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Rev | Author | Line No. | Line |
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8526 | maxcodehac | 1 | /* |
2 | * sound.c |
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3 | * Copyright (C) 1998 Brainchild Design - http://brainchilddesign.com/ |
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4 | * |
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5 | * Copyright (C) 2001 Chuck Mason |
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6 | * |
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7 | * Copyright (C) 2002 Florian Schulze |
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8 | * |
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9 | * This file is part of Jump'n'Bump. |
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10 | * |
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11 | * Jump'n'Bump is free software; you can redistribute it and/or modify |
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12 | * it under the terms of the GNU General Public License as published by |
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13 | * the Free Software Foundation; either version 2 of the License, or |
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14 | * (at your option) any later version. |
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15 | * |
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16 | * Jump'n'Bump is distributed in the hope that it will be useful, |
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17 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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18 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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19 | * GNU General Public License for more details. |
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20 | * |
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21 | * You should have received a copy of the GNU General Public License |
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22 | * along with this program; if not, write to the Free Software |
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23 | * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA |
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24 | */ |
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25 | |||
26 | #include "globals.h" |
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27 | #include |
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28 | #ifndef _MSC_VER |
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29 | #include |
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30 | #endif |
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31 | #include "SDL.h" |
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32 | |||
33 | #ifndef NO_SDL_MIXER |
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34 | #include "SDL_mixer.h" |
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35 | |||
36 | static Mix_Music *current_music = (Mix_Music *) NULL; |
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37 | #endif |
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38 | |||
39 | sfx_data sounds[NUM_SFX]; |
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40 | |||
41 | static int SAMPLECOUNT = 512; |
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42 | |||
43 | #define MAX_CHANNELS 32 |
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44 | |||
45 | typedef struct { |
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46 | /* loop flag */ |
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47 | int loop; |
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48 | /* The channel step amount... */ |
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49 | unsigned int step; |
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50 | /* ... and a 0.16 bit remainder of last step. */ |
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51 | unsigned int stepremainder; |
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52 | unsigned int samplerate; |
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53 | /* The channel data pointers, start and end. */ |
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54 | signed short* data; |
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55 | signed short* startdata; |
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56 | signed short* enddata; |
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57 | /* Hardware left and right channel volume lookup. */ |
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58 | int leftvol; |
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59 | int rightvol; |
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60 | } channel_info_t; |
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61 | |||
62 | channel_info_t channelinfo[MAX_CHANNELS]; |
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63 | |||
64 | /* Sample rate in samples/second */ |
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65 | int audio_rate = 44100; |
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66 | int global_sfx_volume = 0; |
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67 | /* |
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68 | // This function loops all active (internal) sound |
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69 | // channels, retrieves a given number of samples |
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70 | // from the raw sound data, modifies it according |
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71 | // to the current (internal) channel parameters, |
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72 | // mixes the per channel samples into the given |
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73 | // mixing buffer, and clamping it to the allowed |
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74 | // range. |
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75 | // |
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76 | // This function currently supports only 16bit. |
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77 | */ |
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78 | |||
79 | static void stopchan(int i) |
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80 | { |
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81 | if (channelinfo[i].data) { |
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82 | memset(&channelinfo[i], 0, sizeof(channel_info_t)); |
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83 | } |
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84 | } |
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85 | |||
86 | |||
87 | /* |
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88 | // This function adds a sound to the |
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89 | // list of currently active sounds, |
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90 | // which is maintained as a given number |
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91 | // (eight, usually) of internal channels. |
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92 | // Returns a handle. |
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93 | */ |
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94 | int addsfx(signed short *data, int len, int loop, int samplerate, int channel) |
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95 | { |
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96 | stopchan(channel); |
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97 | |||
98 | /* We will handle the new SFX. */ |
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99 | /* Set pointer to raw data. */ |
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100 | channelinfo[channel].data = data; |
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101 | channelinfo[channel].startdata = data; |
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102 | |||
103 | /* Set pointer to end of raw data. */ |
