ffmpeg/libavcodec/aac/aacdec.h

563 lines
18 KiB
C

/*
* AAC decoder 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 decoder definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef AVCODEC_AAC_AACDEC_H
#define AVCODEC_AAC_AACDEC_H
#include <stdint.h>
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/fixed_dsp.h"
#include "libavutil/mem_internal.h"
#include "libavutil/tx.h"
#include "libavcodec/aac.h"
#include "libavcodec/avcodec.h"
#include "libavcodec/mpeg4audio.h"
#include "aacdec_ac.h"
typedef struct AACDecContext AACDecContext;
/**
* 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
};
enum AACOutputChannelOrder {
CHANNEL_ORDER_DEFAULT,
CHANNEL_ORDER_CODED,
};
/**
* The point during decoding at which channel coupling is applied.
*/
enum CouplingPoint {
BEFORE_TNS,
BETWEEN_TNS_AND_IMDCT,
AFTER_IMDCT = 3,
};
enum AACUsacElem {
ID_USAC_SCE = 0,
ID_USAC_CPE = 1,
ID_USAC_LFE = 2,
ID_USAC_EXT = 3,
};
enum ExtensionHeaderType {
ID_CONFIG_EXT_FILL = 0,
ID_CONFIG_EXT_LOUDNESS_INFO = 2,
ID_CONFIG_EXT_STREAM_ID = 7,
};
enum AACUsacExtension {
ID_EXT_ELE_FILL,
ID_EXT_ELE_MPEGS,
ID_EXT_ELE_SAOC,
ID_EXT_ELE_AUDIOPREROLL,
ID_EXT_ELE_UNI_DRC,
};
enum AACUSACLoudnessExt {
UNIDRCLOUDEXT_TERM = 0x0,
UNIDRCLOUDEXT_EQ = 0x1,
};
// Supposed to be equal to AAC_RENAME() in case of USE_FIXED.
#define RENAME_FIXED(name) name ## _fixed
#define INTFLOAT_UNION(name, elems) \
union { \
int RENAME_FIXED(name) elems; \
float name elems; \
}
#define INTFLOAT_ALIGNED_UNION(alignment, name, nb_elems) \
union { \
DECLARE_ALIGNED(alignment, int, RENAME_FIXED(name))[nb_elems]; \
DECLARE_ALIGNED(alignment, float, name)[nb_elems]; \
}
/**
* Long Term Prediction
*/
typedef struct LongTermPrediction {
int8_t present;
int16_t lag;
INTFLOAT_UNION(coef,);
int8_t used[MAX_LTP_LONG_SFB];
} LongTermPrediction;
/* Per channel core mode */
typedef struct AACUsacElemData {
uint8_t core_mode;
uint8_t scale_factor_grouping;
uint8_t tns_data_present;
/* Timewarping ratio */
#define NUM_TW_NODES 16
uint8_t tw_ratio[NUM_TW_NODES];
struct {
uint8_t acelp_core_mode : 3;
uint8_t lpd_mode : 5;
uint8_t bpf_control_info : 1;
uint8_t core_mode_last : 1;
uint8_t fac_data_present : 1;
int last_lpd_mode;
} ldp;
struct {
unsigned int seed;
uint8_t level : 3;
uint8_t offset : 5;
} noise;
struct {
uint8_t gain;
uint32_t kv[8 /* (1024 / 16) / 8 */][8];
} fac;
AACArithState ac;
} AACUsacElemData;
/**
* Individual Channel Stream
*/
typedef struct IndividualChannelStream {
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 sine window.
int num_window_groups;
int prev_num_window_groups; ///< Previous frame's number of 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
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];
uint8_t window_clipping[8]; ///< set if a certain window is near clipping
} IndividualChannelStream;
/**
* Temporal Noise Shaping
*/
typedef struct TemporalNoiseShaping {
int present;
int n_filt[8];
int length[8][4];
int direction[8][4];
int order[8][4];
INTFLOAT_UNION(coef, [8][4][TNS_MAX_ORDER]);
} TemporalNoiseShaping;
/**
* coupling parameters
*/
typedef struct ChannelCoupling {
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
*/
INTFLOAT_UNION(gain, [16][120]);
} ChannelCoupling;
/**
* Single Channel Element - used for both SCE and LFE elements.
