0]);
|
|
if (!common_window)
|
|
put_ics_info(s, &sce->ics);
|
|
encode_band_info(s, sce);
|
|
encode_scale_factors(avctx, s, sce);
|
|
encode_pulses(s, &sce->pulse);
|
|
put_bits(&s->pb, 1, !!sce->tns.present);
|
|
if (s->coder->encode_tns_info)
|
|
s->coder->encode_tns_info(s, sce);
|
|
put_bits(&s->pb, 1, 0); //ssr
|
|
encode_spectral_coeffs(s, sce);
|
|
return 0;
|
|
}
|
|
|
|
/**
|
|
* Write some auxiliary information about the created AAC file.
|
|
*/
|
|
static void put_bitstream_info(AACEncContext *s, const char *name)
|
|
{
|
|
int i, namelen, padbits;
|
|
|
|
namelen = strlen(name) + 2;
|
|
put_bits(&s->pb, 3, TYPE_FIL);
|
|
put_bits(&s->pb, 4, FFMIN(namelen, 15));
|
|
if (namelen >= 15)
|
|
put_bits(&s->pb, 8, namelen - 14);
|
|
put_bits(&s->pb, 4, 0); //extension type - filler
|
|
padbits = -put_bits_count(&s->pb) & 7;
|
|
align_put_bits(&s->pb);
|
|
for (i = 0; i < namelen - 2; i++)
|
|
put_bits(&s->pb, 8, name[i]);
|
|
put_bits(&s->pb, 12 - padbits, 0);
|
|
}
|
|
|
|
/*
|
|
* Copy input samples.
|
|
* Channels are reordered from libavcodec's default order to AAC order.
|
|
*/
|
|
static void copy_input_samples(AACEncContext *s, const AVFrame *frame)
|
|
{
|
|
int ch;
|
|
int end = 2048 + (frame ? frame->nb_samples : 0);
|
|
const uint8_t *channel_map = s->reorder_map;
|
|
|
|
/* copy and remap input samples */
|
|
for (ch = 0; ch < s->channels; ch++) {
|
|
/* copy last 1024 samples of previous frame to the start of the current frame */
|
|
memcpy(&s->planar_samples[ch][1024], &s->planar_samples[ch][2048], 1024 * sizeof(s->planar_samples[0][0]));
|
|
|
|
/* copy new samples and zero any remaining samples */
|
|
if (frame) {
|
|
memcpy(&s->planar_samples[ch][2048],
|
|
frame->extended_data[channel_map[ch]],
|
|
frame->nb_samples * sizeof(s->planar_samples[0][0]));
|
|
}
|
|
memset(&s->planar_samples[ch][end], 0,
|
|
(3072 - end) * sizeof(s->planar_samples[0][0]));
|
|
}
|
|
}
|
|
|
|
static int aac_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
|
|
const AVFrame *frame, int *got_packet_ptr)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
float **samples = s->planar_samples, *samples2, *la, *overlap;
|
|
ChannelElement *cpe;
|
|
SingleChannelElement *sce;
|
|
IndividualChannelStream *ics;
|
|
int i, its, ch, w, chans, tag, start_ch, ret, frame_bits;
|
|
int target_bits, rate_bits, too_many_bits, too_few_bits;
|
|
int ms_mode = 0, is_mode = 0, tns_mode = 0, pred_mode = 0;
|
|
int chan_el_counter[4];
|
|
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
|
|
|
|
/* add current frame to queue */
|
|
if (frame) {
|
|
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
|
|
return ret;
|
|
} else {
|
|
if (!s->afq.remaining_samples || (!s->afq.frame_alloc && !s->afq.frame_count))
|
|
return 0;
|
|
}
|
|
|
|
copy_input_samples(s, frame);
|
|
|
|
if (!avctx->frame_num)
|
|
return 0;
|
|
|
|
start_ch = 0;
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
FFPsyWindowInfo* wi = windows + start_ch;
|
|
tag = s->chan_map[i+1];
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
for (ch = 0; ch < chans; ch++) {
|
|
int k;
|
|
float clip_avoidance_factor;
|
|
sce = &cpe->ch[ch];
|
|
ics = &sce->ics;
|
|
s->cur_channel = start_ch + ch;
|
|
overlap = &samples[s->cur_channel][0];
|
|
samples2 = overlap + 1024;
|
|
la = samples2 + (448+64);
|
|
if (!frame)
|
|
la = NULL;
|
|
if (tag == TYPE_LFE) {
|
|
wi[ch].window_type[0] = wi[ch].window_type[1] = ONLY_LONG_SEQUENCE;
|
|
wi[ch].window_shape = 0;
|
|
wi[ch].num_windows = 1;
|
|
wi[ch].grouping[0] = 1;
|
|
wi[ch].clipping[0] = 0;
|
|
|
|
/* Only the lowest 12 coefficients are used in a LFE channel.
