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1 /* |
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2 SDL - Simple DirectMedia Layer |
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3 Copyright (C) 1997-2006 Sam Lantinga |
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4 |
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5 This library is free software; you can redistribute it and/or |
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6 modify it under the terms of the GNU Lesser General Public |
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7 License as published by the Free Software Foundation; either |
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8 version 2.1 of the License, or (at your option) any later version. |
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9 |
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10 This library is distributed in the hope that it will be useful, |
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11 but WITHOUT ANY WARRANTY; without even the implied warranty of |
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12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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13 Lesser General Public License for more details. |
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14 |
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15 You should have received a copy of the GNU Lesser General Public |
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16 License along with this library; if not, write to the Free Software |
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17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA |
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18 |
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19 Sam Lantinga |
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20 slouken@libsdl.org |
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21 */ |
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22 #include "SDL_config.h" |
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23 |
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24 /* Functions for audio drivers to perform runtime conversion of audio format */ |
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25 |
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26 #include "SDL_audio.h" |
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27 |
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28 |
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29 /* Effectively mix right and left channels into a single channel */ |
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30 void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format) |
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31 { |
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32 int i; |
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33 Sint32 sample; |
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34 |
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35 #ifdef DEBUG_CONVERT |
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36 fprintf(stderr, "Converting to mono\n"); |
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37 #endif |
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38 switch (format&0x8018) { |
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39 |
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40 case AUDIO_U8: { |
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41 Uint8 *src, *dst; |
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42 |
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43 src = cvt->buf; |
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44 dst = cvt->buf; |
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45 for ( i=cvt->len_cvt/2; i; --i ) { |
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46 sample = src[0] + src[1]; |
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47 *dst = (Uint8)(sample / 2); |
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48 src += 2; |
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49 dst += 1; |
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50 } |
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51 } |
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52 break; |
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53 |
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54 case AUDIO_S8: { |
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55 Sint8 *src, *dst; |
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56 |
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57 src = (Sint8 *)cvt->buf; |
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58 dst = (Sint8 *)cvt->buf; |
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59 for ( i=cvt->len_cvt/2; i; --i ) { |
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60 sample = src[0] + src[1]; |
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61 *dst = (Sint8)(sample / 2); |
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62 src += 2; |
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63 dst += 1; |
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64 } |
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65 } |
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66 break; |
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67 |
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68 case AUDIO_U16: { |
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69 Uint8 *src, *dst; |
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70 |
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71 src = cvt->buf; |
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72 dst = cvt->buf; |
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73 if ( (format & 0x1000) == 0x1000 ) { |
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74 for ( i=cvt->len_cvt/4; i; --i ) { |
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75 sample = (Uint16)((src[0]<<8)|src[1])+ |
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76 (Uint16)((src[2]<<8)|src[3]); |
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77 sample /= 2; |
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78 dst[1] = (sample&0xFF); |
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79 sample >>= 8; |
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80 dst[0] = (sample&0xFF); |
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81 src += 4; |
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82 dst += 2; |
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83 } |
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84 } else { |
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85 for ( i=cvt->len_cvt/4; i; --i ) { |
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86 sample = (Uint16)((src[1]<<8)|src[0])+ |
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87 (Uint16)((src[3]<<8)|src[2]); |
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88 sample /= 2; |
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89 dst[0] = (sample&0xFF); |
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90 sample >>= 8; |
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91 dst[1] = (sample&0xFF); |
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92 src += 4; |
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93 dst += 2; |
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94 } |
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95 } |
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96 } |
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97 break; |
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98 |
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99 case AUDIO_S16: { |
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100 Uint8 *src, *dst; |
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101 |
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102 src = cvt->buf; |
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103 dst = cvt->buf; |
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104 if ( (format & 0x1000) == 0x1000 ) { |
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105 for ( i=cvt->len_cvt/4; i; --i ) { |
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106 sample = (Sint16)((src[0]<<8)|src[1])+ |
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107 (Sint16)((src[2]<<8)|src[3]); |
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108 sample /= 2; |
