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https://github.com/FFmpeg/FFmpeg.git
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e8576dc8df
This commit implements intensity stereo coding support to the native aac encoder. This is a way to increase the efficiency of the encoder by zeroing the right channel's spectral coefficients (in a channel pair) and rederiving them in the decoder using information from the scalefactor indices of special band types. This commit confomrs to the official ISO 13818-7 specifications, although due to their ambiguity certain deviations have been taken to ensure maximum sound quality. This commit has been extensively tested and has shown to not result in audiable audio artifacts unless in extreme cases. This commit also adds an option, aac_is, which has the value of 0 by default. Intensity Stereo is part of the scalable aac profile and is thus non-default. The way IS coding works is that it rederives the right channel's spectral coefficients from the left channel via the scalefactor index values left in the right channel. Since an entire band's spectral coefficients do not need to be coded, the encoder's efficiency jumps up and it unzeroes some high frequency values which it previously did not have enough bits to encode. That way less information is lost than the information lost by rederiving the spectral coefficients with some error. This is why the filesize of files encoded with IS do not decrease significantly. Users wishing that IS coding should reduce filesize are expected to reduce their encoding bitrates appropriately. This is V2 of the commit. The old version did not mark ms_mask as 0 since M/S and IS coding are incompactible, which resulted in distortions with M/S coding enabled. This version also improves phase detection by measuring it for every spectral coefficient in the band and using a simple majority rule to determine whether the coefficients are in or out of phase. Also, the energy values per spectral coefficient were changed as to reflect the official specifications. Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
920 lines
33 KiB
C
920 lines
33 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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* add temporal noise shaping
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***********************************/
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#include "libavutil/float_dsp.h"
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "internal.h"
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#include "mpeg4audio.h"
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#include "kbdwin.h"
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#include "sinewin.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 "psymodel.h"
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#define AAC_MAX_CHANNELS 6
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#define ERROR_IF(cond, ...) \
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if (cond) { \
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av_log(avctx, AV_LOG_ERROR, __VA_ARGS__); \
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return AVERROR(EINVAL); \
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}
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#define WARN_IF(cond, ...) \
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if (cond) { \
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av_log(avctx, AV_LOG_WARNING, __VA_ARGS__); \
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}
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float ff_aac_pow34sf_tab[428];
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static const uint8_t swb_size_1024_96[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
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64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_64[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
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12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
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40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
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};
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static const uint8_t swb_size_1024_48[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
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96
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};
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static const uint8_t swb_size_1024_32[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
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};
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static const uint8_t swb_size_1024_24[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
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};
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static const uint8_t swb_size_1024_16[] = {
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8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
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32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
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};
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static const uint8_t swb_size_1024_8[] = {
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12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
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16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
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32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
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};
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static const uint8_t *swb_size_1024[] = {
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swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
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swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
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swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
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swb_size_1024_16, swb_size_1024_16, swb_size_1024_8,
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swb_size_1024_8
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};
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static const uint8_t swb_size_128_96[] = {
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4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
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};
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static const uint8_t swb_size_128_48[] = {
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4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
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};
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static const uint8_t swb_size_128_24[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
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};
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static const uint8_t swb_size_128_16[] = {
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4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
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};
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static const uint8_t swb_size_128_8[] = {
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4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
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};
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static const uint8_t *swb_size_128[] = {
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/* the last entry on the following row is swb_size_128_64 but is a
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duplicate of swb_size_128_96 */
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swb_size_128_96, swb_size_128_96, swb_size_128_96,
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swb_size_128_48, swb_size_128_48, swb_size_128_48,
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swb_size_128_24, swb_size_128_24, swb_size_128_16,
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swb_size_128_16, swb_size_128_16, swb_size_128_8,
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swb_size_128_8
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};
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/** default channel configurations */
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static const uint8_t aac_chan_configs[6][5] = {
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{1, TYPE_SCE}, // 1 channel - single channel element
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{1, TYPE_CPE}, // 2 channels - channel pair
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{2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
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{3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
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{3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
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{4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
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};
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/**
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* Table to remap channels from libavcodec's default order to AAC order.
