Mercurial > almixer_isolated
annotate ALmixer.h @ 3:a929285e1db0
Added CMake build system.
There are problems with the SDL_sound module due to changes in CMake. Right now they just seem to be warnings, but I am unable to suppress them.
Added license.
Added README.
author | Eric Wing <ewing . public |-at-| gmail . com> |
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date | Wed, 27 Oct 2010 20:43:14 -0700 |
parents | SDL_ALmixer.h@279d0427ef26 |
children | 8cb13d89451a |
rev | line source |
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0 | 1 /* |
2
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2 ALmixer: A library to make playing pre-loaded sounds and streams easier |
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3 with high performance and potential access to OpenAL effects. |
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4 Copyright 2002, 2010 Eric Wing <ewing . public @ playcontrol.net> |
0 | 5 |
6 This library is free software; you can redistribute it and/or | |
7 modify it under the terms of the GNU Library General Public | |
8 License as published by the Free Software Foundation; either | |
9 version 2 of the License, or (at your option) any later version. | |
10 | |
11 This library is distributed in the hope that it will be useful, | |
12 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
13 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
14 Library General Public License for more details. | |
15 | |
16 You should have received a copy of the GNU Library General Public | |
17 License along with this library; if not, write to the Free | |
18 Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
19 | |
20 */ | |
2
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21 |
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22 |
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23 /** |
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24 * @mainpage |
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25 * ALmixer (which I sometimes call "SDL-OpenAL-Mixer" or "SDL_ALmixer") is a cross-platform audio library built |
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26 * on top of OpenAL to make playing and managing sounds easier. |
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27 * ALmixer provides a simple API inspired by SDL_mixer to make playing sounds easy |
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28 * with having to worry about directly dealing with OpenAL sources, buffers, |
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29 * and buffer queuing directly. |
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30 * ALmixer currently utilizes SDL_sound behind the scenes to decode |
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31 * various audio formats such as WAV, MP3, AAC, MP4, OGG, etc. |
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32 * |
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33 * This library is targeted towards two major groups: |
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34 * - People who just want an easy, high performance, way to play audio (don't care if its OpenAL or not) |
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35 * - People who want to an easy way to play audio in OpenAL but still want access to OpenAL directly. |
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36 * |
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37 * ALmixer exposes OpenAL sources in the API so you can freely use ALmixer |
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38 * in larger OpenAL applications that need to apply OpenAL 3D effects and features |
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39 * to playing sounds. |
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40 * |
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41 * The API is heavily influenced and inspired by SDL_mixer, though there is one major |
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42 * conceptual design difference. ALmixer doesn't divide sound and music playback into two |
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43 * separate play APIs. Instead, there is one unified play API and you specify via the |
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44 * load API whether you want the audio resource loaded as a stream or completely preloaded. |
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45 * This allows you to have any arbitrary number of streaming sources playing simultaneously |
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46 * (such as music and speech) unlike SDL_mixer where you are limited to only one "music" |
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47 * channel. |
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48 * |
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49 * A less major conceptual design difference is every "Channel" API has a corresponding "Source" API. |
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50 * Every "channel" (in the SDL_mixer definition context) maps to a corresponding OpenAL source id. You can use |
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51 * this source ID directly with OpenAL API commands to utilize OpenAL effects such as position, Doppler, etc. |
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52 * Convenience APIs are provided to let you convert channel numbers to source ids and vice-versa. |
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53 * |
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54 * Another change which is a pet-peev of mine with SDL_mixer is the lack of a user_data parameter in callbacks. |
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55 * ALmixer callbacks allow you to pass user_data (aka context) pointers through the callback functions. |
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56 * |
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57 * @note There are some #defines you can set to change the behavior at compile time. Most you shouldn't touch. |
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58 * The one worth noting is ENABLE_ALMIXER_THREADS. If enabled, ALmixer_Update() is automatically called on a |
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59 * background thread so you no longer have to explicitly call it. (The function turns into a no-op so your existing |
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60 * code won't break.) Having Update run in a separate thread has some advantages, particularly for streaming |
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61 * audio as all the OpenAL buffer queuing happens in this function. It is less likely the background thread will |
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62 * be blocked for long periods and thus less likely your buffer queues will be starved. However, this means you |
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63 * need to be extra careful about what you do in callback functions as they are invoked from the background thread. |
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64 * I still consider this feature a experimental (though I am starting to use it more myself) and there |
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65 * may still be bugs. |
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66 * |
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67 * @author Eric Wing |
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68 */ |
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69 |
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70 /** |
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71 * @file |
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72 * ALmixer (which I sometimes call "SDL-OpenAL-Mixer" or "SDL_ALmixer") is a cross-platform audio library built |
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73 * on top of OpenAL to make playing and managing sounds easier. |
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74 * ALmixer provides a simple API inspired by SDL_mixer to make playing sounds easy |
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75 * with having to worry about directly dealing with OpenAL sources, buffers, |
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76 * and buffer queuing directly. |
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77 * ALmixer currently utilizes SDL_sound behind the scenes to decode |
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78 * various audio formats such as WAV, MP3, AAC, MP4, OGG, etc. |
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79 * |
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80 * This library is targeted towards two major groups: |
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81 * - People who just want an easy, high performance, way to play audio (don't care if its OpenAL or not) |
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82 * - People who want to an easy way to play audio in OpenAL but still want access to OpenAL directly. |
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83 * |
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84 * ALmixer exposes OpenAL sources in the API so you can freely use ALmixer |
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85 * in larger OpenAL applications that need to apply OpenAL 3D effects and features |
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86 * to playing sounds. |
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87 * |
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88 * The API is heavily influenced and inspired by SDL_mixer, though there is one major |
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89 * conceptual design difference. ALmixer doesn't divide sound and music playback into two |
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90 * separate play APIs. Instead, there is one unified play API and you specify via the |
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91 * load API whether you want the audio resource loaded as a stream or completely preloaded. |
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92 * This allows you to have any arbitrary number of streaming sources playing simultaneously |
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93 * (such as music and speech) unlike SDL_mixer where you are limited to only one "music" |
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94 * channel. |
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95 * |
