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annotate SDL_ALmixer.h @ 1:a8a8fe374984

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author Eric Wing <ewing . public |-at-| gmail . com>
date Wed, 27 Oct 2010 16:51:16 -0700
parents 01e39f9f58d5
children 279d0427ef26
rev   line source
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1 /*
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2 SDL_ALmixer: A library to make playing sounds and music easier,
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3 which uses OpenAL to manage sounds and SDL_Sound (by Ryan C. Gordon)
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4 to decode files.
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5 Copyright 2002 Eric Wing
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6
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7 This library is free software; you can redistribute it and/or
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8 modify it under the terms of the GNU Library General Public
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9 License as published by the Free Software Foundation; either
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10 version 2 of the License, or (at your option) any later version.
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11
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12 This library is distributed in the hope that it will be useful,
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13 but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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15 Library General Public License for more details.
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16
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17 You should have received a copy of the GNU Library General Public
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18 License along with this library; if not, write to the Free
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19 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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20
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21 */
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22 #ifndef _SDL_ALMIXER_H_
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23 #define _SDL_ALMIXER_H_
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24
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25 #include "SDL_types.h"
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26 #include "SDL_rwops.h"
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27 #include "SDL_error.h"
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28 #include "SDL_version.h"
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29 /*
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30 #include "SDL_audio.h"
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31 #include "SDL_byteorder.h"
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32 */
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33
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34 /*
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35 #include "begin_code.h"
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36 */
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37
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38 /*
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39 #include "SDL_sound.h"
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40 */
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41 /* Crap! altypes.h is missing from 1.1
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42 #include "altypes.h"
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43 */
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44 #include "al.h"
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45
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46 /* Set up for C function definitions, even when using C++ */
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47 #ifdef __cplusplus
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48 extern "C" {
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49 #endif
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50
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51 /* Printable format: "%d.%d.%d", MAJOR, MINOR, PATCHLEVEL
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52 */
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53 #define ALMIXER_MAJOR_VERSION 0
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54 #define ALMIXER_MINOR_VERSION 1
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55 #define ALMIXER_PATCHLEVEL 0
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56
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57 /* This macro can be used to fill a version structure with the compile-time
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58 * version of the SDL_mixer library.
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59 */
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60 #define ALMIXER_VERSION(X) \
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61 { \
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62 (X)->major = ALMIXER_MAJOR_VERSION; \
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63 (X)->minor = ALMIXER_MINOR_VERSION; \
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64 (X)->patch = ALMIXER_PATCHLEVEL; \
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65 }
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66
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67 /* This function gets the version of the dynamically linked SDL_ALmixer library.
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68 it should NOT be used to fill a version structure, instead you should
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69 use the ALMIXER_VERSION() macro.
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70 */
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71 extern DECLSPEC const SDL_version * SDLCALL ALmixer_Linked_Version();
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72
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73 /*
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74 #define ALmixer_AudioInfo Sound_AudioInfo
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75 */
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76
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77 #define ALMIXER_DEFAULT_FREQUENCY 44100
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78 #define ALMIXER_DEFAULT_REFRESH 0
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79 #define ALMIXER_DEFAULT_NUM_CHANNELS 16
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80 #define ALMIXER_DEFAULT_NUM_SOURCES ALMIXER_DEFAULT_NUM_CHANNELS
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81
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82 #define ALMIXER_DEFAULT_BUFFERSIZE 32768
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83 /* #define ALMIXER_DEFAULT_BUFFERSIZE 16384 */
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84
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85 /* Default Queue Buffers must be at least 2 */
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86 #define ALMIXER_DEFAULT_QUEUE_BUFFERS 5
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87 /* Default startup buffers should be at least 1 */
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88 #define ALMIXER_DEFAULT_STARTUP_BUFFERS 2
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89
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90 /*
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91 #define ALMIXER_DECODE_STREAM 0
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92 #define ALMIXER_DECODE_ALL 1
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93 */
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94
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95
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96 #define ALmixer_GetError SDL_GetError
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97 #define ALmixer_SetError SDL_SetError
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98
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99
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100 /* This is a trick I picked up from Lua. Doing the typedef separately
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101 * (and I guess before the definition) instead of a single
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102 * entry: typedef struct {...} YourName; seems to allow me
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103 * to use forward declarations. Doing it the other way (like SDL)
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104 * seems to prevent me from using forward declarions as I get conflicting
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105 * definition errors. I don't really understand why though.
