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1438 lines
54 KiB
1438 lines
54 KiB
/* |
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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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/** |
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* @file |
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* AAC encoder |
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*/ |
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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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|
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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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|
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#include "psymodel.h" |
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|
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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 = { { 1, 0 }, }, |
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.index = { { 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_3POINT1, |
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.num_ele = { 2, 0, 0, 1 }, |
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.pairing = { { 1, 0 }, }, |
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.index = { { 0, 0 }, { 0 }, { 0 }, { 0 }, }, |
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.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_LFE }, |
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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_4POINT0, |
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.num_ele = { 2, 0, 1, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 0 }, }, |
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.index = { { 0, 0 }, { 0 }, { 1 } }, |
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.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_SCE }, |
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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_4POINT1, |
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.num_ele = { 2, 1, 1, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 0 }, }, |
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.index = { { 0, 0 }, { 1 }, { 2 }, { 0 } }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4 }, |
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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, 1, 0, 0 }, |
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.pairing = { { 1 }, { 1 }, }, |
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.index = { { 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, 1, 0, 0 }, |
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.pairing = { { 1, 0 }, { 1 }, }, |
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.index = { { 0, 0 }, { 1 } }, |
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.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 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, 1, 1, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1 } }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5 }, |
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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 = { { 1, 0 }, { 0 }, { 1 } }, |
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.index = { { 0, 0 }, { 0 }, { 1 } }, |
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.config_map = { 3, TYPE_CPE, TYPE_SCE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 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, 1, 1, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1 } }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5 }, |
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}, |
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{ |
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.layout = AV_CHANNEL_LAYOUT_6POINT0, |
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.num_ele = { 2, 1, 1, 0 }, |
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.pairing = { { 1, 0 }, { 1 }, { 0 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1 } }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5 }, |
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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, 1, 0, 0 }, |
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.pairing = { { 1, 1 }, { 1 } }, |
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.index = { { 1, 0 }, { 2 }, }, |
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.config_map = { 3, TYPE_CPE, TYPE_CPE, TYPE_CPE, }, |
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.reorder_map = { 0, 1, 2, 3, 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 = { { 1, 0 },{ 0 },{ 1, 0 }, }, |
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.index = { { 0, 0 },{ 0 },{ 1, 1 } }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, }, |
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.reorder_map = { 0, 1, 2, 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, 1, 2, 0 }, |
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.pairing = { { 1, 0 },{ 0 },{ 1, 0 }, }, |
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.index = { { 0, 0 },{ 1 },{ 1, 2 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6 }, |
