annotate aac.h @ 9149:955835e2bc99 libavcodec

Fix the channel allocation bug/assumption (issue 800). Approved by Rob on IRC.
author alexc
date Fri, 06 Mar 2009 19:47:01 +0000
parents 9396eb8808ca
children a5c709bba8e4
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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
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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 libavcodec/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 AVCODEC_AAC_H
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31 #define AVCODEC_AAC_H
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32
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33 #include "libavutil/internal.h"
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34 #include "avcodec.h"
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35 #include "dsputil.h"
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36 #include "mpeg4audio.h"
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37
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38 #include <stdint.h>
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39
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40 #define AAC_INIT_VLC_STATIC(num, size) \
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41 INIT_VLC_STATIC(&vlc_spectral[num], 6, ff_aac_spectral_sizes[num], \
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42 ff_aac_spectral_bits[num], sizeof( ff_aac_spectral_bits[num][0]), sizeof( ff_aac_spectral_bits[num][0]), \
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43 ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), sizeof(ff_aac_spectral_codes[num][0]), \
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44 size);
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45
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46 #define MAX_CHANNELS 64
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47 #define MAX_ELEM_ID 16
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48
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49 #define TNS_MAX_ORDER 20
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50
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51 enum AudioObjectType {
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52 AOT_NULL,
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53 // Support? Name
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54 AOT_AAC_MAIN, ///< Y Main
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55 AOT_AAC_LC, ///< Y Low Complexity
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56 AOT_AAC_SSR, ///< N (code in SoC repo) Scalable Sample Rate
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57 AOT_AAC_LTP, ///< N (code in SoC repo) Long Term Prediction
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58 AOT_SBR, ///< N (in progress) Spectral Band Replication
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59 AOT_AAC_SCALABLE, ///< N Scalable
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60 AOT_TWINVQ, ///< N Twin Vector Quantizer
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61 AOT_CELP, ///< N Code Excited Linear Prediction
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62 AOT_HVXC, ///< N Harmonic Vector eXcitation Coding
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63 AOT_TTSI = 12, ///< N Text-To-Speech Interface
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64 AOT_MAINSYNTH, ///< N Main Synthesis
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65 AOT_WAVESYNTH, ///< N Wavetable Synthesis
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66 AOT_MIDI, ///< N General MIDI
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67 AOT_SAFX, ///< N Algorithmic Synthesis and Audio Effects
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68 AOT_ER_AAC_LC, ///< N Error Resilient Low Complexity
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69 AOT_ER_AAC_LTP = 19, ///< N Error Resilient Long Term Prediction
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70 AOT_ER_AAC_SCALABLE, ///< N Error Resilient Scalable
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71 AOT_ER_TWINVQ, ///< N Error Resilient Twin Vector Quantizer
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72 AOT_ER_BSAC, ///< N Error Resilient Bit-Sliced Arithmetic Coding
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73 AOT_ER_AAC_LD, ///< N Error Resilient Low Delay
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74 AOT_ER_CELP, ///< N Error Resilient Code Excited Linear Prediction
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75 AOT_ER_HVXC, ///< N Error Resilient Harmonic Vector eXcitation Coding
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76 AOT_ER_HILN, ///< N Error Resilient Harmonic and Individual Lines plus Noise
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77 AOT_ER_PARAM, ///< N Error Resilient Parametric
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78 AOT_SSC, ///< N SinuSoidal Coding
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79 };
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80
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81 enum RawDataBlockType {
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82 TYPE_SCE,
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83 TYPE_CPE,
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84 TYPE_CCE,
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85 TYPE_LFE,
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86 TYPE_DSE,
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87 TYPE_PCE,
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88 TYPE_FIL,
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89 TYPE_END,
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90 };
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91
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92 enum ExtensionPayloadID {
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93 EXT_FILL,
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94 EXT_FILL_DATA,
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95 EXT_DATA_ELEMENT,
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96 EXT_DYNAMIC_RANGE = 0xb,
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97 EXT_SBR_DATA = 0xd,
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98 EXT_SBR_DATA_CRC = 0xe,
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99 };
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100
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101 enum WindowSequence {
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102 ONLY_LONG_SEQUENCE,
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103 LONG_START_SEQUENCE,
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104 EIGHT_SHORT_SEQUENCE,
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105 LONG_STOP_SEQUENCE,
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106 };
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107
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108 enum BandType {
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109 ZERO_BT = 0, ///< Scalefactors and spectral data are all zero.
