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/*
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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
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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 "libavutil/float_dsp.h"
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#include "avcodec.h"
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#include "dsputil.h"
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#include "fft.h"
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#include "mpeg4audio.h"
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#include "sbr.h"
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#include "fmtconvert.h"
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#include <stdint.h>
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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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#define MAX_LTP_LONG_SFB 40
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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_OFF = 0,
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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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* Output configuration status
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*/
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enum OCStatus {
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OC_NONE, ///< Output unconfigured
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OC_TRIAL_PCE, ///< Output configuration under trial specified by an inband PCE
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OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
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OC_GLOBAL_HDR, ///< Output configuration set in a global header but not yet locked
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OC_LOCKED, ///< Output configuration locked in place
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};
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typedef struct OutputConfiguration {
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MPEG4AudioConfig m4ac;
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uint8_t layout_map[MAX_ELEM_ID*4][3];
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int layout_map_tags;
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int channels;
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uint64_t channel_layout;
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enum OCStatus status;
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} OutputConfiguration;
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/**
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* Predictor State
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*/
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typedef struct PredictorState {
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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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* Long Term Prediction
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*/
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typedef struct LongTermPrediction {
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int8_t present;
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int16_t lag;
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float coef;
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int8_t used[MAX_LTP_LONG_SFB];
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} LongTermPrediction;
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/**
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* Individual Channel Stream
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*/
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typedef struct IndividualChannelStream {
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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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LongTermPrediction ltp;
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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 TemporalNoiseShaping {
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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 DynamicRangeControl {
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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 Pulse {
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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 ChannelCoupling {
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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 SingleChannelElement {
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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(32, float, coeffs)[1024]; ///< coefficients for IMDCT
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DECLARE_ALIGNED(32, float, saved)[1024]; ///< overlap
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DECLARE_ALIGNED(32, float, ret)[2048]; ///< PCM output
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DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
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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 ChannelElement {
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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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SpectralBandReplication sbr;
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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 AACContext {
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AVCodecContext *avctx;
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AVFrame frame;
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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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* @name Channel element related data
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* @{
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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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* @name temporary aligned temporary buffers
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* (We do not want to have these on the stack.)
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* @{
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*/
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DECLARE_ALIGNED(32, float, buf_mdct)[1024];
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/** @} */
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/**
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* @name Computed / set up during initialization
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* @{
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*/
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FFTContext mdct;
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FFTContext mdct_small;
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FFTContext mdct_ltp;
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DSPContext dsp;
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FmtConvertContext fmt_conv;
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AVFloatDSPContext fdsp;
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int random_state;
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/** @} */
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/**
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* @name 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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/** @} */
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aacdec: add support for dual mono in Japanese DTV
Japanese DTV uses some non standard extensions in AAC audio.
One example is 'dual mono', which combines two independent
audio into one stereo stream, storing them in left and right channels
respectively. Historically, dual mono audio has been used for
multi-lingual audio, one for local/native language, and another for english,
and usually the "main" (local language) channel should be output without
any user interactions.
The frames of those dual mono audio are allowed to set
ADTS channel_config field to 0, and just contain two SCE's *WITHOUT* PCE,
which is a non standard extension by Japanese DTV standard.
(ref. ARIB STD-B32 PartII 5.2.3)
This patch adds an AVPacket side data, AV_PKT_DATA_JP_DUALMONO,
which indicates that the AVPacket is likely to contain an audio frame
with the above dual mono extension, and has the parameter to specify
the desired channel selection in that case.
It also makes aacdec to detect dual mono and output just the desired
channel when this side data is attached.
Signed-off-by: Akihiro Tsukada <atsukada@users.sourceforge.net>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
13 years ago
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/**
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* @name Japanese DTV specific extension
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* @{
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*/
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int enable_jp_dmono; ///< enable japanese DTV specific 'dual mono'
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int dmono_mode; ///< select the channel to decode in dual mono.
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/** @} */
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DECLARE_ALIGNED(32, float, temp)[128];
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OutputConfiguration oc[2];
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int warned_num_aac_frames;
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} AACContext;
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#endif /* AVCODEC_AAC_H */
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