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*/
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static const uint8_t aac_chan_maps[AAC_MAX_CHANNELS][AAC_MAX_CHANNELS] = {
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{ 0 },
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{ 0, 1 },
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{ 2, 0, 1 },
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{ 2, 0, 1, 3 },
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{ 2, 0, 1, 3, 4 },
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{ 2, 0, 1, 4, 5, 3 },
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};
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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 void 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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init_put_bits(&pb, avctx->extradata, avctx->extradata_size);
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put_bits(&pb, 5, 2); //object type - AAC-LC
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put_bits(&pb, 4, s->samplerate_index); //sample rate index
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put_bits(&pb, 4, s->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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//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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}
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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;
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const float *swindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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float *out = sce->ret_buf;
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memset(out, 0, sizeof(out[0]) * 448);
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fdsp->vector_fmul(out + 448, audio + 448, swindow, 128);
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memcpy(out + 576, audio + 576, sizeof(out[0]) * 448);
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fdsp->vector_fmul_reverse(out + 1024, audio + 1024, lwindow, 1024);
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}
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WINDOW_FUNC(eight_short)
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{
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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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const float *pwindow = sce->ics.use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
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const float *in = audio + 448;
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float *out = sce->ret_buf;
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int w;
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for (w = 0; w < 8; w++) {
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fdsp->vector_fmul (out, in, w ? pwindow : swindow, 128);
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out += 128;
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in += 128;
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fdsp->vector_fmul_reverse(out, in, swindow, 128);
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out += 128;
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}
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}
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static void (*const apply_window[4])(AVFloatDSPContext *fdsp,
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SingleChannelElement *sce,
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const float *audio) = {
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[ONLY_LONG_SEQUENCE] = apply_only_long_window,
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[LONG_START_SEQUENCE] = apply_long_start_window,
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[EIGHT_SHORT_SEQUENCE] = apply_eight_short_window,
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[LONG_STOP_SEQUENCE] = apply_long_stop_window
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};
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static void apply_window_and_mdct(AACEncContext *s, SingleChannelElement *sce,
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float *audio)
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{
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int i;
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float *output = sce->ret_buf;
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apply_window[sce->ics.window_sequence[0]](s->fdsp, sce, audio);
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if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE)
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s->mdct1024.mdct_calc(&s->mdct1024, sce->coeffs, output);
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else
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for (i = 0; i < 1024; i += 128)
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s->mdct128.mdct_calc(&s->mdct128, sce->coeffs + i, output + i*2);
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memcpy(audio, audio + 1024, sizeof(audio[0]) * 1024);
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memcpy(sce->pcoeffs, sce->coeffs, sizeof(sce->pcoeffs));
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}
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/**
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* Encode ics_info element.
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* @see Table 4.6 (syntax of ics_info)
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*/
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static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
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{
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int w;
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put_bits(&s->pb, 1, 0); // ics_reserved bit
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put_bits(&s->pb, 2, info->window_sequence[0]);
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put_bits(&s->pb, 1, info->use_kb_window[0]);
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if (info->window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
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put_bits(&s->pb, 6, info->max_sfb);
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put_bits(&s->pb, 1, 0); // no prediction
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} else {
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put_bits(&s->pb, 4, info->max_sfb);
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for (w = 1; w < 8; w++)
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put_bits(&s->pb, 1, !info->group_len[w]);
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}
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}
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/**
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* Encode MS data.
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* @see 4.6.8.1 "Joint Coding - M/S Stereo"
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*/
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static void encode_ms_info(PutBitContext *pb, ChannelElement *cpe)
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{
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int i, w;
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put_bits(pb, 2, cpe->ms_mode);
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if (cpe->ms_mode == 1)
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for (w = 0; w < cpe->ch[0].ics.num_windows; w += cpe->ch[0].ics.group_len[w])
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for (i = 0; i < cpe->ch[0].ics.max_sfb; i++)
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put_bits(pb, 1, cpe->ms_mask[w*16 + i]);
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}
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/**
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* Produce integer coefficients from scalefactors provided by the model.