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109 dst[1] = (sample&0xFF); |
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110 sample >>= 8; |
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111 dst[0] = (sample&0xFF); |
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112 src += 4; |
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113 dst += 2; |
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114 } |
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115 } else { |
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116 for ( i=cvt->len_cvt/4; i; --i ) { |
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117 sample = (Sint16)((src[1]<<8)|src[0])+ |
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118 (Sint16)((src[3]<<8)|src[2]); |
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119 sample /= 2; |
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120 dst[0] = (sample&0xFF); |
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121 sample >>= 8; |
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122 dst[1] = (sample&0xFF); |
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123 src += 4; |
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124 dst += 2; |
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125 } |
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126 } |
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127 } |
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128 break; |
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129 } |
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130 cvt->len_cvt /= 2; |
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131 if ( cvt->filters[++cvt->filter_index] ) { |
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132 cvt->filters[cvt->filter_index](cvt, format); |
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133 } |
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134 } |
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135 |
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136 /* Discard top 4 channels */ |
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137 void SDLCALL SDL_ConvertStrip(SDL_AudioCVT *cvt, Uint16 format) |
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138 { |
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139 int i; |
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140 Sint32 lsample, rsample; |
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141 |
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142 #ifdef DEBUG_CONVERT |
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143 fprintf(stderr, "Converting down to stereo\n"); |
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144 #endif |
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145 switch (format&0x8018) { |
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146 |
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147 case AUDIO_U8: { |
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148 Uint8 *src, *dst; |
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149 |
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150 src = cvt->buf; |
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151 dst = cvt->buf; |
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152 for ( i=cvt->len_cvt/6; i; --i ) { |
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153 dst[0] = src[0]; |
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154 dst[1] = src[1]; |
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155 src += 6; |
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156 dst += 2; |
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157 } |
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158 } |
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159 break; |
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160 |
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161 case AUDIO_S8: { |
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162 Sint8 *src, *dst; |
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163 |
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164 src = (Sint8 *)cvt->buf; |
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165 dst = (Sint8 *)cvt->buf; |
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166 for ( i=cvt->len_cvt/6; i; --i ) { |
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167 dst[0] = src[0]; |
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168 dst[1] = src[1]; |
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169 src += 6; |
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170 dst += 2; |
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171 } |
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172 } |
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173 break; |
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174 |
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175 case AUDIO_U16: { |
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176 Uint8 *src, *dst; |
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177 |
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178 src = cvt->buf; |
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179 dst = cvt->buf; |
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180 if ( (format & 0x1000) == 0x1000 ) { |
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181 for ( i=cvt->len_cvt/12; i; --i ) { |
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182 lsample = (Uint16)((src[0]<<8)|src[1]); |
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183 rsample = (Uint16)((src[2]<<8)|src[3]); |
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184 dst[1] = (lsample&0xFF); |
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185 lsample >>= 8; |
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186 dst[0] = (lsample&0xFF); |
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187 dst[3] = (rsample&0xFF); |
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188 rsample >>= 8; |
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189 dst[2] = (rsample&0xFF); |
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190 src += 12; |
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191 dst += 4; |
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192 } |
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193 } else { |
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194 for ( i=cvt->len_cvt/12; i; --i ) { |
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195 lsample = (Uint16)((src[1]<<8)|src[0]); |
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196 rsample = (Uint16)((src[3]<<8)|src[2]); |
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197 dst[0] = (lsample&0xFF); |
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198 lsample >>= 8; |
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199 dst[1] = (lsample&0xFF); |
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200 dst[2] = (rsample&0xFF); |
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201 rsample >>= 8; |
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202 dst[3] = (rsample&0xFF); |
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203 src += 12; |
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204 dst += 4; |
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205 } |
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206 } |
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207 } |
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208 break; |
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209 |
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210 case AUDIO_S16: { |
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211 Uint8 *src, *dst; |
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212 |
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213 src = cvt->buf; |
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214 dst = cvt->buf; |
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215 if ( (format & 0x1000) == 0x1000 ) { |
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216 for ( i=cvt->len_cvt/12; i; --i ) { |
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217 lsample = (Sint16)((src[0]<<8)|src[1]); |
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218 rsample = (Sint16)((src[2]<<8)|src[3]); |
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219 dst[1] = (lsample&0xFF); |
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220 lsample >>= 8; |
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221 dst[0] = (lsample&0xFF); |