|
|
* The expression below results in only the bottom 8 coefficients
|
|
* being used for 11.025kHz to 16kHz sample rates.
|
|
*/
|
|
ics->num_swb = s->samplerate_index >= 8 ? 1 : 3;
|
|
} else {
|
|
wi[ch] = s->psy.model->window(&s->psy, samples2, la, s->cur_channel,
|
|
ics->window_sequence[0]);
|
|
}
|
|
ics->window_sequence[1] = ics->window_sequence[0];
|
|
ics->window_sequence[0] = wi[ch].window_type[0];
|
|
ics->use_kb_window[1] = ics->use_kb_window[0];
|
|
ics->use_kb_window[0] = wi[ch].window_shape;
|
|
ics->num_windows = wi[ch].num_windows;
|
|
ics->swb_sizes = s->psy.bands [ics->num_windows == 8];
|
|
ics->num_swb = tag == TYPE_LFE ? ics->num_swb : s->psy.num_bands[ics->num_windows == 8];
|
|
ics->max_sfb = FFMIN(ics->max_sfb, ics->num_swb);
|
|
ics->swb_offset = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
|
|
ff_swb_offset_128 [s->samplerate_index]:
|
|
ff_swb_offset_1024[s->samplerate_index];
|
|
ics->tns_max_bands = wi[ch].window_type[0] == EIGHT_SHORT_SEQUENCE ?
|
|
ff_tns_max_bands_128 [s->samplerate_index]:
|
|
ff_tns_max_bands_1024[s->samplerate_index];
|
|
|
|
for (w = 0; w < ics->num_windows; w++)
|
|
ics->group_len[w] = wi[ch].grouping[w];
|
|
|
|
/* Calculate input sample maximums and evaluate clipping risk */
|
|
clip_avoidance_factor = 0.0f;
|
|
for (w = 0; w < ics->num_windows; w++) {
|
|
const float *wbuf = overlap + w * 128;
|
|
const int wlen = 2048 / ics->num_windows;
|
|
float max = 0;
|
|
int j;
|
|
/* mdct input is 2 * output */
|
|
for (j = 0; j < wlen; j++)
|
|
max = FFMAX(max, fabsf(wbuf[j]));
|
|
wi[ch].clipping[w] = max;
|
|
}
|
|
for (w = 0; w < ics->num_windows; w++) {
|
|
if (wi[ch].clipping[w] > CLIP_AVOIDANCE_FACTOR) {
|
|
ics->window_clipping[w] = 1;
|
|
clip_avoidance_factor = FFMAX(clip_avoidance_factor, wi[ch].clipping[w]);
|
|
} else {
|
|
ics->window_clipping[w] = 0;
|
|
}
|
|
}
|
|
if (clip_avoidance_factor > CLIP_AVOIDANCE_FACTOR) {
|
|
ics->clip_avoidance_factor = CLIP_AVOIDANCE_FACTOR / clip_avoidance_factor;
|
|
} else {
|
|
ics->clip_avoidance_factor = 1.0f;
|
|
}
|
|
|
|
apply_window_and_mdct(s, sce, overlap);
|
|
|
|
for (k = 0; k < 1024; k++) {
|
|
if (!(fabs(cpe->ch[ch].coeffs[k]) < 1E16)) { // Ensure headroom for energy calculation
|
|
av_log(avctx, AV_LOG_ERROR, "Input contains (near) NaN/+-Inf\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
}
|
|
avoid_clipping(s, sce);
|
|
}
|
|
start_ch += chans;
|
|
}
|
|
if ((ret = ff_alloc_packet(avctx, avpkt, 8192 * s->channels)) < 0)
|
|
return ret;
|
|
frame_bits = its = 0;
|
|
do {
|
|
init_put_bits(&s->pb, avpkt->data, avpkt->size);
|
|
|
|
if ((avctx->frame_num & 0xFF)==1 && !(avctx->flags & AV_CODEC_FLAG_BITEXACT))
|
|
put_bitstream_info(s, LIBAVCODEC_IDENT);
|
|
start_ch = 0;
|
|
target_bits = 0;
|
|
memset(chan_el_counter, 0, sizeof(chan_el_counter));
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
FFPsyWindowInfo* wi = windows + start_ch;
|
|
const float *coeffs[2];
|
|
tag = s->chan_map[i+1];
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
cpe->common_window = 0;
|
|
memset(cpe->is_mask, 0, sizeof(cpe->is_mask));
|
|
memset(cpe->ms_mask, 0, sizeof(cpe->ms_mask));
|
|
put_bits(&s->pb, 3, tag);
|
|
put_bits(&s->pb, 4, chan_el_counter[tag]++);
|
|