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*/
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static void adjust_frame_information(ChannelElement *cpe, int chans)
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{
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int i, w, w2, g, ch;
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int maxsfb, cmaxsfb;
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IndividualChannelStream *ics;
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if (cpe->common_window) {
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ics = &cpe->ch[0].ics;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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int start = (w+w2) * 128;
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for (g = 0; g < ics->num_swb; g++) {
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//apply Intensity stereo coeffs transformation
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if (cpe->is_mask[w*16 + g]) {
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int p = -1 + 2 * (cpe->ch[1].band_type[w*16+g] - 14);
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float scale = cpe->ch[0].is_ener[w*16+g];
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + p*cpe->ch[1].pcoeffs[start+i]) * scale;
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cpe->ch[1].coeffs[start+i] = 0.0f;
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}
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} else if (cpe->ms_mask[w*16 + g] &&
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cpe->ch[0].band_type[w*16 + g] < NOISE_BT &&
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cpe->ch[1].band_type[w*16 + g] < NOISE_BT) {
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for (i = 0; i < ics->swb_sizes[g]; i++) {
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cpe->ch[0].coeffs[start+i] = (cpe->ch[0].pcoeffs[start+i] + cpe->ch[1].pcoeffs[start+i]) * 0.5f;
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cpe->ch[1].coeffs[start+i] = cpe->ch[0].coeffs[start+i] - cpe->ch[1].pcoeffs[start+i];
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}
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}
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start += ics->swb_sizes[g];
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}
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}
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}
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}
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for (ch = 0; ch < chans; ch++) {
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IndividualChannelStream *ics = &cpe->ch[ch].ics;
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maxsfb = 0;
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cpe->ch[ch].pulse.num_pulse = 0;
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (w2 = 0; w2 < ics->group_len[w]; w2++) {
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for (cmaxsfb = ics->num_swb; cmaxsfb > 0 && cpe->ch[ch].zeroes[w*16+cmaxsfb-1]; cmaxsfb--)
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;
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maxsfb = FFMAX(maxsfb, cmaxsfb);
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}
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}
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ics->max_sfb = maxsfb;
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//adjust zero bands for window groups
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for (w = 0; w < ics->num_windows; w += ics->group_len[w]) {
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for (g = 0; g < ics->max_sfb; g++) {
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i = 1;
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for (w2 = w; w2 < w + ics->group_len[w]; w2++) {
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if (!cpe->ch[ch].zeroes[w2*16 + g]) {
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i = 0;
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break;
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}
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}
