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author | Robert Swain <robert.swain@gmail.com> | 2008-08-05 19:27:17 +0000 |
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committer | Robert Swain <robert.swain@gmail.com> | 2008-08-05 19:27:17 +0000 |
commit | 589ce6e63de479832478f23ba941c0eac6e0c79e (patch) | |
tree | 5f62aa2ea434aaa52b3c9dbba2f3b9ed2ea02ea5 /libavcodec/aac.h | |
parent | 2ccd25d02b332ea727236a16899f24043e8221e2 (diff) | |
download | ffmpeg-streaming-589ce6e63de479832478f23ba941c0eac6e0c79e.zip ffmpeg-streaming-589ce6e63de479832478f23ba941c0eac6e0c79e.tar.gz |
AAC definitions and structures that have been OKed from the SoC AAC decoder
code
Originally committed as revision 14624 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavcodec/aac.h')
-rw-r--r-- | libavcodec/aac.h | 88 |
1 files changed, 88 insertions, 0 deletions
diff --git a/libavcodec/aac.h b/libavcodec/aac.h new file mode 100644 index 0000000..164362f --- /dev/null +++ b/libavcodec/aac.h @@ -0,0 +1,88 @@ +/* + * AAC definitions and structures + * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) + * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) + * + * 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 + */ + +/** + * @file aac.h + * AAC definitions and structures + * @author Oded Shimon ( ods15 ods15 dyndns org ) + * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) + */ + +#ifndef FFMPEG_AAC_H +#define FFMPEG_AAC_H + +#include "avcodec.h" +#include "dsputil.h" +#include "mpeg4audio.h" + +#include <stdint.h> + +#define AAC_INIT_VLC_STATIC(num, size) \ + INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \ + ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ + ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ + size); + +#define IVQUANT_SIZE 1024 + +enum WindowSequence { + ONLY_LONG_SEQUENCE, + LONG_START_SEQUENCE, + EIGHT_SHORT_SEQUENCE, + LONG_STOP_SEQUENCE, +}; + +enum ChannelType { + AAC_CHANNEL_FRONT = 1, + AAC_CHANNEL_SIDE = 2, + AAC_CHANNEL_BACK = 3, + AAC_CHANNEL_LFE = 4, + AAC_CHANNEL_CC = 5, +}; + +/** + * main AAC context + */ +typedef struct { + AVCodecContext * avccontext; + + /** + * @defgroup tables Computed / set up during initialization. + * @{ + */ + MDCTContext mdct; + MDCTContext mdct_small; + DSPContext dsp; + /** @} */ + + /** + * @defgroup output Members used for output interleaving and down-mixing. + * @{ + */ + float add_bias; ///< offset for dsp.float_to_int16 + float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16. + int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16 + /** @} */ + +} AACContext; + +#endif /* FFMPEG_AAC_H */ |