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106 */
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107 typedef struct ALmixer_Data ALmixer_Data;
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108 typedef struct ALmixer_AudioInfo ALmixer_AudioInfo;
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109
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110 /**
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111 * Equvialent to the Sound_AudioInfo struct in SDL_sound.
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112 * Originally, I just used the Sound_AudioInfo directly, but
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113 * I've been trying to reduce the header dependencies for this file.
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114 * But more to the point, I've been interested in dealing with the
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115 * WinMain override problem Josh faced when trying to use SDL components
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116 * in an MFC app which didn't like losing control of WinMain.
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117 * My theory is that if I can purge the header of any thing that
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118 * #include's SDL_main.h, then this might work.
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119 * So I am now introducing my own AudioInfo struct.
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120 */
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121 struct ALmixer_AudioInfo
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122 {
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123 Uint16 format; /**< Equivalent of SDL_AudioSpec.format. */
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124 Uint8 channels; /**< Number of sound channels. 1 == mono, 2 == stereo. */
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125 Uint32 rate; /**< Sample rate; frequency of sample points per second. */
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126 };
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127
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128
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129 #if 0
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130 typedef struct {
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131 Sound_Sample* sample;
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132 Mix_Chunk** chunk; /* provide two chunks for double buffering */
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133 Uint8** double_buffer; /* Only used for streaming */
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134 Uint8 active_buffer; /* used to index the above chunk */
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135 void (*channel_done_callback)(int channel);
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136 } ALmixer_Chunk;
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137 #endif
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138
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139 /*
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140 extern DECLSPEC int SDLCALL ALmixer_Init(int frequency, Uint16 format, int channels, int chunksize);
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141 */
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142 /* Frequency == 0 means ALMIXER_DEFAULT_FREQUENCY */
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143 /* This is the recommended Init function. This will initialize the context, SDL_sound,
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144 * and the mixer system. If you attempt to bypass this function, you do so at
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145 * your own risk.
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146 */
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147 extern DECLSPEC Sint32 SDLCALL ALmixer_Init(Uint32 frequency, Sint32 num_sources, Uint32 refresh);
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148
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149 /* This is a backdoor in case you need to initialize the AL context and
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150 * the mixer system separately. I strongly recommend avoiding these two functions
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151 * and use the normal Init() function.
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152 */
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153 /* Init_Context will only initialize the OpenAL context (and not the mixer part).
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154 * Note that SDL_Sound is also initialized here because load order matters
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155 * because SDL audio will conflict with OpenAL when using SMPEG. This is only
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156 * provided as a backdoor and is not recommended.
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157 */
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158 extern DECLSPEC Sint32 SDLCALL ALmixer_Init_Context(Uint32 frequency, Uint32 refresh);
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159 /* Init_Mixer will only initialize the Mixer system. This is provided in the case
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160 * that you need control over the loading of the context. You may load the context
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161 * yourself, and then call this function. This is not recommended practice, but is
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162 * provided as a backdoor in case you have good reason to
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163 * do this. Be warned that if ALmixer_Init_Mixer() fails,
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164 * it will not clean up the AL context. Also be warned that Quit() still does try to
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165 * clean up everything.