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cpe->ch[ch].zeroes[w*16 + g] = i;
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}
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}
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}
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if (chans > 1 && cpe->common_window) {
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IndividualChannelStream *ics0 = &cpe->ch[0].ics;
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IndividualChannelStream *ics1 = &cpe->ch[1].ics;
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int msc = 0;
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ics0->max_sfb = FFMAX(ics0->max_sfb, ics1->max_sfb);
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ics1->max_sfb = ics0->max_sfb;
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for (w = 0; w < ics0->num_windows*16; w += 16)
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for (i = 0; i < ics0->max_sfb; i++)
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if (cpe->ms_mask[w+i])
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msc++;
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if (msc == 0 || ics0->max_sfb == 0)
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cpe->ms_mode = 0;
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else
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cpe->ms_mode = msc < ics0->max_sfb * ics0->num_windows ? 1 : 2;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* Encode scalefactor band coding type.
|
|
*/
|
|
static void encode_band_info(AACEncContext *s, SingleChannelElement *sce)
|
|
{
|
|
int w;
|
|
|
|
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,
|
|
sce->ics.swb_sizes[i],
|
|
sce->sf_idx[w*16 + i],
|
|
sce->band_type[w*16 + i],
|
|
s->lambda);
|
|
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, 0); //tns
|
|
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;
|
|
avpriv_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 = aac_chan_maps[s->channels - 1];
|
|
|
|
/* 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;
|
|
int i, ch, w, g, chans, tag, start_ch, ret, ms_mode = 0, is_mode = 0;
|
|
int chan_el_counter[4];
|
|
FFPsyWindowInfo windows[AAC_MAX_CHANNELS];
|
|
|
|
if (s->last_frame == 2)
|
|
return 0;
|
|
|
|
/* add current frame to queue */
|
|
if (frame) {
|
|
if ((ret = ff_af_queue_add(&s->afq, frame)) < 0)
|
|
return ret;
|
|
}
|
|
|
|
copy_input_samples(s, frame);
|
|
if (s->psypp)
|
|
ff_psy_preprocess(s->psypp, s->planar_samples, s->channels);
|
|
|
|
if (!avctx->frame_number)
|
|
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++) {
|
|
IndividualChannelStream *ics = &cpe->ch[ch].ics;
|
|
int cur_channel = start_ch + ch;
|
|
overlap = &samples[cur_channel][0];
|
|
samples2 = overlap + 1024;
|
|
la = samples2 + (448+64);
|
|
if (!frame)
|
|
la = NULL;
|
|
if (tag == TYPE_LFE) {
|
|
wi[ch].window_type[0] = ONLY_LONG_SEQUENCE;
|
|
wi[ch].window_shape = 0;
|
|
wi[ch].num_windows = 1;
|
|
wi[ch].grouping[0] = 1;
|
|
|
|
/* 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, 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];
|
|
for (w = 0; w < ics->num_windows; w++)
|
|
ics->group_len[w] = wi[ch].grouping[w];
|
|
|
|
apply_window_and_mdct(s, &cpe->ch[ch], overlap);
|
|
if (isnan(cpe->ch->coeffs[0])) {
|
|
av_log(avctx, AV_LOG_ERROR, "Input contains NaN\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
}
|
|
start_ch += chans;
|
|
}
|
|
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192 * s->channels)) < 0)
|
|
return ret;
|
|
do {
|
|
int frame_bits;
|
|
|
|
init_put_bits(&s->pb, avpkt->data, avpkt->size);
|
|
|
|
if ((avctx->frame_number & 0xFF)==1 && !(avctx->flags & CODEC_FLAG_BITEXACT))
|
|
put_bitstream_info(s, LIBAVCODEC_IDENT);
|
|
start_ch = 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];
|
|
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++)
|
|
coeffs[ch] = cpe->ch[ch].coeffs;
|
|
s->psy.model->analyze(&s->psy, start_ch, coeffs, wi);
|
|
for (ch = 0; ch < chans; ch++) {
|
|
s->cur_channel = start_ch + ch;
|
|
s->coder->search_for_quantizers(avctx, s, &cpe->ch[ch], s->lambda);
|
|
}
|
|
cpe->common_window = 0;
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
if (s->options.pns && s->coder->search_for_pns) {