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222 dst[3] = (rsample&0xFF); |
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223 rsample >>= 8; |
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224 dst[2] = (rsample&0xFF); |
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225 src += 12; |
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226 dst += 4; |
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227 } |
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228 } else { |
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229 for ( i=cvt->len_cvt/12; i; --i ) { |
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230 lsample = (Sint16)((src[1]<<8)|src[0]); |
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231 rsample = (Sint16)((src[3]<<8)|src[2]); |
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232 dst[0] = (lsample&0xFF); |
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233 lsample >>= 8; |
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234 dst[1] = (lsample&0xFF); |
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235 dst[2] = (rsample&0xFF); |
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236 rsample >>= 8; |
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237 dst[3] = (rsample&0xFF); |
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238 src += 12; |
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239 dst += 4; |
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240 } |
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241 } |
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242 } |
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243 break; |
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244 } |
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245 cvt->len_cvt /= 3; |
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246 if ( cvt->filters[++cvt->filter_index] ) { |
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247 cvt->filters[cvt->filter_index](cvt, format); |
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248 } |
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249 } |
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250 |
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251 |
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252 /* Discard top 2 channels of 6 */ |
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253 void SDLCALL SDL_ConvertStrip_2(SDL_AudioCVT *cvt, Uint16 format) |
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254 { |
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255 int i; |
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256 Sint32 lsample, rsample; |
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257 |
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258 #ifdef DEBUG_CONVERT |
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259 fprintf(stderr, "Converting 6 down to quad\n"); |
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260 #endif |
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261 switch (format&0x8018) { |
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262 |
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263 case AUDIO_U8: { |
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264 Uint8 *src, *dst; |
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265 |
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266 src = cvt->buf; |
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267 dst = cvt->buf; |
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268 for ( i=cvt->len_cvt/4; i; --i ) { |
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269 dst[0] = src[0]; |
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270 dst[1] = src[1]; |
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271 src += 4; |
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272 dst += 2; |
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273 } |
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274 } |
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275 break; |
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276 |
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277 case AUDIO_S8: { |
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278 Sint8 *src, *dst; |
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279 |
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280 src = (Sint8 *)cvt->buf; |
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281 dst = (Sint8 *)cvt->buf; |
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282 for ( i=cvt->len_cvt/4; i; --i ) { |
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283 dst[0] = src[0]; |
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284 dst[1] = src[1]; |
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285 src += 4; |
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286 dst += 2; |
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287 } |
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288 } |
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289 break; |
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290 |
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291 case AUDIO_U16: { |
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292 Uint8 *src, *dst; |
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293 |
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294 src = cvt->buf; |
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295 dst = cvt->buf; |
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296 if ( (format & 0x1000) == 0x1000 ) { |
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297 for ( i=cvt->len_cvt/8; i; --i ) { |
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298 lsample = (Uint16)((src[0]<<8)|src[1]); |
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299 rsample = (Uint16)((src[2]<<8)|src[3]); |
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300 dst[1] = (lsample&0xFF); |
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301 lsample >>= 8; |
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302 dst[0] = (lsample&0xFF); |
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303 dst[3] = (rsample&0xFF); |
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304 rsample >>= 8; |
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305 dst[2] = (rsample&0xFF); |
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306 src += 8; |
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307 dst += 4; |
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308 } |
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309 } else { |
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310 for ( i=cvt->len_cvt/8; i; --i ) { |
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311 lsample = (Uint16)((src[1]<<8)|src[0]); |
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312 rsample = (Uint16)((src[3]<<8)|src[2]); |
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313 dst[0] = (lsample&0xFF); |
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314 lsample >>= 8; |
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315 dst[1] = (lsample&0xFF); |
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316 dst[2] = (rsample&0xFF); |
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317 rsample >>= 8; |
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318 dst[3] = (rsample&0xFF); |
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319 src += 8; |
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320 dst += 4; |
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321 } |
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322 } |
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323 } |
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324 break; |
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325 |
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326 case AUDIO_S16: { |
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327 Uint8 *src, *dst; |
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328 |
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329 src = cvt->buf; |
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330 dst = cvt->buf; |
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331 if ( (format & 0x1000) == 0x1000 ) { |
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332 for ( i=cvt->len_cvt/8; i; --i ) { |
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333 lsample = (Sint16)((src[0]<<8)|src[1]); |
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334 rsample = (Sint16)((src[2]<<8)|src[3]); |