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110 FIRST_PAIR_BT = 5, ///< This and later band types encode two values (rather than four) with one code word.
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111 ESC_BT = 11, ///< Spectral data are coded with an escape sequence.
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112 NOISE_BT = 13, ///< Spectral data are scaled white noise not coded in the bitstream.
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113 INTENSITY_BT2 = 14, ///< Scalefactor data are intensity stereo positions.
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114 INTENSITY_BT = 15, ///< Scalefactor data are intensity stereo positions.
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115 };
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116
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117 #define IS_CODEBOOK_UNSIGNED(x) ((x - 1) & 10)
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118
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119 enum ChannelPosition {
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120 AAC_CHANNEL_FRONT = 1,
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121 AAC_CHANNEL_SIDE = 2,
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122 AAC_CHANNEL_BACK = 3,
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123 AAC_CHANNEL_LFE = 4,
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124 AAC_CHANNEL_CC = 5,
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125 };
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126
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127 /**
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128 * The point during decoding at which channel coupling is applied.
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129 */
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130 enum CouplingPoint {
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131 BEFORE_TNS,
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132 BETWEEN_TNS_AND_IMDCT,
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133 AFTER_IMDCT = 3,
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134 };
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135
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136 /**
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137 * Predictor State
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138 */
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139 typedef struct {
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140 float cor0;
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141 float cor1;
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142 float var0;
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143 float var1;
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144 float r0;
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145 float r1;
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146 } PredictorState;
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147
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148 #define MAX_PREDICTORS 672
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149
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150 /**
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151 * Individual Channel Stream
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152 */
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153 typedef struct {
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154 uint8_t max_sfb; ///< number of scalefactor bands per group
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155 enum WindowSequence window_sequence[2];
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156 uint8_t use_kb_window[2]; ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
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157 int num_window_groups;
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158 uint8_t group_len[8];
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159 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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160 int num_swb; ///< number of scalefactor window bands
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161 int num_windows;
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162 int tns_max_bands;
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163 int predictor_present;
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164 int predictor_initialized;
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165 int predictor_reset_group;
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166 uint8_t prediction_used[41];
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167 } IndividualChannelStream;
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168
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169 /**
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170 * Temporal Noise Shaping
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171 */
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172 typedef struct {
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173 int present;
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174 int n_filt[8];
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175 int length[8][4];
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176 int direction[8][4];
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177 int order[8][4];
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178 float coef[8][4][TNS_MAX_ORDER];
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179 } TemporalNoiseShaping;
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180
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181 /**
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182 * Dynamic Range Control - decoded from the bitstream but not processed further.
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183 */
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184 typedef struct {
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185 int pce_instance_tag; ///< Indicates with which program the DRC info is associated.
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186 int dyn_rng_sgn[17]; ///< DRC sign information; 0 - positive, 1 - negative
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187 int dyn_rng_ctl[17]; ///< DRC magnitude information
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188 int exclude_mask[MAX_CHANNELS]; ///< Channels to be excluded from DRC processing.
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189 int band_incr; ///< Number of DRC bands greater than 1 having DRC info.
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190 int interpolation_scheme; ///< Indicates the interpolation scheme used in the SBR QMF domain.
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191 int band_top[17]; ///< Indicates the top of the i-th DRC band in units of 4 spectral lines.
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192 int prog_ref_level; /**< A reference level for the long-term program audio level for all
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193 * channels combined.