for (ch = 0; ch < chans; ch++) {
|
|
sce = &cpe->ch[ch];
|
|
coeffs[ch] = sce->coeffs;
|
|
memset(&sce->tns, 0, sizeof(TemporalNoiseShaping));
|
|
for (w = 0; w < 128; w++)
|
|
if (sce->band_type[w] > RESERVED_BT)
|
|
sce->band_type[w] = 0;
|
|
}
|
|
s->psy.bitres.alloc = -1;
|
|
s->psy.bitres.bits = s->last_frame_pb_count / s->channels;
|
|
s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
|
|
if (s->psy.bitres.alloc > 0) {
|
|
/* Lambda unused here on purpose, we need to take psy's unscaled allocation */
|
|
target_bits += s->psy.bitres.alloc
|
|
* (s->lambda / (avctx->global_quality ? avctx->global_quality : 120));
|
|
s->psy.bitres.alloc /= chans;
|
|
}
|
|
s->cur_type = tag;
|
|
if (chans > 1
|
|
&& wi[0].window_type[0] == wi[1].window_type[0]
|
|
&& wi[0].window_shape == wi[1].window_shape) {
|
|
|
|
cpe->common_window = 1;
|
|
for (w = 0; w < wi[0].num_windows; w++) {
|
|
if (wi[0].grouping[w] != wi[1].grouping[w]) {
|
|
cpe->common_window = 0;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
const int use_tns = s->options.tns && s->coder->search_for_tns &&
|
|
s->coder->apply_tns_filt;
|
|
|
|
/* The NMR coder rate-controls itself and never re-quantizes, so TNS must run
|
|
* before the quantizer */
|
|
const int tns_first = s->options.coder == AAC_CODER_NMR;
|
|
if (tns_first && use_tns) {
|
|
for (ch = 0; ch < chans; ch++) {
|
|
sce = &cpe->ch[ch];
|
|
s->cur_channel = start_ch + ch;
|
|
/* mono: mark_pns before TNS so the region cap sees PNS bands. Stereo
|
|
* PNS is marked in its own block (below) after the stereo decision. */
|
|
if (chans == 1 && s->options.pns && s->coder->mark_pns)
|
|
s->coder->mark_pns(s, avctx, sce);
|
|
s->coder->search_for_tns(s, sce);
|
|
s->coder->apply_tns_filt(s, sce);
|
|
if (sce->tns.present)
|
|
tns_mode = 1;
|
|
}
|
|
}
|
|
|
|
/* NMR stereo PNS (imaging-safe). Mark each channel's noise-like bands on the
|
|
* original L/R psy, then keep PNS only where BOTH channels are noise-like. */
|
|
if (chans == 2 && cpe->common_window && tns_first &&
|
|
s->options.pns && s->coder->mark_pns) {
|
|
s->cur_channel = start_ch; s->coder->mark_pns(s, avctx, &cpe->ch[0]);
|
|
s->cur_channel = start_ch + 1; s->coder->mark_pns(s, avctx, &cpe->ch[1]);
|
|
for (int b = 0; b < 128; b++)
|
|
if (!cpe->ch[0].can_pns[b] || !cpe->ch[1].can_pns[b])
|
|
cpe->ch[0].can_pns[b] = cpe->ch[1].can_pns[b] = 0;
|
|
}
|
|
|
|
/* The NMR coder decides I/S and M/S BEFORE quantization, from the psy model,
|
|
* and the trellis then allocates natively on the coeffs actually coded. */
|
|
if (chans == 2 && cpe->common_window && s->options.coder == AAC_CODER_NMR &&
|
|
(s->options.mid_side || s->options.intensity_stereo)) {
|
|
s->cur_channel = start_ch;
|
|
nmr_decide_stereo(s, cpe);
|
|
}
|
|
for (ch = 0; ch < chans; ch++) {
|
|
s->cur_channel = start_ch + ch;
|
|
/* NMR PNS is mono-only */
|
|
if (s->options.pns && s->coder->mark_pns && !tns_first)
|
|
s->coder->mark_pns(s, avctx, &cpe->ch[ch]);
|
|
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
|
|
}
|
|
for (ch = 0; ch < chans; ch++) { /* TNS (non-NMR) and PNS */
|
|
sce = &cpe->ch[ch];
|
|
s->cur_channel = start_ch + ch;
|
|
if (!tns_first && use_tns) {
|
|
s->coder->search_for_tns(s, sce);
|
|
s->coder->apply_tns_filt(s, sce);
|
|
