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281 lines
9.4 KiB
281 lines
9.4 KiB
/* |
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* AAC definitions and structures |
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* Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) |
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* Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) |
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* |
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* This file is part of FFmpeg. |
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* |
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* FFmpeg is free software; you can redistribute it and/or |
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* modify it under the terms of the GNU Lesser General Public |
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* License as published by the Free Software Foundation; either |
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* version 2.1 of the License, or (at your option) any later version. |
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* |
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* FFmpeg is distributed in the hope that it will be useful, |
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* but WITHOUT ANY WARRANTY; without even the implied warranty of |
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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* Lesser General Public License for more details. |
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* |
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* You should have received a copy of the GNU Lesser General Public |
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* License along with FFmpeg; if not, write to the Free Software |
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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*/ |
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/** |
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* @file libavcodec/aac.h |
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* AAC definitions and structures |
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* @author Oded Shimon ( ods15 ods15 dyndns org ) |
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* @author Maxim Gavrilov ( maxim.gavrilov gmail com ) |
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*/ |
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#ifndef AVCODEC_AAC_H |
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#define AVCODEC_AAC_H |
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#include "avcodec.h" |
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#include "dsputil.h" |
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#include "mpeg4audio.h" |
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#include <stdint.h> |
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#define AAC_INIT_VLC_STATIC(num, size) \ |
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INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \ |
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ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \ |
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ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \ |
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size); |
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#define MAX_CHANNELS 64 |
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#define MAX_ELEM_ID 16 |
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#define TNS_MAX_ORDER 20 |
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enum RawDataBlockType { |
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TYPE_SCE, |
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TYPE_CPE, |
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TYPE_CCE, |
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TYPE_LFE, |
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TYPE_DSE, |
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TYPE_PCE, |
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TYPE_FIL, |
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TYPE_END, |
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}; |
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enum ExtensionPayloadID { |
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EXT_FILL, |
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EXT_FILL_DATA, |
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EXT_DATA_ELEMENT, |
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EXT_DYNAMIC_RANGE = 0xb, |
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EXT_SBR_DATA = 0xd, |
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EXT_SBR_DATA_CRC = 0xe, |
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}; |
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enum WindowSequence { |
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ONLY_LONG_SEQUENCE, |
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LONG_START_SEQUENCE, |
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EIGHT_SHORT_SEQUENCE, |
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LONG_STOP_SEQUENCE, |
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}; |
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enum BandType { |
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ZERO_BT = 0, ///< Scalefactors and spectral data are all zero. |
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FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word. |
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ESC_BT = 11, ///< Spectral data are coded with an escape sequence. |
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NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream. |
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INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions. |
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INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions. |
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}; |
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#define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10) |
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enum ChannelPosition { |
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AAC_CHANNEL_FRONT = 1, |
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AAC_CHANNEL_SIDE = 2, |
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AAC_CHANNEL_BACK = 3, |
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AAC_CHANNEL_LFE = 4, |
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AAC_CHANNEL_CC = 5, |
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}; |
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/** |
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* The point during decoding at which channel coupling is applied. |
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*/ |
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enum CouplingPoint { |
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BEFORE_TNS, |
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BETWEEN_TNS_AND_IMDCT, |
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AFTER_IMDCT = 3, |
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}; |
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/** |
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* Predictor State |
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*/ |
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typedef struct { |
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float cor0; |
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float cor1; |
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float var0; |
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float var1; |
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float r0; |
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float r1; |
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} PredictorState; |
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#define MAX_PREDICTORS 672 |
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#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times |
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#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0 |
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#define SCALE_MAX_POS 255 ///< scalefactor index maximum value |
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#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard |
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#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference |
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/** |
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* Individual Channel Stream |
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*/ |
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typedef struct { |
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uint8_t max_sfb; ///< number of scalefactor bands per group |
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enum WindowSequence window_sequence[2]; |
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uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window. |
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int num_window_groups; |
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uint8_t group_len[8]; |
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const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window |
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const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window |
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int num_swb; ///< number of scalefactor window bands |
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int num_windows; |
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int tns_max_bands; |
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int predictor_present; |
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int predictor_initialized; |
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int predictor_reset_group; |
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uint8_t prediction_used[41]; |
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} IndividualChannelStream; |