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96 * A less major conceptual design difference is every "Channel" API has a corresponding "Source" API. |
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97 * Every "channel" (in the SDL_mixer definition context) maps to a corresponding OpenAL source id. You can use |
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98 * this source ID directly with OpenAL API commands to utilize OpenAL effects such as position, Doppler, etc. |
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99 * Convenience APIs are provided to let you convert channel numbers to source ids and vice-versa. |
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100 * |
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101 * Another change which is a pet-peev of mine with SDL_mixer is the lack of a user_data parameter in callbacks. |
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102 * ALmixer callbacks allow you to pass user_data (aka context) pointers through the callback functions. |
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103 * |
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104 * @note There are some #defines you can set to change the behavior at compile time. Most you shouldn't touch. |
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105 * The one worth noting is ENABLE_ALMIXER_THREADS. If enabled, ALmixer_Update() is automatically called on a |
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106 * background thread so you no longer have to explicitly call it. (The function turns into a no-op so your existing |
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107 * code won't break.) Having Update run in a separate thread has some advantages, particularly for streaming |
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108 * audio as all the OpenAL buffer queuing happens in this function. It is less likely the background thread will |
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109 * be blocked for long periods and thus less likely your buffer queues will be starved. However, this means you |
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110 * need to be extra careful about what you do in callback functions as they are invoked from the background thread. |
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111 * I still consider this feature a experimental (though I am starting to use it more myself) and there |
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112 * may still be bugs. |
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113 * |
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114 * @author Eric Wing |
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115 */ |
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116 |
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117 |
0 | 118 #ifndef _SDL_ALMIXER_H_ |
119 #define _SDL_ALMIXER_H_ | |
120 | |
2
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121 |
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122 #ifndef DOXYGEN_SHOULD_IGNORE_THIS |
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123 /** @cond DOXYGEN_SHOULD_IGNORE_THIS */ |
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124 |
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125 /* Note: For Doxygen to produce clean output, you should set the |
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126 * PREDEFINED option to remove ALMIXER_DECLSPEC, ALMIXER_CALL, and |
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127 * the DOXYGEN_SHOULD_IGNORE_THIS blocks. |
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128 * PREDEFINED = DOXYGEN_SHOULD_IGNORE_THIS=1 ALMIXER_DECLSPEC= ALMIXER_CALL= |
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129 */ |
0 | 130 |
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131 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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132 #if defined(_WIN32) |
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133 #if defined(ALMIXER_BUILD_LIBRARY) |
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134 #define ALMIXER_DECLSPEC __declspec(dllexport) |
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135 #else |
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136 #define ALMIXER_DECLSPEC __declspec(dllimport) |
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137 #endif |
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138 #else |
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139 #if defined(ALMIXER_BUILD_LIBRARY) |
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140 #if defined (__GNUC__) && __GNUC__ >= 4 |
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141 #define ALMIXER_DECLSPEC __attribute__((visibility("default"))) |
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142 #else |
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143 #define ALMIXER_DECLSPEC |
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144 #endif |
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145 #else |
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146 #define ALMIXER_DECLSPEC |
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147 #endif |
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148 #endif |
0 | 149 |
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150 #if defined(_WIN32) |
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151 #define ALMIXER_CALL __cdecl |
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152 #else |
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153 #define ALMIXER_CALL |
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154 #endif |
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155 #else |
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156 #include "SDL_types.h" /* will include begin_code.h which is what I really want */ |
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157 #define ALMIXER_DECLSPEC DECLSPEC |
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158 #define ALMIXER_CALL SDLCALL |
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159 #endif |
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160 |
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161 /** @endcond DOXYGEN_SHOULD_IGNORE_THIS */ |
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162 #endif /* DOXYGEN_SHOULD_IGNORE_THIS */ |
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163 |
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164 |
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165 |
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166 /* Needed for OpenAL types since altypes.h was removed in 1.1 */ |
1 | 167 #include "al.h" |
0 | 168 |
169 /* Set up for C function definitions, even when using C++ */ | |
170 #ifdef __cplusplus | |
171 extern "C" { | |
172 #endif | |
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173 |
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174 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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175 /** |
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176 * Struct that contains the version information of this library. |
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177 * This represents the library's version as three levels: major revision |
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178 * (increments with massive changes, additions, and enhancements), |
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179 * minor revision (increments with backwards-compatible changes to the |
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180 * major revision), and patchlevel (increments with fixes to the minor |
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181 * revision). |
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182 * @see ALMIXER_VERSION, ALmixer_GetLinkedVersion |
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183 */ |
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184 typedef struct ALmixer_version |
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185 { |
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186 ALubyte major; |
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187 ALubyte minor; |
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188 ALubyte patch; |
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189 } ALmixer_version; |
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190 #else |
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191 #include "SDL_version.h" |
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192 #define ALmixer_version SDL_version |
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193 #endif |
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194 |
0 | 195 /* Printable format: "%d.%d.%d", MAJOR, MINOR, PATCHLEVEL |
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196 */ |
0 | 197 #define ALMIXER_MAJOR_VERSION 0 |
198 #define ALMIXER_MINOR_VERSION 1 | |
199 #define ALMIXER_PATCHLEVEL 0 | |
200 | |
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201 |
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202 /** |
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203 * @defgroup CoreOperation Initialization, Tear-down, and Core Operational Commands |
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204 * @{ |
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205 * Functions for setting up and using ALmixer. |
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206 */ |
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207 |
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208 |
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209 /** |
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210 * This macro fills in a version structure with the version of the |
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211 * library you compiled against. This is determined by what header the |
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212 * compiler uses. Note that if you dynamically linked the library, you might |
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213 * have a slightly newer or older version at runtime. That version can be |
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214 * determined with ALmixer_GetLinkedVersion(), which, unlike |
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215 * ALMIXER_GET_COMPILED_VERSION, is not a macro. |
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216 * |