if (sce->tns.present)
|
|
tns_mode = 1;
|
|
}
|
|
if (s->options.pns && s->coder->search_for_pns)
|
|
s->coder->search_for_pns(s, avctx, sce);
|
|
}
|
|
s->cur_channel = start_ch;
|
|
if (s->options.intensity_stereo) { /* Intensity Stereo */
|
|
if (s->options.coder != AAC_CODER_NMR) { /* NMR: decided pre-search */
|
|
if (s->coder->search_for_is)
|
|
s->coder->search_for_is(s, avctx, cpe);
|
|
apply_intensity_stereo(cpe);
|
|
}
|
|
if (cpe->is_mode) is_mode = 1;
|
|
}
|
|
if (s->options.mid_side && s->options.coder != AAC_CODER_NMR) { /* Mid/Side stereo */
|
|
if (s->options.mid_side == -1 && s->coder->search_for_ms)
|
|
s->coder->search_for_ms(s, cpe);
|
|
else if (cpe->common_window)
|
|
memset(cpe->ms_mask, 1, sizeof(cpe->ms_mask));
|
|
apply_mid_side_stereo(cpe);
|
|
}
|
|
adjust_frame_information(cpe, chans);
|
|
if (chans == 2) {
|
|
put_bits(&s->pb, 1, cpe->common_window);
|
|
if (cpe->common_window) {
|
|
put_ics_info(s, &cpe->ch[0].ics);
|
|
encode_ms_info(&s->pb, cpe);
|
|
if (cpe->ms_mode) ms_mode = 1;
|
|
}
|
|
}
|
|
for (ch = 0; ch < chans; ch++) {
|
|
s->cur_channel = start_ch + ch;
|
|
encode_individual_channel(avctx, s, &cpe->ch[ch], cpe->common_window);
|
|
}
|
|
start_ch += chans;
|
|
}
|
|
|
|
if (avctx->flags & AV_CODEC_FLAG_QSCALE) {
|
|
/* When using a constant Q-scale, don't mess with lambda */
|
|
break;
|
|
}
|
|
|
|
frame_bits = put_bits_count(&s->pb);
|
|
|
|
/* The NMR coder rate-controls itself (global-lambda reservoir servo):
|
|
* per-frame bits intentionally float around the nominal rate, so skip
|
|
* the lambda rate loop and only intervene on a hard overflow. */
|
|
if (s->options.coder == AAC_CODER_NMR && avctx->bit_rate_tolerance != 0 &&
|
|
frame_bits < 6144 * s->channels - 3)
|
|
break;
|
|
|
|
/* rate control stuff
|
|
* allow between the nominal bitrate, and what psy's bit reservoir says to target
|
|
* but drift towards the nominal bitrate always
|
|
*/
|
|
rate_bits = avctx->bit_rate * 1024 / avctx->sample_rate;
|
|
rate_bits = FFMIN(rate_bits, 6144 * s->channels - 3);
|
|
too_many_bits = FFMAX(target_bits, rate_bits);
|
|
too_many_bits = FFMIN(too_many_bits, 6144 * s->channels - 3);
|
|
too_few_bits = FFMIN(FFMAX(rate_bits - rate_bits/4, target_bits), too_many_bits);
|
|
|
|
/* When strict bit-rate control is demanded */
|
|
if (avctx->bit_rate_tolerance == 0) {
|
|
if (rate_bits < frame_bits) {
|
|
float ratio = ((float)rate_bits) / frame_bits;
|
|
s->lambda *= FFMIN(0.9f, ratio);
|
|
continue;
|
|
}
|
|
/* reset lambda when solution is found */
|
|
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
|
|
break;
|
|
}
|
|
|
|
/* When using ABR, be strict (but only for increasing) */
|
|
too_few_bits = too_few_bits - too_few_bits/8;
|
|
too_many_bits = too_many_bits + too_many_bits/2;
|
|
|
|
if ( its == 0 /* for steady-state Q-scale tracking */
|
|
|| (its < 5 && (frame_bits < too_few_bits || frame_bits > too_many_bits))
|
|
|| frame_bits >= 6144 * s->channels - 3 )
|
|
{
|
|
float ratio = ((float)rate_bits) / frame_bits;
|
|
|
|
if (frame_bits >= too_few_bits && frame_bits <= too_many_bits) {
|
|
/*
|
|
* This path is for steady-state Q-scale tracking
|
|
* When frame bits fall within the stable range, we still need to adjust
|
|
