mirror of
https://github.com/nyanmisaka/ffmpeg-rockchip.git
synced 2026-04-25 08:56:59 +08:00
Rostislav Pehlivanov
44ddee945a
This commit completely alters the algorithm of prediction. The original commit which introduced prediction was completely incorrect to even remotely care about what the actual coefficients contain or whether any options were enabled. Not my actual fault. This commit treats prediction the way the decoder does and expects to do: like lossy encryption. Everything related to prediction now happens at the very end but just before quantization and encoding of coefficients. On the decoder side, prediction happens before anything has had a chance to even access the coefficients. Also the original implementation had problems because it actually touched the band_type of special bands which already had their scalefactor indices marked and it's a wonder the asserion wasn't triggered when transmitting those. Overall, this now drastically increases audio quality and you should think about enabling it if you don't plan on playing anything encoded on really old low power ultra-embedded devices since they might not support decoding of prediction or AAC-Main. Though the specifications were written ages ago and as times change so do the FLOPS. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
113 lines
4.2 KiB
C
113 lines
4.2 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 stereo_mode;
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int aac_coder;
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int pns;
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int tns;
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int pred;
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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_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
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void (*adjust_common_prediction)(struct AACEncContext *s, ChannelElement *cpe);
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void (*apply_main_pred)(struct AACEncContext *s, SingleChannelElement *sce);
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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 (*search_for_tns)(struct AACEncContext *s, SingleChannelElement *sce);
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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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/**
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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[6]; ///< 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;
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int last_frame;
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float lambda;
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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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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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#endif /* AVCODEC_AACENC_H */
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