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335 dst[1] = (lsample&0xFF); |
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336 lsample >>= 8; |
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337 dst[0] = (lsample&0xFF); |
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338 dst[3] = (rsample&0xFF); |
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339 rsample >>= 8; |
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340 dst[2] = (rsample&0xFF); |
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341 src += 8; |
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342 dst += 4; |
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343 } |
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344 } else { |
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345 for ( i=cvt->len_cvt/8; i; --i ) { |
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346 lsample = (Sint16)((src[1]<<8)|src[0]); |
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347 rsample = (Sint16)((src[3]<<8)|src[2]); |
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348 dst[0] = (lsample&0xFF); |
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349 lsample >>= 8; |
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350 dst[1] = (lsample&0xFF); |
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351 dst[2] = (rsample&0xFF); |
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352 rsample >>= 8; |
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353 dst[3] = (rsample&0xFF); |
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354 src += 8; |
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355 dst += 4; |
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356 } |
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357 } |
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358 } |
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359 break; |
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360 } |
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361 cvt->len_cvt /= 2; |
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362 if ( cvt->filters[++cvt->filter_index] ) { |
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363 cvt->filters[cvt->filter_index](cvt, format); |
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364 } |
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365 } |
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366 |
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367 /* Duplicate a mono channel to both stereo channels */ |
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368 void SDLCALL SDL_ConvertStereo(SDL_AudioCVT *cvt, Uint16 format) |
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369 { |
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370 int i; |
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371 |
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372 #ifdef DEBUG_CONVERT |
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373 fprintf(stderr, "Converting to stereo\n"); |
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374 #endif |
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375 if ( (format & 0xFF) == 16 ) { |
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376 Uint16 *src, *dst; |
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377 |
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378 src = (Uint16 *)(cvt->buf+cvt->len_cvt); |
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379 dst = (Uint16 *)(cvt->buf+cvt->len_cvt*2); |
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380 for ( i=cvt->len_cvt/2; i; --i ) { |
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381 dst -= 2; |
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382 src -= 1; |
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383 dst[0] = src[0]; |
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384 dst[1] = src[0]; |
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385 } |
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386 } else { |
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387 Uint8 *src, *dst; |
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388 |
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389 src = cvt->buf+cvt->len_cvt; |
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390 dst = cvt->buf+cvt->len_cvt*2; |
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391 for ( i=cvt->len_cvt; i; --i ) { |
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392 dst -= 2; |
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393 src -= 1; |
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394 dst[0] = src[0]; |
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395 dst[1] = src[0]; |
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396 } |
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397 } |
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398 cvt->len_cvt *= 2; |
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399 if ( cvt->filters[++cvt->filter_index] ) { |
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400 cvt->filters[cvt->filter_index](cvt, format); |
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401 } |
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402 } |
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403 |
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404 |
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405 /* Duplicate a stereo channel to a pseudo-5.1 stream */ |
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406 void SDLCALL SDL_ConvertSurround(SDL_AudioCVT *cvt, Uint16 format) |
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407 { |
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408 int i; |
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409 |
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410 #ifdef DEBUG_CONVERT |
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411 fprintf(stderr, "Converting stereo to surround\n"); |
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412 #endif |
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413 switch (format&0x8018) { |
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414 |
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415 case AUDIO_U8: { |
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416 Uint8 *src, *dst, lf, rf, ce; |
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417 |
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418 src = (Uint8 *)(cvt->buf+cvt->len_cvt); |
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419 dst = (Uint8 *)(cvt->buf+cvt->len_cvt*3); |
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420 for ( i=cvt->len_cvt; i; --i ) { |
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421 dst -= 6; |
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422 src -= 2; |
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423 lf = src[0]; |
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424 rf = src[1]; |
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425 ce = (lf/2) + (rf/2); |
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426 dst[0] = lf; |
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427 dst[1] = rf; |
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428 dst[2] = lf - ce; |
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429 dst[3] = rf - ce; |
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430 dst[4] = ce; |
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431 dst[5] = ce; |
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432 } |
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433 } |
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434 break; |
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435 |
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436 case AUDIO_S8: { |
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437 Sint8 *src, *dst, lf, rf, ce; |
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438 |
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439 src = (Sint8 *)cvt->buf+cvt->len_cvt; |
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440 dst = (Sint8 *)cvt->buf+cvt->len_cvt*3; |
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441 for ( i=cvt->len_cvt; i; --i ) { |
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442 dst -= 6; |
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443 src -= 2; |
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444 lf = src[0]; |
