Mercurial > libavcodec.hg
view aac.h @ 7577:ed956c3c2cf3 libavcodec
The ff_sine_#[] should be aligned as they will commonly be used in dsputil
functions
author | superdump |
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date | Thu, 14 Aug 2008 22:12:19 +0000 |
parents | 4fca7939ad48 |
children | a05954c505ab |
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/* * 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 MAX_CHANNELS 64 #define MAX_ELEM_ID 16 #define IVQUANT_SIZE 1024 enum AudioObjectType { AOT_NULL, // Support? Name AOT_AAC_MAIN, ///< Y Main AOT_AAC_LC, ///< Y Low Complexity AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction AOT_SBR, ///< N (in progress) Spectral Band Replication AOT_AAC_SCALABLE, ///< N Scalable AOT_TWINVQ, ///< N Twin Vector Quantizer AOT_CELP, ///< N Code Excited Linear Prediction AOT_HVXC, ///< N Harmonic Vector eXcitation Coding AOT_TTSI = 12, ///< N Text-To-Speech Interface AOT_MAINSYNTH, ///< N Main Synthesis AOT_WAVESYNTH, ///< N Wavetable Synthesis AOT_MIDI, ///< N General MIDI AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding AOT_ER_AAC_LD, ///< N Error Resilient Low Delay AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise AOT_ER_PARAM, ///< N Error Resilient Parametric AOT_SSC, ///< N SinuSoidal Coding }; enum RawDataBlockType { TYPE_SCE, TYPE_CPE, TYPE_CCE, TYPE_LFE, TYPE_DSE, TYPE_PCE, TYPE_FIL, TYPE_END, }; enum ExtensionPayloadID { EXT_FILL, EXT_FILL_DATA, EXT_DATA_ELEMENT, EXT_DYNAMIC_RANGE = 0xb, EXT_SBR_DATA = 0xd, EXT_SBR_DATA_CRC = 0xe, }; enum WindowSequence { ONLY_LONG_SEQUENCE, LONG_START_SEQUENCE, EIGHT_SHORT_SEQUENCE, LONG_STOP_SEQUENCE, }; enum BandType { ZERO_BT = 0, ///< Scalefactors and spectral data are all zero. FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word. ESC_BT = 11, ///< Spectral data are coded with an escape sequence. NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream. INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions. INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions. }; #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10) enum ChannelPosition { AAC_CHANNEL_FRONT = 1, AAC_CHANNEL_SIDE = 2, AAC_CHANNEL_BACK = 3, AAC_CHANNEL_LFE = 4, AAC_CHANNEL_CC = 5, }; /** * The point during decoding at which channel coupling is applied. */ enum CouplingPoint { BEFORE_TNS, BETWEEN_TNS_AND_IMDCT, AFTER_IMDCT = 3, }; /** * Individual Channel Stream */ /** * Dynamic Range Control - decoded from the bitstream but not processed further. */ typedef struct { int pce_instance_tag; ///< Indicates with which program the DRC info is associated. int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative int dyn_rng_ctl[17]; ///< DRC magnitude information int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing. int band_incr; ///< Number of DRC bands greater than 1 having DRC info. int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain. int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines. int prog_ref_level; /**< A reference level for the long-term program audio level for all * channels combined. */ } DynamicRangeControl; typedef struct { int num_pulse; int start; int offset[4]; int amp[4]; } Pulse; /** * coupling parameters */ typedef struct { /** * main AAC context */ typedef struct { AVCodecContext * avccontext; MPEG4AudioConfig m4ac; int is_saved; ///< Set if elements have stored overlap from previous frame. DynamicRangeControl che_drc; /** * @defgroup elements * @{ */ enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the * first index as the first 4 raw data block types */ ChannelElement * che[4][MAX_ELEM_ID]; /** @} */ /** * @defgroup tables Computed / set up during initialization. * @{ */ MDCTContext mdct; MDCTContext mdct_small; DSPContext dsp; int random_state; /** @} */ /** * @defgroup output Members used for output interleaving. * @{ */ float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output). 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 */