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217 * @note When compiled with SDL, this macro can be used to fill a version structure |
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218 * compatible with SDL_version. |
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219 * |
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220 * @param X A pointer to a ALmixer_version struct to initialize. |
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221 * |
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222 * @see ALmixer_version, ALmixer_GetLinkedVersion |
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223 */ |
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224 #define ALMIXER_GET_COMPILED_VERSION(X) \ |
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225 { \ |
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226 (X)->major = ALMIXER_MAJOR_VERSION; \ |
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227 (X)->minor = ALMIXER_MINOR_VERSION; \ |
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228 (X)->patch = ALMIXER_PATCHLEVEL; \ |
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229 } |
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230 |
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231 /** |
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232 * Gets the library version of the dynamically linked ALmixer you are using. |
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233 * This gets the version of ALmixer that is linked against your program. |
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234 * If you are using a shared library (DLL) version of ALmixer, then it is |
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235 * possible that it will be different than the version you compiled against. |
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236 * |
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237 * This is a real function; the macro ALMIXER_GET_COMPILED_VERSION |
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238 * tells you what version of tErrorLib you compiled against: |
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239 * |
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240 * @code |
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241 * ALmixer_version compiled; |
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242 * ALmixer_version linked; |
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243 * |
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244 * ALMIXER_GET_COMPILED_VERSION(&compiled); |
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245 * ALmixer_GetLinkedVersion(&linked); |
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246 * printf("We compiled against tError version %d.%d.%d ...\n", |
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247 * compiled.major, compiled.minor, compiled.patch); |
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248 * printf("But we linked against tError version %d.%d.%d.\n", |
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249 * linked.major, linked.minor, linked.patch); |
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250 * @endcode |
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251 * |
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252 * @see ALmixer_version, ALMIXER_GET_COMPILED_VERSION |
0 | 253 */ |
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254 extern ALMIXER_DECLSPEC const ALmixer_version* ALMIXER_CALL ALmixer_GetLinkedVersion(void); |
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255 |
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256 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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257 /** |
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258 * Gets the last error string that was set by the system and clears the error. |
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259 * |
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260 * @note When compiled with SDL, this directly uses SDL_GetError. |
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261 * |
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262 * @return Returns a string containing the last error or "" when no error is set. |
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263 */ |
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264 extern ALMIXER_DECLSPEC const char* ALMIXER_CALL ALmixer_GetError(void); |
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265 /** |
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266 * Sets an error string that can be retrieved by ALmixer_GetError. |
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267 * |
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268 * @note When compiled with SDL, this directly uses SDL_SetError. |
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269 * |
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270 * param The error string to set. |
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271 */ |
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272 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetError(const char *fmt, ...); |
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273 #else |
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274 #include "SDL_error.h" |
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275 /** |
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276 * Gets the last error string that was set by the system and clears the error. |
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277 * |
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278 * @note When compiled with SDL, this directly uses SDL_GetError. |
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279 * |
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280 * @return Returns a string containing the last error or "" when no error is set. |
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281 */ |
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282 #define ALmixer_GetError SDL_GetError |
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283 /** |
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284 * Sets an error string that can be retrieved by ALmixer_GetError. |
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285 * |
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286 * @note When compiled with SDL, this directly uses SDL_SetError. |
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287 * |
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288 * param The error string to set. |
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289 */ |
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290 #define ALmixer_SetError SDL_SetError |
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291 #endif |
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292 |
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293 |
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294 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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295 #include "ALmixer_rwops.h" |
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296 #else |
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297 #include "SDL_rwops.h" |
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298 /** |
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299 * A struct that mimicks the SDL_RWops structure. |
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300 * |
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301 * @note When compiled with SDL, this directly uses SDL_RWops. |
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302 */ |
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303 #define ALmixer_RWops SDL_RWops |
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304 #endif |
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305 |
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306 |
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307 #define ALMIXER_DEFAULT_FREQUENCY 0 |
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308 #define ALMIXER_DEFAULT_REFRESH 0 |
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309 #define ALMIXER_DEFAULT_NUM_CHANNELS 16 |
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310 #define ALMIXER_DEFAULT_NUM_SOURCES ALMIXER_DEFAULT_NUM_CHANNELS |
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311 |
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312 /** |
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313 * This is the recommended Init function. This will initialize the context, SDL_sound, |
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314 * and the mixer system. You should call this in the setup of your code, after SDL_Init. |
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315 * If you attempt to bypass this function, you do so at your own risk. |
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316 * |
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317 * @note ALmixer expects the SDL audio subsystem to be disabled. In some cases, an enabled |
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318 * SDL audio subsystem will interfere and cause problems in your app. This Init method explicitly |
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319 * disables the SDL subsystem if SDL is compiled in. |
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320 * |
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321 * @note The maximum number of sources is OpenAL implementation dependent. |
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322 * Currently 16 is lowest common denominator for all OpenAL implementations in current use. |
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323 * 32 is currently the second lowest common denominator. |
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324 * If you try to allocate more sources than are actually available, this function may return false depending |
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325 * if the OpenAL implementation returns an error or not. It is possible for OpenAL to silently fail |
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326 * so be very careful about picking too many sources. |
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327 * |
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328 * @param playback_frequency The sample rate you want OpenAL to play at, e.g. 44100 |
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329 * Note that OpenAL is not required to actually respect this value. |
