1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/psymodel.h
Claudio Freire 59216e0525 AAC Encoder: clipping avoidance
Avoid clipping due to quantization noise to produce audible
artifacts, by detecting near-clipping signals and both attenuating
them a little and encoding escape-encoded bands (usually the
loudest) rounding towards zero instead of nearest, which tends to
decrease overall energy and thus clipping.

Currently fate tests measure numerical error so this change makes
tests using asynth (which are near clipping) report higher error
not less, because of window attenuation. Yet, they sound better,
not worse (albeit subtle, other samples aren't subtle at all).
Only measuring psychoacoustically weighted error would make for
a representative test, so that will be left for a future patch.

Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2015-07-27 19:13:48 +02:00

190 lines
6.7 KiB
C

/*
* audio encoder psychoacoustic model
* Copyright (C) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_PSYMODEL_H
#define AVCODEC_PSYMODEL_H
#include "avcodec.h"
/** maximum possible number of bands */
#define PSY_MAX_BANDS 128
/** maximum number of channels */
#define PSY_MAX_CHANS 20
#define AAC_CUTOFF(s) ((s)->bit_rate ? FFMIN3(4000 + (s)->bit_rate/8, 12000 + (s)->bit_rate/32, (s)->sample_rate / 2) : ((s)->sample_rate / 2))
/**
* single band psychoacoustic information
*/
typedef struct FFPsyBand {
int bits;
float energy;
float threshold;
float spread; /* Energy spread over the band */
} FFPsyBand;
/**
* single channel psychoacoustic information
*/
typedef struct FFPsyChannel {
FFPsyBand psy_bands[PSY_MAX_BANDS]; ///< channel bands information
float entropy; ///< total PE for this channel
} FFPsyChannel;
/**
* psychoacoustic information for an arbitrary group of channels
*/
typedef struct FFPsyChannelGroup {
FFPsyChannel *ch[PSY_MAX_CHANS]; ///< pointers to the individual channels in the group
uint8_t num_ch; ///< number of channels in this group
uint8_t coupling[PSY_MAX_BANDS]; ///< allow coupling for this band in the group
} FFPsyChannelGroup;
/**
* windowing related information
*/
typedef struct FFPsyWindowInfo {
int window_type[3]; ///< window type (short/long/transitional, etc.) - current, previous and next
int window_shape; ///< window shape (sine/KBD/whatever)
int num_windows; ///< number of windows in a frame
int grouping[8]; ///< window grouping (for e.g. AAC)
float clipping[8]; ///< maximum absolute normalized intensity in the given window for clip avoidance
int *window_sizes; ///< sequence of window sizes inside one frame (for eg. WMA)
} FFPsyWindowInfo;
/**
* context used by psychoacoustic model
*/
typedef struct FFPsyContext {
AVCodecContext *avctx; ///< encoder context
const struct FFPsyModel *model; ///< encoder-specific model functions
FFPsyChannel *ch; ///< single channel information
FFPsyChannelGroup *group; ///< channel group information
int num_groups; ///< number of channel groups
uint8_t **bands; ///< scalefactor band sizes for possible frame sizes
int *num_bands; ///< number of scalefactor bands for possible frame sizes
int num_lens; ///< number of scalefactor band sets
struct {
int size; ///< size of the bitresevoir in bits
int bits; ///< number of bits used in the bitresevoir
} bitres;
void* model_priv_data; ///< psychoacoustic model implementation private data
} FFPsyContext;
/**
* codec-specific psychoacoustic model implementation
*/
typedef struct FFPsyModel {
const char *name;
int (*init) (FFPsyContext *apc);
/**
* Suggest window sequence for channel.
*
* @param ctx model context
* @param audio samples for the current frame
* @param la lookahead samples (NULL when unavailable)
* @param channel number of channel element to analyze
* @param prev_type previous window type
*
* @return suggested window information in a structure
*/
FFPsyWindowInfo (*window)(FFPsyContext *ctx, const float *audio, const float *la, int channel, int prev_type);
/**
* Perform psychoacoustic analysis and set band info (threshold, energy) for a group of channels.
*
* @param ctx model context
* @param channel channel number of the first channel in the group to perform analysis on
* @param coeffs array of pointers to the transformed coefficients
* @param wi window information for the channels in the group
*/
void (*analyze)(FFPsyContext *ctx, int channel, const float **coeffs, const FFPsyWindowInfo *wi);
void (*end) (FFPsyContext *apc);
} FFPsyModel;
/**
* Initialize psychoacoustic model.
*
* @param ctx model context
* @param avctx codec context
* @param num_lens number of possible frame lengths
* @param bands scalefactor band lengths for all frame lengths
* @param num_bands number of scalefactor bands for all frame lengths
* @param num_groups number of channel groups
* @param group_map array with # of channels in group - 1, for each group
*
* @return zero if successful, a negative value if not
*/
int ff_psy_init(FFPsyContext *ctx, AVCodecContext *avctx, int num_lens,
const uint8_t **bands, const int *num_bands,
int num_groups, const uint8_t *group_map);
/**
* Determine what group a channel belongs to.
*
* @param ctx psymodel context
* @param channel channel to locate the group for
*
* @return pointer to the FFPsyChannelGroup this channel belongs to
*/
FFPsyChannelGroup *ff_psy_find_group(FFPsyContext *ctx, int channel);
/**
* Cleanup model context at the end.
*
* @param ctx model context
*/
void ff_psy_end(FFPsyContext *ctx);
/**************************************************************************
* Audio preprocessing stuff. *
* This should be moved into some audio filter eventually. *
**************************************************************************/
struct FFPsyPreprocessContext;
/**
* psychoacoustic model audio preprocessing initialization
*/
struct FFPsyPreprocessContext *ff_psy_preprocess_init(AVCodecContext *avctx);
/**
* Preprocess several channel in audio frame in order to compress it better.
*
* @param ctx preprocessing context
* @param audio samples to be filtered (in place)
* @param channels number of channel to preprocess
*/
void ff_psy_preprocess(struct FFPsyPreprocessContext *ctx, float **audio, int channels);
/**
* Cleanup audio preprocessing module.
*/
void ff_psy_preprocess_end(struct FFPsyPreprocessContext *ctx);
#endif /* AVCODEC_PSYMODEL_H */