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166 */
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167 extern DECLSPEC Sint32 SDLCALL ALmixer_Init_Mixer(Sint32 num_sources);
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168
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169 extern DECLSPEC void SDLCALL ALmixer_Quit();
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170 extern DECLSPEC SDL_bool SDLCALL ALmixer_IsInitialized();
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171
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172 extern DECLSPEC Uint32 SDLCALL ALmixer_GetFrequency();
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173
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174 extern DECLSPEC Sint32 SDLCALL ALmixer_AllocateChannels(Sint32 numchans);
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175 extern DECLSPEC Sint32 SDLCALL ALmixer_ReserveChannels(Sint32 num);
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176
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177 extern DECLSPEC ALmixer_Data * SDLCALL ALmixer_LoadSample_RW(SDL_RWops* rwops, const char* fileext, Uint32 buffersize, SDL_bool decode_mode_is_predecoded, Uint32 max_queue_buffers, Uint32 num_startup_buffers, SDL_bool access_data);
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178
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179
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180 #define ALmixer_LoadStream_RW(rwops,fileext,buffersize,max_queue_buffers,num_startup_buffers,access_data) ALmixer_LoadSample_RW(rwops,fileext,buffersize, SDL_FALSE, max_queue_buffers, num_startup_buffers,access_data)
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181
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182 #define ALmixer_LoadAll_RW(rwops,fileext,buffersize,access_data) ALmixer_LoadSample_RW(rwops,fileext,buffersize, SDL_TRUE, 0, 0,access_data)
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183
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184
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185
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186 extern DECLSPEC ALmixer_Data * SDLCALL ALmixer_LoadSample(const char* filename, Uint32 buffersize, SDL_bool decode_mode_is_predecoded, Uint32 max_queue_buffers, Uint32 num_startup_buffers, SDL_bool access_data);
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187
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188
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189 #define ALmixer_LoadStream(filename,buffersize,max_queue_buffers,num_startup_buffers,access_data) ALmixer_LoadSample(filename,buffersize, SDL_FALSE, max_queue_buffers, num_startup_buffers,access_data)
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190
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191 #define ALmixer_LoadAll(filename,buffersize,access_data) ALmixer_LoadSample(filename,buffersize, SDL_TRUE, 0, 0,access_data)
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192
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193
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194 extern DECLSPEC ALmixer_Data * SDLCALL ALmixer_LoadSample_RAW_RW(SDL_RWops* rwops, const char* fileext, ALmixer_AudioInfo* desired, Uint32 buffersize, SDL_bool decode_mode_is_predecoded, Uint32 max_queue_buffers, Uint32 num_startup_buffers, SDL_bool access_data);
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195
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196 #define ALmixer_LoadStream_RAW_RW(rwops,fileext,desired,buffersize,max_queue_buffers,num_startup_buffers,access_data) ALmixer_LoadSample_RAW_RW(rwops,fileext,desired,buffersize, SDL_FALSE, max_queue_buffers, num_startup_buffers,access_data)
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197
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198 #define ALmixer_LoadAll_RAW_RW(rwops,fileext,desired,buffersize,access_data) ALmixer_LoadSample_RAW_RW(rwops,fileext,desired,buffersize, SDL_TRUE, 0, 0,access_data)
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199
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200 extern DECLSPEC ALmixer_Data * SDLCALL ALmixer_LoadSample_RAW(const char* filename, ALmixer_AudioInfo* desired, Uint32 buffersize, SDL_bool decode_mode_is_predecoded, Uint32 max_queue_buffers, Uint32 num_startup_buffers, SDL_bool access_data);
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201
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202
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203
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204 extern DECLSPEC void SDLCALL ALmixer_FreeData(ALmixer_Data* data);
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205
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206 extern DECLSPEC Sint32 SDLCALL ALmixer_GetTotalTime(ALmixer_Data* data);
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207
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208
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209 /* If not using threads, this function must be periodically called
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210 * to poll ALmixer to force streamed music and other events to
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211 * take place. If threads are enabled, then this function just
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212 * returns 0.
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213 */
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214 extern DECLSPEC Sint32 SDLCALL ALmixer_Update();
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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 /* Play a sound on a channel with a time limit */
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220 extern DECLSPEC Sint32 SDLCALL ALmixer_PlayChannelTimed(Sint32 channel, ALmixer_Data* data, Sint32 loops, Sint32 ticks);
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221
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222 /* The same as above, but the sound is played without time limits */
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223 #define ALmixer_PlayChannel(channel,data,loops) ALmixer_PlayChannelTimed(channel,data,loops,-1)
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224 /* These functions are the same as PlayChannel*(), but use sources
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225 * instead of channels
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226 */
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227 extern DECLSPEC ALuint SDLCALL ALmixer_PlaySourceTimed(ALuint source, ALmixer_Data* data, Sint32 loops, Sint32 ticks);
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228
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229 #define ALmixer_PlaySource(source,data,loops) ALmixer_PlaySourceTimed(source,data,loops,-1)
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230
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231 /* This function will look up the source for the corresponding channel.
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232 * If -1 is supplied, it will try to return a source not in use
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233 * Must return 0 on error instead of -1 because of unsigned int
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234 */
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235 extern DECLSPEC ALuint SDLCALL ALmixer_GetSource(Sint32 channel);
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236 /* This function will look up the channel for the corresponding source.
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237 * If -1 is supplied, it will try to return the first channel not in use.
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238 * Returns -1 on error, or the channel.