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330 * Pass in 0 or ALMIXER_DEFAULT_FREQUENCY to specify you want to use your implementation's default value. |
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331 * @param num_sources The number of OpenAL sources (also can be thought of as |
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332 * SDL_mixer channels) you wish to allocate. |
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333 * Pass in 0 or ALMIXER_DEFAULT_NUM_SOURCES to use ALmixer's default value. |
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334 * @param refresh_rate The refresh rate you want OpenAL to operate at. |
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335 * Note that OpenAL is not required to respect this value. |
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336 * Pass in 0 or ALMIXER_DEFAULT_REFRESH to use OpenAL default behaviors. |
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337 * @return Returns AL_FALSE on a failure or AL_TRUE if successfully initialized. |
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338 */ |
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339 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_Init(ALuint playback_frequency, ALint num_sources, ALuint refresh_rate); |
0 | 340 |
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341 /** |
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342 * InitContext will only initialize the OpenAL context (and not the mixer part). |
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343 * Note that SDL_Sound is also initialized here because load order matters |
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344 * because SDL audio will conflict with OpenAL when using SMPEG. This is only |
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345 * provided as a backdoor and is not recommended. |
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346 * |
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347 * @note This is a backdoor in case you need to initialize the AL context and |
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348 * the mixer system separately. I strongly recommend avoiding these two functions |
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349 * and use the normal Init() function. |
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350 */ |
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351 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_InitContext(ALuint playback_frequency, ALuint refresh_rate); |
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352 |
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353 /** |
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354 * InitMixer will only initialize the Mixer system. This is provided in the case |
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355 * that you need control over the loading of the context. You may load the context |
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356 * yourself, and then call this function. This is not recommended practice, but is |
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357 * provided as a backdoor in case you have good reason to |
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358 * do this. Be warned that if ALmixer_InitMixer() fails, |
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359 * it will not clean up the AL context. Also be warned that Quit() still does try to |
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360 * clean up everything. |
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361 * |
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362 * @note This is a backdoor in case you need to initialize the AL context and |
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363 * the mixer system separately. I strongly recommend avoiding these two functions |
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364 * and use the normal Init() function. |
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365 */ |
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366 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_InitMixer(ALint num_sources); |
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367 |
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368 /** |
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369 * This shuts down ALmixer. Please remember to free your ALmixer_Data* instances |
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370 * before calling this method. |
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371 */ |
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372 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_Quit(void); |
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373 /** |
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374 * Returns whether ALmixer has been initializatized (via Init) or not. |
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375 * @return Returns true for initialized and false for not initialized. |
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376 */ |
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377 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_IsInitialized(void); |
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378 |
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379 /** |
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380 * Returns the frequency that OpenAL is set to. |
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381 * @note This function is not guaranteed to give correct information and is OpenAL implementation dependent. |
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382 * @return Returns the frequency, e.g. 44100. |
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383 */ |
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384 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_GetFrequency(void); |
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385 |
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386 /** |
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387 * Let's you change the maximum number of channels/sources available. |
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388 * This function is not heavily tested. It is probably better to simply initialize |
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389 * ALmixer with the number of sources you want when you initialize it instead of |
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390 * dynamically changing it later. |
0 | 391 */ |
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392 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_AllocateChannels(ALint num_chans); |
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393 |
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394 /** |
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395 * Allows you to reserve a certain number of channels so they won't be automatically |
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396 * allocated to play on. |
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397 * This function will effectively block off a certain number of channels so they won't |
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398 * be automatically assigned to be played on when you call various play functions |
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399 * (applies to both play-channel and play-source functions since they are the same under the hood). |
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400 * The lowest number channels will always be blocked off first. |
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401 * For example, if there are 16 channels available, and you pass 2 into this function, |
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402 * channels 0 and 1 will be reserved so they won't be played on automatically when you specify |
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403 * you want to play a sound on any available channel/source. You can |
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404 * still play on channels 0 and 1 if you explicitly designate you want to play on their channel |
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405 * number or source id. |
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406 * Setting back to 0 will clear all the reserved channels so all will be available again for |
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407 * auto-assignment. |
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408 * As an example, this feature can be useful if you always want your music to be on channel 0 and |
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409 * speech on channel 1 and you don't want sound effects to ever occupy those channels. This allows |
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410 * you to build in certain assumptions about your code, perhaps for deciding which data you want |
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411 * to analyze in a data callback. |
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412 * Specifying the number of reserve channels to the maximum number of channels will effectively |
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413 * disable auto-assignment. |
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414 * @param number_of_reserve_channels The number of channels/sources to reserve. |
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415 * Or pass -1 to find out how many channels are currently reserved. |
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416 * @return Returns the number of currently reserved channels. |
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417 */ |
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418 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ReserveChannels(ALint number_of_reserve_channels); |
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419 |
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420 |
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421 /** |
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422 * The update function that allows ALmixer to update its internal state. |
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423 * If not compiled with/using threads, this function must be periodically called |
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424 * to poll ALmixer to force streamed music and other events to |
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425 * take place. |
