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Diffstat (limited to 'ffmpeg1/libavcodec/aacenc.h')
| -rw-r--r-- | ffmpeg1/libavcodec/aacenc.h | 90 |
1 files changed, 90 insertions, 0 deletions
diff --git a/ffmpeg1/libavcodec/aacenc.h b/ffmpeg1/libavcodec/aacenc.h new file mode 100644 index 0000000..cebdd18 --- /dev/null +++ b/ffmpeg1/libavcodec/aacenc.h @@ -0,0 +1,90 @@ +/* + * 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 "libavutil/float_dsp.h" +#include "avcodec.h" +#include "put_bits.h" + +#include "aac.h" +#include "audio_frame_queue.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 + AVFloatDSPContext fdsp; + 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; + AudioFrameQueue afq; + 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]; + +void ff_aac_coder_init_mips(AACEncContext *c); + +#endif /* AVCODEC_AACENC_H */ |
