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366 lines
12 KiB
C
366 lines
12 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 aacenc.c
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* AAC encoder
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*/
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/***********************************
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* TODOs:
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* psy model selection with some option
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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 "avcodec.h"
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#include "bitstream.h"
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#include "dsputil.h"
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#include "mpeg4audio.h"
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#include "aacpsy.h"
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#include "aac.h"
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#include "aactab.h"
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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 * const 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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};
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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 * const 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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};
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/** bits needed to code codebook run value for long windows */
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static const uint8_t run_value_bits_long[64] = {
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5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
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5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
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10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
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10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
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};
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/** bits needed to code codebook run value for short windows */
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static const uint8_t run_value_bits_short[16] = {
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3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
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};
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static const uint8_t* const run_value_bits[2] = {
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run_value_bits_long, run_value_bits_short
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};
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/** 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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* structure used in optimal codebook search
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*/
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typedef struct BandCodingPath {
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int prev_idx; ///< pointer to the previous path point
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int codebook; ///< codebook for coding band run
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int bits; ///< number of bit needed to code given number of bands
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} BandCodingPath;
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/**
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* AAC encoder context
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*/
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typedef struct {
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PutBitContext pb;
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MDCTContext mdct1024; ///< long (1024 samples) frame transform context
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MDCTContext mdct128; ///< short (128 samples) frame transform context
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DSPContext dsp;
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DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
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int16_t* samples; ///< saved preprocessed input
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int samplerate_index; ///< MPEG-4 samplerate index
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ChannelElement *cpe; ///< channel elements
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AACPsyContext psy; ///< psychoacoustic model context
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int last_frame;
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} AACEncContext;
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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*8);
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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, avctx->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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flush_put_bits(&pb);
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}
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static av_cold int aac_encode_init(AVCodecContext *avctx)
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{
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AACEncContext *s = avctx->priv_data;
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int i;
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avctx->frame_size = 1024;
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for(i = 0; i < 16; i++)
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if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
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break;
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if(i == 16){
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av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
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return -1;
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}
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if(avctx->channels > 6){
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av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
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return -1;
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}
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s->samplerate_index = i;
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dsputil_init(&s->dsp, avctx);
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ff_mdct_init(&s->mdct1024, 11, 0);
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ff_mdct_init(&s->mdct128, 8, 0);
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// window init
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ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
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ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
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ff_sine_window_init(ff_sine_1024, 1024);
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ff_sine_window_init(ff_sine_128, 128);
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s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
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s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
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if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
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aac_chan_configs[avctx->channels-1][0], 0,
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swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){
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av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
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return -1;
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}
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avctx->extradata = av_malloc(2);
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avctx->extradata_size = 2;
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put_audio_specific_config(avctx);
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return 0;
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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 i;
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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(i = 1; i < info->num_windows; i++)
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put_bits(&s->pb, 1, info->group_len[i]);
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}
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}
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/**
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* Calculate the number of bits needed to code all coefficient signs in current band.
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*/
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static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
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int group_len, int start, int size)
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{
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int bits = 0;
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int i, w;
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for(w = 0; w < group_len; w++){
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for(i = 0; i < size; i++){
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if(sce->icoefs[start + i])
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bits++;
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}
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start += 128;
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}
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return bits;
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}
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/**
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* Encode pulse data.
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*/
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static void encode_pulses(AACEncContext *s, Pulse *pulse)
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{
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int i;
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put_bits(&s->pb, 1, !!pulse->num_pulse);
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if(!pulse->num_pulse) return;
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put_bits(&s->pb, 2, pulse->num_pulse - 1);
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put_bits(&s->pb, 6, pulse->start);
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for(i = 0; i < pulse->num_pulse; i++){
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put_bits(&s->pb, 5, pulse->pos[i]);
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put_bits(&s->pb, 4, pulse->amp[i]);
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}
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}
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/**
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* Encode spectral coefficients processed by psychoacoustic model.
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*/
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static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
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{
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int start, i, w, w2, wg;
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w = 0;
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for(wg = 0; wg < sce->ics.num_window_groups; wg++){
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start = 0;
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for(i = 0; i < sce->ics.max_sfb; i++){
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if(sce->zeroes[w*16 + i]){
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start += sce->ics.swb_sizes[i];
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continue;
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}
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for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){
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encode_band_coeffs(s, sce, start + w2*128,
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sce->ics.swb_sizes[i],
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sce->band_type[w*16 + i]);
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}
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start += sce->ics.swb_sizes[i];
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}
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w += sce->ics.group_len[wg];
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}
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}
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/**
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* Write some auxiliary information about the created AAC file.
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*/
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static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
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{
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int i, namelen, padbits;
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namelen = strlen(name) + 2;
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put_bits(&s->pb, 3, TYPE_FIL);
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put_bits(&s->pb, 4, FFMIN(namelen, 15));
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if(namelen >= 15)
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put_bits(&s->pb, 8, namelen - 16);
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put_bits(&s->pb, 4, 0); //extension type - filler
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padbits = 8 - (put_bits_count(&s->pb) & 7);
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align_put_bits(&s->pb);
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for(i = 0; i < namelen - 2; i++)
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put_bits(&s->pb, 8, name[i]);
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put_bits(&s->pb, 12 - padbits, 0);
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}
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static av_cold int aac_encode_end(AVCodecContext *avctx)
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{
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AACEncContext *s = avctx->priv_data;
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ff_mdct_end(&s->mdct1024);
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ff_mdct_end(&s->mdct128);
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ff_aac_psy_end(&s->psy);
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av_freep(&s->samples);
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av_freep(&s->cpe);
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return 0;
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}
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AVCodec aac_encoder = {
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"aac",
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CODEC_TYPE_AUDIO,
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CODEC_ID_AAC,
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sizeof(AACEncContext),
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aac_encode_init,
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aac_encode_frame,
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aac_encode_end,
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.capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
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.sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
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.long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
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};
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