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426 * The typical place to put this function is in your main-loop. |
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427 * If threads are enabled, then this function just |
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428 * returns 0 and is effectively a no-op. With threads, it is not necessary to call this function |
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429 * because updates are handled internally on another thread. However, because threads are still considered |
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430 * experimental, it is recommended you call this function in a proper place in your code in case |
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431 * future versions of this library need to abandon threads. |
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432 * @return Returns 0 if using threads. If not using threads, for debugging purposes, it returns |
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433 * the number of buffers queued during the loop, or a negative value indicating the numer of errors encountered. |
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434 * This is subject to change and should not be relied on. |
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435 */ |
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436 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_Update(void); |
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437 |
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438 /** |
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439 * @} |
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440 */ |
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441 |
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442 /** |
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443 * @defgroup LoadAPI Load Audio Functions |
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444 * @{ |
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445 * Functions for loading and unloading audio data. |
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446 */ |
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447 |
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448 |
0 | 449 |
450 /* | |
451 #define ALmixer_AudioInfo Sound_AudioInfo | |
452 */ | |
453 | |
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454 /* |
0 | 455 #define ALMIXER_DEFAULT_BUFFERSIZE 32768 |
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456 #define ALMIXER_DEFAULT_BUFFERSIZE 4096 |
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457 */ |
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458 #define ALMIXER_DEFAULT_BUFFERSIZE 16384 |
0 | 459 |
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460 /* You probably never need to use these macros directly. */ |
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461 #ifndef ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION |
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462 #define ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE ALMIXER_DEFAULT_BUFFERSIZE * 4 |
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463 #else |
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464 /* I'm picking a smaller buffer because ALmixer will try to create a new larger buffer |
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465 * based on the length of the audio. So creating a large block up-front might just be a waste. |
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466 * However, if my attempts fail for some reason, this buffer size becomes a fallback. |
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467 * Having too small of a buffer might cause performance bottlenecks. |
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468 */ |
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469 #define ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE 1024 |
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470 #endif |
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471 |
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472 /** |
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473 * Specifies the maximum number of queue buffers to use for a sound stream. |
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474 * Default Queue Buffers must be at least 2. |
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475 */ |
0 | 476 #define ALMIXER_DEFAULT_QUEUE_BUFFERS 5 |
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477 /** |
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478 * Specifies the number of queue buffers initially filled when first loading a stream. |
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479 * Default startup buffers should be at least 1. */ |
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480 #define ALMIXER_DEFAULT_STARTUP_BUFFERS 2 |
0 | 481 |
1 | 482 /* |
0 | 483 #define ALMIXER_DECODE_STREAM 0 |
484 #define ALMIXER_DECODE_ALL 1 | |
1 | 485 */ |
0 | 486 |
1 | 487 /* This is a trick I picked up from Lua. Doing the typedef separately |
488 * (and I guess before the definition) instead of a single | |
489 * entry: typedef struct {...} YourName; seems to allow me | |
490 * to use forward declarations. Doing it the other way (like SDL) | |
491 * seems to prevent me from using forward declarions as I get conflicting | |
492 * definition errors. I don't really understand why though. | |
493 */ | |
494 typedef struct ALmixer_Data ALmixer_Data; | |
495 typedef struct ALmixer_AudioInfo ALmixer_AudioInfo; | |
0 | 496 |
1 | 497 /** |
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498 * Roughly the equvialent to the Sound_AudioInfo struct in SDL_sound. |
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499 * Types have been changed to use AL types because I know those are available. |
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500 * This is different than SDL which uses fixed types so there might be subtle |
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501 * things you need to pay attention to.. |
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502 * @note Originally, I just used the Sound_AudioInfo directly, but |
1 | 503 * I've been trying to reduce the header dependencies for this file. |
504 * But more to the point, I've been interested in dealing with the | |
505 * WinMain override problem Josh faced when trying to use SDL components | |
506 * in an MFC app which didn't like losing control of WinMain. | |
507 * My theory is that if I can purge the header of any thing that | |
508 * #include's SDL_main.h, then this might work. | |
509 * So I am now introducing my own AudioInfo struct. | |
510 */ | |
511 struct ALmixer_AudioInfo | |
512 { | |
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513 ALushort format; /**< Equivalent of SDL_AudioSpec.format. */ |
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514 ALubyte channels; /**< Number of sound channels. 1 == mono, 2 == stereo. */ |
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515 ALuint rate; /**< Sample rate; frequency of sample points per second. */ |
1 | 516 }; |
0 | 517 |
518 | |
519 | |
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520 /** |
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521 * This is a general loader function to load an audio resource from an RWops. |
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522 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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523 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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524 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which |
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525 * decoder to use. |
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526 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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527 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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528 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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529 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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530 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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531 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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532 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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533 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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534 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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535 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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536 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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537 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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538 */ |
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539 extern ALMIXER_DECLSPEC ALmixer_Data* ALMIXER_CALL ALmixer_LoadSample_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
0 | 540 |
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541 #ifdef DOXYGEN_ONLY |
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542 /** |
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543 * This is the loader function to load an audio resource from an RWops as a stream. |
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544 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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545 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which |
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546 * decoder to use. |