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445 rf = src[1]; |
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446 ce = (lf/2) + (rf/2); |
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447 dst[0] = lf; |
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448 dst[1] = rf; |
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449 dst[2] = lf - ce; |
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450 dst[3] = rf - ce; |
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451 dst[4] = ce; |
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452 dst[5] = ce; |
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453 } |
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454 } |
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455 break; |
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456 |
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457 case AUDIO_U16: { |
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458 Uint8 *src, *dst; |
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459 Uint16 lf, rf, ce, lr, rr; |
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460 |
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461 src = cvt->buf+cvt->len_cvt; |
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462 dst = cvt->buf+cvt->len_cvt*3; |
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463 |
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464 if ( (format & 0x1000) == 0x1000 ) { |
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465 for ( i=cvt->len_cvt/4; i; --i ) { |
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466 dst -= 12; |
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467 src -= 4; |
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468 lf = (Uint16)((src[0]<<8)|src[1]); |
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469 rf = (Uint16)((src[2]<<8)|src[3]); |
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470 ce = (lf/2) + (rf/2); |
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471 rr = lf - ce; |
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472 lr = rf - ce; |
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473 dst[1] = (lf&0xFF); |
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474 dst[0] = ((lf>>8)&0xFF); |
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475 dst[3] = (rf&0xFF); |
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476 dst[2] = ((rf>>8)&0xFF); |
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477 |
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478 dst[1+4] = (lr&0xFF); |
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479 dst[0+4] = ((lr>>8)&0xFF); |
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480 dst[3+4] = (rr&0xFF); |
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481 dst[2+4] = ((rr>>8)&0xFF); |
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482 |
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483 dst[1+8] = (ce&0xFF); |
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484 dst[0+8] = ((ce>>8)&0xFF); |
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485 dst[3+8] = (ce&0xFF); |
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486 dst[2+8] = ((ce>>8)&0xFF); |
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487 } |
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488 } else { |
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489 for ( i=cvt->len_cvt/4; i; --i ) { |
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490 dst -= 12; |
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491 src -= 4; |
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492 lf = (Uint16)((src[1]<<8)|src[0]); |
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493 rf = (Uint16)((src[3]<<8)|src[2]); |
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494 ce = (lf/2) + (rf/2); |
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495 rr = lf - ce; |
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496 lr = rf - ce; |
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497 dst[0] = (lf&0xFF); |
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498 dst[1] = ((lf>>8)&0xFF); |
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499 dst[2] = (rf&0xFF); |
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500 dst[3] = ((rf>>8)&0xFF); |
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501 |
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502 dst[0+4] = (lr&0xFF); |
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503 dst[1+4] = ((lr>>8)&0xFF); |
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504 dst[2+4] = (rr&0xFF); |
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505 dst[3+4] = ((rr>>8)&0xFF); |
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506 |
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507 dst[0+8] = (ce&0xFF); |
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508 dst[1+8] = ((ce>>8)&0xFF); |
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509 dst[2+8] = (ce&0xFF); |
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510 dst[3+8] = ((ce>>8)&0xFF); |
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511 } |
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512 } |
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513 } |
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514 break; |
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515 |
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516 case AUDIO_S16: { |
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517 Uint8 *src, *dst; |
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518 Sint16 lf, rf, ce, lr, rr; |
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519 |
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520 src = cvt->buf+cvt->len_cvt; |
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521 dst = cvt->buf+cvt->len_cvt*3; |
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522 |
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523 if ( (format & 0x1000) == 0x1000 ) { |
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524 for ( i=cvt->len_cvt/4; i; --i ) { |
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525 dst -= 12; |
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526 src -= 4; |
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527 lf = (Sint16)((src[0]<<8)|src[1]); |
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528 rf = (Sint16)((src[2]<<8)|src[3]); |
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529 ce = (lf/2) + (rf/2); |
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530 rr = lf - ce; |
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531 lr = rf - ce; |
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532 dst[1] = (lf&0xFF); |
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533 dst[0] = ((lf>>8)&0xFF); |
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534 dst[3] = (rf&0xFF); |
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535 dst[2] = ((rf>>8)&0xFF); |
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536 |
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537 dst[1+4] = (lr&0xFF); |
|
538 dst[0+4] = ((lr>>8)&0xFF); |
|
539 dst[3+4] = (rr&0xFF); |
|
540 dst[2+4] = ((rr>>8)&0xFF); |
|
541 |
|
542 dst[1+8] = (ce&0xFF); |
|
543 dst[0+8] = ((ce>>8)&0xFF); |
|
544 dst[3+8] = (ce&0xFF); |
|
545 dst[2+8] = ((ce>>8)&0xFF); |
|
546 } |
|
547 } else { |
|
548 for ( i=cvt->len_cvt/4; i; --i ) { |
|
549 dst -= 12; |
|
550 src -= 4; |
|
551 lf = (Sint16)((src[1]<<8)|src[0]); |
|
552 rf = (Sint16)((src[3]<<8)|src[2]); |
|
553 ce = (lf/2) + (rf/2); |
|
554 rr = lf - ce; |
|
555 lr = rf - ce; |
|
556 dst[0] = (lf&0xFF); |
|
557 dst[1] = ((lf>>8)&0xFF); |
|
558 dst[2] = (rf&0xFF); |
|
559 dst[3] = ((rf>>8)&0xFF); |
|
560 |
|
561 dst[0+4] = (lr&0xFF); |
|
562 dst[1+4] = ((lr>>8)&0xFF); |
|
563 dst[2+4] = (rr&0xFF); |
|
564 dst[3+4] = ((rr>>8)&0xFF); |
|
565 |
|
566 dst[0+8] = (ce&0xFF); |
|
567 dst[1+8] = ((ce>>8)&0xFF); |
|
568 dst[2+8] = (ce&0xFF); |
|
569 dst[3+8] = ((ce>>8)&0xFF); |
|
570 } |
|
571 } |
|
572 } |
|
573 break; |
|
574 } |
|
575 cvt->len_cvt *= 3; |
|
576 if ( cvt->filters[++cvt->filter_index] ) { |
|
577 cvt->filters[cvt->filter_index](cvt, format); |
|
578 } |
|
579 } |
|
580 |
|
581 |
|
582 /* Duplicate a stereo channel to a pseudo-4.0 stream */ |
|
583 void SDLCALL SDL_ConvertSurround_4(SDL_AudioCVT *cvt, Uint16 format) |
|
584 { |
|
585 int i; |
|
586 |
|
587 #ifdef DEBUG_CONVERT |
|
588 fprintf(stderr, "Converting stereo to quad\n"); |
|
589 #endif |
|
590 switch (format&0x8018) { |