*/
typedef struct SingleChannelElement {
IndividualChannelStream ics;
AACUsacElemData ue; ///< USAC element data
TemporalNoiseShaping tns;
enum BandType band_type[128]; ///< band types
int sfo[128]; ///< scalefactor offsets
INTFLOAT_UNION(sf, [128]); ///< scalefactors (8 windows * 16 sfb max)
INTFLOAT_ALIGNED_UNION(32, coeffs, 1024); ///< coefficients for IMDCT, maybe processed
INTFLOAT_ALIGNED_UNION(32, prev_coeffs, 1024); ///< unscaled previous contents of coeffs[] for USAC
INTFLOAT_ALIGNED_UNION(32, saved, 1536); ///< overlap
INTFLOAT_ALIGNED_UNION(32, ret_buf, 2048); ///< PCM output buffer
INTFLOAT_ALIGNED_UNION(16, ltp_state, 3072); ///< time signal for LTP
union {
struct PredictorStateFixed *RENAME_FIXED(predictor_state);
struct PredictorState *predictor_state;
};
union {
float *output; ///< PCM output
int *RENAME_FIXED(output); ///< PCM output
};
} SingleChannelElement;
typedef struct AACUsacStereo {
uint8_t common_window;
uint8_t common_tw;
uint8_t tns_on_lr; ///< Apply TNS before M/S and stereo prediction
uint8_t ms_mask_mode;
uint8_t config_idx;
/* Complex prediction */
uint8_t use_prev_frame;
uint8_t pred_dir;
uint8_t complex_coef;
uint8_t pred_used[128];
INTFLOAT_ALIGNED_UNION(32, alpha_q_re, 1024);
INTFLOAT_ALIGNED_UNION(32, alpha_q_im, 1024);
INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_re, 1024);
INTFLOAT_ALIGNED_UNION(32, prev_alpha_q_im, 1024);
INTFLOAT_ALIGNED_UNION(32, dmix_re, 1024);
INTFLOAT_ALIGNED_UNION(32, prev_dmix_re, 1024); /* Recalculated on every frame */
INTFLOAT_ALIGNED_UNION(32, dmix_im, 1024); /* Final prediction data */
} AACUsacStereo;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct ChannelElement {
int present;
// CPE specific
uint8_t max_sfb_ste; ///< (USAC) Maximum of both max_sfb values
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;
// USAC stereo coupling data
AACUsacStereo us;
} ChannelElement;
typedef struct AACUSACLoudnessInfo {
uint8_t drc_set_id : 6;
uint8_t downmix_id : 7;
struct {
uint16_t lvl : 12;
uint8_t present : 1;
} sample_peak;
struct {
uint16_t lvl : 12;
uint8_t measurement : 4;
uint8_t reliability : 2;
uint8_t present : 1;
} true_peak;
uint8_t nb_measurements : 4;
struct {
uint8_t method_def : 4;
uint8_t method_val;
uint8_t measurement : 4;
uint8_t reliability : 2;
} measurements[16];
} AACUSACLoudnessInfo;
typedef struct AACUsacElemConfig {
enum AACUsacElem type;
uint8_t tw_mdct : 1;
uint8_t noise_fill : 1;
uint8_t stereo_config_index;
struct {
int ratio;
uint8_t harmonic_sbr : 1; /* harmonicSBR */
uint8_t bs_intertes : 1; /* bs_interTes */
uint8_t bs_pvc : 1; /* bs_pvc */
struct {
uint8_t start_freq; /* dflt_start_freq */
uint8_t stop_freq; /* dflt_stop_freq */
uint8_t freq_scale; /* dflt_freq_scale */
uint8_t alter_scale : 1; /* dflt_alter_scale */
uint8_t noise_bands; /* dflt_noise_bands */
uint8_t limiter_bands; /* dflt_limiter_bands */
uint8_t limiter_gains; /* dflt_limiter_gains */
uint8_t interpol_freq : 1; /* dflt_interpol_freq */
uint8_t smoothing_mode : 1; /* dflt_smoothing_mode */
} dflt;
} sbr;
struct {
uint8_t freq_res; /* bsFreqRes */
uint8_t fixed_gain; /* bsFixedGainDMX */
uint8_t temp_shape_config; /* bsTempShapeConfig */
uint8_t decorr_config; /* bsDecorrConfig */
uint8_t high_rate_mode : 1; /* bsHighRateMode */
uint8_t phase_coding : 1; /* bsPhaseCoding */
uint8_t otts_bands_phase; /* bsOttBandsPhase */
