annotate aac.h @ 7543:f04ff5a6fb55 libavcodec

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author lorenm
date Tue, 12 Aug 2008 00:27:21 +0000
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children a05954c505ab
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1 /*
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2 * AAC definitions and structures
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3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
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4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
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5 *
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6 * This file is part of FFmpeg.
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7 *
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8 * FFmpeg is free software; you can redistribute it and/or
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9 * modify it under the terms of the GNU Lesser General Public
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10 * License as published by the Free Software Foundation; either
3eec98e5ab8f AAC definitions and structures that have been OKed from the SoC AAC decoder
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11 * version 2.1 of the License, or (at your option) any later version.
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12 *
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13 * FFmpeg is distributed in the hope that it will be useful,
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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16 * Lesser General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU Lesser General Public
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19 * License along with FFmpeg; if not, write to the Free Software
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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21 */
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22
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23 /**
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24 * @file aac.h
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25 * AAC definitions and structures
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26 * @author Oded Shimon ( ods15 ods15 dyndns org )
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27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
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28 */
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29
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30 #ifndef FFMPEG_AAC_H
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31 #define FFMPEG_AAC_H
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32
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33 #include "avcodec.h"
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34 #include "dsputil.h"
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35 #include "mpeg4audio.h"
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36
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37 #include <stdint.h>
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38
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39 #define AAC_INIT_VLC_STATIC(num, size) \
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40 INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
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41 ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
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42 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
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43 size);
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44
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45 #define MAX_CHANNELS 64
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46 #define MAX_ELEM_ID 16
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47
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48 #define IVQUANT_SIZE 1024
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49
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50 enum AudioObjectType {
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51 AOT_NULL,
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52 // Support? Name
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53 AOT_AAC_MAIN, ///< Y Main
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54 AOT_AAC_LC, ///< Y Low Complexity
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55 AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
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56 AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
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57 AOT_SBR, ///< N (in progress) Spectral Band Replication
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58 AOT_AAC_SCALABLE, ///< N Scalable
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59 AOT_TWINVQ, ///< N Twin Vector Quantizer
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60 AOT_CELP, ///< N Code Excited Linear Prediction
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61 AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
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62 AOT_TTSI = 12, ///< N Text-To-Speech Interface
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63 AOT_MAINSYNTH, ///< N Main Synthesis
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64 AOT_WAVESYNTH, ///< N Wavetable Synthesis
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65 AOT_MIDI, ///< N General MIDI
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66 AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
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67 AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
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68 AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
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69 AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
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70 AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
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71 AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
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72 AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
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73 AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
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74 AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
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75 AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
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76 AOT_ER_PARAM, ///< N Error Resilient Parametric
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77 AOT_SSC, ///< N SinuSoidal Coding
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78 };
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79
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80 enum RawDataBlockType {
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81 TYPE_SCE,
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82 TYPE_CPE,
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83 TYPE_CCE,
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84 TYPE_LFE,
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85 TYPE_DSE,
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86 TYPE_PCE,
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87 TYPE_FIL,
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88 TYPE_END,
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89 };
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90
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91 enum ExtensionPayloadID {
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92 EXT_FILL,
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93 EXT_FILL_DATA,
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94 EXT_DATA_ELEMENT,
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95 EXT_DYNAMIC_RANGE = 0xb,
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96 EXT_SBR_DATA = 0xd,
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97 EXT_SBR_DATA_CRC = 0xe,
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98 };
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99
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100 enum WindowSequence {
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101 ONLY_LONG_SEQUENCE,
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102 LONG_START_SEQUENCE,
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103 EIGHT_SHORT_SEQUENCE,
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104 LONG_STOP_SEQUENCE,
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105 };
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106
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107 enum BandType {
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108 ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
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109 FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
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110 ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
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111 NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
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112 INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions.
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113 INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions.
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114 };
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115
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116 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
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117
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118 enum ChannelPosition {
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119 AAC_CHANNEL_FRONT = 1,
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120 AAC_CHANNEL_SIDE = 2,
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121 AAC_CHANNEL_BACK = 3,
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122 AAC_CHANNEL_LFE = 4,
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123 AAC_CHANNEL_CC = 5,
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124 };
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125
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126 /**
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127 * The point during decoding at which channel coupling is applied.
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128 */
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129 enum CouplingPoint {
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130 BEFORE_TNS,
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131 BETWEEN_TNS_AND_IMDCT,
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132 AFTER_IMDCT = 3,
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133 };
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134
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135 /**
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136 * Individual Channel Stream
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137 */
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138
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139 /**
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140 * Dynamic Range Control - decoded from the bitstream but not processed further.
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141 */
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142 typedef struct {
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143 int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
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144 int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
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145 int dyn_rng_ctl[17]; ///< DRC magnitude information
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146 int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
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147 int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
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148 int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
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149 int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
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150 int prog_ref_level; /**< A reference level for the long-term program audio level for all
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151 * channels combined.
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152 */
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153 } DynamicRangeControl;
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154
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155 typedef struct {
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156 int num_pulse;
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157 int start;
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158 int offset[4];
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159 int amp[4];
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160 } Pulse;
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161
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162 /**
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163 * coupling parameters
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164 */
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165 typedef struct {
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166
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167 /**
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168 * main AAC context
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169 */
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170 typedef struct {
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171 AVCodecContext * avccontext;
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172
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173 MPEG4AudioConfig m4ac;
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174
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175 int is_saved; ///< Set if elements have stored overlap from previous frame.
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176 DynamicRangeControl che_drc;
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177
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178 /**
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179 * @defgroup elements
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180 * @{
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181 */
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182 enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
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183 * first index as the first 4 raw data block types
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184 */
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185 ChannelElement * che[4][MAX_ELEM_ID];
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186 /** @} */
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187
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188 /**
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189 * @defgroup tables Computed / set up during initialization.
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190 * @{
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191 */
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192 MDCTContext mdct;
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193 MDCTContext mdct_small;
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194 DSPContext dsp;
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195 int random_state;
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196 /** @} */
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197
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198 /**
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199 * @defgroup output Members used for output interleaving.
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200 * @{
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201 */
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202 float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
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203 float add_bias; ///< offset for dsp.float_to_int16
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204 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
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205 int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
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206 /** @} */
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207
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208 } AACContext;
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209
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210 #endif /* FFMPEG_AAC_H */