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104 | channelinfo[channel].enddata = channelinfo[channel].data + len - 1; |
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105 | channelinfo[channel].samplerate = samplerate; |
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106 | |||
107 | channelinfo[channel].loop = loop; |
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108 | channelinfo[channel].stepremainder = 0; |
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109 | |||
110 | return channel; |
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111 | } |
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112 | |||
113 | |||
114 | static void updateSoundParams(int slot, int volume) |
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115 | { |
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116 | int rightvol; |
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117 | int leftvol; |
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118 | |||
119 | /* |
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120 | // Set stepping |
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121 | // MWM 2000-12-24: Calculates proportion of channel samplerate |
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122 | // to global samplerate for mixing purposes. |
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123 | // Patched to shift left *then* divide, to minimize roundoff errors |
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124 | // as well as to use SAMPLERATE as defined above, not to assume 11025 Hz |
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125 | */ |
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126 | channelinfo[slot].step = ((channelinfo[slot].samplerate<<16)/audio_rate); |
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127 | |||
128 | leftvol = volume; |
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129 | rightvol= volume; |
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130 | |||
131 | /* Sanity check, clamp volume. */ |
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132 | if (rightvol < 0) |
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133 | rightvol = 0; |
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134 | if (rightvol > 127) |
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135 | rightvol = 127; |
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136 | |||
137 | if (leftvol < 0) |
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138 | leftvol = 0; |
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139 | if (leftvol > 127) |
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140 | leftvol = 127; |
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141 | |||
142 | channelinfo[slot].leftvol = leftvol; |
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143 | channelinfo[slot].rightvol = rightvol; |
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144 | } |
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145 | |||
146 | |||
147 | void mix_sound(void *unused, Uint8 *stream, int len) |
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148 | { |
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149 | /* Mix current sound data. */ |
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150 | /* Data, from raw sound, for right and left. */ |
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151 | register int sample; |
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152 | register int dl; |
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153 | register int dr; |
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154 | |||
155 | /* Pointers in audio stream, left, right, end. */ |
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156 | signed short* leftout; |
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157 | signed short* rightout; |
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158 | signed short* leftend; |
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159 | /* Step in stream, left and right, thus two. */ |
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160 | int step; |
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161 | |||
162 | /* Mixing channel index. */ |
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163 | int chan; |
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164 | |||
165 | /* Left and right channel */ |
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166 | /* are in audio stream, alternating. */ |
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167 | leftout = (signed short *)stream; |
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168 | rightout = ((signed short *)stream)+1; |
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169 | step = 2; |
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170 | |||
171 | /* Determine end, for left channel only */ |
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172 | /* (right channel is implicit). */ |
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173 | leftend = leftout + (len/4)*step; |
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174 | |||
175 | /* Mix sounds into the mixing buffer. */ |
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176 | /* Loop over step*SAMPLECOUNT, */ |
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177 | /* that is 512 values for two channels. */ |
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178 | while (leftout != leftend) { |
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179 | /* Reset left/right value. */ |
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180 | dl = *leftout * 256; |
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181 | dr = *rightout * 256; |
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182 | |||
183 | /* Love thy L2 chache - made this a loop. */ |
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184 | /* Now more channels could be set at compile time */ |
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185 | /* as well. Thus loop those channels. */ |
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186 | for ( chan = 0; chan < MAX_CHANNELS; chan++ ) { |
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187 | /* Check channel, if active. */ |
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188 | if (channelinfo[chan].data) { |
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189 | /* Get the raw data from the channel. */ |
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190 | /* no filtering */ |
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191 | /* sample = *channelinfo[chan].data; */ |
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192 | /* linear filtering */ |
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193 | sample = (int)(((int)channelinfo[chan].data[0] * (int)(0x10000 - channelinfo[chan].stepremainder)) |
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194 | + ((int)channelinfo[chan].data[1] * (int)(channelinfo[chan].stepremainder))) >> 16; |
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195 | |||
196 | /* Add left and right part */ |
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197 | /* for this channel (sound) */ |
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198 | /* to the current data. */ |
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199 | /* Adjust volume accordingly. */ |
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200 | dl += sample * (channelinfo[chan].leftvol * global_sfx_volume) / 128; |
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201 | dr += sample * (channelinfo[chan].rightvol * global_sfx_volume) / 128; |
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202 | /* Increment index ??? */ |
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203 | channelinfo[chan].stepremainder += channelinfo[chan].step; |
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204 | /* MSB is next sample??? */ |
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205 | channelinfo[chan].data += channelinfo[chan].stepremainder >> 16; |