uint8_t residual_coding; /* bsResidualCoding */
uint8_t residual_bands; /* bsResidualBands */
uint8_t pseudo_lr : 1; /* bsPseudoLr */
uint8_t env_quant_mode : 1; /* bsEnvQuantMode */
} mps;
struct {
enum AACUsacExtension type;
uint8_t payload_frag;
uint32_t default_len;
uint32_t pl_data_offset;
uint8_t *pl_data;
} ext;
} AACUsacElemConfig;
typedef struct AACUSACConfig {
uint8_t core_sbr_frame_len_idx; /* coreSbrFrameLengthIndex */
uint16_t core_frame_len;
uint16_t stream_identifier;
AACUsacElemConfig elems[64];
int nb_elems;
struct {
uint8_t nb_album;
AACUSACLoudnessInfo album_info[64];
uint8_t nb_info;
AACUSACLoudnessInfo info[64];
} loudness;
} AACUSACConfig;
typedef struct OutputConfiguration {
MPEG4AudioConfig m4ac;
uint8_t layout_map[MAX_ELEM_ID*4][3];
int layout_map_tags;
AVChannelLayout ch_layout;
enum OCStatus status;
AACUSACConfig usac;
} OutputConfiguration;
/**
* Dynamic Range Control - decoded from the bitstream but not processed further.
*/
typedef struct DynamicRangeControl {
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;
/**
* Decode-specific primitives
*/
typedef struct AACDecProc {
int (*decode_spectrum_and_dequant)(AACDecContext *ac,
GetBitContext *gb,
const Pulse *pulse,
SingleChannelElement *sce);
int (*decode_cce)(AACDecContext *ac, GetBitContext *gb, ChannelElement *che);
int (*sbr_ctx_alloc_init)(AACDecContext *ac, ChannelElement **che, int id_aac);
int (*sbr_decode_extension)(AACDecContext *ac, ChannelElement *che,
GetBitContext *gb, int crc, int cnt, int id_aac);
void (*sbr_apply)(AACDecContext *ac, ChannelElement *che,
int id_aac, void /* INTFLOAT */ *L, void /* INTFLOAT */ *R);
void (*sbr_ctx_close)(ChannelElement *che);
} AACDecProc;
/**
* DSP-specific primitives
*/
typedef struct AACDecDSP {
void (*dequant_scalefactors)(SingleChannelElement *sce);
void (*apply_mid_side_stereo)(AACDecContext *ac, ChannelElement *cpe);
void (*apply_intensity_stereo)(AACDecContext *ac, ChannelElement *cpe,
int ms_present);
void (*apply_tns)(void *_coef_param, TemporalNoiseShaping *tns,
IndividualChannelStream *ics, int decode);
void (*apply_ltp)(AACDecContext *ac, SingleChannelElement *sce);
void (*update_ltp)(AACDecContext *ac, SingleChannelElement *sce);
void (*apply_prediction)(AACDecContext *ac, SingleChannelElement *sce);
void (*apply_dependent_coupling)(AACDecContext *ac,
SingleChannelElement *target,
ChannelElement *cce, int index);
void (*apply_independent_coupling)(AACDecContext *ac,
SingleChannelElement *target,
ChannelElement *cce, int index);
void (*imdct_and_windowing)(AACDecContext *ac, SingleChannelElement *sce);
void (*imdct_and_windowing_768)(AACDecContext *ac, SingleChannelElement *sce);
void (*imdct_and_windowing_960)(AACDecContext *ac, SingleChannelElement *sce);
void (*imdct_and_windowing_ld)(AACDecContext *ac, SingleChannelElement *sce);
void (*imdct_and_windowing_eld)(AACDecContext *ac, SingleChannelElement *sce);
void (*clip_output)(AACDecContext *ac, ChannelElement *che, int type, int samples);
} AACDecDSP;
/**
* main AAC decoding context
*/
struct AACDecContext {
const struct AVClass *class;
struct AVCodecContext *avctx;
AACDecDSP dsp;
AACDecProc proc;
struct 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;
int warned_remapping_once;
/** @} */
/**
* @name temporary aligned temporary buffers
* (We do not want to have these on the stack.)
