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d0ee50216d
the EIGHT_SHORT window type. Fixes issue 664. Patch by Alex Converse (alex converse gmail com) Originally committed as revision 15776 to svn://svn.ffmpeg.org/ffmpeg/trunk
275 lines
10 KiB
C
275 lines
10 KiB
C
/*
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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 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 AudioObjectType {
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AOT_NULL,
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// Support? Name
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AOT_AAC_MAIN, ///< Y Main
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AOT_AAC_LC, ///< Y Low Complexity
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AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
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AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
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AOT_SBR, ///< N (in progress) Spectral Band Replication
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AOT_AAC_SCALABLE, ///< N Scalable
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AOT_TWINVQ, ///< N Twin Vector Quantizer
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AOT_CELP, ///< N Code Excited Linear Prediction
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AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
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AOT_TTSI = 12, ///< N Text-To-Speech Interface
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AOT_MAINSYNTH, ///< N Main Synthesis
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AOT_WAVESYNTH, ///< N Wavetable Synthesis
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AOT_MIDI, ///< N General MIDI
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AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
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AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
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AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
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AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
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AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
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AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
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AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
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AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
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AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
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AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
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AOT_ER_PARAM, ///< N Error Resilient Parametric
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AOT_SSC, ///< N SinuSoidal Coding
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};
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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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* 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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} 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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} 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
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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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/** @} */
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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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} AACContext;
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#endif /* AVCODEC_AAC_H */
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