* lambda to maintain it like so in a stable fashion (large jumps in lambda
|
|
* create artifacts and should be avoided), but slowly
|
|
*/
|
|
ratio = sqrtf(sqrtf(ratio));
|
|
ratio = av_clipf(ratio, 0.9f, 1.1f);
|
|
} else {
|
|
/* Not so fast though */
|
|
ratio = sqrtf(ratio);
|
|
}
|
|
s->lambda = av_clipf(s->lambda * ratio, FLT_EPSILON, 65536.f);
|
|
|
|
/* Keep iterating if we must reduce and lambda is in the sky */
|
|
if (ratio > 0.9f && ratio < 1.1f) {
|
|
break;
|
|
} else {
|
|
if (is_mode || ms_mode || tns_mode || pred_mode) {
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
// Must restore coeffs
|
|
chans = tag == TYPE_CPE ? 2 : 1;
|
|
cpe = &s->cpe[i];
|
|
for (ch = 0; ch < chans; ch++)
|
|
memcpy(cpe->ch[ch].coeffs, cpe->ch[ch].pcoeffs, sizeof(cpe->ch[ch].coeffs));
|
|
}
|
|
}
|
|
its++;
|
|
}
|
|
} else {
|
|
break;
|
|
}
|
|
} while (1);
|
|
|
|
/* tool-usage stats over the final per-band decisions of this frame */
|
|
for (i = 0; i < s->chan_map[0]; i++) {
|
|
int etag = s->chan_map[i + 1], echans = etag == TYPE_CPE ? 2 : 1;
|
|
ChannelElement *ce = &s->cpe[i];
|
|
IndividualChannelStream *ics = &ce->ch[0].ics;
|
|
for (ch = 0; ch < echans; ch++) { /* per-channel frame stats */
|
|
int is_short = ce->ch[ch].ics.window_sequence[0] == EIGHT_SHORT_SEQUENCE;
|
|
s->stat_chans++;
|
|
if (is_short)
|
|
s->stat_short++;
|
|
if (ce->ch[ch].tns.present) {
|
|
if (is_short) s->stat_tns_short++;
|
|
else s->stat_tns_long++;
|
|
}
|
|
}
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
|
|
for (int g = 0; g < ics->num_swb; g++) {
|
|
int idx = w*16 + g, coded = 0;
|
|
for (ch = 0; ch < echans; ch++) {
|
|
SingleChannelElement *sce = &ce->ch[ch];
|
|
if (sce->zeroes[idx] && sce->band_type[idx] == 0)
|
|
continue;
|
|
s->stat_ch_bands++;
|
|
if (sce->band_type[idx] == NOISE_BT)
|
|
s->stat_pns++;
|
|
coded = 1;
|
|
}
|
|
if (etag == TYPE_CPE && coded) {
|
|
s->stat_cpe_bands++;
|
|
if (ce->ms_mask[idx]) s->stat_ms++;
|
|
if (ce->is_mask[idx]) s->stat_is++;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
put_bits(&s->pb, 3, TYPE_END);
|
|
flush_put_bits(&s->pb);
|
|
|
|
s->last_frame_pb_count = put_bits_count(&s->pb);
|
|
|
|
/* NMR rate accounting: how many bits the frame really took beyond what the
|
|
* trellis counted; feeds the next frame's budget correction */
|
|
if (s->nmr) {
|
|
int counted = 0;
|
|
for (i = 0; i < s->channels; i++)
|
|
counted += s->nmr->counted[i];
|
|
if (counted > 0) {
|
|
float side = (float)s->last_frame_pb_count - counted;
|
|
if (s->nmr->side_inited) {
|
|
s->nmr->side_ema += 0.125f * (side - s->nmr->side_ema);
|
|
} else {
|
|
s->nmr->side_ema = side;
|
|
s->nmr->side_inited = 1;
|
|
}
|
|
}
|
|
}
|
|
avpkt->size = put_bytes_output(&s->pb);
|
|
|
|
s->lambda_sum += (s->nmr && s->nmr->lam_rc > 0.0f) ? s->nmr->lam_rc : s->lambda;
|
|
s->lambda_count++;
|
|
|
|
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
|
|
&avpkt->duration);
|
|
|
|
int discard_padding = avctx->frame_size - ff_samples_from_time_base(avctx, avpkt->duration);
|
|
if (discard_padding > 0) {
|
|
uint8_t *side_data =
|
|
av_packet_new_side_data(avpkt, AV_PKT_DATA_SKIP_SAMPLES, 10);
|
|
if (!side_data)
|
|
return AVERROR(ENOMEM);
|
|
AV_WL32(side_data + 4, discard_padding);
|
|
}
|
|
|
|
