annotate aacenc.c @ 8643:1fa3e8a72ca5 libavcodec

Do not allocate RoqTempData on the stack
author vitor
date Sat, 24 Jan 2009 08:15:43 +0000
parents dc1a7a6ec58d
children e9d9d946f213
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1 /*
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2 * AAC encoder
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3 * Copyright (C) 2008 Konstantin Shishkov
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4 *
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5 * This file is part of FFmpeg.
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6 *
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7 * FFmpeg is free software; you can redistribute it and/or
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8 * modify it under the terms of the GNU Lesser General Public
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9 * License as published by the Free Software Foundation; either
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10 * version 2.1 of the License, or (at your option) any later version.
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11 *
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12 * FFmpeg is distributed in the hope that it will be useful,
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 * Lesser General Public License for more details.
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16 *
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17 * You should have received a copy of the GNU Lesser General Public
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18 * License along with FFmpeg; if not, write to the Free Software
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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20 */
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21
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22 /**
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23 * @file aacenc.c
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24 * AAC encoder
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25 */
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26
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27 /***********************************
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28 * TODOs:
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29 * psy model selection with some option
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30 * add sane pulse detection
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31 * add temporal noise shaping
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32 ***********************************/
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33
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34 #include "avcodec.h"
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35 #include "bitstream.h"
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36 #include "dsputil.h"
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37 #include "mpeg4audio.h"
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38
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39 #include "aacpsy.h"
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40 #include "aac.h"
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41 #include "aactab.h"
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42
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43 static const uint8_t swb_size_1024_96[] = {
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44 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8,
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45 12, 12, 12, 12, 12, 16, 16, 24, 28, 36, 44,
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46 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64
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47 };
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48
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49 static const uint8_t swb_size_1024_64[] = {
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50 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8,
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51 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36,
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52 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40
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53 };
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54
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55 static const uint8_t swb_size_1024_48[] = {
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56 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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57 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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58 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32,
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59 96
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60 };
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61
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62 static const uint8_t swb_size_1024_32[] = {
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63 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8,
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64 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28,
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65 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32
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66 };
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67
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68 static const uint8_t swb_size_1024_24[] = {
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69 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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70 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, 28, 28,
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71 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64
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72 };
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73
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74 static const uint8_t swb_size_1024_16[] = {
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75 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8,
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76 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, 28,
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77 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64
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78 };
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79
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80 static const uint8_t swb_size_1024_8[] = {
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81 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12,
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82 16, 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, 28,
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83 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80
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84 };
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85
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86 static const uint8_t * const swb_size_1024[] = {
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87 swb_size_1024_96, swb_size_1024_96, swb_size_1024_64,
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88 swb_size_1024_48, swb_size_1024_48, swb_size_1024_32,
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89 swb_size_1024_24, swb_size_1024_24, swb_size_1024_16,
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90 swb_size_1024_16, swb_size_1024_16, swb_size_1024_8
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91 };
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92
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93 static const uint8_t swb_size_128_96[] = {
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94 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36
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95 };
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96
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97 static const uint8_t swb_size_128_48[] = {
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98 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16
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99 };
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100