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239 */
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240 extern DECLSPEC Sint32 SDLCALL ALmixer_GetChannel(ALuint source);
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241
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242 extern DECLSPEC Sint32 SDLCALL ALmixer_FindFreeChannel(Sint32 start_channel);
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243
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244 extern DECLSPEC void SDLCALL ALmixer_ChannelFinished(void (*channel_finished)(Sint32 channel, void* userdata), void* userdata);
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245
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246 /*
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247 extern DECLSPEC void SDLCALL ALmixer_ChannelData(void (*channel_data)(Sint32 which_chan, Uint8* data, Uint32 num_bytes, Uint32 frequency, Uint8 channels, Uint8 bitdepth, Uint16 format, Uint8 decode_mode));
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248 */
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249 /**
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250 * Audio data callback system.
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251 * This is a callback function pointer that when set, will trigger a function
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252 * anytime there is new data loaded for a sample. The appropriate load
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253 * parameter must be set in order for a sample to appear here.
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254 * Keep in mind the the current backend implementation must do an end run
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255 * around OpenAL because OpenAL lacks support for this kind of thing.
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256 * As such, buffers are copied at decode time, and there is no attempt to do
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257 * fine grained timing syncronization. You will be provided the entire buffer
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258 * that is decoded regardless of length. So if you predecoded the entire
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259 * audio file, the entire data buffer will be provided in a single callback.
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260 * If you stream the data, you will be getting chunk sizes that are the same as
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261 * what you specified the decode size to be. Unfortunely, this means if you
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262 * pick smaller buffers, you get finer detail at the expense/risk of buffer
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263 * underruns. If you decode more data, you have to deal with the syncronization
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264 * issues if you want to display the data during playback in something like an
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265 * oscilloscope.
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266 *
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267 * @param which_chan The ALmixer channel that the data is currently playing on.
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268 * @param data This is a pointer to the data buffer containing ALmixer's
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269 * version of the decoded data. Consider this data as read-only. In the
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270 * non-threaded backend, this data will persist until potentially the next call
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271 * to Update(). Currently, data buffers are preallocated and not destroyed
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272 * until FreeData() is called (though this behavior is subject to change),
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273 * but the contents will change when the buffer needs to be reused for a
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274 * future callback. The buffer reuse is tied to the amount of buffers that
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275 * may be queued.
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276 * But assuming I don't change this, this may allow for some optimization
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277 * so you can try referencing data from these buffers without worrying
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278 * about crashing. (You still need to be aware that the data could be
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279 * modified behind the scenes on an Update().)
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280 *
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281 * The data type listed is an Unsigned 8-bit format, but the real data may
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282 * not actually be this. Uint8 was chosen as a convenience. If you have
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283 * a 16 bit format, you will want to cast the data and also divide the num_bytes
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284 * by 2. Typically, data is either Sint16 or Uint8. This seems to be a
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285 * convention audio people seem to follow though I'm not sure what the
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286 * underlying reasons (if any) are for this. I suspect that there may be
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287 * some nice alignment/conversion property if you need to cast from Uint8
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288 * to Sint16.
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289 *
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290 * @param num_bytes This is the total length of the data buffer. It presumes
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291 * that this length is measured for Uint8. So if you have Sint16 data, you
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292 * should divide num_bytes by two if you access the data as Sint16.
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293 *
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294 * @param frequency The frequency the data was decoded at.
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295 *
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296 * @param channels 1 for mono, 2 for stereo.
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297 *
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298 * @param bit_depth Bits per sample. This is expected to be 8 or 16. This
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299 * number will tell you if you if you need to treat the data buffer as
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300 * 16 bit or not.
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301 *
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302 * @param is_unsigned 1 if the data is unsigned, 0 if signed. Using this
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303 * combined with bit_depth will tell you if you need to treat the data
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304 * as Uint8, Sint8, Uint32, or Sint32.
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305 *
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306 * @param decode_mode_is_predecoded This is here to tell you if the data was totally
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307 * predecoded or loaded as a stream. If predecoded, you will only get
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308 * one data callback per playback instance. (This might also be true for
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309 * looping the same sample...I don't remember how it was implemented.
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310 * Maybe this should be fixed.)
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311 * 0 (ALMIXER_DECODE_STREAM) for streamed.
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312 * 1 (ALMIXER_DECODE_ALL) for predecoded.
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313 *
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314 * @param length_in_msec This returns the total length (time) of the data
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315 * buffer in milliseconds. This could be computed yourself, but is provided
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316 * as a convenince.