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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, 1, 2, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 1, 0 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1, 2 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6 }, |
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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, 1, 2, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 1, 0 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1, 2 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6 }, |
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}, |
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{ |
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.layout = AV_CHANNEL_LAYOUT_7POINT0, |
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.num_ele = { 2, 1, 1, 0 }, |
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.pairing = { { 1, 0 }, { 1 }, { 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 2 }, }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 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 = { 2, 1, 1, 0 }, |
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.pairing = { { 1, 0 }, { 1 }, { 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 2 }, }, |
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.config_map = { 4, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 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, 1, 2, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 1, 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 }, |
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}, |
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{ |
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.layout = AV_CHANNEL_LAYOUT_7POINT1_WIDE, |
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.num_ele = { 2, 1, 2, 0 }, |
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.pairing = { { 1, 0 }, { 0 },{ 1, 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 }, |
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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 = { 2, 1, 2, 0 }, |
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.pairing = { { 1, 0 }, { 0 }, { 1, 1 }, }, |
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.index = { { 0, 0 }, { 1 }, { 1, 2 }, { 0 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_SCE, TYPE_CPE, TYPE_CPE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 }, |
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}, |
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{ |
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.layout = AV_CHANNEL_LAYOUT_OCTAGONAL, |
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.num_ele = { 2, 1, 2, 0 }, |
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.pairing = { { 1, 0 }, { 1 }, { 1, 0 }, }, |
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.index = { { 0, 0 }, { 1 }, { 2, 1 } }, |
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.config_map = { 5, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7 }, |
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}, |
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{ /* Meant for order 2/mixed ambisonics */ |
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.layout = { .order = AV_CHANNEL_ORDER_NATIVE, .nb_channels = 9, |
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.u.mask = AV_CH_LAYOUT_OCTAGONAL | AV_CH_TOP_CENTER }, |
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.num_ele = { 2, 2, 2, 0 }, |
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.pairing = { { 1, 0 }, { 1, 0 }, { 1, 0 }, }, |
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.index = { { 0, 0 }, { 1, 1 }, { 2, 2 } }, |
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.config_map = { 6, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8 }, |
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}, |
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{ /* Meant for order 2/mixed ambisonics */ |
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.layout = { .order = AV_CHANNEL_ORDER_NATIVE, .nb_channels = 10, |
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.u.mask = AV_CH_LAYOUT_6POINT0_FRONT | AV_CH_BACK_CENTER | |
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AV_CH_BACK_LEFT | AV_CH_BACK_RIGHT | AV_CH_TOP_CENTER }, |
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.num_ele = { 2, 2, 2, 0 }, |
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.pairing = { { 1, 1 }, { 1, 0 }, { 1, 0 }, }, |
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.index = { { 0, 1 }, { 2, 0 }, { 3, 1 } }, |
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.config_map = { 6, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 }, |
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}, |
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{ |
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.layout = AV_CHANNEL_LAYOUT_HEXADECAGONAL, |
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.num_ele = { 4, 2, 4, 0 }, |
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.pairing = { { 1, 0, 1, 0 }, { 1, 1 }, { 1, 0, 1, 0 }, }, |
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.index = { { 0, 0, 1, 1 }, { 2, 3 }, { 4, 2, 5, 3 } }, |
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.config_map = { 10, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_CPE, TYPE_SCE, TYPE_CPE, TYPE_SCE }, |
