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365 lines
12 KiB
365 lines
12 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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* @file libavcodec/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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