38fd4c2e66
This commit finalizes the PNS implementation previously added to the encoder by moving it to a seperate function search_for_pns() and thus making it coder-generic. This new implementation makes use of the spread field of the psy bands and the lambda quality feedback paremeter. The spread of the spectrum in a band prevents PNS from being used excessively and thus preserve more phase information in high frequencies. The lambda parameter allows the number of PNS-marked bands to vary based on the lambda parameter and the amount of bits available, making better choices on which bands are to be marked as noise. Comparisons with the previous PNS implementation can be found here: https://trac.ffmpeg.org/attachment/wiki/Encode/AAC/ This is V2 of the patch, the changes from the previous version being that this version uses the new band->spread metric from aacpsy and normalizes the energy using the group size. These changes were suggested by Claudio Freire on the mailing list. Another change is the use of lambda to alter the frequency threshold. This change makes the actual threshold frequencies vary between +-2Khz of what's specified, depending on frame encoding performance. Reviewed-by: Claudio Freire <klaussfreire@gmail.com> Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
101 lines
3.5 KiB
C
101 lines
3.5 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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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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} 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, int size,
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int scale_idx, int cb, const float lambda);
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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, const float lambda);
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void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
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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 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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extern float ff_aac_pow34sf_tab[428];
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void ff_aac_coder_init_mips(AACEncContext *c);
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#endif /* AVCODEC_AACENC_H */
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