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101 static const uint8_t swb_size_128_24[] = {
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102 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20
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103 };
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104
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105 static const uint8_t swb_size_128_16[] = {
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106 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20
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107 };
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108
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109 static const uint8_t swb_size_128_8[] = {
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110 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20
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111 };
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112
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113 static const uint8_t * const swb_size_128[] = {
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114 /* the last entry on the following row is swb_size_128_64 but is a
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115 duplicate of swb_size_128_96 */
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116 swb_size_128_96, swb_size_128_96, swb_size_128_96,
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117 swb_size_128_48, swb_size_128_48, swb_size_128_48,
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118 swb_size_128_24, swb_size_128_24, swb_size_128_16,
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119 swb_size_128_16, swb_size_128_16, swb_size_128_8
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120 };
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121
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122 /** bits needed to code codebook run value for long windows */
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123 static const uint8_t run_value_bits_long[64] = {
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124 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
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125 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 5, 10,
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126 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10,
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127 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 10, 15
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128 };
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129
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130 /** bits needed to code codebook run value for short windows */
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131 static const uint8_t run_value_bits_short[16] = {
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132 3, 3, 3, 3, 3, 3, 3, 6, 6, 6, 6, 6, 6, 6, 6, 9
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133 };
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134
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135 static const uint8_t* const run_value_bits[2] = {
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136 run_value_bits_long, run_value_bits_short
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137 };
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138
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139 /** default channel configurations */
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140 static const uint8_t aac_chan_configs[6][5] = {
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141 {1, TYPE_SCE}, // 1 channel - single channel element
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142 {1, TYPE_CPE}, // 2 channels - channel pair
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143 {2, TYPE_SCE, TYPE_CPE}, // 3 channels - center + stereo
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144 {3, TYPE_SCE, TYPE_CPE, TYPE_SCE}, // 4 channels - front center + stereo + back center
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145 {3, TYPE_SCE, TYPE_CPE, TYPE_CPE}, // 5 channels - front center + stereo + back stereo
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146 {4, TYPE_SCE, TYPE_CPE, TYPE_CPE, TYPE_LFE}, // 6 channels - front center + stereo + back stereo + LFE
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147 };
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148
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149 /**
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150 * structure used in optimal codebook search
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151 */
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152 typedef struct BandCodingPath {
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153 int prev_idx; ///< pointer to the previous path point
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154 int codebook; ///< codebook for coding band run
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155 int bits; ///< number of bit needed to code given number of bands
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156 } BandCodingPath;
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157
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158 /**
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159 * AAC encoder context
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160 */
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161 typedef struct {
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162 PutBitContext pb;
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163 MDCTContext mdct1024; ///< long (1024 samples) frame transform context
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164 MDCTContext mdct128; ///< short (128 samples) frame transform context
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165 DSPContext dsp;
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166 DECLARE_ALIGNED_16(FFTSample, output[2048]); ///< temporary buffer for MDCT input coefficients
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167 int16_t* samples; ///< saved preprocessed input
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168
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169 int samplerate_index; ///< MPEG-4 samplerate index
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170
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171 ChannelElement *cpe; ///< channel elements
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172 AACPsyContext psy; ///< psychoacoustic model context
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173 int last_frame;
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174 } AACEncContext;
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175
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176 /**
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177 * Make AAC audio config object.
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178 * @see 1.6.2.1 "Syntax - AudioSpecificConfig"
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179 */
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180 static void put_audio_specific_config(AVCodecContext *avctx)
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181 {
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182 PutBitContext pb;
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183 AACEncContext *s = avctx->priv_data;
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184
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185 init_put_bits(&pb, avctx->extradata, avctx->extradata_size*8);
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186 put_bits(&pb, 5, 2); //object type - AAC-LC
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187 put_bits(&pb, 4, s->samplerate_index); //sample rate index
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188 put_bits(&pb, 4, avctx->channels);
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189 //GASpecificConfig
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190 put_bits(&pb, 1, 0); //frame length - 1024 samples
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191 put_bits(&pb, 1, 0); //does not depend on core coder
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192 put_bits(&pb, 1, 0); //is not extension
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193 flush_put_bits(&pb);
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194 }
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195