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317 *
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318 *
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319 */
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320 extern DECLSPEC void SDLCALL ALmixer_ChannelData(void (*channel_data)(Sint32 which_chan, Uint8* data, Uint32 num_bytes, Uint32 frequency, Uint8 channels, Uint8 bit_depth, SDL_bool is_unsigned, SDL_bool decode_mode_is_predecoded, Uint32 length_in_msec, void* user_data), void* user_data);
0
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321
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322
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323 extern DECLSPEC Sint32 SDLCALL ALmixer_HaltChannel(Sint32 channel);
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324 extern DECLSPEC Sint32 SDLCALL ALmixer_HaltSource(ALuint source);
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325
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326
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327 extern DECLSPEC Sint32 SDLCALL ALmixer_RewindData(ALmixer_Data* data);
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328
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329 /* If decoded all, rewind will instantly rewind it. Data is not
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330 * affected, so it will start at the "Seek"'ed positiond.
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331 * Streamed data will rewind the actual data, but the effect
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332 * will not be noticed until the currently buffered data is played.
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333 * Use Halt before this call for instantaneous changes
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334 */
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335 extern DECLSPEC Sint32 SDLCALL ALmixer_RewindChannel(Sint32 channel);
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336 extern DECLSPEC Sint32 SDLCALL ALmixer_RewindSource(ALuint source);
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337
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338 extern DECLSPEC Sint32 SDLCALL ALmixer_PauseChannel(Sint32 channel);
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339 extern DECLSPEC Sint32 SDLCALL ALmixer_PauseSource(ALuint source);
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340
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341 extern DECLSPEC Sint32 SDLCALL ALmixer_ResumeChannel(Sint32 channel);
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342 extern DECLSPEC Sint32 SDLCALL ALmixer_ResumeSource(ALuint source);
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343
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344 extern DECLSPEC Sint32 SDLCALL ALmixer_Seek(ALmixer_Data* data, Uint32 msec);
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345
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346
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347 extern DECLSPEC Sint32 SDLCALL ALmixer_FadeInChannelTimed(Sint32 channel, ALmixer_Data* data, Sint32 loops, Uint32 fade_ticks, Sint32 expire_ticks);
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348
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349 #define ALmixer_FadeInChannel(channel,data,loops,fade_ticks) ALmixer_FadeInChannelTimed(channel,data,loops,fade_ticks,-1)
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350
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351 extern DECLSPEC ALuint SDLCALL ALmixer_FadeInSourceTimed(ALuint source, ALmixer_Data* data, Sint32 loops, Uint32 fade_ticks, Sint32 expire_ticks);
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352
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353 #define ALmixer_FadeInSource(source,data,loops,fade_ticks) ALmixer_FadeInSourceTimed(source,data,loops,fade_ticks,-1)
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354
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355 extern DECLSPEC Sint32 SDLCALL ALmixer_FadeOutChannel(Sint32 channel, Uint32 ticks);
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356 extern DECLSPEC Sint32 SDLCALL ALmixer_FadeOutSource(ALuint source, Uint32 ticks);
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357
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358 extern DECLSPEC Sint32 SDLCALL ALmixer_FadeChannel(Sint32 channel, Uint32 ticks, ALfloat volume);
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359 extern DECLSPEC Sint32 SDLCALL ALmixer_FadeSource(ALuint source, Uint32 ticks, ALfloat volume);
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360
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361 extern DECLSPEC Sint32 SDLCALL ALmixer_SetMaxVolumeChannel(Sint32 channel, ALfloat volume);
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362 extern DECLSPEC Sint32 SDLCALL ALmixer_SetMaxVolumeSource(ALuint source, ALfloat volume);
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363 extern DECLSPEC ALfloat SDLCALL ALmixer_GetMaxVolumeChannel(Sint32 channel);
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364 extern DECLSPEC ALfloat SDLCALL ALmixer_GetMaxVolumeSource(ALuint source);
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365
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366 extern DECLSPEC Sint32 SDLCALL ALmixer_SetMinVolumeChannel(Sint32 channel, ALfloat volume);
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367 extern DECLSPEC Sint32 SDLCALL ALmixer_SetMinVolumeSource(ALuint source, ALfloat volume);
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368 extern DECLSPEC ALfloat SDLCALL ALmixer_GetMinVolumeChannel(Sint32 channel);
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369 extern DECLSPEC ALfloat SDLCALL ALmixer_GetMinVolumeSource(ALuint source);
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370
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371
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372 extern DECLSPEC Sint32 SDLCALL ALmixer_SetMasterVolume(ALfloat volume);
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373 extern DECLSPEC ALfloat SDLCALL ALmixer_GetMasterVolume();
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374
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375
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376 extern DECLSPEC Sint32 SDLCALL ALmixer_ExpireChannel(Sint32 channel, Sint32 ticks);
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377 extern DECLSPEC Sint32 SDLCALL ALmixer_ExpireSource(ALuint source, Sint32 ticks);
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378
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379 extern DECLSPEC Sint32 SDLCALL ALmixer_QueryChannel(Sint32 channel);
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380 extern DECLSPEC Sint32 SDLCALL ALmixer_QuerySource(ALuint source);
