annotate g726.c @ 11710:8e3a3bc4fb24 libavcodec

Factorize READ_FLIP_SIGN() optimization out
author michael
date Tue, 11 May 2010 21:20:33 +0000
parents 8a4984c5cacc
children dde20597f15e
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1 /*
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2 * G.726 ADPCM audio codec
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3 * Copyright (c) 2004 Roman Shaposhnik
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4 *
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5 * This is a very straightforward rendition of the G.726
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6 * Section 4 "Computational Details".
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7 *
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8 * This file is part of FFmpeg.
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9 *
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10 * FFmpeg is free software; you can redistribute it and/or
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11 * modify it under the terms of the GNU Lesser General Public
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12 * License as published by the Free Software Foundation; either
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13 * version 2.1 of the License, or (at your option) any later version.
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14 *
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15 * FFmpeg is distributed in the hope that it will be useful,
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16 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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18 * Lesser General Public License for more details.
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19 *
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20 * You should have received a copy of the GNU Lesser General Public
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21 * License along with FFmpeg; if not, write to the Free Software
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22 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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23 */
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24 #include <limits.h>
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25 #include "avcodec.h"
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26 #include "get_bits.h"
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27 #include "put_bits.h"
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28
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29 /**
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30 * G.726 11bit float.
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31 * G.726 Standard uses rather odd 11bit floating point arithmentic for
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32 * numerous occasions. It's a mistery to me why they did it this way
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33 * instead of simply using 32bit integer arithmetic.
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34 */
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35 typedef struct Float11 {
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36 uint8_t sign; /**< 1bit sign */
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37 uint8_t exp; /**< 4bit exponent */
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38 uint8_t mant; /**< 6bit mantissa */
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39 } Float11;
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40
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41 static inline Float11* i2f(int i, Float11* f)
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42 {
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43 f->sign = (i < 0);
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44 if (f->sign)
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45 i = -i;
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46 f->exp = av_log2_16bit(i) + !!i;
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47 f->mant = i? (i<<6) >> f->exp : 1<<5;
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48 return f;
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49 }
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50
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51 static inline int16_t mult(Float11* f1, Float11* f2)
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52 {
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53 int res, exp;
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54
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55 exp = f1->exp + f2->exp;
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56 res = (((f1->mant * f2->mant) + 0x30) >> 4);
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57 res = exp > 19 ? res << (exp - 19) : res >> (19 - exp);
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58 return (f1->sign ^ f2->sign) ? -res : res;
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59 }
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60
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61 static inline int sgn(int value)
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62 {
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63 return (value < 0) ? -1 : 1;
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64 }
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65
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66 typedef struct G726Tables {
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67 const int* quant; /**< quantization table */
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68 const int16_t* iquant; /**< inverse quantization table */
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69 const int16_t* W; /**< special table #1 ;-) */
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70 const uint8_t* F; /**< special table #2 */
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71 } G726Tables;
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72
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73 typedef struct G726Context {
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74 G726Tables tbls; /**< static tables needed for computation */
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75
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76 Float11 sr[2]; /**< prev. reconstructed samples */
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77 Float11 dq[6]; /**< prev. difference */
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78 int a[2]; /**< second order predictor coeffs */
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79 int b[6]; /**< sixth order predictor coeffs */
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80 int pk[2]; /**< signs of prev. 2 sez + dq */
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81
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82 int ap; /**< scale factor control */
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83 int yu; /**< fast scale factor */
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84 int yl; /**< slow scale factor */
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85 int dms; /**< short average magnitude of F[i] */
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86 int dml; /**< long average magnitude of F[i] */
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87 int td; /**< tone detect */
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88