avpkt->flags |= AV_PKT_FLAG_KEY;
|
|
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int aac_encode_end(AVCodecContext *avctx)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
|
|
av_log(avctx, AV_LOG_INFO,
|
|
"Qavg: %.3f Tr: %.1f%% TNS(L): %.1f%% TNS(S): %.1f%% M/S: %.1f%% I/S: %.1f%% PNS: %.1f%%\n",
|
|
s->lambda_count ? s->lambda_sum / s->lambda_count : NAN,
|
|
s->stat_chans ? 100.0 * s->stat_short / s->stat_chans : 0.0,
|
|
s->stat_chans - s->stat_short ? 100.0 * s->stat_tns_long / (s->stat_chans - s->stat_short) : 0.0,
|
|
s->stat_short ? 100.0 * s->stat_tns_short / s->stat_short : 0.0,
|
|
s->stat_cpe_bands ? 100.0 * s->stat_ms / s->stat_cpe_bands : 0.0,
|
|
s->stat_cpe_bands ? 100.0 * s->stat_is / s->stat_cpe_bands : 0.0,
|
|
s->stat_ch_bands ? 100.0 * s->stat_pns / s->stat_ch_bands : 0.0);
|
|
|
|
av_tx_uninit(&s->mdct1024);
|
|
av_tx_uninit(&s->mdct128);
|
|
ff_psy_end(&s->psy);
|
|
ff_lpc_end(&s->lpc);
|
|
av_freep(&s->buffer.samples);
|
|
av_freep(&s->cpe);
|
|
av_freep(&s->fdsp);
|
|
av_freep(&s->nmr);
|
|
ff_af_queue_close(&s->afq);
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int dsp_init(AVCodecContext *avctx, AACEncContext *s)
|
|
{
|
|
int ret = 0;
|
|
float scale = 32768.0f;
|
|
|
|
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
|
|
if (!s->fdsp)
|
|
return AVERROR(ENOMEM);
|
|
|
|
if ((ret = av_tx_init(&s->mdct1024, &s->mdct1024_fn, AV_TX_FLOAT_MDCT, 0,
|
|
1024, &scale, 0)) < 0)
|
|
return ret;
|
|
if ((ret = av_tx_init(&s->mdct128, &s->mdct128_fn, AV_TX_FLOAT_MDCT, 0,
|
|
128, &scale, 0)) < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
|
|
{
|
|
int ch;
|
|
if (!FF_ALLOCZ_TYPED_ARRAY(s->buffer.samples, s->channels * 3 * 1024) ||
|
|
!FF_ALLOCZ_TYPED_ARRAY(s->cpe, s->chan_map[0]))
|
|
return AVERROR(ENOMEM);
|
|
|
|
for(ch = 0; ch < s->channels; ch++)
|
|
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
|
|
|
|
if (s->options.coder == AAC_CODER_NMR) {
|
|
s->nmr = av_mallocz(sizeof(*s->nmr));
|
|
if (!s->nmr)
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int aac_encode_init(AVCodecContext *avctx)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
int i, ret = 0;
|
|
int chcfg;
|
|
const uint8_t *sizes[2];
|
|
uint8_t grouping[AAC_MAX_CHANNELS];
|
|
int lengths[2];
|
|
|
|
/* Constants */
|
|
s->last_frame_pb_count = 0;
|
|
avctx->frame_size = 1024;
|
|
avctx->initial_padding = 1024;
|
|
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
|
|
|
|
/* Channel map and unspecified bitrate guessing */
|
|
s->channels = avctx->ch_layout.nb_channels;
|
|
|
|
s->needs_pce = 1;
|
|
for (chcfg = 1; chcfg < FF_ARRAY_ELEMS(aac_normal_chan_layouts); chcfg++) {
|
|
if (!av_channel_layout_compare(&avctx->ch_layout, &aac_normal_chan_layouts[chcfg])) {
|
|
s->needs_pce = s->options.pce;
|
|
break;
|
|
}
|
|
}
|
|
|
|
if (s->needs_pce) {
|
|
char buf[64];
|
|
for (i = 0; i < FF_ARRAY_ELEMS(aac_pce_configs); i++)
|
|
if (!av_channel_layout_compare(&avctx->ch_layout, &aac_pce_configs[i].layout))
|
|
break;
|
|
av_channel_layout_describe(&avctx->ch_layout, buf, sizeof(buf));
|
|
if (i == FF_ARRAY_ELEMS(aac_pce_configs)) {
|
|
av_log(avctx, AV_LOG_ERROR, "Unsupported channel layout \"%s\"\n", buf);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