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547 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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548 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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549 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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550 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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551 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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552 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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553 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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554 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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555 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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556 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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557 */ |
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558 ALmixer_Data* ALmixer_LoadStream_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
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559 #else |
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560 #define ALmixer_LoadStream_RW(rw_ops, file_ext, buffer_size, max_queue_buffers, num_startup_buffers, access_data) ALmixer_LoadSample_RW(rw_ops,file_ext, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data) |
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561 #endif |
0 | 562 |
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563 #ifdef DOXYGEN_ONLY |
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564 /** |
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565 * This is the loader function to completely preload an audio resource from an RWops into RAM. |
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566 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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567 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which |
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568 * decoder to use. |
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569 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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570 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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571 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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572 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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573 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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574 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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575 */ |
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576 ALmixer_Data* ALmixer_LoadAll_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALboolean access_data); |
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577 #else |
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578 #define ALmixer_LoadAll_RW(rw_ops, file_ext, access_data) ALmixer_LoadSample_RW(rw_ops, fileext, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data) |
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579 #endif |
0 | 580 |
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581 /** |
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582 * This is a general loader function to load an audio resource from a file. |
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583 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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584 * @param file_name The file of the audio resource you want to load. |
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585 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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586 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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587 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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588 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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589 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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590 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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591 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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592 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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593 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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594 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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595 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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596 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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597 */ |
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598 extern ALMIXER_DECLSPEC ALmixer_Data * ALMIXER_CALL ALmixer_LoadSample(const char* file_name, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
0 | 599 |
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600 #ifdef DOXYGEN_ONLY |
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601 /** |
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602 * This is the loader function to load an audio resource from a file. |
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603 * @param file_name The file to the audio resource you want to load. |
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604 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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605 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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606 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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607 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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608 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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609 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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610 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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611 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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612 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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613 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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614 */ |
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615 ALmixer_Data* ALmixer_LoadStream(const char* file_name, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
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616 #else |
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617 #define ALmixer_LoadStream(file_name, buffer_size, max_queue_buffers, num_startup_buffers,access_data) ALmixer_LoadSample(file_name, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data) |
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618 #endif |
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619 |
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620 #ifdef DOXYGEN_ONLY |
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621 /** |
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622 * This is the loader function to completely preload an audio resource from a file into RAM. |
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623 * @param file_name The file to the audio resource you want to load. |
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624 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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625 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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626 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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627 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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628 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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629 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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630 */ |
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631 ALmixer_Data* ALmixer_LoadAll(const char* file_name, ALboolean access_data); |
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632 #else |
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633 #define ALmixer_LoadAll(file_name, access_data) ALmixer_LoadSample(file_name, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data) |
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634 #endif |
0 | 635 |
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636 /** |
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637 * This is a back door general loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field. |
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638 * Use at your own risk. |
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639 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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640 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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641 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which |
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642 * decoder to use. Pass "raw" for raw formats. |
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643 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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644 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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645 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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646 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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647 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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648 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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649 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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650 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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651 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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652 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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653 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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654 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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655 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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656 */ |