|
591 |
|
592 case AUDIO_U8: { |
|
593 Uint8 *src, *dst, lf, rf, ce; |
|
594 |
|
595 src = (Uint8 *)(cvt->buf+cvt->len_cvt); |
|
596 dst = (Uint8 *)(cvt->buf+cvt->len_cvt*2); |
|
597 for ( i=cvt->len_cvt; i; --i ) { |
|
598 dst -= 4; |
|
599 src -= 2; |
|
600 lf = src[0]; |
|
601 rf = src[1]; |
|
602 ce = (lf/2) + (rf/2); |
|
603 dst[0] = lf; |
|
604 dst[1] = rf; |
|
605 dst[2] = lf - ce; |
|
606 dst[3] = rf - ce; |
|
607 } |
|
608 } |
|
609 break; |
|
610 |
|
611 case AUDIO_S8: { |
|
612 Sint8 *src, *dst, lf, rf, ce; |
|
613 |
|
614 src = (Sint8 *)cvt->buf+cvt->len_cvt; |
|
615 dst = (Sint8 *)cvt->buf+cvt->len_cvt*2; |
|
616 for ( i=cvt->len_cvt; i; --i ) { |
|
617 dst -= 4; |
|
618 src -= 2; |
|
619 lf = src[0]; |
|
620 rf = src[1]; |
|
621 ce = (lf/2) + (rf/2); |
|
622 dst[0] = lf; |
|
623 dst[1] = rf; |
|
624 dst[2] = lf - ce; |
|
625 dst[3] = rf - ce; |
|
626 } |
|
627 } |
|
628 break; |
|
629 |
|
630 case AUDIO_U16: { |
|
631 Uint8 *src, *dst; |
|
632 Uint16 lf, rf, ce, lr, rr; |
|
633 |
|
634 src = cvt->buf+cvt->len_cvt; |
|
635 dst = cvt->buf+cvt->len_cvt*2; |
|
636 |
|
637 if ( (format & 0x1000) == 0x1000 ) { |
|
638 for ( i=cvt->len_cvt/4; i; --i ) { |
|
639 dst -= 8; |
|
640 src -= 4; |
|
641 lf = (Uint16)((src[0]<<8)|src[1]); |
|
642 rf = (Uint16)((src[2]<<8)|src[3]); |
|
643 ce = (lf/2) + (rf/2); |
|
644 rr = lf - ce; |
|
645 lr = rf - ce; |
|
646 dst[1] = (lf&0xFF); |
|
647 dst[0] = ((lf>>8)&0xFF); |
|
648 dst[3] = (rf&0xFF); |
|
649 dst[2] = ((rf>>8)&0xFF); |
|
650 |
|
651 dst[1+4] = (lr&0xFF); |
|
652 dst[0+4] = ((lr>>8)&0xFF); |
|
653 dst[3+4] = (rr&0xFF); |
|
654 dst[2+4] = ((rr>>8)&0xFF); |
|
655 } |
|
656 } else { |
|
657 for ( i=cvt->len_cvt/4; i; --i ) { |
|
658 dst -= 8; |
|
659 src -= 4; |
|
660 lf = (Uint16)((src[1]<<8)|src[0]); |
|
661 rf = (Uint16)((src[3]<<8)|src[2]); |
|
662 ce = (lf/2) + (rf/2); |
|
663 rr = lf - ce; |
|
664 lr = rf - ce; |
|
665 dst[0] = (lf&0xFF); |
|
666 dst[1] = ((lf>>8)&0xFF); |
|
667 dst[2] = (rf&0xFF); |
|
668 dst[3] = ((rf>>8)&0xFF); |
|
669 |
|
670 dst[0+4] = (lr&0xFF); |
|
671 dst[1+4] = ((lr>>8)&0xFF); |
|
672 dst[2+4] = (rr&0xFF); |
|
673 dst[3+4] = ((rr>>8)&0xFF); |
|
674 } |
|
675 } |
|
676 } |
|
677 break; |
|
678 |
|
679 case AUDIO_S16: { |
|
680 Uint8 *src, *dst; |
|
681 Sint16 lf, rf, ce, lr, rr; |
|
682 |
|
683 src = cvt->buf+cvt->len_cvt; |
|
684 dst = cvt->buf+cvt->len_cvt*2; |
|
685 |
|
686 if ( (format & 0x1000) == 0x1000 ) { |
|
687 for ( i=cvt->len_cvt/4; i; --i ) { |
|
688 dst -= 8; |
|
689 src -= 4; |
|
690 lf = (Sint16)((src[0]<<8)|src[1]); |
|
691 rf = (Sint16)((src[2]<<8)|src[3]); |
|
692 ce = (lf/2) + (rf/2); |
|
693 rr = lf - ce; |
|
694 lr = rf - ce; |
|
695 dst[1] = (lf&0xFF); |
|
696 dst[0] = ((lf>>8)&0xFF); |
|
697 dst[3] = (rf&0xFF); |
|
698 dst[2] = ((rf>>8)&0xFF); |
|
699 |
|
700 dst[1+4] = (lr&0xFF); |
|
701 dst[0+4] = ((lr>>8)&0xFF); |
|
702 dst[3+4] = (rr&0xFF); |
|
703 dst[2+4] = ((rr>>8)&0xFF); |
|
704 } |
|
705 } else { |
|
706 for ( i=cvt->len_cvt/4; i; --i ) { |
|
707 dst -= 8; |
|
708 src -= 4; |
|
709 lf = (Sint16)((src[1]<<8)|src[0]); |
|
710 rf = (Sint16)((src[3]<<8)|src[2]); |
|
711 ce = (lf/2) + (rf/2); |
|
712 rr = lf - ce; |
|
713 lr = rf - ce; |
|
714 dst[0] = (lf&0xFF); |
|
715 dst[1] = ((lf>>8)&0xFF); |
|
716 dst[2] = (rf&0xFF); |
|
717 dst[3] = ((rf>>8)&0xFF); |
|
718 |
|
719 dst[0+4] = (lr&0xFF); |
|
720 dst[1+4] = ((lr>>8)&0xFF); |
|
721 dst[2+4] = (rr&0xFF); |
|
722 dst[3+4] = ((rr>>8)&0xFF); |
|
723 } |
|
724 } |
|
725 } |
|
726 break; |
|
727 } |
|
728 cvt->len_cvt *= 2; |
|
729 if ( cvt->filters[++cvt->filter_index] ) { |
|
730 cvt->filters[cvt->filter_index](cvt, format); |
|
731 } |
|
732 } |
|
733 |
|
734 |
|
735 /* Convert 8-bit to 16-bit - LSB */ |
|
736 void SDLCALL SDL_Convert16LSB(SDL_AudioCVT *cvt, Uint16 format) |
|
737 { |
|
738 int i; |
|
739 Uint8 *src, *dst; |
|
740 |
|
741 #ifdef DEBUG_CONVERT |
|
742 fprintf(stderr, "Converting to 16-bit LSB\n"); |
|
743 #endif |
|
744 src = cvt->buf+cvt->len_cvt; |
|
745 dst = cvt->buf+cvt->len_cvt*2; |
|
746 for ( i=cvt->len_cvt; i; --i ) { |
|
747 src -= 1; |
|
748 dst -= 2; |
|
749 dst[1] = *src; |
|
750 dst[0] = 0; |
|
751 } |
|
752 format = ((format & ~0x0008) | AUDIO_U16LSB); |
|
753 cvt->len_cvt *= 2; |
|
754 if ( cvt->filters[++cvt->filter_index] ) { |
|
755 cvt->filters[cvt->filter_index](cvt, format); |
|
756 } |
|
757 } |
|
758 /* Convert 8-bit to 16-bit - MSB */ |
|
759 void SDLCALL SDL_Convert16MSB(SDL_AudioCVT *cvt, Uint16 format) |
|
760 { |
|
761 int i; |
|
762 Uint8 *src, *dst; |
|
763 |
|
764 #ifdef DEBUG_CONVERT |
|
765 fprintf(stderr, "Converting to 16-bit MSB\n"); |
|
766 #endif |
|
767 src = cvt->buf+cvt->len_cvt; |
|
768 dst = cvt->buf+cvt->len_cvt*2; |
|
769 for ( i=cvt->len_cvt; i; --i ) { |
|
770 src -= 1; |
|
771 dst -= 2; |
|
772 dst[0] = *src; |
|
773 dst[1] = 0; |
|
774 } |
|
775 format = ((format & ~0x0008) | AUDIO_U16MSB); |
|
776 cvt->len_cvt *= 2; |
|
777 if ( cvt->filters[++cvt->filter_index] ) { |
|
778 cvt->filters[cvt->filter_index](cvt, format); |
|
779 } |
|
780 } |
|
781 |
|
782 /* Convert 16-bit to 8-bit */ |
|
783 void SDLCALL SDL_Convert8(SDL_AudioCVT *cvt, Uint16 format) |
|
784 { |
|
785 int i; |
|
786 Uint8 *src, *dst; |
|
787 |
|
788 #ifdef DEBUG_CONVERT |
|
789 fprintf(stderr, "Converting to 8-bit\n"); |
|
790 #endif |
|
791 src = cvt->buf; |
|
792 dst = cvt->buf; |
|
793 if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ |
|
794 ++src; |
|
795 } |
|
796 for ( i=cvt->len_cvt/2; i; --i ) { |
|
797 *dst = *src; |
|
798 src += 2; |
|
799 dst += 1; |
|
800 } |
|
801 format = ((format & ~0x9010) | AUDIO_U8); |
|
802 cvt->len_cvt /= 2; |
|
803 if ( cvt->filters[++cvt->filter_index] ) { |
|
804 cvt->filters[cvt->filter_index](cvt, format); |
|
805 } |
|
806 } |
|
807 |
|
808 /* Toggle signed/unsigned */ |
|
809 void SDLCALL SDL_ConvertSign(SDL_AudioCVT *cvt, Uint16 format) |
|
810 { |
|
811 int i; |
|
812 Uint8 *data; |
|
813 |
|
814 #ifdef DEBUG_CONVERT |
|
815 fprintf(stderr, "Converting audio signedness\n"); |
|
816 #endif |
|
817 data = cvt->buf; |
|
818 if ( (format & 0xFF) == 16 ) { |
|
819 if ( (format & 0x1000) != 0x1000 ) { /* Little endian */ |
|
820 ++data; |
|
821 } |
|
822 for ( i=cvt->len_cvt/2; i; --i ) { |
|
823 *data ^= 0x80; |
|
824 data += 2; |
|
825 } |
|
826 } else { |
|
827 for ( i=cvt->len_cvt; i; --i ) { |
|
828 *data++ ^= 0x80; |
|
829 } |
|
830 } |
|
831 format = (format ^ 0x8000); |
|
832 if ( cvt->filters[++cvt->filter_index] ) { |
|
833 cvt->filters[cvt->filter_index](cvt, format); |
|
834 } |
|
835 } |
|
836 |
|
837 /* Toggle endianness */ |
|
838 void SDLCALL SDL_ConvertEndian(SDL_AudioCVT *cvt, Uint16 format) |
|
839 { |
|
840 int i; |
|
841 Uint8 *data, tmp; |
|
842 |
|
843 #ifdef DEBUG_CONVERT |
|
844 fprintf(stderr, "Converting audio endianness\n"); |
|
845 #endif |
|
846 data = cvt->buf; |
|
847 for ( i=cvt->len_cvt/2; i; --i ) { |
|
848 tmp = data[0]; |
|
849 data[0] = data[1]; |
|
850 data[1] = tmp; |
|
851 data += 2; |
|
852 } |
|
853 format = (format ^ 0x1000); |
|
854 if ( cvt->filters[++cvt->filter_index] ) { |
|
855 cvt->filters[cvt->filter_index](cvt, format); |
|
856 } |
|
857 } |
|
858 |
|
859 /* Convert rate up by multiple of 2 */ |
|
860 void SDLCALL SDL_RateMUL2(SDL_AudioCVT *cvt, Uint16 format) |
|
861 { |
|
862 int i; |
|
863 Uint8 *src, *dst; |
|
864 |
|
865 #ifdef DEBUG_CONVERT |
|
866 fprintf(stderr, "Converting audio rate * 2\n"); |
|
867 #endif |
|
868 src = cvt->buf+cvt->len_cvt; |
|
869 dst = cvt->buf+cvt->len_cvt*2; |
|
870 switch (format & 0xFF) { |
|
871 case 8: |
|
872 for ( i=cvt->len_cvt; i; --i ) { |
|
873 src -= 1; |
|
874 dst -= 2; |
|
875 dst[0] = src[0]; |
|
876 dst[1] = src[0]; |
|
877 } |
|
878 break; |
|
879 case 16: |
|
880 for ( i=cvt->len_cvt/2; i; --i ) { |
|
881 src -= 2; |
|
882 dst -= 4; |
|
883 dst[0] = src[0]; |
|
884 dst[1] = src[1]; |
|
885 dst[2] = src[0]; |
|
886 dst[3] = src[1]; |
|
887 } |
|
888 break; |
|
889 } |
|
890 cvt->len_cvt *= 2; |
|
891 if ( cvt->filters[++cvt->filter_index] ) { |
|
892 cvt->filters[cvt->filter_index](cvt, format); |
|
893 } |
|
894 } |
|
895 |
|
896 |
|
897 /* Convert rate up by multiple of 2, for stereo */ |
|
898 void SDLCALL SDL_RateMUL2_c2(SDL_AudioCVT *cvt, Uint16 format) |
|
899 { |
|
900 int i; |
|
901 Uint8 *src, *dst; |
|
902 |
|
903 #ifdef DEBUG_CONVERT |
|
904 fprintf(stderr, "Converting audio rate * 2\n"); |
|
905 #endif |
|
906 src = cvt->buf+cvt->len_cvt; |
|
907 dst = cvt->buf+cvt->len_cvt*2; |
|
908 switch (format & 0xFF) { |
|
909 case 8: |
|
910 for ( i=cvt->len_cvt/2; i; --i ) { |