av_log(avctx, AV_LOG_INFO, "Using a PCE to encode channel layout \"%s\"\n", buf);
|
|
s->pce = aac_pce_configs[i];
|
|
s->reorder_map = s->pce.reorder_map;
|
|
s->chan_map = s->pce.config_map;
|
|
chcfg = 0;
|
|
} else {
|
|
s->reorder_map = aac_chan_maps[chcfg - 1];
|
|
s->chan_map = aac_chan_configs[chcfg - 1];
|
|
}
|
|
|
|
if (!avctx->bit_rate) {
|
|
for (i = 1; i <= s->chan_map[0]; i++) {
|
|
avctx->bit_rate += s->chan_map[i] == TYPE_CPE ? 128000 : /* Pair */
|
|
s->chan_map[i] == TYPE_LFE ? 16000 : /* LFE */
|
|
69000 ; /* SCE */
|
|
}
|
|
}
|
|
|
|
/* Samplerate */
|
|
for (int i = 0;; i++) {
|
|
av_assert1(i < 13);
|
|
if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) {
|
|
s->samplerate_index = i;
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* Bitrate limiting */
|
|
WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
|
|
"Too many bits %f > %d per frame requested, clamping to max\n",
|
|
1024.0 * avctx->bit_rate / avctx->sample_rate,
|
|
6144 * s->channels);
|
|
avctx->bit_rate = (int64_t)FFMIN(6144 * s->channels / 1024.0 * avctx->sample_rate,
|
|
avctx->bit_rate);
|
|
|
|
/* Profile and option setting */
|
|
avctx->profile = avctx->profile == AV_PROFILE_UNKNOWN ? AV_PROFILE_AAC_LOW :
|
|
avctx->profile;
|
|
for (i = 0; i < FF_ARRAY_ELEMS(aacenc_profiles); i++)
|
|
if (avctx->profile == aacenc_profiles[i])
|
|
break;
|
|
ERROR_IF(i == FF_ARRAY_ELEMS(aacenc_profiles), "Profile not supported!\n");
|
|
if (avctx->profile == AV_PROFILE_MPEG2_AAC_LOW) {
|
|
avctx->profile = AV_PROFILE_AAC_LOW;
|
|
WARN_IF(s->options.pns,
|
|
"PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n");
|
|
s->options.pns = 0;
|
|
}
|
|
s->profile = avctx->profile;
|
|
|
|
/* Coder limitations */
|
|
s->coder = &ff_aac_coders[s->options.coder];
|
|
|
|
/* M/S introduces horrible artifacts with multichannel files, this is temporary */
|
|
if (s->channels > 3)
|
|
s->options.mid_side = 0;
|
|
|
|
/* Coding bandwidth, fixed at init time */
|
|
if (avctx->cutoff > 0) {
|
|
s->bandwidth = avctx->cutoff;
|
|
} else {
|
|
int frame_br = (avctx->flags & AV_CODEC_FLAG_QSCALE) ?
|
|
(avctx->bit_rate / 2.0f * (s->lambda / 120.f) * 1.5f) :
|
|
(avctx->bit_rate / avctx->ch_layout.nb_channels);
|
|
|
|
/* For NMR, the rate to bandwidth conversion was tuned to maximize metrics
|
|
* over a variable cutoff x bitrate combo */
|
|
if (s->options.coder == AAC_CODER_NMR && frame_br >= 32000) {
|
|
static const int rates[] = { 32000, 48000, 64000, 96000, 192000 };
|
|
static const int bws[] = { 14000, 15000, 16000, 18000, 20000 };
|
|
int bw_i = 0;
|
|
for (; bw_i < FF_ARRAY_ELEMS(rates) - 2 && frame_br > rates[bw_i + 1]; bw_i++);
|
|
s->bandwidth = bws[bw_i] + (int)((int64_t)(bws[bw_i + 1] - bws[bw_i]) *
|
|
(frame_br - rates[bw_i]) / (rates[bw_i + 1] - rates[bw_i]));
|
|
s->bandwidth = FFMIN3(s->bandwidth, 22000, avctx->sample_rate / 2);
|
|
} else {
|
|
if (s->options.pns || s->options.intensity_stereo)
|
|
frame_br *= 1.15f;
|
|
s->bandwidth = FFMAX(3000, AAC_CUTOFF_FROM_BITRATE(frame_br, 1,
|
|
avctx->sample_rate));
|
|
}
|
|
|
|
s->bandwidth = FFMIN(FFMAX(s->bandwidth, 8000), avctx->sample_rate / 2);
|
|
}
|
|
|
|
// Initialize static tables
|
|
ff_aac_float_common_init();
|
|
|
|
if ((ret = dsp_init(avctx, s)) < 0)