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/** |
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* Temporal Noise Shaping |
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*/ |
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typedef struct { |
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int present; |
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int n_filt[8]; |
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int length[8][4]; |
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int direction[8][4]; |
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int order[8][4]; |
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float coef[8][4][TNS_MAX_ORDER]; |
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} TemporalNoiseShaping; |
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/** |
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* Dynamic Range Control - decoded from the bitstream but not processed further. |
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*/ |
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typedef struct { |
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int pce_instance_tag; ///< Indicates with which program the DRC info is associated. |
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int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative |
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int dyn_rng_ctl[17]; ///< DRC magnitude information |
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int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing. |
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int band_incr; ///< Number of DRC bands greater than 1 having DRC info. |
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int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain. |
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int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines. |
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int prog_ref_level; /**< A reference level for the long-term program audio level for all |
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* channels combined. |
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*/ |
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} DynamicRangeControl; |
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typedef struct { |
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int num_pulse; |
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int start; |
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int pos[4]; |
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int amp[4]; |
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} Pulse; |
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/** |
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* coupling parameters |
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*/ |
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typedef struct { |
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enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied. |
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int num_coupled; ///< number of target elements |
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enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE. |
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int id_select[8]; ///< element id |
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int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel; |
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* [2] list of gains for left channel; [3] lists of gains for both channels |
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*/ |
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float gain[16][120]; |
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} ChannelCoupling; |
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/** |
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* Single Channel Element - used for both SCE and LFE elements. |
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*/ |
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typedef struct { |
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IndividualChannelStream ics; |
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TemporalNoiseShaping tns; |
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Pulse pulse; |
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enum BandType band_type[128]; ///< band types |
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int band_type_run_end[120]; ///< band type run end points |
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float sf[120]; ///< scalefactors |
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int sf_idx[128]; ///< scalefactor indices (used by encoder) |
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uint8_t zeroes[128]; ///< band is not coded (used by encoder) |
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DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT |
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DECLARE_ALIGNED_16(float, saved[1024]); ///< overlap |
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DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output |
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PredictorState predictor_state[MAX_PREDICTORS]; |
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} SingleChannelElement; |
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/** |
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* channel element - generic struct for SCE/CPE/CCE/LFE |
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*/ |
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typedef struct { |
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// CPE specific |
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int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream. |
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int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder) |
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uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band |
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// shared |
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SingleChannelElement ch[2]; |
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// CCE specific |
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ChannelCoupling coup; |
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} ChannelElement; |
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/** |
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* main AAC context |
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*/ |
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typedef struct { |
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AVCodecContext * avccontext; |
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MPEG4AudioConfig m4ac; |
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int is_saved; ///< Set if elements have stored overlap from previous frame. |
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DynamicRangeControl che_drc; |
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/** |
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* @defgroup elements Channel element related data. |
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* @{ |
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*/ |
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enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the |
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* first index as the first 4 raw data block types |
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*/ |
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ChannelElement * che[4][MAX_ELEM_ID]; |
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ChannelElement * tag_che_map[4][MAX_ELEM_ID]; |
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int tags_mapped; |
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/** @} */ |
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/** |
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* @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.) |
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* @{ |
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*/ |
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DECLARE_ALIGNED_16(float, buf_mdct[1024]); |
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/** @} */ |
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/** |
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* @defgroup tables Computed / set up during initialization. |
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* @{ |
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*/ |
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MDCTContext mdct; |
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MDCTContext mdct_small; |
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DSPContext dsp; |
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int random_state; |
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/** @} */ |
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/** |
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* @defgroup output Members used for output interleaving. |
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* @{ |
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*/ |
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float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output). |
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float add_bias; ///< offset for dsp.float_to_int16 |
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float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16. |
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int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16 |
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/** @} */ |
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DECLARE_ALIGNED(16, float, temp[128]); |
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int output_configured; |
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} AACContext; |
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#endif /* AVCODEC_AAC_H */
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