|
911 src -= 2; |
|
912 dst -= 4; |
|
913 dst[0] = src[0]; |
|
914 dst[1] = src[1]; |
|
915 dst[2] = src[0]; |
|
916 dst[3] = src[1]; |
|
917 } |
|
918 break; |
|
919 case 16: |
|
920 for ( i=cvt->len_cvt/4; i; --i ) { |
|
921 src -= 4; |
|
922 dst -= 8; |
|
923 dst[0] = src[0]; |
|
924 dst[1] = src[1]; |
|
925 dst[2] = src[2]; |
|
926 dst[3] = src[3]; |
|
927 dst[4] = src[0]; |
|
928 dst[5] = src[1]; |
|
929 dst[6] = src[2]; |
|
930 dst[7] = src[3]; |
|
931 } |
|
932 break; |
|
933 } |
|
934 cvt->len_cvt *= 2; |
|
935 if ( cvt->filters[++cvt->filter_index] ) { |
|
936 cvt->filters[cvt->filter_index](cvt, format); |
|
937 } |
|
938 } |
|
939 |
|
940 /* Convert rate up by multiple of 2, for quad */ |
|
941 void SDLCALL SDL_RateMUL2_c4(SDL_AudioCVT *cvt, Uint16 format) |
|
942 { |
|
943 int i; |
|
944 Uint8 *src, *dst; |
|
945 |
|
946 #ifdef DEBUG_CONVERT |
|
947 fprintf(stderr, "Converting audio rate * 2\n"); |
|
948 #endif |
|
949 src = cvt->buf+cvt->len_cvt; |
|
950 dst = cvt->buf+cvt->len_cvt*2; |
|
951 switch (format & 0xFF) { |
|
952 case 8: |
|
953 for ( i=cvt->len_cvt/4; i; --i ) { |
|
954 src -= 4; |
|
955 dst -= 8; |
|
956 dst[0] = src[0]; |
|
957 dst[1] = src[1]; |
|
958 dst[2] = src[2]; |
|
959 dst[3] = src[3]; |
|
960 dst[4] = src[0]; |
|
961 dst[5] = src[1]; |
|
962 dst[6] = src[2]; |
|
963 dst[7] = src[3]; |
|
964 } |
|
965 break; |
|
966 case 16: |
|
967 for ( i=cvt->len_cvt/8; i; --i ) { |
|
968 src -= 8; |
|
969 dst -= 16; |
|
970 dst[0] = src[0]; |
|
971 dst[1] = src[1]; |
|
972 dst[2] = src[2]; |
|
973 dst[3] = src[3]; |
|
974 dst[4] = src[4]; |
|
975 dst[5] = src[5]; |
|
976 dst[6] = src[6]; |
|
977 dst[7] = src[7]; |
|
978 dst[8] = src[0]; |
|
979 dst[9] = src[1]; |
|
980 dst[10] = src[2]; |
|
981 dst[11] = src[3]; |
|
982 dst[12] = src[4]; |
|
983 dst[13] = src[5]; |
|
984 dst[14] = src[6]; |
|
985 dst[15] = src[7]; |
|
986 } |
|
987 break; |
|
988 } |
|
989 cvt->len_cvt *= 2; |
|
990 if ( cvt->filters[++cvt->filter_index] ) { |
|
991 cvt->filters[cvt->filter_index](cvt, format); |
|
992 } |
|
993 } |
|
994 |
|
995 |
|
996 /* Convert rate up by multiple of 2, for 5.1 */ |
|
997 void SDLCALL SDL_RateMUL2_c6(SDL_AudioCVT *cvt, Uint16 format) |
|
998 { |
|
999 int i; |
|
1000 Uint8 *src, *dst; |
|
1001 |
|
1002 #ifdef DEBUG_CONVERT |
|
1003 fprintf(stderr, "Converting audio rate * 2\n"); |
|
1004 #endif |
|
1005 src = cvt->buf+cvt->len_cvt; |
|
1006 dst = cvt->buf+cvt->len_cvt*2; |
|
1007 switch (format & 0xFF) { |
|
1008 case 8: |
|
1009 for ( i=cvt->len_cvt/6; i; --i ) { |
|
1010 src -= 6; |
|
1011 dst -= 12; |
|
1012 dst[0] = src[0]; |
|
1013 dst[1] = src[1]; |
|
1014 dst[2] = src[2]; |
|
1015 dst[3] = src[3]; |
|
1016 dst[4] = src[4]; |
|
1017 dst[5] = src[5]; |
|
1018 dst[6] = src[0]; |
|
1019 dst[7] = src[1]; |
|
1020 dst[8] = src[2]; |
|
1021 dst[9] = src[3]; |
|
1022 dst[10] = src[4]; |
|
1023 dst[11] = src[5]; |
|
1024 } |
|
1025 break; |
|
1026 case 16: |
|
1027 for ( i=cvt->len_cvt/12; i; --i ) { |
|
1028 src -= 12; |
|
1029 dst -= 24; |
|
1030 dst[0] = src[0]; |
|
1031 dst[1] = src[1]; |
|
1032 dst[2] = src[2]; |
|
1033 dst[3] = src[3]; |
|
1034 dst[4] = src[4]; |
|
1035 dst[5] = src[5]; |
|
1036 dst[6] = src[6]; |
|
1037 dst[7] = src[7]; |
|
1038 dst[8] = src[8]; |
|
1039 dst[9] = src[9]; |
|
1040 dst[10] = src[10]; |
|
1041 dst[11] = src[11]; |
|
1042 dst[12] = src[0]; |
|
1043 dst[13] = src[1]; |
|
1044 dst[14] = src[2]; |
|
1045 dst[15] = src[3]; |
|
1046 dst[16] = src[4]; |
|
1047 dst[17] = src[5]; |
|
1048 dst[18] = src[6]; |
|
1049 dst[19] = src[7]; |
|
1050 dst[20] = src[8]; |
|
1051 dst[21] = src[9]; |
|
1052 dst[22] = src[10]; |
|
1053 dst[23] = src[11]; |
|
1054 } |
|
1055 break; |
|
1056 } |
|
1057 cvt->len_cvt *= 2; |
|
1058 if ( cvt->filters[++cvt->filter_index] ) { |
|
1059 cvt->filters[cvt->filter_index](cvt, format); |
|
1060 } |
|
1061 } |
|
1062 |
|
1063 /* Convert rate down by multiple of 2 */ |
|
1064 void SDLCALL SDL_RateDIV2(SDL_AudioCVT *cvt, Uint16 format) |
|
1065 { |
|
1066 int i; |
|
1067 Uint8 *src, *dst; |
|
1068 |
|
1069 #ifdef DEBUG_CONVERT |
|
1070 fprintf(stderr, "Converting audio rate / 2\n"); |
|
1071 #endif |
|
1072 src = cvt->buf; |
|
1073 dst = cvt->buf; |
|
1074 switch (format & 0xFF) { |
|
1075 case 8: |
|
1076 for ( i=cvt->len_cvt/2; i; --i ) { |
|
1077 dst[0] = src[0]; |
|
1078 src += 2; |
|
1079 dst += 1; |
|
1080 } |
|
1081 break; |
|
1082 case 16: |
|
1083 for ( i=cvt->len_cvt/4; i; --i ) { |
|
1084 dst[0] = src[0]; |
|
1085 dst[1] = src[1]; |
|
1086 src += 4; |
|
1087 dst += 2; |
|
1088 } |
|
1089 break; |
|
1090 } |
|
1091 cvt->len_cvt /= 2; |
|
1092 if ( cvt->filters[++cvt->filter_index] ) { |
|
1093 cvt->filters[cvt->filter_index](cvt, format); |
|
1094 } |
|
1095 } |
|
1096 |
|
1097 |
|
1098 /* Convert rate down by multiple of 2, for stereo */ |
|
1099 void SDLCALL SDL_RateDIV2_c2(SDL_AudioCVT *cvt, Uint16 format) |
|
1100 { |
|
1101 int i; |
|
1102 Uint8 *src, *dst; |
|
1103 |
|
1104 #ifdef DEBUG_CONVERT |
|
1105 fprintf(stderr, "Converting audio rate / 2\n"); |
|
1106 #endif |
|
1107 src = cvt->buf; |
|
1108 dst = cvt->buf; |
|
1109 switch (format & 0xFF) { |
|
1110 case 8: |
|
1111 for ( i=cvt->len_cvt/4; i; --i ) { |
|
1112 dst[0] = src[0]; |
|
1113 dst[1] = src[1]; |
|
1114 src += 4; |
|
1115 dst += 2; |
|
1116 } |
|
1117 break; |
|
1118 case 16: |
|
1119 for ( i=cvt->len_cvt/8; i; --i ) { |
|
1120 dst[0] = src[0]; |
|
1121 dst[1] = src[1]; |
|
1122 dst[2] = src[2]; |
|
1123 dst[3] = src[3]; |
|
1124 src += 8; |
|
1125 dst += 4; |
|
1126 } |
|
1127 break; |
|
1128 } |
|
1129 cvt->len_cvt /= 2; |
|
1130 if ( cvt->filters[++cvt->filter_index] ) { |
|
1131 cvt->filters[cvt->filter_index](cvt, format); |
|
1132 } |
|
1133 } |
|
1134 |
|
1135 |
|
1136 /* Convert rate down by multiple of 2, for quad */ |
|
1137 void SDLCALL SDL_RateDIV2_c4(SDL_AudioCVT *cvt, Uint16 format) |
|
1138 { |
|
1139 int i; |
|
1140 Uint8 *src, *dst; |
|
1141 |
|
1142 #ifdef DEBUG_CONVERT |
|
1143 fprintf(stderr, "Converting audio rate / 2\n"); |
|
1144 #endif |
|
1145 src = cvt->buf; |
|
1146 dst = cvt->buf; |
|
1147 switch (format & 0xFF) { |
|
1148 case 8: |
|
1149 for ( i=cvt->len_cvt/8; i; --i ) { |
|
1150 dst[0] = src[0]; |
|
1151 dst[1] = src[1]; |
|
1152 dst[2] = src[2]; |
|
1153 dst[3] = src[3]; |
|
1154 src += 8; |
|
1155 dst += 4; |
|
1156 } |
|
1157 break; |
|
1158 case 16: |
|
1159 for ( i=cvt->len_cvt/16; i; --i ) { |
|
1160 dst[0] = src[0]; |
|
1161 dst[1] = src[1]; |
|
1162 dst[2] = src[2]; |
|
1163 dst[3] = src[3]; |
|
1164 dst[4] = src[4]; |
|
1165 dst[5] = src[5]; |
|
1166 dst[6] = src[6]; |
|
1167 dst[7] = src[7]; |
|
1168 src += 16; |
|
1169 dst += 8; |
|
1170 } |
|
1171 break; |
|
1172 } |
|
1173 cvt->len_cvt /= 2; |
|
1174 if ( cvt->filters[++cvt->filter_index] ) { |
|
1175 cvt->filters[cvt->filter_index](cvt, format); |
|
1176 } |
|
1177 } |
|
1178 |
|
1179 /* Convert rate down by multiple of 2, for 5.1 */ |
|
1180 void SDLCALL SDL_RateDIV2_c6(SDL_AudioCVT *cvt, Uint16 format) |
|
1181 { |
|
1182 int i; |
|
1183 Uint8 *src, *dst; |
|
1184 |
|
1185 #ifdef DEBUG_CONVERT |
|
1186 fprintf(stderr, "Converting audio rate / 2\n"); |
|
1187 #endif |
|
1188 src = cvt->buf; |
|
1189 dst = cvt->buf; |
|
1190 switch (format & 0xFF) { |
|
1191 case 8: |
|
1192 for ( i=cvt->len_cvt/12; i; --i ) { |
|
1193 dst[0] = src[0]; |
|
1194 dst[1] = src[1]; |
|
1195 dst[2] = src[2]; |
|
1196 dst[3] = src[3]; |
|
1197 dst[4] = src[4]; |
|
1198 dst[5] = src[5]; |
|
1199 src += 12; |
|
1200 dst += 6; |
|
1201 } |
|
1202 break; |
|
1203 case 16: |
|
1204 for ( i=cvt->len_cvt/24; i; --i ) { |
|
1205 dst[0] = src[0]; |
|
1206 dst[1] = src[1]; |
|
1207 dst[2] = src[2]; |
|
1208 dst[3] = src[3]; |
|
1209 dst[4] = src[4]; |
|
1210 dst[5] = src[5]; |
|
1211 dst[6] = src[6]; |
|
1212 dst[7] = src[7]; |
|
1213 dst[8] = src[8]; |
|
1214 dst[9] = src[9]; |
|
1215 dst[10] = src[10]; |
|
1216 dst[11] = src[11]; |
|
1217 src += 24; |
|
1218 dst += 12; |
|
1219 } |
|
1220 break; |
|
1221 } |
|
1222 cvt->len_cvt /= 2; |
|
1223 if ( cvt->filters[++cvt->filter_index] ) { |
|
1224 cvt->filters[cvt->filter_index](cvt, format); |
|
1225 } |
|
1226 } |
|
1227 |
|
1228 /* Very slow rate conversion routine */ |
|
1229 void SDLCALL SDL_RateSLOW(SDL_AudioCVT *cvt, Uint16 format) |
|
1230 { |
|
1231 double ipos; |
|
1232 int i, clen; |
|
1233 |
|
1234 #ifdef DEBUG_CONVERT |
|
1235 fprintf(stderr, "Converting audio rate * %4.4f\n", 1.0/cvt->rate_incr); |
|
1236 #endif |
|
1237 clen = (int)((double)cvt->len_cvt / cvt->rate_incr); |
|
1238 if ( cvt->rate_incr > 1.0 ) { |
|
1239 switch (format & 0xFF) { |