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89 int se; /**< estimated signal for the next iteration */
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90 int sez; /**< estimated second order prediction */
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91 int y; /**< quantizer scaling factor for the next iteration */
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92 int code_size;
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93 } G726Context;
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94
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95 static const int quant_tbl16[] = /**< 16kbit/s 2bits per sample */
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96 { 260, INT_MAX };
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97 static const int16_t iquant_tbl16[] =
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98 { 116, 365, 365, 116 };
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99 static const int16_t W_tbl16[] =
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100 { -22, 439, 439, -22 };
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101 static const uint8_t F_tbl16[] =
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102 { 0, 7, 7, 0 };
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103
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104 static const int quant_tbl24[] = /**< 24kbit/s 3bits per sample */
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105 { 7, 217, 330, INT_MAX };
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106 static const int16_t iquant_tbl24[] =
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107 { INT16_MIN, 135, 273, 373, 373, 273, 135, INT16_MIN };
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108 static const int16_t W_tbl24[] =
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109 { -4, 30, 137, 582, 582, 137, 30, -4 };
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110 static const uint8_t F_tbl24[] =
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111 { 0, 1, 2, 7, 7, 2, 1, 0 };
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112
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113 static const int quant_tbl32[] = /**< 32kbit/s 4bits per sample */
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114 { -125, 79, 177, 245, 299, 348, 399, INT_MAX };
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115 static const int16_t iquant_tbl32[] =
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116 { INT16_MIN, 4, 135, 213, 273, 323, 373, 425,
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117 425, 373, 323, 273, 213, 135, 4, INT16_MIN };
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118 static const int16_t W_tbl32[] =
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119 { -12, 18, 41, 64, 112, 198, 355, 1122,
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120 1122, 355, 198, 112, 64, 41, 18, -12};
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121 static const uint8_t F_tbl32[] =
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122 { 0, 0, 0, 1, 1, 1, 3, 7, 7, 3, 1, 1, 1, 0, 0, 0 };
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123
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124 static const int quant_tbl40[] = /**< 40kbit/s 5bits per sample */
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125 { -122, -16, 67, 138, 197, 249, 297, 338,
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126 377, 412, 444, 474, 501, 527, 552, INT_MAX };
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127 static const int16_t iquant_tbl40[] =
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128 { INT16_MIN, -66, 28, 104, 169, 224, 274, 318,
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129 358, 395, 429, 459, 488, 514, 539, 566,
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130 566, 539, 514, 488, 459, 429, 395, 358,
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131 318, 274, 224, 169, 104, 28, -66, INT16_MIN };
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132 static const int16_t W_tbl40[] =
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133 { 14, 14, 24, 39, 40, 41, 58, 100,
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134 141, 179, 219, 280, 358, 440, 529, 696,
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135 696, 529, 440, 358, 280, 219, 179, 141,
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136 100, 58, 41, 40, 39, 24, 14, 14 };
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137 static const uint8_t F_tbl40[] =
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138 { 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 2, 3, 4, 5, 6, 6,
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139 6, 6, 5, 4, 3, 2, 1, 1, 1, 1, 1, 0, 0, 0, 0, 0 };
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140
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141 static const G726Tables G726Tables_pool[] =
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142 {{ quant_tbl16, iquant_tbl16, W_tbl16, F_tbl16 },
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143 { quant_tbl24, iquant_tbl24, W_tbl24, F_tbl24 },
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144 { quant_tbl32, iquant_tbl32, W_tbl32, F_tbl32 },
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145 { quant_tbl40, iquant_tbl40, W_tbl40, F_tbl40 }};
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146
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147
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148 /**
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149 * Para 4.2.2 page 18: Adaptive quantizer.
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150 */
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151 static inline uint8_t quant(G726Context* c, int d)
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152 {
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153 int sign, exp, i, dln;
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154
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155 sign = i = 0;
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156 if (d < 0) {
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157 sign = 1;
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158 d = -d;
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159 }
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160 exp = av_log2_16bit(d);
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161 dln = ((exp<<7) + (((d<<7)>>exp)&0x7f)) - (c->y>>2);
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162
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163 while (c->tbls.quant[i] < INT_MAX && c->tbls.quant[i] < dln)
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164 ++i;
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165
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166 if (sign)
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167 i = ~i;
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168 if (c->code_size != 2 && i == 0) /* I'm not sure this is a good idea */
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169 i = 0xff;
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170
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171 return i;
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172 }
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173
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174 /**
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175 * Para 4.2.3 page 22: Inverse adaptive quantizer.
