1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-11-26 19:01:44 +02:00
FFmpeg/libavcodec/aac.h
Robert Swain 9cc04edff9 More OKed hunks of the AAC decoder from SoC
Originally committed as revision 14694 to svn://svn.ffmpeg.org/ffmpeg/trunk
2008-08-11 11:16:06 +00:00

211 lines
7.9 KiB
C

/*
* 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.h
* AAC definitions and structures
* @author Oded Shimon ( ods15 ods15 dyndns org )
* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
*/
#ifndef FFMPEG_AAC_H
#define FFMPEG_AAC_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 IVQUANT_SIZE 1024
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,
};
/**
* Individual Channel Stream
*/
/**
* 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 offset[4];
int amp[4];
} Pulse;
/**
* coupling parameters
*/
typedef struct {
/**
* 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
* @{
*/
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 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
/** @} */
} AACContext;
#endif /* FFMPEG_AAC_H */