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27d23ae074
Long Term Prediction allows for prediction of spectral coefficients via the previously decoded time-dependent samples. This feature works well with harmonic content 2 or more frames long, like speech, human or non-human, piano music or any constant tones at very low bitrates. It should be noted that the current coder is highly efficient and the rate control system is unable to encode files at extremely low bitrates (less than 14kbps seems to be impossible) so this extension isn't capable of optimum operation. Dramatic difference is observable with some types of audio and speech but for the most part the audiable differences are subtle. The spectrum looks better however so the encoder is able to harvest the additional bits that this feature provies, should the user choose to enable it. So it's best to enable this feature only if encoding at the absolutely lowest bitrate that the encoder is capable of.
139 lines
5.5 KiB
C
139 lines
5.5 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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#ifndef AVCODEC_AACENC_H
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#define AVCODEC_AACENC_H
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#include "libavutil/float_dsp.h"
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#include "avcodec.h"
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#include "put_bits.h"
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#include "aac.h"
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#include "audio_frame_queue.h"
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#include "psymodel.h"
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#include "lpc.h"
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typedef enum AACCoder {
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AAC_CODER_FAAC = 0,
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AAC_CODER_ANMR,
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AAC_CODER_TWOLOOP,
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AAC_CODER_FAST,
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AAC_CODER_NB,
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}AACCoder;
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typedef struct AACEncOptions {
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int coder;
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int pns;
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int tns;
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int ltp;
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int pred;
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int mid_side;
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int intensity_stereo;
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} AACEncOptions;
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struct AACEncContext;
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typedef struct AACCoefficientsEncoder {
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void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
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SingleChannelElement *sce, const float lambda);
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void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
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int win, int group_len, const float lambda);
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void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, float *out, int size,
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int scale_idx, int cb, const float lambda, int rtz);
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void (*encode_tns_info)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*encode_ltp_info)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
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void (*encode_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*adjust_common_pred)(struct AACEncContext *s, ChannelElement *cpe);
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void (*adjust_common_ltp)(struct AACEncContext *s, ChannelElement *cpe);
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void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*apply_tns_filt)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*update_ltp)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*ltp_insert_new_frame)(struct AACEncContext *s);
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void (*set_special_band_scalefactors)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*search_for_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
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void (*mark_pns)(struct AACEncContext *s, AVCodecContext *avctx, SingleChannelElement *sce);
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void (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*search_for_ltp)(struct AACEncContext *s, SingleChannelElement *sce, int common_window);
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void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe);
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void (*search_for_is)(struct AACEncContext *s, AVCodecContext *avctx, ChannelElement *cpe);
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void (*search_for_pred)(struct AACEncContext *s, SingleChannelElement *sce);
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} AACCoefficientsEncoder;
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extern AACCoefficientsEncoder ff_aac_coders[];
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typedef struct AACQuantizeBandCostCacheEntry {
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float rd;
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float energy;
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int bits; ///< -1 means uninitialized entry
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char cb;
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char rtz;
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char padding[2]; ///< Keeps the entry size a multiple of 32 bits
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} AACQuantizeBandCostCacheEntry;
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/**
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* AAC encoder context
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*/
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typedef struct AACEncContext {
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AVClass *av_class;
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AACEncOptions options; ///< encoding options
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PutBitContext pb;
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FFTContext mdct1024; ///< long (1024 samples) frame transform context
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FFTContext mdct128; ///< short (128 samples) frame transform context
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AVFloatDSPContext *fdsp;
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float *planar_samples[8]; ///< saved preprocessed input
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int profile; ///< copied from avctx
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LPCContext lpc; ///< used by TNS
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int samplerate_index; ///< MPEG-4 samplerate index
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int channels; ///< channel count
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const uint8_t *chan_map; ///< channel configuration map
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ChannelElement *cpe; ///< channel elements
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FFPsyContext psy;
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struct FFPsyPreprocessContext* psypp;
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AACCoefficientsEncoder *coder;
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int cur_channel; ///< current channel for coder context
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int last_frame;
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int random_state;
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float lambda;
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float lambda_sum; ///< sum(lambda), for Qvg reporting
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int lambda_count; ///< count(lambda), for Qvg reporting
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enum RawDataBlockType cur_type; ///< channel group type cur_channel belongs to
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AudioFrameQueue afq;
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DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
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DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
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AACQuantizeBandCostCacheEntry quantize_band_cost_cache[256][128]; ///< memoization area for quantize_band_cost
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struct {
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float *samples;
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} buffer;
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} AACEncContext;
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void ff_aac_coder_init_mips(AACEncContext *c);
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void ff_quantize_band_cost_cache_init(struct AACEncContext *s);
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#endif /* AVCODEC_AACENC_H */
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