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176 */
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177 static inline int16_t inverse_quant(G726Context* c, int i)
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178 {
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179 int dql, dex, dqt;
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180
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181 dql = c->tbls.iquant[i] + (c->y >> 2);
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182 dex = (dql>>7) & 0xf; /* 4bit exponent */
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183 dqt = (1<<7) + (dql & 0x7f); /* log2 -> linear */
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184 return (dql < 0) ? 0 : ((dqt<<dex) >> 7);
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185 }
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186
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187 static int16_t g726_decode(G726Context* c, int I)
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188 {
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189 int dq, re_signal, pk0, fa1, i, tr, ylint, ylfrac, thr2, al, dq0;
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190 Float11 f;
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191 int I_sig= I >> (c->code_size - 1);
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192
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193 dq = inverse_quant(c, I);
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194
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195 /* Transition detect */
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196 ylint = (c->yl >> 15);
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197 ylfrac = (c->yl >> 10) & 0x1f;
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198 thr2 = (ylint > 9) ? 0x1f << 10 : (0x20 + ylfrac) << ylint;
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199 tr= (c->td == 1 && dq > ((3*thr2)>>2));
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200
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201 if (I_sig) /* get the sign */
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202 dq = -dq;
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203 re_signal = c->se + dq;
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204
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205 /* Update second order predictor coefficient A2 and A1 */
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206 pk0 = (c->sez + dq) ? sgn(c->sez + dq) : 0;
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207 dq0 = dq ? sgn(dq) : 0;
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208 if (tr) {
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209 c->a[0] = 0;
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210 c->a[1] = 0;
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211 for (i=0; i<6; i++)
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212 c->b[i] = 0;
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213 } else {
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214 /* This is a bit crazy, but it really is +255 not +256 */
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215 fa1 = av_clip((-c->a[0]*c->pk[0]*pk0)>>5, -256, 255);
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216
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217 c->a[1] += 128*pk0*c->pk[1] + fa1 - (c->a[1]>>7);
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218 c->a[1] = av_clip(c->a[1], -12288, 12288);
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219 c->a[0] += 64*3*pk0*c->pk[0] - (c->a[0] >> 8);
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220 c->a[0] = av_clip(c->a[0], -(15360 - c->a[1]), 15360 - c->a[1]);
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221
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222 for (i=0; i<6; i++)
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223 c->b[i] += 128*dq0*sgn(-c->dq[i].sign) - (c->b[i]>>8);
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224 }
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225
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226 /* Update Dq and Sr and Pk */
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227 c->pk[1] = c->pk[0];
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228 c->pk[0] = pk0 ? pk0 : 1;
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229 c->sr[1] = c->sr[0];
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230 i2f(re_signal, &c->sr[0]);
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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231 for (i=5; i>0; i--)
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232 c->dq[i] = c->dq[i-1];
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233 i2f(dq, &c->dq[0]);
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234 c->dq[0].sign = I_sig; /* Isn't it crazy ?!?! */
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235
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236 c->td = c->a[1] < -11776;
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237
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238 /* Update Ap */
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239 c->dms += (c->tbls.F[I]<<4) + ((- c->dms) >> 5);
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240 c->dml += (c->tbls.F[I]<<4) + ((- c->dml) >> 7);
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241 if (tr)
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242 c->ap = 256;
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243 else {
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244 c->ap += (-c->ap) >> 4;
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245 if (c->y <= 1535 || c->td || abs((c->dms << 2) - c->dml) >= (c->dml >> 3))
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246 c->ap += 0x20;
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247 }
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248
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249 /* Update Yu and Yl */
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250 c->yu = av_clip(c->y + c->tbls.W[I] + ((-c->y)>>5), 544, 5120);
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251 c->yl += c->yu + ((-c->yl)>>6);
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252
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253 /* Next iteration for Y */
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254 al = (c->ap >= 256) ? 1<<6 : c->ap >> 2;
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255 c->y = (c->yl + (c->yu - (c->yl>>6))*al) >> 6;
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256
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257 /* Next iteration for SE and SEZ */
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258 c->se = 0;
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259 for (i=0; i<6; i++)
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260 c->se += mult(i2f(c->b[i] >> 2, &f), &c->dq[i]);
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261 c->sez = c->se >> 1;
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262 for (i=0; i<2; i++)
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263 c->se += mult(i2f(c->a[i] >> 2, &f), &c->sr[i]);
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264 c->se >>= 1;
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265
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266 return av_clip(re_signal << 2, -0xffff, 0xffff);
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267 }
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268
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269 static av_cold int g726_reset(G726Context* c, int index)
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270 {
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271 int i;
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272
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273 c->tbls = G726Tables_pool[index];
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274 for (i=0; i<2; i++) {
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275 c->sr[i].mant = 1<<5;
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276 c->pk[i] = 1;
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277 }
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278 for (i=0; i<6; i++) {
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279 c->dq[i].mant = 1<<5;
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280 }