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657 extern ALMIXER_DECLSPEC ALmixer_Data * ALMIXER_CALL ALmixer_LoadSample_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
0 | 658 |
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659 #ifdef DOXYGEN_ONLY |
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660 /** |
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661 * This is a back door stream loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field. |
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662 * Use at your own risk. |
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663 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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664 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which |
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665 * decoder to use. Pass "raw" for raw formats. |
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666 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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667 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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668 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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669 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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670 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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671 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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672 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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673 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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674 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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675 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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676 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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677 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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678 */ |
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679 ALmixer_Data* ALmixer_LoadStream_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
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680 #else |
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681 #define ALmixer_LoadStream_RAW_RW(rw_ops, file_ext, desired_format, buffer_size, max_queue_buffers, num_startup_buffers, access_data) ALmixer_LoadSample_RAW_RW(rw_ops, file_ext, desired_format, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data) |
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682 #endif |
0 | 683 |
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684 #ifdef DOXYGEN_ONLY |
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685 /** |
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686 * This is a back door loader function for complete preloading RAW samples into RAM or if you need to specify the ALmixer_AudioInfo field. |
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687 * Use at your own risk. |
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688 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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689 * @param file_ext The file extension of your audio type which is used as a hint by the backend to decide which |
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690 * decoder to use. Pass "raw" for raw formats. |
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691 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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692 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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693 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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694 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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695 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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696 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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697 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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698 */ |
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699 ALmixer_Data* ALmixer_LoadAll_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALboolean access_data); |
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700 #else |
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701 #define ALmixer_LoadAll_RAW_RW(rw_ops, file_ext, desired_format, access_data) ALmixer_LoadSample_RAW_RW(rw_ops, file_ext, desired_format, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data) |
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702 #endif |
0 | 703 |
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704 /** |
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705 * This is a back door general loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field. |
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706 * Use at your own risk. |
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707 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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708 * @param file_name The file to the audio resource you want to load. Extension should be "raw" for raw formats. |
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709 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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710 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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711 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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712 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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713 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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714 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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715 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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716 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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717 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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718 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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719 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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720 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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721 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
0 | 722 */ |
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723 extern ALMIXER_DECLSPEC ALmixer_Data * ALMIXER_CALL ALmixer_LoadSample_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALboolean decode_mode_is_predecoded, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
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724 |
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725 #ifdef DOXYGEN_ONLY |
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726 /** |
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727 * This is a back door stream loader function for RAW samples or if you need to specify the ALmixer_AudioInfo field. |
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728 * Use at your own risk. |
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729 * @param file_name The file to the audio resource you want to load.Extension should be "raw" for raw formats. |
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730 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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731 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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732 * the valid frame sizes of your audio data. If the data is streamed, the data will be read in buffer_size chunks. |
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733 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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734 * @param max_queue_buffers For streamed data, specifies the maximum number of buffers that can be queued at any given time. |
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735 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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736 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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737 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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738 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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739 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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740 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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741 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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742 */ |
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743 ALmixer_Data* ALmixer_LoadStream_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALboolean access_data); |
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744 #else |
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745 #define ALmixer_LoadStream_RAW(file_name, desired_format, buffer_size, max_queue_buffers, num_startup_buffers, access_data) ALmixer_LoadSample_RAW(file_name, desired_format, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, access_data) |
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746 #endif |
0 | 747 |
2
