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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>
188 lines
6.5 KiB
C
188 lines
6.5 KiB
C
/*
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* audio encoder psychoacoustic model
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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_PSYMODEL_H
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#define AVCODEC_PSYMODEL_H
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#include "avcodec.h"
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/** maximum possible number of bands */
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#define PSY_MAX_BANDS 128
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/** maximum number of channels */
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#define PSY_MAX_CHANS 20
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/**
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* single band psychoacoustic information
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*/
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typedef struct FFPsyBand {
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int bits;
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float energy;
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float threshold;
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float distortion;
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float perceptual_weight;
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} FFPsyBand;
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/**
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* single channel psychoacoustic information
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*/
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typedef struct FFPsyChannel {
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FFPsyBand psy_bands[PSY_MAX_BANDS]; ///< channel bands information
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float entropy; ///< total PE for this channel
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} FFPsyChannel;
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/**
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* psychoacoustic information for an arbitrary group of channels
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*/
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typedef struct FFPsyChannelGroup {
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FFPsyChannel *ch[PSY_MAX_CHANS]; ///< pointers to the individual channels in the group
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uint8_t num_ch; ///< number of channels in this group
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uint8_t coupling[PSY_MAX_BANDS]; ///< allow coupling for this band in the group
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} FFPsyChannelGroup;
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/**
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* windowing related information
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*/
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typedef struct FFPsyWindowInfo {
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int window_type[3]; ///< window type (short/long/transitional, etc.) - current, previous and next
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int window_shape; ///< window shape (sine/KBD/whatever)
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int num_windows; ///< number of windows in a frame
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int grouping[8]; ///< window grouping (for e.g. AAC)
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int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
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} FFPsyWindowInfo;
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/**
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* context used by psychoacoustic model
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*/
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typedef struct FFPsyContext {
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AVCodecContext *avctx; ///< encoder context
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const struct FFPsyModel *model; ///< encoder-specific model functions
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FFPsyChannel *ch; ///< single channel information
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FFPsyChannelGroup *group; ///< channel group information
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int num_groups; ///< number of channel groups
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uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
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int *num_bands; ///< number of scalefactor bands for possible frame sizes
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int num_lens; ///< number of scalefactor band sets
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struct {
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int size; ///< size of the bitresevoir in bits
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int bits; ///< number of bits used in the bitresevoir
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} bitres;
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void* model_priv_data; ///< psychoacoustic model implementation private data
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} FFPsyContext;
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/**
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* codec-specific psychoacoustic model implementation
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*/
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typedef struct FFPsyModel {
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const char *name;
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int (*init) (FFPsyContext *apc);
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/**
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* Suggest window sequence for channel.
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*
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* @param ctx model context
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* @param audio samples for the current frame
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* @param la lookahead samples (NULL when unavailable)
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* @param channel number of channel element to analyze
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* @param prev_type previous window type
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*
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* @return suggested window information in a structure
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*/
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FFPsyWindowInfo (*window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type);
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/**
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* Perform psychoacoustic analysis and set band info (threshold, energy) for a group of channels.
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*
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* @param ctx model context
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* @param channel channel number of the first channel in the group to perform analysis on
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* @param coeffs array of pointers to the transformed coefficients
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* @param wi window information for the channels in the group
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*/
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void (*analyze)(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi);
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void (*end) (FFPsyContext *apc);
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} FFPsyModel;
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/**
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* Initialize psychoacoustic model.
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*
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* @param ctx model context
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* @param avctx codec context
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* @param num_lens number of possible frame lengths
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* @param bands scalefactor band lengths for all frame lengths
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* @param num_bands number of scalefactor bands for all frame lengths
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* @param num_groups number of channel groups
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* @param group_map array with # of channels in group - 1, for each group
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*
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* @return zero if successful, a negative value if not
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*/
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av_cold int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
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const uint8_t **bands, const int* num_bands,
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int num_groups, const uint8_t *group_map);
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/**
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* Determine what group a channel belongs to.
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*
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* @param ctx psymodel context
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* @param channel channel to locate the group for
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*
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* @return pointer to the FFPsyChannelGroup this channel belongs to
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*/
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FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel);
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/**
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* Cleanup model context at the end.
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*
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* @param ctx model context
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*/
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av_cold void ff_psy_end(FFPsyContext *ctx);
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/**************************************************************************
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* Audio preprocessing stuff. *
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* This should be moved into some audio filter eventually. *
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**************************************************************************/
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struct FFPsyPreprocessContext;
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/**
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* psychoacoustic model audio preprocessing initialization
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*/
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av_cold struct FFPsyPreprocessContext* ff_psy_preprocess_init(AVCodecContext *avctx);
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/**
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* Preprocess several channel in audio frame in order to compress it better.
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*
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* @param ctx preprocessing context
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* @param audio samples to be filtered (in place)
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* @param channels number of channel to preprocess
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*/
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void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels);
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/**
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* Cleanup audio preprocessing module.
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*/
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av_cold void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);
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#endif /* AVCODEC_PSYMODEL_H */
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