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281 c->yu = 544;
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282 c->yl = 34816;
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283
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284 c->y = 544;
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285
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286 return 0;
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287 }
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288
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289 #if CONFIG_ADPCM_G726_ENCODER
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290 static int16_t g726_encode(G726Context* c, int16_t sig)
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291 {
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292 uint8_t i;
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293
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294 i = quant(c, sig/4 - c->se) & ((1<<c->code_size) - 1);
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295 g726_decode(c, i);
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296 return i;
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297 }
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298 #endif
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299
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300 /* Interfacing to the libavcodec */
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301
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302 static av_cold int g726_init(AVCodecContext * avctx)
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303 {
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304 G726Context* c = avctx->priv_data;
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305 unsigned int index;
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306
a570a1e80400 Prevent a division by 0 in the g726 decoder when the configured samplerate is 0.
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307 if (avctx->sample_rate <= 0) {
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308 av_log(avctx, AV_LOG_ERROR, "Samplerate is invalid\n");
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309 return -1;
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310 }
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311
a570a1e80400 Prevent a division by 0 in the g726 decoder when the configured samplerate is 0.
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312 index = (avctx->bit_rate + avctx->sample_rate/2) / avctx->sample_rate - 2;
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313
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314 if (avctx->bit_rate % avctx->sample_rate && avctx->codec->encode) {
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315 av_log(avctx, AV_LOG_ERROR, "Bitrate - Samplerate combination is invalid\n");
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316 return -1;
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317 }
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318 if(avctx->channels != 1){
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319 av_log(avctx, AV_LOG_ERROR, "Only mono is supported\n");
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320 return -1;
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321 }
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322 if(index>3){
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323 av_log(avctx, AV_LOG_ERROR, "Unsupported number of bits %d\n", index+2);
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324 return -1;
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325 }
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326 g726_reset(c, index);
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327 c->code_size = index+2;
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328
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329 avctx->coded_frame = avcodec_alloc_frame();
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330 if (!avctx->coded_frame)
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331 return AVERROR(ENOMEM);
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332 avctx->coded_frame->key_frame = 1;
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333
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334 if (avctx->codec->decode)
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335 avctx->sample_fmt = SAMPLE_FMT_S16;
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336
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337 return 0;
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338 }
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339
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340 static av_cold int g726_close(AVCodecContext *avctx)
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341 {
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342 av_freep(&avctx->coded_frame);
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343 return 0;
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344 }
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345
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346 #if CONFIG_ADPCM_G726_ENCODER
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347 static int g726_encode_frame(AVCodecContext *avctx,
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348 uint8_t *dst, int buf_size, void *data)
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349 {
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350 G726Context *c = avctx->priv_data;
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351 short *samples = data;
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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352 PutBitContext pb;
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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353
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354 init_put_bits(&pb, dst, 1024*1024);
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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355
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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356 for (; buf_size; buf_size--)
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357 put_bits(&pb, c->code_size, g726_encode(c, *samples++));
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358
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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359 flush_put_bits(&pb);
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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360
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361 return put_bits_count(&pb)>>3;
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362 }
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363 #endif
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364
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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365 static int g726_decode_frame(AVCodecContext *avctx,
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366 void *data, int *data_size,
9355
54bc8a2727b0 Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
rbultje
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367 AVPacket *avpkt)
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368 {
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rbultje
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369 const uint8_t *buf = avpkt->data;
54bc8a2727b0 Implement avcodec_decode_video2(), _audio3() and _subtitle2() which takes an
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diff changeset
370 int buf_size = avpkt->size;
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371 G726Context *c = avctx->priv_data;
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372 short *samples = data;
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373 GetBitContext gb;
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374
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375 init_get_bits(&gb, buf, buf_size * 8);
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376
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5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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377 while (get_bits_count(&gb) + c->code_size <= buf_size*8)
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fb6038ffd2a9 Get rid of the redundant AVG726Context.