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206 | /* Limit to LSB??? */ |
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207 | channelinfo[chan].stepremainder &= 0xffff; |
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208 | |||
209 | /* Check whether we are done. */ |
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210 | if (channelinfo[chan].data >= channelinfo[chan].enddata) { |
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211 | if (channelinfo[chan].loop) { |
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212 | channelinfo[chan].data = channelinfo[chan].startdata; |
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213 | } else { |
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214 | stopchan(chan); |
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215 | } |
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216 | } |
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217 | } |
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218 | } |
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219 | |||
220 | /* Clamp to range. Left hardware channel. */ |
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221 | /* Has been char instead of short. */ |
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222 | /* if (dl > 127) *leftout = 127; */ |
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223 | /* else if (dl < -128) *leftout = -128; */ |
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224 | /* else *leftout = dl; */ |
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225 | |||
226 | dl = dl / 256; |
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227 | dr = dr / 256; |
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228 | |||
229 | if (dl > SHRT_MAX) |
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230 | *leftout = SHRT_MAX; |
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231 | else if (dl < SHRT_MIN) |
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232 | *leftout = SHRT_MIN; |
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233 | else |
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234 | *leftout = (signed short)dl; |
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235 | |||
236 | /* Same for right hardware channel. */ |
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237 | if (dr > SHRT_MAX) |
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238 | *rightout = SHRT_MAX; |
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239 | else if (dr < SHRT_MIN) |
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240 | *rightout = SHRT_MIN; |
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241 | else |
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242 | *rightout = (signed short)dr; |
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243 | |||
244 | /* Increment current pointers in stream */ |
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245 | leftout += step; |
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246 | rightout += step; |
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247 | } |
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248 | } |
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249 | |||
250 | /* misc handling */ |
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251 | |||
252 | char dj_init(void) |
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253 | { |
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254 | Uint16 audio_format = MIX_DEFAULT_FORMAT; |
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255 | int audio_channels = 2; |
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256 | int audio_buffers = 4096; |
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257 | |||
258 | open_screen(); |
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259 | |||
260 | if (main_info.no_sound) |
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261 | return 0; |
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262 | |||
263 | audio_buffers = SAMPLECOUNT*audio_rate/11025; |
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264 | |||
265 | memset(channelinfo, 0, sizeof(channelinfo)); |
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266 | memset(sounds, 0, sizeof(sounds)); |
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267 | |||
268 | #ifndef NO_SDL_MIXER |
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269 | if (Mix_OpenAudio(audio_rate, audio_format, audio_channels, audio_buffers) < 0) { |
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270 | fprintf(stderr, "Couldn't open audio: %s\n", SDL_GetError()); |
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271 | main_info.no_sound = 1; |
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272 | return 1; |
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273 | } |
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274 | |||
275 | Mix_QuerySpec(&audio_rate, &audio_format, &audio_channels); |
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276 | printf("Opened audio at %dHz %dbit %s, %d bytes audio buffer\n", audio_rate, (audio_format & 0xFF), (audio_channels > 1) ? "stereo" : "mono", audio_buffers); |
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277 | |||
278 | Mix_SetMusicCMD(getenv("MUSIC_CMD")); |
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279 | |||
280 | Mix_SetPostMix(mix_sound, NULL); |
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281 | #else |
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282 | main_info.no_sound = 1; |
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283 | return 1; |
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284 | #endif |
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285 | |||
286 | return 0; |
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287 | } |
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288 | |||
289 | void dj_deinit(void) |
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290 | { |
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291 | if (main_info.no_sound) |
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292 | return; |
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293 | |||
294 | #ifndef NO_SDL_MIXER |
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295 | Mix_HaltMusic(); |
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296 | if (current_music) |
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297 | Mix_FreeMusic(current_music); |
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298 | current_music = NULL; |
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299 | |||
300 | Mix_CloseAudio(); |
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301 | #endif |
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302 | |||
303 | SDL_Quit(); |
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304 | } |
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305 | |||
306 | void dj_start(void) |
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307 | { |
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308 | } |
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309 | |||
310 | void dj_stop(void) |
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311 | { |
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312 | } |
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313 | |||
314 | char dj_autodetect_sd(void) |
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315 | { |
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316 | return 0; |
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317 | } |
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318 | |||
319 | char dj_set_stereo(char flag) |
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320 | { |
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321 | return 0; |
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322 | } |
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323 | |||
324 | void dj_set_auto_mix(char flag) |
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325 | { |