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196 static av_cold int aac_encode_init(AVCodecContext *avctx)
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197 {
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198 AACEncContext *s = avctx->priv_data;
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199 int i;
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200
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201 avctx->frame_size = 1024;
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202
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203 for(i = 0; i < 16; i++)
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204 if(avctx->sample_rate == ff_mpeg4audio_sample_rates[i])
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205 break;
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206 if(i == 16){
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207 av_log(avctx, AV_LOG_ERROR, "Unsupported sample rate %d\n", avctx->sample_rate);
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208 return -1;
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209 }
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210 if(avctx->channels > 6){
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211 av_log(avctx, AV_LOG_ERROR, "Unsupported number of channels: %d\n", avctx->channels);
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212 return -1;
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213 }
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214 s->samplerate_index = i;
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215
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216 dsputil_init(&s->dsp, avctx);
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217 ff_mdct_init(&s->mdct1024, 11, 0);
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218 ff_mdct_init(&s->mdct128, 8, 0);
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219 // window init
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220 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
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221 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
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222 ff_sine_window_init(ff_sine_1024, 1024);
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223 ff_sine_window_init(ff_sine_128, 128);
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224
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225 s->samples = av_malloc(2 * 1024 * avctx->channels * sizeof(s->samples[0]));
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226 s->cpe = av_mallocz(sizeof(ChannelElement) * aac_chan_configs[avctx->channels-1][0]);
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227 if(ff_aac_psy_init(&s->psy, avctx, AAC_PSY_3GPP,
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228 aac_chan_configs[avctx->channels-1][0], 0,
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229 swb_size_1024[i], ff_aac_num_swb_1024[i], swb_size_128[i], ff_aac_num_swb_128[i]) < 0){
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230 av_log(avctx, AV_LOG_ERROR, "Cannot initialize selected model.\n");
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231 return -1;
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232 }
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233 avctx->extradata = av_malloc(2);
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234 avctx->extradata_size = 2;
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235 put_audio_specific_config(avctx);
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236 return 0;
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237 }
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238
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239 /**
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240 * Encode ics_info element.
58f6bb760994 Okayed parts of AAC encoder
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241 * @see Table 4.6 (syntax of ics_info)
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242 */
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243 static void put_ics_info(AACEncContext *s, IndividualChannelStream *info)
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244 {
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245 int i;
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246
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247 put_bits(&s->pb, 1, 0); // ics_reserved bit
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248 put_bits(&s->pb, 2, info->window_sequence[0]);
58f6bb760994 Okayed parts of AAC encoder
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249 put_bits(&s->pb, 1, info->use_kb_window[0]);
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250 if(info->window_sequence[0] != EIGHT_SHORT_SEQUENCE){
58f6bb760994 Okayed parts of AAC encoder
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251 put_bits(&s->pb, 6, info->max_sfb);
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252 put_bits(&s->pb, 1, 0); // no prediction
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253 }else{
58f6bb760994 Okayed parts of AAC encoder
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254 put_bits(&s->pb, 4, info->max_sfb);
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255 for(i = 1; i < info->num_windows; i++)
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256 put_bits(&s->pb, 1, info->group_len[i]);
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257 }
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258 }
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259
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260 /**
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261 * Calculate the number of bits needed to code all coefficient signs in current band.
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262 */
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263 static int calculate_band_sign_bits(AACEncContext *s, SingleChannelElement *sce,
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264 int group_len, int start, int size)
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265 {
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266 int bits = 0;
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267 int i, w;
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268 for(w = 0; w < group_len; w++){
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269 for(i = 0; i < size; i++){
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270 if(sce->icoefs[start + i])
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271 bits++;
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272 }
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273 start += 128;
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274 }
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275 return bits;
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276 }
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277
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278 /**
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279 * Encode pulse data.
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280 */
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281 static void encode_pulses(AACEncContext *s, Pulse *pulse)
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282 {
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283 int i;
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284
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285 put_bits(&s->pb, 1, !!pulse->num_pulse);
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286 if(!pulse->num_pulse) return;
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287
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288 put_bits(&s->pb, 2, pulse->num_pulse - 1);
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289 put_bits(&s->pb, 6, pulse->start);
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290 for(i = 0; i < pulse->num_pulse; i++){
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291 put_bits(&s->pb, 5, pulse->pos[i]);
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292 put_bits(&s->pb, 4, pulse->amp[i]);
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293 }
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294 }
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295
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296 /**
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297 * Encode spectral coefficients processed by psychoacoustic model.