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.reorder_map = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15 }, |
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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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|
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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) |
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{ |
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PutBitContext pb; |
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AACEncContext *s = avctx->priv_data; |
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int channels = (!s->needs_pce)*(s->channels - (s->channels == 8 ? 1 : 0)); |
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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, channels); |
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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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|
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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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|
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return 0; |
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} |
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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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|
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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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|
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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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|
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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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|
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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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|
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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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|
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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; |
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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, !!info->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]; |
|
} |
|
} |
|
} |
|
} |
|
|
|
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); |
|
if (s->coder->encode_main_pred) |
|
s->coder->encode_main_pred(s, sce); |
|
if (s->coder->encode_ltp_info) |
|
s->coder->encode_ltp_info(s, sce, 0); |
|
} |
|
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 (s->psypp) |
|
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels); |
|
|
|
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); |
|
|
|
if (s->options.ltp && s->coder->update_ltp) { |
|
s->coder->update_ltp(s, sce); |
|
apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, &sce->ltp_state[0]); |
|
s->mdct1024_fn(s->mdct1024, sce->lcoeffs, sce->ret_buf, sizeof(float)); |
|
} |
|
|
|
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; |
|
sce->ics.predictor_present = 0; |
|
sce->ics.ltp.present = 0; |
|
memset(sce->ics.ltp.used, 0, sizeof(sce->ics.ltp.used)); |
|
memset(sce->ics.prediction_used, 0, sizeof(sce->ics.prediction_used)); |
|
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; |
|
for (ch = 0; ch < chans; ch++) { |
|
s->cur_channel = start_ch + ch; |
|
if (s->options.pns && s->coder->mark_pns) |
|
s->coder->mark_pns(s, avctx, &cpe->ch[ch]); |
|
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda); |
|
} |
|
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; |
|
} |
|
} |
|
} |
|
for (ch = 0; ch < chans; ch++) { /* TNS and PNS */ |
|
sce = &cpe->ch[ch]; |
|
s->cur_channel = start_ch + ch; |
|
if (s->options.tns && s->coder->search_for_tns) |
|
s->coder->search_for_tns(s, sce); |
|
if (s->options.tns && s->coder->apply_tns_filt) |
|
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->coder->search_for_is) |
|
s->coder->search_for_is(s, avctx, cpe); |
|
if (cpe->is_mode) is_mode = 1; |
|
apply_intensity_stereo(cpe); |
|
} |
|
if (s->options.pred) { /* Prediction */ |
|
for (ch = 0; ch < chans; ch++) { |
|
sce = &cpe->ch[ch]; |
|
s->cur_channel = start_ch + ch; |
|
if (s->options.pred && s->coder->search_for_pred) |
|
s->coder->search_for_pred(s, sce); |
|
if (cpe->ch[ch].ics.predictor_present) pred_mode = 1; |
|
} |
|
if (s->coder->adjust_common_pred) |
|
s->coder->adjust_common_pred(s, cpe); |
|
for (ch = 0; ch < chans; ch++) { |
|
sce = &cpe->ch[ch]; |
|
s->cur_channel = start_ch + ch; |
|
if (s->options.pred && s->coder->apply_main_pred) |
|
s->coder->apply_main_pred(s, sce); |
|
} |
|
s->cur_channel = start_ch; |
|
} |
|
if (s->options.mid_side) { /* 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 (s->options.ltp) { /* LTP */ |
|
for (ch = 0; ch < chans; ch++) { |
|
sce = &cpe->ch[ch]; |
|
s->cur_channel = start_ch + ch; |
|
if (s->coder->search_for_ltp) |
|
s->coder->search_for_ltp(s, sce, cpe->common_window); |
|
if (sce->ics.ltp.present) pred_mode = 1; |
|
} |
|
s->cur_channel = start_ch; |
|
if (s->coder->adjust_common_ltp) |
|
s->coder->adjust_common_ltp(s, cpe); |
|
} |
|
if (chans == 2) { |
|
put_bits(&s->pb, 1, cpe->common_window); |