|
|
return ret;
|
|
|
|
if ((ret = alloc_buffers(avctx, s)) < 0)
|
|
return ret;
|
|
|
|
if ((ret = put_audio_specific_config(avctx, chcfg)))
|
|
return ret;
|
|
|
|
sizes[0] = ff_aac_swb_size_1024[s->samplerate_index];
|
|
sizes[1] = ff_aac_swb_size_128[s->samplerate_index];
|
|
lengths[0] = ff_aac_num_swb_1024[s->samplerate_index];
|
|
lengths[1] = ff_aac_num_swb_128[s->samplerate_index];
|
|
for (i = 0; i < s->chan_map[0]; i++)
|
|
grouping[i] = s->chan_map[i + 1] == TYPE_CPE;
|
|
if ((ret = ff_psy_init(&s->psy, avctx, 2, sizes, lengths,
|
|
s->chan_map[0], grouping, s->bandwidth)) < 0)
|
|
return ret;
|
|
ff_lpc_init(&s->lpc, 2*avctx->frame_size, TNS_MAX_ORDER, FF_LPC_TYPE_LEVINSON);
|
|
s->random_state = 0x1f2e3d4c;
|
|
|
|
ff_aacenc_dsp_init(&s->aacdsp);
|
|
|
|
ff_af_queue_init(avctx, &s->afq);
|
|
|
|
return 0;
|
|
}
|
|
|
|
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
|
|
static const AVOption aacenc_options[] = {
|
|
{"aac_coder", "Coding algorithm", offsetof(AACEncContext, options.coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_NMR}, 0, AAC_CODER_NB-1, AACENC_FLAGS, .unit = "coder"},
|
|
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, .unit = "coder"},
|
|
{"fast", "Fast search", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, .unit = "coder"},
|
|
{"nmr", "Noise-to-mask ratio scalefactor trellis", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_NMR}, INT_MIN, INT_MAX, AACENC_FLAGS, .unit = "coder"},
|
|
{"aac_ms", "Force M/S stereo coding", offsetof(AACEncContext, options.mid_side), AV_OPT_TYPE_BOOL, {.i64 = -1}, -1, 1, AACENC_FLAGS},
|
|
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
|
|
{"aac_pns", "Perceptual noise substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
|
|
{"aac_tns", "Temporal noise shaping", offsetof(AACEncContext, options.tns), AV_OPT_TYPE_BOOL, {.i64 = 1}, -1, 1, AACENC_FLAGS},
|
|
{"aac_pce", "Forces the use of PCEs", offsetof(AACEncContext, options.pce), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS},
|
|
{"aac_nmr_speed", "NMR coder speed level: 0 = slowest/best, higher trades quality for speed", offsetof(AACEncContext, options.nmr_speed), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 4, AACENC_FLAGS},
|
|
FF_AAC_PROFILE_OPTS
|
|
{NULL}
|
|
};
|
|
|
|
static const AVClass aacenc_class = {
|
|
.class_name = "AAC encoder",
|
|
.item_name = av_default_item_name,
|
|
.option = aacenc_options,
|
|
.version = LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
static const FFCodecDefault aac_encode_defaults[] = {
|
|
{ "b", "0" },
|
|
{ NULL }
|
|
};
|
|
|
|
const FFCodec ff_aac_encoder = {
|
|
.p.name = "aac",
|
|
CODEC_LONG_NAME("AAC (Advanced Audio Coding)"),
|
|
.p.type = AVMEDIA_TYPE_AUDIO,
|
|
.p.id = AV_CODEC_ID_AAC,
|
|
.p.capabilities = AV_CODEC_CAP_DR1 | AV_CODEC_CAP_DELAY |
|
|
AV_CODEC_CAP_SMALL_LAST_FRAME,
|
|
.priv_data_size = sizeof(AACEncContext),
|
|
.init = aac_encode_init,
|
|
FF_CODEC_ENCODE_CB(aac_encode_frame),
|
|
.close = aac_encode_end,
|
|
.defaults = aac_encode_defaults,
|
|
CODEC_SAMPLERATES_ARRAY(ff_mpeg4audio_sample_rates),
|
|
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP,
|
|
CODEC_SAMPLEFMTS(AV_SAMPLE_FMT_FLTP),
|
|
.p.priv_class = &aacenc_class,
|
|
};
|