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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 definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef AVCODEC_AAC_H
#define AVCODEC_AAC_H
#include "libavutil/float_dsp.h"
#include "avcodec.h"
#include "dsputil.h"
#include "fft.h"
#include "mpeg4audio.h"
#include "sbr.h"
#include "fmtconvert.h"
#include <stdint.h>
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
#define MAX_LTP_LONG_SFB 40
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_OFF = 0,
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,
};
/**
* Output configuration status
*/
enum OCStatus {
OC_NONE, ///< Output unconfigured
OC_TRIAL_PCE, ///< Output configuration under trial specified by an inband PCE
OC_TRIAL_FRAME, ///< Output configuration under trial specified by a frame header
OC_GLOBAL_HDR, ///< Output configuration set in a global header but not yet locked
OC_LOCKED, ///< Output configuration locked in place
};
typedef struct {
MPEG4AudioConfig m4ac;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags;
int channels;
uint64_t channel_layout;
enum OCStatus status;
} OutputConfiguration;
/**
* Predictor State
*/
typedef struct {
float cor0;
float cor1;
float var0;
float var1;
float r0;
float r1;
} PredictorState;
#define MAX_PREDICTORS 672
#define SCALE_DIV_512 36 ///< scalefactor difference that corresponds to scale difference in 512 times
#define SCALE_ONE_POS 140 ///< scalefactor index that corresponds to scale=1.0
#define SCALE_MAX_POS 255 ///< scalefactor index maximum value
#define SCALE_MAX_DIFF 60 ///< maximum scalefactor difference allowed by standard
#define SCALE_DIFF_ZERO 60 ///< codebook index corresponding to zero scalefactor indices difference
#define POW_SF2_ZERO 200 ///< ff_aac_pow2sf_tab index corresponding to pow(2, 0);
/**
* Long Term Prediction
*/
typedef struct {
int8_t present;
int16_t lag;
float coef;
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
/**
* Individual Channel Stream
*/
typedef struct {
uint8_t max_sfb; ///< number of scalefactor bands per group
enum WindowSequence window_sequence[2];
uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
int num_window_groups;
uint8_t group_len[8];
LongTermPrediction ltp;
const uint16_t *swb_offset; ///< table of offsets to the lowest spectral coefficient of a scalefactor band, sfb, for a particular window
const uint8_t *swb_sizes; ///< table of scalefactor band sizes for a particular window
int num_swb; ///< number of scalefactor window bands
int num_windows;
int tns_max_bands;
int predictor_present;
int predictor_initialized;
int predictor_reset_group;
uint8_t prediction_used[41];
} IndividualChannelStream;
/**
* Temporal Noise Shaping
*/
typedef struct {
int present;
int n_filt[8];
int length[8][4];
int direction[8][4];
int order[8][4];
float coef[8][4][TNS_MAX_ORDER];
} TemporalNoiseShaping;
/**
* 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 pos[4];
int amp[4];
} Pulse;
/**
* coupling parameters
*/
typedef struct {
enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
int num_coupled; ///< number of target elements
enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
int id_select[8]; ///< element id
int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
* [2] list of gains for left channel; [3] lists of gains for both channels
*/
float gain[16][120];
} ChannelCoupling;
/**
* Single Channel Element - used for both SCE and LFE elements.
*/
typedef struct {
IndividualChannelStream ics;
TemporalNoiseShaping tns;
Pulse pulse;
enum BandType band_type[128]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
int sf_idx[128]; ///< scalefactor indices (used by encoder)
uint8_t zeroes[128]; ///< band is not coded (used by encoder)
DECLARE_ALIGNED(32, float, coeffs)[1024]; ///< coefficients for IMDCT
DECLARE_ALIGNED(32, float, saved)[1024]; ///< overlap
DECLARE_ALIGNED(32, float, ret)[2048]; ///< PCM output
DECLARE_ALIGNED(16, float, ltp_state)[3072]; ///< time signal for LTP
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct {
// CPE specific
int common_window; ///< Set if channels share a common 'IndividualChannelStream' in bitstream.
int ms_mode; ///< Signals mid/side stereo flags coding mode (used by encoder)
uint8_t ms_mask[128]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
SpectralBandReplication sbr;
} ChannelElement;
/**
* main AAC context
*/
typedef struct {
AVCodecContext *avctx;
AVFrame frame;
int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
/**
* @name Channel element related data
* @{
*/
ChannelElement *che[4][MAX_ELEM_ID];
ChannelElement *tag_che_map[4][MAX_ELEM_ID];
int tags_mapped;
/** @} */
/**
* @name temporary aligned temporary buffers
* (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED(32, float, buf_mdct)[1024];
/** @} */
/**
* @name Computed / set up during initialization
* @{
*/
FFTContext mdct;
FFTContext mdct_small;
FFTContext mdct_ltp;
DSPContext dsp;
FmtConvertContext fmt_conv;
AVFloatDSPContext fdsp;
int random_state;
/** @} */
/**
* @name Members used for output interleaving
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
/** @} */
DECLARE_ALIGNED(32, float, temp)[128];
OutputConfiguration oc[2];
int warned_num_aac_frames;
} AACContext;
#endif /* AVCODEC_AAC_H */
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