|
|
for (ch = 0; ch < chans; ch++) {
|
|
s->cur_channel = start_ch + ch;
|
|
s->coder->search_for_pns(s, avctx, &cpe->ch[ch], s->lambda);
|
|
}
|
|
}
|
|
s->cur_channel = start_ch;
|
|
if (s->options.stereo_mode && cpe->common_window) {
|
|
if (s->options.stereo_mode > 0) {
|
|
IndividualChannelStream *ics = &cpe->ch[0].ics;
|
|
for (w = 0; w < ics->num_windows; w += ics->group_len[w])
|
|
for (g = 0; g < ics->num_swb; g++)
|
|
cpe->ms_mask[w*16+g] = 1;
|
|
} else if (s->coder->search_for_ms) {
|
|
s->coder->search_for_ms(s, cpe, s->lambda);
|
|
}
|
|
}
|
|
if (chans > 1 && s->options.intensity_stereo && s->coder->search_for_is) {
|
|
s->coder->search_for_is(s, avctx, cpe, s->lambda);
|
|
if (cpe->is_mode) is_mode = 1;
|
|
}
|
|
if (s->coder->set_special_band_scalefactors)
|
|
for (ch = 0; ch < chans; ch++)
|
|
s->coder->set_special_band_scalefactors(s, &cpe->ch[ch]);
|
|
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;
|
|
}
|
|
|
|
frame_bits = put_bits_count(&s->pb);
|
|
if (frame_bits <= 6144 * s->channels - 3) {
|
|
s->psy.bitres.bits = frame_bits / s->channels;
|
|
break;
|
|
}
|
|
if (is_mode || ms_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));
|
|
}
|
|
}
|
|
|
|
s->lambda *= avctx->bit_rate * 1024.0f / avctx->sample_rate / frame_bits;
|
|
|
|
} while (1);
|
|
|
|
put_bits(&s->pb, 3, TYPE_END);
|
|
flush_put_bits(&s->pb);
|
|
avctx->frame_bits = put_bits_count(&s->pb);
|
|
|
|
// rate control stuff
|
|
if (!(avctx->flags & CODEC_FLAG_QSCALE)) {
|
|
float ratio = avctx->bit_rate * 1024.0f / avctx->sample_rate / avctx->frame_bits;
|
|
s->lambda *= ratio;
|
|
s->lambda = FFMIN(s->lambda, 65536.f);
|
|
}
|
|
|
|
if (!frame)
|
|
s->last_frame++;
|
|
|
|
ff_af_queue_remove(&s->afq, avctx->frame_size, &avpkt->pts,
|
|
&avpkt->duration);
|
|
|
|
avpkt->size = put_bits_count(&s->pb) >> 3;
|
|
*got_packet_ptr = 1;
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int aac_encode_end(AVCodecContext *avctx)
|
|
{
|
|
AACEncContext *s = avctx->priv_data;
|
|
|
|
ff_mdct_end(&s->mdct1024);
|
|
ff_mdct_end(&s->mdct128);
|
|
ff_psy_end(&s->psy);
|
|
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;
|
|
|
|
s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
|
|
if (!s->fdsp)
|
|
return AVERROR(ENOMEM);
|
|
|
|
// window init
|
|
ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
|
|
ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
|
|
ff_init_ff_sine_windows(10);
|
|
ff_init_ff_sine_windows(7);
|
|
|
|
if ((ret = ff_mdct_init(&s->mdct1024, 11, 0, 32768.0)) < 0)
|
|
return ret;
|
|
if ((ret = ff_mdct_init(&s->mdct128, 8, 0, 32768.0)) < 0)
|
|
return ret;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static av_cold int alloc_buffers(AVCodecContext *avctx, AACEncContext *s)
|
|
{
|
|
int ch;
|
|
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->buffer.samples, s->channels, 3 * 1024 * sizeof(s->buffer.samples[0]), alloc_fail);
|
|
FF_ALLOCZ_ARRAY_OR_GOTO(avctx, s->cpe, s->chan_map[0], sizeof(ChannelElement), alloc_fail);
|
|
FF_ALLOCZ_OR_GOTO(avctx, avctx->extradata, 5 + FF_INPUT_BUFFER_PADDING_SIZE, alloc_fail);
|
|
|
|
for(ch = 0; ch < s->channels; ch++)
|
|
s->planar_samples[ch] = s->buffer.samples + 3 * 1024 * ch;
|
|
|
|
return 0;
|
|
alloc_fail:
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
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];
|
|
|
|
avctx->frame_size = 1024;
|
|
|
|
for (i = 0; i < 16; i++)
|
|
if (avctx->sample_rate == avpriv_mpeg4audio_sample_rates[i])
|
|
break;
|
|
|
|
s->channels = avctx->channels;
|
|
|
|
ERROR_IF(i == 16
|
|
|| i >= (sizeof(swb_size_1024) / sizeof(*swb_size_1024))
|
|
|| i >= (sizeof(swb_size_128) / sizeof(*swb_size_128)),
|
|
"Unsupported sample rate %d\n", avctx->sample_rate);
|
|
ERROR_IF(s->channels > AAC_MAX_CHANNELS,
|
|
"Unsupported number of channels: %d\n", s->channels);
|
|
ERROR_IF(avctx->profile != FF_PROFILE_UNKNOWN && avctx->profile != FF_PROFILE_AAC_LOW,
|
|
"Unsupported profile %d\n", avctx->profile);
|
|
WARN_IF(1024.0 * avctx->bit_rate / avctx->sample_rate > 6144 * s->channels,
|
|
"Too many bits per frame requested, clamping to max\n");
|
|
|
|
avctx->bit_rate = (int)FFMIN(
|
|
6144 * s->channels / 1024.0 * avctx->sample_rate,
|
|
avctx->bit_rate);
|
|
|
|
