RepoMirrors
/
ffmpeg
mirror of https://git.ffmpeg.org/ffmpeg.git
1
0
Fork 0
Code Issues Projects Releases Wiki Activity
ffmpeg/libavcodec/aac.h

296 lines
11 KiB
C
Raw Blame History

/*
* 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 libavcodec/aac.h
* 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/internal.h"
#include "avcodec.h"
#include "dsputil.h"
#include "mpeg4audio.h"
#include <stdint.h>
#define AAC_INIT_VLC_STATIC(num, size) \
INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
size);
#define MAX_CHANNELS 64
#define MAX_ELEM_ID 16
#define TNS_MAX_ORDER 20
enum AudioObjectType {
AOT_NULL,
// Support? Name
AOT_AAC_MAIN, ///< Y Main
AOT_AAC_LC, ///< Y Low Complexity
AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
AOT_SBR, ///< N (in progress) Spectral Band Replication
AOT_AAC_SCALABLE, ///< N Scalable
AOT_TWINVQ, ///< N Twin Vector Quantizer
AOT_CELP, ///< N Code Excited Linear Prediction
AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
AOT_TTSI = 12, ///< N Text-To-Speech Interface
AOT_MAINSYNTH, ///< N Main Synthesis
AOT_WAVESYNTH, ///< N Wavetable Synthesis
AOT_MIDI, ///< N General MIDI
AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
AOT_ER_PARAM, ///< N Error Resilient Parametric
AOT_SSC, ///< N SinuSoidal Coding
};
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_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,
};
/**
* Predictor State
*/
typedef struct {
float cor0;
float cor1;
float var0;
float var1;
float r0;
float r1;
} PredictorState;
#define MAX_PREDICTORS 672
/**
* 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];
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];
} 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 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;
enum BandType band_type[120]; ///< band types
int band_type_run_end[120]; ///< band type run end points
float sf[120]; ///< scalefactors
DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
PredictorState predictor_state[MAX_PREDICTORS];
} SingleChannelElement;
/**
* channel element - generic struct for SCE/CPE/CCE/LFE
*/
typedef struct {
// CPE specific
uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
// shared
SingleChannelElement ch[2];
// CCE specific
ChannelCoupling coup;
} ChannelElement;
/**
* main AAC context
*/
typedef struct {
AVCodecContext * avccontext;
MPEG4AudioConfig m4ac;
int is_saved; ///< Set if elements have stored overlap from previous frame.
DynamicRangeControl che_drc;
/**
* @defgroup elements Channel element related data.
* @{
*/
enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
* first index as the first 4 raw data block types
*/
ChannelElement * che[4][MAX_ELEM_ID];
/** @} */
/**
* @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
* @{
*/
DECLARE_ALIGNED_16(float, buf_mdct[1024]);
/** @} */
/**
* @defgroup tables Computed / set up during initialization.
* @{
*/
MDCTContext mdct;
MDCTContext mdct_small;
DSPContext dsp;
int random_state;
/** @} */
/**
* @defgroup output Members used for output interleaving.
* @{
*/
float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
float add_bias; ///< offset for dsp.float_to_int16
float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
/** @} */
DECLARE_ALIGNED(16, float, temp[128]);
} AACContext;
#endif /* AVCODEC_AAC_H */
Reference in New Issue View Git Blame Copy Permalink