* @{
*/
INTFLOAT_ALIGNED_UNION(32, buf_mdct, 1024);
INTFLOAT_ALIGNED_UNION(32, temp, 128);
/** @} */
/**
* @name Computed / set up during initialization
* @{
*/
AVTXContext *mdct96;
AVTXContext *mdct120;
AVTXContext *mdct128;
AVTXContext *mdct480;
AVTXContext *mdct512;
AVTXContext *mdct768;
AVTXContext *mdct960;
AVTXContext *mdct1024;
AVTXContext *mdct_ltp;
av_tx_fn mdct96_fn;
av_tx_fn mdct120_fn;
av_tx_fn mdct128_fn;
av_tx_fn mdct480_fn;
av_tx_fn mdct512_fn;
av_tx_fn mdct768_fn;
av_tx_fn mdct960_fn;
av_tx_fn mdct1024_fn;
av_tx_fn mdct_ltp_fn;
union {
AVFixedDSPContext *RENAME_FIXED(fdsp);
AVFloatDSPContext *fdsp;
};
int random_state;
/** @} */
/**
* @name Members used for output
* @{
*/
SingleChannelElement *output_element[MAX_CHANNELS]; ///< Points to each SingleChannelElement
/** @} */
/**
* @name Japanese DTV specific extension
* @{
*/
int force_dmono_mode;///< 0->not dmono, 1->use first channel, 2->use second channel
int dmono_mode; ///< 0->not dmono, 1->use first channel, 2->use second channel
/** @} */
enum AACOutputChannelOrder output_channel_order;
OutputConfiguration oc[2];
int warned_num_aac_frames;
int warned_960_sbr;
unsigned warned_71_wide;
int warned_gain_control;
int warned_he_aac_mono;
int is_fixed;
};
#if defined(USE_FIXED) && USE_FIXED
#define fdsp RENAME_FIXED(fdsp)
#endif
int ff_aac_decode_init(AVCodecContext *avctx);
int ff_aac_decode_init_float(AVCodecContext *avctx);
int ff_aac_decode_init_fixed(AVCodecContext *avctx);
int ff_aac_decode_ics(AACDecContext *ac, SingleChannelElement *sce,
GetBitContext *gb, int common_window, int scale_flag);
int ff_aac_decode_tns(AACDecContext *ac, TemporalNoiseShaping *tns,
GetBitContext *gb, const IndividualChannelStream *ics);
int ff_aac_set_default_channel_config(AACDecContext *ac, AVCodecContext *avctx,
uint8_t (*layout_map)[3],
int *tags,
int channel_config);
int ff_aac_output_configure(AACDecContext *ac,
uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
enum OCStatus oc_type, int get_new_frame);
ChannelElement *ff_aac_get_che(AACDecContext *ac, int type, int elem_id);
#endif /* AVCODEC_AAC_AACDEC_H */