s->samplerate_index = i;
|
|
|
|
s->chan_map = aac_chan_configs[s->channels-1];
|
|
|
|
if ((ret = dsp_init(avctx, s)) < 0)
|
|
goto fail;
|
|
|
|
if ((ret = alloc_buffers(avctx, s)) < 0)
|
|
goto fail;
|
|
|
|
avctx->extradata_size = 5;
|
|
put_audio_specific_config(avctx);
|
|
|
|
sizes[0] = swb_size_1024[i];
|
|
sizes[1] = swb_size_128[i];
|
|
lengths[0] = ff_aac_num_swb_1024[i];
|
|
lengths[1] = ff_aac_num_swb_128[i];
|
|
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)
|
|
goto fail;
|
|
s->psypp = ff_psy_preprocess_init(avctx);
|
|
s->coder = &ff_aac_coders[s->options.aac_coder];
|
|
|
|
if (HAVE_MIPSDSPR1)
|
|
ff_aac_coder_init_mips(s);
|
|
|
|
s->lambda = avctx->global_quality > 0 ? avctx->global_quality : 120;
|
|
|
|
ff_aac_tableinit();
|
|
|
|
for (i = 0; i < 428; i++)
|
|
ff_aac_pow34sf_tab[i] = sqrt(ff_aac_pow2sf_tab[i] * sqrt(ff_aac_pow2sf_tab[i]));
|
|
|
|
avctx->initial_padding = 1024;
|
|
ff_af_queue_init(avctx, &s->afq);
|
|
|
|
return 0;
|
|
fail:
|
|
aac_encode_end(avctx);
|
|
return ret;
|
|
}
|
|
|
|
#define AACENC_FLAGS AV_OPT_FLAG_ENCODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
|
|
static const AVOption aacenc_options[] = {
|
|
{"stereo_mode", "Stereo coding method", offsetof(AACEncContext, options.stereo_mode), AV_OPT_TYPE_INT, {.i64 = 0}, -1, 1, AACENC_FLAGS, "stereo_mode"},
|
|
{"auto", "Selected by the Encoder", 0, AV_OPT_TYPE_CONST, {.i64 = -1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
|
|
{"ms_off", "Disable Mid/Side coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
|
|
{"ms_force", "Force Mid/Side for the whole frame if possible", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "stereo_mode"},
|
|
{"aac_coder", "", offsetof(AACEncContext, options.aac_coder), AV_OPT_TYPE_INT, {.i64 = AAC_CODER_TWOLOOP}, 0, AAC_CODER_NB-1, AACENC_FLAGS, "aac_coder"},
|
|
{"faac", "FAAC-inspired method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAAC}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"anmr", "ANMR method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_ANMR}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"twoloop", "Two loop searching method", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_TWOLOOP}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"fast", "Constant quantizer", 0, AV_OPT_TYPE_CONST, {.i64 = AAC_CODER_FAST}, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_coder"},
|
|
{"aac_pns", "Perceptual Noise Substitution", offsetof(AACEncContext, options.pns), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "aac_pns"},
|
|
{"disable", "Disable PNS", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
|
|
{"enable", "Enable PNS (Proof of concept)", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, INT_MIN, INT_MAX, AACENC_FLAGS, "aac_pns"},
|
|
{"aac_is", "Intensity stereo coding", offsetof(AACEncContext, options.intensity_stereo), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AACENC_FLAGS, "intensity_stereo"},
|
|
{"disable", "Disable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 0}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
|
|
{"enable", "Enable intensity stereo coding", 0, AV_OPT_TYPE_CONST, {.i64 = 1}, INT_MIN, INT_MAX, AACENC_FLAGS, "intensity_stereo"},
|
|
{NULL}
|
|
};
|
|
|
|
static const AVClass aacenc_class = {
|
|
"AAC encoder",
|
|
av_default_item_name,
|
|
aacenc_options,
|
|
LIBAVUTIL_VERSION_INT,
|
|
};
|
|
|
|
/* duplicated from avpriv_mpeg4audio_sample_rates to avoid shared build
|
|
* failures */
|
|
static const int mpeg4audio_sample_rates[16] = {
|
|
96000, 88200, 64000, 48000, 44100, 32000,
|
|
24000, 22050, 16000, 12000, 11025, 8000, 7350
|
|
};
|
|
|
|
AVCodec ff_aac_encoder = {
|
|
.name = "aac",
|
|
.long_name = NULL_IF_CONFIG_SMALL("AAC (Advanced Audio Coding)"),
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.id = AV_CODEC_ID_AAC,
|
|
.priv_data_size = sizeof(AACEncContext),
|
|
.init = aac_encode_init,
|
|
.encode2 = aac_encode_frame,
|
|
.close = aac_encode_end,
|
|
.supported_samplerates = mpeg4audio_sample_rates,
|
|
.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY |
|
|
CODEC_CAP_EXPERIMENTAL,
|
|
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_FLTP,
|
|
AV_SAMPLE_FMT_NONE },
|
|
.priv_class = &aacenc_class,
|
|
};
|