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381 extern DECLSPEC Sint32 SDLCALL ALmixer_PlayingChannel(Sint32 channel);
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382 extern DECLSPEC Sint32 SDLCALL ALmixer_PlayingSource(ALuint source);
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383 extern DECLSPEC Sint32 SDLCALL ALmixer_PausedChannel(Sint32 channel);
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384 extern DECLSPEC Sint32 SDLCALL ALmixer_PausedSource(ALuint source);
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385
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386 extern DECLSPEC Sint32 SDLCALL ALmixer_CountAllFreeChannels();
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387 extern DECLSPEC Sint32 SDLCALL ALmixer_CountUnreservedFreeChannels();
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388 extern DECLSPEC Sint32 SDLCALL ALmixer_CountAllUsedChannels();
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389 extern DECLSPEC Sint32 SDLCALL ALmixer_CountUnreservedUsedChannels();
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390 #define ALmixer_CountTotalChannels() ALmixer_AllocateChannels(-1)
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391 #define ALmixer_CountReservedChannels() ALmixer_ReserveChannels(-1)
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392
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393 extern DECLSPEC SDL_bool SDLCALL ALmixer_IsPredecoded(ALmixer_Data* data);
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394
0
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395
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396
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397 /* For testing */
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398 #if 0
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399 extern DECLSPEC void SDLCALL ALmixer_Output_Attributes();
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400 #endif
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401 extern DECLSPEC void SDLCALL ALmixer_Output_Decoders();
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402 extern DECLSPEC void SDLCALL ALmixer_Output_OpenAL_Info();
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403
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404 #if 0
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405
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406
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407
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408 extern DECLSPEC Uint32 SDLCALL ALmixer_Volume(Sint32 channel, Sint32 volume);
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409
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410
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411 /* I'm going to blindly throw in the Mixer effects sections and
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412 * hope they work.
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413 */
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414 #define ALmixer_EffectFunc_t Mix_EffectFunc_t
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415 #define ALmixer_EffectDone_t Mix_EffectDone_t
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416 /*
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417 #define ALmixer_RegisterEffect Mix_RegisterEffect
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418 #define ALmixer_UnregisterEffect Mix_UnregisterEffect
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419 #define ALmixer_UnregisterAllEffects Mix_RegisterEffect
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420 */
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421
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422 #define ALmixer_SetPostMix Mix_SetPostMix
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423 #define ALmixer_SetPanning Mix_SetPanning
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424 #define ALmixer_SetDistance Mix_SetDistance
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425 #define ALmixer_SetPosition Mix_SetPosition
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426 #define ALmixer_SetReverseStereo Mix_SetReverseStereo
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427
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428 /* Unfortunately, effects have a nasty behavior of unregistering
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429 * themselves after the channel finishes. This is incompatible
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430 * with the streaming system that this library uses.
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431 * Implementing a proper effects system will take more time.
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432 * For now, I need to be able to retrieve the playing data
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433 * for an oscilloscope, so I am hacking together a 1 effect
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434 * system. You can't have more than one.
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435 */
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436
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437 extern DECLSPEC Sint32 SDLCALL ALmixer_RegisterEffect(Sint32 chan, ALmixer_EffectFunc_t f, ALmixer_EffectDone_t d, void* arg);
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438
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439 extern DECLSPEC Sint32 SDLCALL ALmixer_UnregisterEffect(Sint32 chan, ALmixer_EffectFunc_t f);
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440
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441 extern DECLSPEC Sint32 SDLCALL ALmixer_UnregisterAllEffects(Sint32 chan);
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442
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443 #endif
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444
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445
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446
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447
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448 /* Ends C function definitions when using C++ */
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449 #ifdef __cplusplus
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450 }
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451 #endif
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452 /*
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453 #include "close_code.h"
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454 */
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455
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456 #endif /* _SDL_ALMIXER_H_ */
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457
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458 /* end of SDL_ALmixer.h ... */
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459
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460