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298 */
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299 static void encode_spectral_coeffs(AACEncContext *s, SingleChannelElement *sce)
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300 {
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301 int start, i, w, w2, wg;
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302
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303 w = 0;
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304 for(wg = 0; wg < sce->ics.num_window_groups; wg++){
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305 start = 0;
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306 for(i = 0; i < sce->ics.max_sfb; i++){
1c01b74dc78c Do not pass unnecessary arguments to AAC encoder functions
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307 if(sce->zeroes[w*16 + i]){
1c01b74dc78c Do not pass unnecessary arguments to AAC encoder functions
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308 start += sce->ics.swb_sizes[i];
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309 continue;
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310 }
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311 for(w2 = w; w2 < w + sce->ics.group_len[wg]; w2++){
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312 encode_band_coeffs(s, sce, start + w2*128,
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313 sce->ics.swb_sizes[i],
1c01b74dc78c Do not pass unnecessary arguments to AAC encoder functions
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314 sce->band_type[w*16 + i]);
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315 }
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316 start += sce->ics.swb_sizes[i];
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317 }
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318 w += sce->ics.group_len[wg];
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319 }
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320 }
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321
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322 /**
7571
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323 * Write some auxiliary information about the created AAC file.
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324 */
58f6bb760994 Okayed parts of AAC encoder
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325 static void put_bitstream_info(AVCodecContext *avctx, AACEncContext *s, const char *name)
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326 {
58f6bb760994 Okayed parts of AAC encoder
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327 int i, namelen, padbits;
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328
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329 namelen = strlen(name) + 2;
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330 put_bits(&s->pb, 3, TYPE_FIL);
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331 put_bits(&s->pb, 4, FFMIN(namelen, 15));
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332 if(namelen >= 15)
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333 put_bits(&s->pb, 8, namelen - 16);
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334 put_bits(&s->pb, 4, 0); //extension type - filler
58f6bb760994 Okayed parts of AAC encoder
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335 padbits = 8 - (put_bits_count(&s->pb) & 7);
58f6bb760994 Okayed parts of AAC encoder
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336 align_put_bits(&s->pb);
58f6bb760994 Okayed parts of AAC encoder
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337 for(i = 0; i < namelen - 2; i++)
58f6bb760994 Okayed parts of AAC encoder
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338 put_bits(&s->pb, 8, name[i]);
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339 put_bits(&s->pb, 12 - padbits, 0);
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340 }
58f6bb760994 Okayed parts of AAC encoder
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341
58f6bb760994 Okayed parts of AAC encoder
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342 static av_cold int aac_encode_end(AVCodecContext *avctx)
58f6bb760994 Okayed parts of AAC encoder
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343 {
58f6bb760994 Okayed parts of AAC encoder
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344 AACEncContext *s = avctx->priv_data;
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345
58f6bb760994 Okayed parts of AAC encoder
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346 ff_mdct_end(&s->mdct1024);
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347 ff_mdct_end(&s->mdct128);
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348 ff_aac_psy_end(&s->psy);
58f6bb760994 Okayed parts of AAC encoder
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349 av_freep(&s->samples);
58f6bb760994 Okayed parts of AAC encoder
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350 av_freep(&s->cpe);
58f6bb760994 Okayed parts of AAC encoder
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351 return 0;
58f6bb760994 Okayed parts of AAC encoder
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352 }
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353
58f6bb760994 Okayed parts of AAC encoder
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354 AVCodec aac_encoder = {
58f6bb760994 Okayed parts of AAC encoder
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355 "aac",
58f6bb760994 Okayed parts of AAC encoder
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356 CODEC_TYPE_AUDIO,
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357 CODEC_ID_AAC,
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358 sizeof(AACEncContext),
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359 aac_encode_init,
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360 aac_encode_frame,
58f6bb760994 Okayed parts of AAC encoder
kostya
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diff changeset
361 aac_encode_end,
58f6bb760994 Okayed parts of AAC encoder
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362 .capabilities = CODEC_CAP_SMALL_LAST_FRAME | CODEC_CAP_DELAY,
58f6bb760994 Okayed parts of AAC encoder
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363 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
58f6bb760994 Okayed parts of AAC encoder
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364 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
58f6bb760994 Okayed parts of AAC encoder
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parents:
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365 };