|
if (cpe->common_window) { |
|
put_ics_info(s, &cpe->ch[0].ics); |
|
if (s->coder->encode_main_pred) |
|
s->coder->encode_main_pred(s, &cpe->ch[0]); |
|
if (s->coder->encode_ltp_info) |
|
s->coder->encode_ltp_info(s, &cpe->ch[0], 1); |
|
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; |
|
} |
|
|
|
/* rate control stuff |
|
* allow between the nominal bitrate, and what psy's bit reservoir says to target |
|
* but drift towards the nominal bitrate always |
|
*/ |
|
frame_bits = put_bits_count(&s->pb); |
|
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); |
|
|
|
if (s->options.ltp && s->coder->ltp_insert_new_frame) |
|
s->coder->ltp_insert_new_frame(s); |
|
|
|
put_bits(&s->pb, 3, TYPE_END); |
|
flush_put_bits(&s->pb); |
|
|
|
s->last_frame_pb_count = put_bits_count(&s->pb); |
|
avpkt->size = put_bytes_output(&s->pb); |
|
|
|
s->lambda_sum += s->lambda; |
|
s->lambda_count++; |
|
|
|
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts, |
|
&avpkt->duration); |
|
|
|
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\n", s->lambda_count ? s->lambda_sum / s->lambda_count : NAN); |
|
|
|
av_tx_uninit(&s->mdct1024); |
|
av_tx_uninit(&s->mdct128); |
|
ff_psy_end(&s->psy); |
|
ff_lpc_end(&s->lpc); |
|
if (s->psypp) |
|
ff_psy_preprocess_end(s->psypp); |
|
av_freep(&s->buffer.samples); |
|
av_freep(&s->cpe); |
|
av_freep(&s->fdsp); |
|
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; |
|
|
|
return 0; |
|
} |
|
|
|
static av_cold int aac_encode_init(AVCodecContext *avctx) |
|
{ |
|
AACEncContext *s = avctx->priv_data; |
|
int i, ret = 0; |
|
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 (i = 0; i < FF_ARRAY_ELEMS(aac_normal_chan_layouts); i++) { |
|
if (!av_channel_layout_compare(&avctx->ch_layout, &aac_normal_chan_layouts[i])) { |
|
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; |
|
} else { |
|
s->reorder_map = aac_chan_maps[s->channels - 1]; |
|
s->chan_map = aac_chan_configs[s->channels - 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 (i = 0; i < 16; i++) |
|
if (avctx->sample_rate == ff_mpeg4audio_sample_rates[i]) |
|
break; |
|
s->samplerate_index = i; |
|
ERROR_IF(s->samplerate_index == 16 || |
|
s->samplerate_index >= ff_aac_swb_size_1024_len || |
|
s->samplerate_index >= ff_aac_swb_size_128_len, |
|
"Unsupported sample rate %d\n", avctx->sample_rate); |
|
|
|
/* 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; |
|
if (avctx->profile == AV_PROFILE_MPEG2_AAC_LOW) { |
|
avctx->profile = AV_PROFILE_AAC_LOW; |
|
ERROR_IF(s->options.pred, |
|
"Main prediction unavailable in the \"mpeg2_aac_low\" profile\n"); |
|
ERROR_IF(s->options.ltp, |
|
"LTP prediction unavailable in the \"mpeg2_aac_low\" profile\n"); |
|
WARN_IF(s->options.pns, |
|
"PNS unavailable in the \"mpeg2_aac_low\" profile, turning off\n"); |
|
s->options.pns = 0; |
|
} else if (avctx->profile == AV_PROFILE_AAC_LTP) { |
|
s->options.ltp = 1; |
|
ERROR_IF(s->options.pred, |
|
"Main prediction unavailable in the \"aac_ltp\" profile\n"); |
|
} else if (avctx->profile == AV_PROFILE_AAC_MAIN) { |
|
s->options.pred = 1; |
|
ERROR_IF(s->options.ltp, |
|
"LTP prediction unavailable in the \"aac_main\" profile\n"); |
|
} else if (s->options.ltp) { |
|
avctx->profile = AV_PROFILE_AAC_LTP; |
|
WARN_IF(1, |
|
"Chainging profile to \"aac_ltp\"\n"); |
|
ERROR_IF(s->options.pred, |
|
"Main prediction unavailable in the \"aac_ltp\" profile\n"); |
|
} else if (s->options.pred) { |
|
avctx->profile = AV_PROFILE_AAC_MAIN; |
|
WARN_IF(1, |
|
"Chainging profile to \"aac_main\"\n"); |
|
ERROR_IF(s->options.ltp, |
|
"LTP prediction unavailable in the \"aac_main\" profile\n"); |
|
} |
|
s->profile = avctx->profile; |
|
|
|
/* Coder limitations */ |
|
s->coder = &ff_aac_coders[s->options.coder]; |
|
if (s->options.coder == AAC_CODER_ANMR) { |
|
ERROR_IF(avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, |
|
"The ANMR coder is considered experimental, add -strict -2 to enable!\n"); |
|
s->options.intensity_stereo = 0; |
|
s->options.pns = 0; |
|
} |
|
ERROR_IF(s->options.ltp && avctx->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL, |
|
"The LPT profile requires experimental compliance, add -strict -2 to enable!\n"); |
|
|
|
/* M/S introduces horrible artifacts with multichannel files, this is temporary */ |
|
if (s->channels > 3) |
|
s->options.mid_side = 0; |
|
|
|
// 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))) |
|
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)) < 0) |
|
return ret; |
|
s->psypp = ff_psy_preprocess_init(avctx); |
|
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_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, .unit = "coder"}, |
|
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, 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"}, |
|
{"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_ltp", "Long term prediction", offsetof(AACEncContext, options.ltp), AV_OPT_TYPE_BOOL, {.i64 = 0}, -1, 1, AACENC_FLAGS}, |
|
{"aac_pred", "AAC-Main prediction", offsetof(AACEncContext, options.pred), AV_OPT_TYPE_BOOL, {.i64 = 0}, -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}, |
|
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, |
|
.p.supported_samplerates = ff_mpeg4audio_sample_rates, |
|
.caps_internal = FF_CODEC_CAP_INIT_CLEANUP, |
|
.p.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP, |
|
AV_SAMPLE_FMT_NONE }, |
|
.p.priv_class = &aacenc_class, |
|
};
|
|
|