|
1240 case 8: { |
|
1241 Uint8 *output; |
|
1242 |
|
1243 output = cvt->buf; |
|
1244 ipos = 0.0; |
|
1245 for ( i=clen; i; --i ) { |
|
1246 *output = cvt->buf[(int)ipos]; |
|
1247 ipos += cvt->rate_incr; |
|
1248 output += 1; |
|
1249 } |
|
1250 } |
|
1251 break; |
|
1252 |
|
1253 case 16: { |
|
1254 Uint16 *output; |
|
1255 |
|
1256 clen &= ~1; |
|
1257 output = (Uint16 *)cvt->buf; |
|
1258 ipos = 0.0; |
|
1259 for ( i=clen/2; i; --i ) { |
|
1260 *output=((Uint16 *)cvt->buf)[(int)ipos]; |
|
1261 ipos += cvt->rate_incr; |
|
1262 output += 1; |
|
1263 } |
|
1264 } |
|
1265 break; |
|
1266 } |
|
1267 } else { |
|
1268 switch (format & 0xFF) { |
|
1269 case 8: { |
|
1270 Uint8 *output; |
|
1271 |
|
1272 output = cvt->buf+clen; |
|
1273 ipos = (double)cvt->len_cvt; |
|
1274 for ( i=clen; i; --i ) { |
|
1275 ipos -= cvt->rate_incr; |
|
1276 output -= 1; |
|
1277 *output = cvt->buf[(int)ipos]; |
|
1278 } |
|
1279 } |
|
1280 break; |
|
1281 |
|
1282 case 16: { |
|
1283 Uint16 *output; |
|
1284 |
|
1285 clen &= ~1; |
|
1286 output = (Uint16 *)(cvt->buf+clen); |
|
1287 ipos = (double)cvt->len_cvt/2; |
|
1288 for ( i=clen/2; i; --i ) { |
|
1289 ipos -= cvt->rate_incr; |
|
1290 output -= 1; |
|
1291 *output=((Uint16 *)cvt->buf)[(int)ipos]; |
|
1292 } |
|
1293 } |
|
1294 break; |
|
1295 } |
|
1296 } |
|
1297 cvt->len_cvt = clen; |
|
1298 if ( cvt->filters[++cvt->filter_index] ) { |
|
1299 cvt->filters[cvt->filter_index](cvt, format); |
|
1300 } |
|
1301 } |
|
1302 |
|
1303 int SDL_ConvertAudio(SDL_AudioCVT *cvt) |
|
1304 { |
|
1305 /* Make sure there's data to convert */ |
|
1306 if ( cvt->buf == NULL ) { |
|
1307 SDL_SetError("No buffer allocated for conversion"); |
|
1308 return(-1); |
|
1309 } |
|
1310 /* Return okay if no conversion is necessary */ |
|
1311 cvt->len_cvt = cvt->len; |
|
1312 if ( cvt->filters[0] == NULL ) { |
|
1313 return(0); |
|
1314 } |
|
1315 |
|
1316 /* Set up the conversion and go! */ |
|
1317 cvt->filter_index = 0; |
|
1318 cvt->filters[0](cvt, cvt->src_format); |
|
1319 return(0); |
|
1320 } |
|
1321 |
|
1322 /* Creates a set of audio filters to convert from one format to another. |
|
1323 Returns -1 if the format conversion is not supported, or 1 if the |
|
1324 audio filter is set up. |
|
1325 */ |
|
1326 |
|
1327 int SDL_BuildAudioCVT(SDL_AudioCVT *cvt, |
|
1328 Uint16 src_format, Uint8 src_channels, int src_rate, |
|
1329 Uint16 dst_format, Uint8 dst_channels, int dst_rate) |
|
1330 { |
|
1331 /*printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n", |
|
1332 src_format, dst_format, src_channels, dst_channels, src_rate, dst_rate);*/ |
|
1333 /* Start off with no conversion necessary */ |
|
1334 cvt->needed = 0; |
|
1335 cvt->filter_index = 0; |
|
1336 cvt->filters[0] = NULL; |
|
1337 cvt->len_mult = 1; |
|
1338 cvt->len_ratio = 1.0; |
|
1339 |
|
1340 /* First filter: Endian conversion from src to dst */ |
|
1341 if ( (src_format & 0x1000) != (dst_format & 0x1000) |
|
1342 && ((src_format & 0xff) == 16) && ((dst_format & 0xff) == 16)) { |
|
1343 cvt->filters[cvt->filter_index++] = SDL_ConvertEndian; |
|
1344 } |
|
1345 |
|
1346 /* Second filter: Sign conversion -- signed/unsigned */ |
|
1347 if ( (src_format & 0x8000) != (dst_format & 0x8000) ) { |
|
1348 cvt->filters[cvt->filter_index++] = SDL_ConvertSign; |
|
1349 } |
|
1350 |
|
1351 /* Next filter: Convert 16 bit <--> 8 bit PCM */ |
|
1352 if ( (src_format & 0xFF) != (dst_format & 0xFF) ) { |
|
1353 switch (dst_format&0x10FF) { |
|
1354 case AUDIO_U8: |
|
1355 cvt->filters[cvt->filter_index++] = |
|
1356 SDL_Convert8; |
|
1357 cvt->len_ratio /= 2; |
|
1358 break; |
|
1359 case AUDIO_U16LSB: |
|
1360 cvt->filters[cvt->filter_index++] = |
|
1361 SDL_Convert16LSB; |
|
1362 cvt->len_mult *= 2; |
|
1363 cvt->len_ratio *= 2; |
|
1364 break; |
|
1365 case AUDIO_U16MSB: |
|
1366 cvt->filters[cvt->filter_index++] = |
|
1367 SDL_Convert16MSB; |
|
1368 cvt->len_mult *= 2; |
|
1369 cvt->len_ratio *= 2; |
|
1370 break; |
|
1371 } |
|
1372 } |
|
1373 |
|
1374 /* Last filter: Mono/Stereo conversion */ |
|
1375 if ( src_channels != dst_channels ) { |
|
1376 if ( (src_channels == 1) && (dst_channels > 1) ) { |
|
1377 cvt->filters[cvt->filter_index++] = |
|
1378 SDL_ConvertStereo; |
|
1379 cvt->len_mult *= 2; |
|
1380 src_channels = 2; |
|
1381 cvt->len_ratio *= 2; |
|
1382 } |
|
1383 if ( (src_channels == 2) && |
|
1384 (dst_channels == 6) ) { |
|
1385 cvt->filters[cvt->filter_index++] = |
|
1386 SDL_ConvertSurround; |
|
1387 src_channels = 6; |
|
1388 cvt->len_mult *= 3; |
|
1389 cvt->len_ratio *= 3; |
|
1390 } |
|
1391 if ( (src_channels == 2) && |
|
1392 (dst_channels == 4) ) { |
|
1393 cvt->filters[cvt->filter_index++] = |
|
1394 SDL_ConvertSurround_4; |
|
1395 src_channels = 4; |
|
1396 cvt->len_mult *= 2; |
|
1397 cvt->len_ratio *= 2; |
|
1398 } |
|
1399 while ( (src_channels*2) <= dst_channels ) { |
|
1400 cvt->filters[cvt->filter_index++] = |
|
1401 SDL_ConvertStereo; |
|
1402 cvt->len_mult *= 2; |
|
1403 src_channels *= 2; |
|
1404 cvt->len_ratio *= 2; |
|
1405 } |
|
1406 if ( (src_channels == 6) && |
|
1407 (dst_channels <= 2) ) { |
|
1408 cvt->filters[cvt->filter_index++] = |
|
1409 SDL_ConvertStrip; |
|
1410 src_channels = 2; |
|
1411 cvt->len_ratio /= 3; |
|
1412 } |
|
1413 if ( (src_channels == 6) && |
|
1414 (dst_channels == 4) ) { |
|
1415 cvt->filters[cvt->filter_index++] = |
|
1416 SDL_ConvertStrip_2; |
|
1417 src_channels = 4; |
|
1418 cvt->len_ratio /= 2; |
|
1419 } |
|
1420 /* This assumes that 4 channel audio is in the format: |
|
1421 Left {front/back} + Right {front/back} |
|
1422 so converting to L/R stereo works properly. |
|
1423 */ |
|
1424 while ( ((src_channels%2) == 0) && |
|
1425 ((src_channels/2) >= dst_channels) ) { |
|
1426 cvt->filters[cvt->filter_index++] = |
|
1427 SDL_ConvertMono; |
|
1428 src_channels /= 2; |
|
1429 cvt->len_ratio /= 2; |
|
1430 } |
|
1431 if ( src_channels != dst_channels ) { |
|
1432 /* Uh oh.. */; |
|
1433 } |
|
1434 } |
|
1435 |
|
1436 /* Do rate conversion */ |
|
1437 cvt->rate_incr = 0.0; |
|
1438 if ( (src_rate/100) != (dst_rate/100) ) { |
|
1439 Uint32 hi_rate, lo_rate; |
|
1440 int len_mult; |
|
1441 double len_ratio; |
|
1442 void (SDLCALL *rate_cvt)(SDL_AudioCVT *cvt, Uint16 format); |
|
1443 |
|
1444 if ( src_rate > dst_rate ) { |
|
1445 hi_rate = src_rate; |
|
1446 lo_rate = dst_rate; |
|
1447 switch (src_channels) { |
|
1448 case 1: rate_cvt = SDL_RateDIV2; break; |
|
1449 case 2: rate_cvt = SDL_RateDIV2_c2; break; |
|
1450 case 4: rate_cvt = SDL_RateDIV2_c4; break; |
|
1451 case 6: rate_cvt = SDL_RateDIV2_c6; break; |
|
1452 default: return -1; |
|
1453 } |
|
1454 len_mult = 1; |
|
1455 len_ratio = 0.5; |
|
1456 } else { |
|
1457 hi_rate = dst_rate; |
|
1458 lo_rate = src_rate; |
|
1459 switch (src_channels) { |
|
1460 case 1: rate_cvt = SDL_RateMUL2; break; |
|
1461 case 2: rate_cvt = SDL_RateMUL2_c2; break; |
|
1462 case 4: rate_cvt = SDL_RateMUL2_c4; break; |
|
1463 case 6: rate_cvt = SDL_RateMUL2_c6; break; |
|
1464 default: return -1; |
|
1465 } |
|
1466 len_mult = 2; |
|
1467 len_ratio = 2.0; |
|
1468 } |
|
1469 /* If hi_rate = lo_rate*2^x then conversion is easy */ |
|
1470 while ( ((lo_rate*2)/100) <= (hi_rate/100) ) { |
|
1471 cvt->filters[cvt->filter_index++] = rate_cvt; |
|
1472 cvt->len_mult *= len_mult; |
|
1473 lo_rate *= 2; |
|
1474 cvt->len_ratio *= len_ratio; |
|
1475 } |
|
1476 /* We may need a slow conversion here to finish up */ |
|
1477 if ( (lo_rate/100) != (hi_rate/100) ) { |
|
1478 #if 1 |
|
1479 /* The problem with this is that if the input buffer is |
|
1480 say 1K, and the conversion rate is say 1.1, then the |
|
1481 output buffer is 1.1K, which may not be an acceptable |
|
1482 buffer size for the audio driver (not a power of 2) |
|
1483 */ |
|
1484 /* For now, punt and hope the rate distortion isn't great. |
|
1485 */ |
|
1486 #else |
|
1487 if ( src_rate < dst_rate ) { |
|
1488 cvt->rate_incr = (double)lo_rate/hi_rate; |
|
1489 cvt->len_mult *= 2; |
|
1490 cvt->len_ratio /= cvt->rate_incr; |
|
1491 } else { |
|
1492 cvt->rate_incr = (double)hi_rate/lo_rate; |
|
1493 cvt->len_ratio *= cvt->rate_incr; |
|
1494 } |
|
1495 cvt->filters[cvt->filter_index++] = SDL_RateSLOW; |
|
1496 #endif |
|
1497 } |
|
1498 } |
|
1499 |
|
1500 /* Set up the filter information */ |
|
1501 if ( cvt->filter_index != 0 ) { |
|
1502 cvt->needed = 1; |
|
1503 cvt->src_format = src_format; |
|
1504 cvt->dst_format = dst_format; |
|
1505 cvt->len = 0; |
|
1506 cvt->buf = NULL; |
|
1507 cvt->filters[cvt->filter_index] = NULL; |
|
1508 } |
|
1509 return(cvt->needed); |
|
1510 } |