ffmpeg/libavcodec/aacenc.h
Michael Niedermayer 721be99371 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  cosmetics: fix some then/than typos
  doxygen: Include libavcodec and libavformat examples into the documentation
  avutil: elaborate documentation for av_get_random_seed
  Add support for aac streams in mp4/mov without extradata.
  aes: whitespace cosmetics
  adler32: whitespace cosmetics
  swscale: fix another yuv range conversion overflow in 16bit scaling.
  Fix cpu flags test program
  opt-test: Add missing braces to silence compiler warnings.
  build: Eliminate obsolete test targets.
  udp: Fix a compilation warning
  swscale: Unbreak build with --enable-small
  base64: add fate test
  aes: improve test program and add fate test
  adler32: make test program more useful and add fate test
  swscale: fix yuv range correction when using 16-bit scaling.
  aacenc: Make chan_map const correct

Conflicts:
	Makefile
	doc/examples/muxing-example.c
	libavformat/udp.c
	libavutil/random_seed.h

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-07-01 05:35:26 +02:00

78 lines
2.8 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"
typedef struct AACEncOptions {
int stereo_mode;
} 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;
int16_t *samples; ///< saved preprocessed input
int samplerate_index; ///< MPEG-4 samplerate index
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
} AACEncContext;
#endif /* AVCODEC_AACENC_H */