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https://github.com/FFmpeg/FFmpeg.git
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f20b67173c
This commit abandons the way the specifications state to quantize the coefficients, makes use of the new LPC float functions and is much better. The original way of converting non-normalized float samples to int32_t which out LPC system expects was wrong and it was wrong to assume the coefficients that are generated are also valid. It was essentially a full garbage-in, garbage-out system and it definitely shows when looking at spectrals and listening. The high frequencies were very overattenuated. The new LPC function performs the analysis directly. The specifications state to quantize the coefficients into four bit index values using an asin() function which of course had to have ugly ternary operators because the function turns negative if the coefficients are negative which when encoding causes invalid bitstream to get generated. This deviates from this by using the direct TNS tables, which are fairly small since you only have 4 bits at most for index values. The LPC values are directly quantized against the tables and are then used to perform filtering after the requantization, which simply fetches the array values. The end result is that TNS works much better now and doesn't attenuate anything but the actual signal, e.g. TNS removes quantization errors and does it's job correctly now. It might be enabled by default soon since it doesn't hurt and helps reduce nastyness at low bitrates. Signed-off-by: Rostislav Pehlivanov <atomnuker@gmail.com>
114 lines
4.2 KiB
C
114 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 (*apply_tns_filt)(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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