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194 */
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195 } DynamicRangeControl;
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196
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197 typedef struct {
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198 int num_pulse;
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199 int pos[4];
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200 int amp[4];
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201 } Pulse;
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202
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203 /**
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204 * coupling parameters
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205 */
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206 typedef struct {
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207 enum CouplingPoint coupling_point; ///< The point during decoding at which coupling is applied.
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208 int num_coupled; ///< number of target elements
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209 enum RawDataBlockType type[8]; ///< Type of channel element to be coupled - SCE or CPE.
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210 int id_select[8]; ///< element id
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211 int ch_select[8]; /**< [0] shared list of gains; [1] list of gains for right channel;
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212 * [2] list of gains for left channel; [3] lists of gains for both channels
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213 */
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214 float gain[16][120];
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215 } ChannelCoupling;
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216
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217 /**
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218 * Single Channel Element - used for both SCE and LFE elements.
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219 */
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220 typedef struct {
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221 IndividualChannelStream ics;
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222 TemporalNoiseShaping tns;
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223 enum BandType band_type[120]; ///< band types
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224 int band_type_run_end[120]; ///< band type run end points
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225 float sf[120]; ///< scalefactors
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226 DECLARE_ALIGNED_16(float, coeffs[1024]); ///< coefficients for IMDCT
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227 DECLARE_ALIGNED_16(float, saved[512]); ///< overlap
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228 DECLARE_ALIGNED_16(float, ret[1024]); ///< PCM output
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229 PredictorState predictor_state[MAX_PREDICTORS];
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230 } SingleChannelElement;
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231
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232 /**
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233 * channel element - generic struct for SCE/CPE/CCE/LFE
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234 */
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235 typedef struct {
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236 // CPE specific
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237 uint8_t ms_mask[120]; ///< Set if mid/side stereo is used for each scalefactor window band
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238 // shared
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239 SingleChannelElement ch[2];
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240 // CCE specific
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241 ChannelCoupling coup;
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242 } ChannelElement;
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243
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244 /**
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245 * main AAC context
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246 */
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247 typedef struct {
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248 AVCodecContext * avccontext;
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249
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250 MPEG4AudioConfig m4ac;
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251
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252 int is_saved; ///< Set if elements have stored overlap from previous frame.
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253 DynamicRangeControl che_drc;
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254
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255 /**
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256 * @defgroup elements Channel element related data.
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257 * @{
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258 */
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259 enum ChannelPosition che_pos[4][MAX_ELEM_ID]; /**< channel element channel mapping with the
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260 * first index as the first 4 raw data block types
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261 */
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262 ChannelElement * che[4][MAX_ELEM_ID];
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263 ChannelElement * tag_che_map[4][MAX_ELEM_ID];
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264 int tags_mapped;
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265 /** @} */
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266
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267 /**
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268 * @defgroup temporary aligned temporary buffers (We do not want to have these on the stack.)
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269 * @{
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270 */
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271 DECLARE_ALIGNED_16(float, buf_mdct[1024]);
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272 /** @} */
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273
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274 /**
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275 * @defgroup tables Computed / set up during initialization.
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276 * @{
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277 */
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278 MDCTContext mdct;
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279 MDCTContext mdct_small;
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280 DSPContext dsp;
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281 int random_state;
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282 /** @} */
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283
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284 /**
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285 * @defgroup output Members used for output interleaving.
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286 * @{
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287 */
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288 float *output_data[MAX_CHANNELS]; ///< Points to each element's 'ret' buffer (PCM output).
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289 float add_bias; ///< offset for dsp.float_to_int16
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290 float sf_scale; ///< Pre-scale for correct IMDCT and dsp.float_to_int16.
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291 int sf_offset; ///< offset into pow2sf_tab as appropriate for dsp.float_to_int16
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292 /** @} */
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293
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294 DECLARE_ALIGNED(16, float, temp[128]);
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295 } AACContext;
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296
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297 #endif /* AVCODEC_AAC_H */