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* qatar/master: Remove ffmpeg. aacenc: Simplify windowing aacenc: Move saved overlap samples to the beginning of the same buffer as incoming samples. aacenc: Deinterleave input samples before processing. aacenc: Store channel count in AACEncContext. aacenc: Move Q^3/4 calculation to it's own table aacenc: Request normalized float samples instead of converting s16 samples to float. aacpsy: Replace an if with FFMAX in LAME windowing. aacenc: cosmetics, replace 'rd' with 'bits' in codebook_trellis_rate to make it more clear what is being calculated. aacpsy: cosmetics, change a FIXME to a NOTE about subshort comparisons aacenc: cosmetics: move init() and end() to the bottom of the file. aacenc: aac_encode_init() cleanup XWD encoder and decoder vc1: don't read the interpfrm and bfraction elements for interlaced frames mxfdec: fix memleak on mxf_read_close() westwood: split the AUD and VQA demuxers into separate files. Conflicts: .gitignore Changelog Makefile configure doc/ffmpeg.texi ffmpeg.c libavcodec/Makefile libavcodec/aacenc.c libavcodec/allcodecs.c libavcodec/avcodec.h libavcodec/version.h libavformat/Makefile libavformat/img2.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
88 lines
3.0 KiB
C
88 lines
3.0 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 "avcodec.h"
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#include "put_bits.h"
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#include "dsputil.h"
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#include "aac.h"
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#include "psymodel.h"
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#define AAC_CODER_NB 4
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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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} 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 (*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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DSPContext dsp;
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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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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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#endif /* AVCODEC_AACENC_H */
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