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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 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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/**
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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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    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 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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    enum BandType band_type[120];             ///< 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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    DECLARE_ALIGNED_16(float, coeffs[1024]);  ///< coefficients for IMDCT
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    DECLARE_ALIGNED_16(float, saved[512]);    ///< 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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    uint8_t ms_mask[120];     ///< 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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} AACContext;
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#endif /* AVCODEC_AAC_H */