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378 *samples++ = g726_decode(c, get_bits(&gb, c->code_size));
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379
7061
3e51aa540377 Remove the truncated bitstream handling from our g726 decoder.
michael
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diff changeset
380 if(buf_size*8 != get_bits_count(&gb))
3e51aa540377 Remove the truncated bitstream handling from our g726 decoder.
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381 av_log(avctx, AV_LOG_ERROR, "Frame invalidly split, missing parser?\n");
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382
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383 *data_size = (uint8_t*)samples - (uint8_t*)data;
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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384 return buf_size;
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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385 }
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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386
8590
7a463923ecd1 Change semantic of CONFIG_*, HAVE_* and ARCH_*.
aurel
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diff changeset
387 #if CONFIG_ADPCM_G726_ENCODER
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5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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388 AVCodec adpcm_g726_encoder = {
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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389 "g726",
11560
8a4984c5cacc Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents: 10145
diff changeset
390 AVMEDIA_TYPE_AUDIO,
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5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
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diff changeset
391 CODEC_ID_ADPCM_G726,
7062
fb6038ffd2a9 Get rid of the redundant AVG726Context.
michael
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diff changeset
392 sizeof(G726Context),
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
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diff changeset
393 g726_init,
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
394 g726_encode_frame,
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fac680cf3008 * gotta setup coded_frame for encoding. avcodec.h says that for decoding
romansh
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diff changeset
395 g726_close,
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
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diff changeset
396 NULL,
10145
7955db355703 Make sample_fmts and channel_layouts compound literals const to reduce size of
reimar
parents: 9428
diff changeset
397 .sample_fmts = (const enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
7040
e943e1409077 Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
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diff changeset
398 .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
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diff changeset
399 };
7782
6efb15a24e91 Replace generic CONFIG_ENCODERS preprocessor conditionals by more specific
diego
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diff changeset
400 #endif
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
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diff changeset
401
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
402 AVCodec adpcm_g726_decoder = {
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
403 "g726",
11560
8a4984c5cacc Define AVMediaType enum, and use it instead of enum CodecType, which
stefano
parents: 10145
diff changeset
404 AVMEDIA_TYPE_AUDIO,
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
405 CODEC_ID_ADPCM_G726,
7062
fb6038ffd2a9 Get rid of the redundant AVG726Context.
michael
parents: 7061
diff changeset
406 sizeof(G726Context),
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
407 g726_init,
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
408 NULL,
1797
fac680cf3008 * gotta setup coded_frame for encoding. avcodec.h says that for decoding
romansh
parents: 1792
diff changeset
409 g726_close,
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
410 g726_decode_frame,
7040
e943e1409077 Make AVCodec long_names definition conditional depending on CONFIG_SMALL.
stefano
parents: 6710
diff changeset
411 .long_name = NULL_IF_CONFIG_SMALL("G.726 ADPCM"),
1791
5e5c3d4a1e82 * Initial implementation of the G.726 ADPCM audio codec.
romansh
parents:
diff changeset
412 };