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326 | } |
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327 | |||
328 | unsigned short dj_set_mixing_freq(unsigned short freq) |
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329 | { |
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330 | return freq; |
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331 | } |
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332 | |||
333 | void dj_set_dma_time(unsigned short time) |
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334 | { |
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335 | } |
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336 | |||
337 | void dj_set_nosound(char flag) |
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338 | { |
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339 | } |
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340 | |||
341 | /* mix handling */ |
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342 | |||
343 | void dj_mix(void) |
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344 | { |
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345 | } |
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346 | |||
347 | /* sfx handling */ |
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348 | |||
349 | char dj_set_num_sfx_channels(char num_channels) |
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350 | { |
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351 | return num_channels; |
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352 | } |
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353 | |||
354 | void dj_set_sfx_volume(char volume) |
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355 | { |
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356 | if (main_info.no_sound) |
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357 | return; |
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358 | |||
359 | SDL_LockAudio(); |
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360 | global_sfx_volume = volume*2; |
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361 | SDL_UnlockAudio(); |
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362 | } |
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363 | |||
364 | void dj_play_sfx(unsigned char sfx_num, unsigned short freq, char volume, char panning, unsigned short delay, char channel) |
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365 | { |
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366 | int slot; |
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367 | |||
368 | if (main_info.music_no_sound || main_info.no_sound) |
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369 | return; |
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370 | |||
371 | if (channel<0) { |
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372 | for (slot=0; slot |
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373 | if (channelinfo[slot].data==NULL) |
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374 | break; |
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375 | if (slot>=MAX_CHANNELS) |
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376 | return; |
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377 | } else |
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378 | slot = channel; |
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379 | |||
380 | SDL_LockAudio(); |
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381 | addsfx((short *)sounds[sfx_num].buf, sounds[sfx_num].length, sounds[sfx_num].loop, freq, slot); |
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382 | updateSoundParams(slot, volume*2); |
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383 | SDL_UnlockAudio(); |
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384 | } |
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385 | |||
386 | char dj_get_sfx_settings(unsigned char sfx_num, sfx_data *data) |
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387 | { |
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388 | if (main_info.no_sound) |
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389 | return 0; |
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390 | |||
391 | memcpy(data, &sounds[sfx_num], sizeof(sfx_data)); |
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392 | return 0; |
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393 | } |
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394 | |||
395 | char dj_set_sfx_settings(unsigned char sfx_num, sfx_data *data) |
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396 | { |
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397 | if (main_info.no_sound) |
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398 | return 0; |
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399 | |||
400 | memcpy(&sounds[sfx_num], data, sizeof(sfx_data)); |
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401 | return 0; |
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402 | } |
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403 | |||
404 | void dj_set_sfx_channel_volume(char channel_num, char volume) |
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405 | { |
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406 | if (main_info.no_sound) |
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407 | return; |
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408 | |||
409 | SDL_LockAudio(); |
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410 | updateSoundParams(channel_num, volume*2); |
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411 | SDL_UnlockAudio(); |
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412 | } |
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413 | |||
414 | void dj_stop_sfx_channel(char channel_num) |
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415 | { |
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416 | if (main_info.no_sound) |
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417 | return; |
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418 | |||
419 | SDL_LockAudio(); |
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420 | stopchan(channel_num); |
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421 | SDL_UnlockAudio(); |
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422 | } |
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423 | |||
424 | char dj_load_sfx(unsigned char * file_handle, char *filename, int file_length, char sfx_type, unsigned char sfx_num) |
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425 | { |
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426 | unsigned int i; |
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427 | unsigned char *src; |
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428 | unsigned short *dest; |
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429 | |||
430 | if (main_info.no_sound) |
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431 | return 0; |
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432 | |||
433 | sounds[sfx_num].buf = malloc(file_length); |
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434 | |||
435 | memcpy(sounds[sfx_num].buf, file_handle, file_length); |
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436 | |||
437 | sounds[sfx_num].length = file_length / 2; |
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438 | src = sounds[sfx_num].buf; |
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439 | dest = (unsigned short *)sounds[sfx_num].buf; |
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440 | for (i=0; i |
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441 | { |
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442 | unsigned short temp; |
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443 | temp = src[0] + (src[1] << 8); |
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444 | *dest = temp; |
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445 | src += 2; |
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446 | dest++; |
