ffmpeg/libavcodec/aacenc.h
Michael Niedermayer 0bb57f8bf0 Merge remote-tracking branch 'qatar/master'
* 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>
2012-01-24 02:41:53 +01:00

88 lines
3.0 KiB
C

/*
* AAC encoder
* 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_AACENC_H
#define AVCODEC_AACENC_H
#include "avcodec.h"
#include "put_bits.h"
#include "dsputil.h"
#include "aac.h"
#include "psymodel.h"
#define AAC_CODER_NB 4
typedef struct AACEncOptions {
int stereo_mode;
int aac_coder;
} AACEncOptions;
struct AACEncContext;
typedef struct AACCoefficientsEncoder {
void (*search_for_quantizers)(AVCodecContext *avctx, struct AACEncContext *s,
SingleChannelElement *sce, const float lambda);
void (*encode_window_bands_info)(struct AACEncContext *s, SingleChannelElement *sce,
int win, int group_len, const float lambda);
void (*quantize_and_encode_band)(struct AACEncContext *s, PutBitContext *pb, const float *in, int size,
int scale_idx, int cb, const float lambda);
void (*search_for_ms)(struct AACEncContext *s, ChannelElement *cpe, const float lambda);
} AACCoefficientsEncoder;
extern AACCoefficientsEncoder ff_aac_coders[];
/**
* AAC encoder context
*/
typedef struct AACEncContext {
AVClass *av_class;
AACEncOptions options; ///< encoding options
PutBitContext pb;
FFTContext mdct1024; ///< long (1024 samples) frame transform context
FFTContext mdct128; ///< short (128 samples) frame transform context
DSPContext dsp;
float *planar_samples[6]; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
int channels; ///< channel count
const uint8_t *chan_map; ///< channel configuration map
ChannelElement *cpe; ///< channel elements
FFPsyContext psy;
struct FFPsyPreprocessContext* psypp;
AACCoefficientsEncoder *coder;
int cur_channel;
int last_frame;
float lambda;
DECLARE_ALIGNED(16, int, qcoefs)[96]; ///< quantized coefficients
DECLARE_ALIGNED(32, float, scoefs)[1024]; ///< scaled coefficients
struct {
float *samples;
} buffer;
} AACEncContext;
extern float ff_aac_pow34sf_tab[428];
#endif /* AVCODEC_AACENC_H */