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447 | } |
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448 | return 0; |
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449 | } |
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450 | |||
451 | void dj_free_sfx(unsigned char sfx_num) |
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452 | { |
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453 | if (main_info.no_sound) |
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454 | return; |
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455 | |||
456 | free(sounds[sfx_num].buf); |
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457 | memset(&sounds[sfx_num], 0, sizeof(sfx_data)); |
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458 | } |
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459 | |||
460 | /* mod handling */ |
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461 | |||
462 | char dj_ready_mod(char mod_num) |
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463 | { |
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464 | #ifndef NO_SDL_MIXER |
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465 | FILE *tmp; |
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466 | # if ((defined _MSC_VER) || (defined __MINGW32__)) |
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467 | char filename[] = "jnb.tmpmusic.mod"; |
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468 | # else |
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469 | char filename[] = "/tmp/jnb.tmpmusic.mod"; |
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470 | # endif |
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471 | unsigned char *fp; |
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472 | int len; |
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473 | |||
474 | if (main_info.no_sound) |
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475 | return 0; |
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476 | |||
477 | switch (mod_num) { |
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478 | case MOD_MENU: |
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479 | fp = dat_open("jump.mod"); |
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480 | len = dat_filelen("jump.mod"); |
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481 | break; |
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482 | case MOD_GAME: |
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483 | fp = dat_open("bump.mod"); |
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484 | len = dat_filelen("bump.mod"); |
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485 | break; |
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486 | case MOD_SCORES: |
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487 | fp = dat_open("scores.mod"); |
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488 | len = dat_filelen("scores.mod"); |
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489 | break; |
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490 | default: |
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491 | fprintf(stderr, "bogus parameter to dj_ready_mod()\n"); |
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492 | fp = NULL; |
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493 | len = 0; |
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494 | break; |
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495 | } |
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496 | |||
497 | if (Mix_PlayingMusic()) |
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498 | Mix_FadeOutMusic(1500); |
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499 | |||
500 | if (current_music) { |
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501 | Mix_FreeMusic(current_music); |
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502 | current_music = NULL; |
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503 | } |
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504 | |||
505 | if (fp == NULL) { |
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506 | return 0; |
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507 | } |
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508 | |||
509 | tmp = fopen(filename, "wb"); |
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510 | if (tmp) { |
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511 | fwrite(fp, len, 1, tmp); |
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512 | fflush(tmp); |
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513 | fclose(tmp); |
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514 | } |
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515 | |||
516 | current_music = Mix_LoadMUS(filename); |
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517 | unlink(filename); |
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518 | if (current_music == NULL) { |
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519 | fprintf(stderr, "Couldn't load music: %s\n", SDL_GetError()); |
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520 | return 0; |
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521 | } |
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522 | |||
523 | #endif |
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524 | |||
525 | return 0; |
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526 | } |
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527 | |||
528 | char dj_start_mod(void) |
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529 | { |
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530 | #ifndef NO_SDL_MIXER |
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531 | if (main_info.no_sound) |
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532 | return 0; |
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533 | |||
534 | Mix_VolumeMusic(0); |
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535 | Mix_PlayMusic(current_music, -1); |
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536 | #endif |
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537 | |||
538 | return 0; |
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539 | } |
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540 | |||
541 | void dj_stop_mod(void) |
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542 | { |
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543 | #ifndef NO_SDL_MIXER |
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544 | if (main_info.no_sound) |
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545 | return; |
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546 | |||
547 | Mix_HaltMusic(); |
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548 | #endif |
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549 | } |
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550 | |||
551 | void dj_set_mod_volume(char volume) |
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552 | { |
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553 | #ifndef NO_SDL_MIXER |
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554 | if (main_info.no_sound) |
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555 | return; |
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556 | |||
557 | Mix_VolumeMusic(volume); |
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558 | #endif |
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559 | } |
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560 | |||
561 | char dj_load_mod(unsigned char * file_handle, char *filename, char mod_num) |
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562 | { |
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563 | return 0; |
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564 | } |
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565 | |||
566 | void dj_free_mod(char mod_num) |
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567 | { |
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568 | }><>0)>>>>>>>>16)/audio_rate); |