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748 #ifdef DOXYGEN_ONLY |
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749 /** |
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750 * This is a back door loader function for complete preloading RAW samples into RAM or if you need to specify the ALmixer_AudioInfo field. |
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751 * Use at your own risk. |
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752 * @param file_name The file to the audio resource you want to load. Extension should be "raw" for raw formats. |
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753 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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754 * @param access_data A boolean that specifies if you want the data contained in the currently playing buffer to be handed |
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755 * to you in a callback function. Note that for predecoded data, you get back the entire buffer in one callback when the |
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756 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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757 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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758 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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759 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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760 */ |
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761 ALmixer_Data* ALmixer_LoadAll_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALboolean access_data); |
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762 #else |
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763 #define ALmixer_LoadAll_RAW(file_name, desired_format, access_data) ALmixer_LoadSample_RAW(file_name, desired_format, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, access_data) |
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764 #endif |
0 | 765 |
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766 /** |
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767 * Frees an ALmixer_Data. |
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768 * Releases the memory associated with a ALmixer_Data. Use this when you are done playing the audio sample |
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769 * and wish to release the memory. |
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770 * @warning Do not try releasing data that is currently in use (e.g. playing, paused). |
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771 * @warning Make sure to free your data before calling ALmixer_Quit. Do not free data aftter ALmixer_Quit(). |
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772 * @param almixer_data The ALmixer_Data* you want to free. |
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773 */ |
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774 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_FreeData(ALmixer_Data* almixer_data); |
0 | 775 |
776 | |
2
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777 /** |
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778 * Returns true if the almixer_data was completely loaded into memory or false if it was loaded |
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779 * as a stream. |
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780 * @param almixer_data The audio resource you want to know about. |
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781 * @return AL_TRUE is predecoded, or AL_FALSE if streamed. |
0 | 782 */ |
2
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783 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_IsPredecoded(ALmixer_Data* almixer_data); |
0 | 784 |
2
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785 /** |
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786 * @} |
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787 */ |
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788 |
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789 /** |
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790 * @defgroup CallbackAPI Callbacks |
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791 * @{ |
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792 * Functions for callbacks |
0 | 793 */ |
794 | |
2
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795 /** |
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796 * Allows you to set a callback for when a sound has finished playing on a channel/source. |
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797 * @param playback_finished_callback The function you want to be invoked when a sound finishes. |
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798 * The callback function will pass you back the channel number which just finished playing, |
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799 * the OpenAL source id associated with the channel, the ALmixer_Data* that was played, |
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800 * a boolean telling you whether a sound finished playing because it ended normally or because |
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801 * something interrupted the playback (such as the user calling ALmixer_Halt*), and the |
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802 * user_data supplied as the second parameter to this function. |
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803 * @param which_chan The ALmixer channel that the data is currently playing on. |
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804 * @param al_source The OpenAL source that the data is currently playing on. |
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805 * @param almixer_data The ALmixer_Data that was played. |
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806 * @param finished_naturally AL_TRUE if the sound finished playing because it ended normally |
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807 * or AL_FALSE because something interrupted playback (such as the user calling ALmixer_Halt*). |
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808 * @param user_data This will be passed back to you in the callback. |
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809 * |
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810 * @warning You should not call other ALmixer functions in this callback. |
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811 * Particularly in the case of when compiled with threads, recursive locking |
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812 * will occur which will lead to deadlocks. Also be aware that particularly in the |
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813 * threaded case, the callbacks may (and currently do) occur on a background thread. |
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814 * One typical thread safe strategy is to set flags or schedule events to occur on the |
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815 * main thread. |
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816 * One possible exception to the no-calling ALmixer functions rule is ALmixer_Free. ALmixer_Free |
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817 * currently does not lock so it might okay to call this to free your data. However, this is not |
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818 * tested and not the expected pattern to be used. |
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819 */ |
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820 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetPlaybackFinishedCallback(void (*playback_finished_callback)(ALint which_channel, ALuint al_source, ALmixer_Data* almixer_data, ALboolean finished_naturally, void* user_data), void* user_data); |
1 | 821 |
822 /** | |
2
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823 * Allows you to set a callback for getting audio data. |
1 | 824 * This is a callback function pointer that when set, will trigger a function |
825 * anytime there is new data loaded for a sample. The appropriate load | |
826 * parameter must be set in order for a sample to appear here. | |
827 * Keep in mind the the current backend implementation must do an end run | |
828 * around OpenAL because OpenAL lacks support for this kind of thing. | |
829 * As such, buffers are copied at decode time, and there is no attempt to do | |
830 * fine grained timing syncronization. You will be provided the entire buffer | |
831 * that is decoded regardless of length. So if you predecoded the entire | |
832 * audio file, the entire data buffer will be provided in a single callback. | |
833 * If you stream the data, you will be getting chunk sizes that are the same as | |
834 * what you specified the decode size to be. Unfortunely, this means if you | |
835 * pick smaller buffers, you get finer detail at the expense/risk of buffer | |
836 * underruns. If you decode more data, you have to deal with the syncronization | |
837 * issues if you want to display the data during playback in something like an | |
838 * oscilloscope. | |
2
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839 * |
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840 * @warning You should not call other ALmixer functions in this callback. |
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841 * Particularly in the case of when compiled with threads, recursive locking |
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842 * will occur which will lead to deadlocks. Also be aware that particularly in the |
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843 * threaded case, the callbacks may (and currently do) occur on a background thread. |
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844 * One typical thread safe strategy is to set flags or schedule events to occur on the |
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845 * main thread. |
1 | 846 * |
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847 * @param playback_data_callback The function you want called back. |
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848 * @param which_channel The ALmixer channel that the data is currently playing on. |
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849 * @param al_source The OpenAL source that the data is currently playing on. |
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850 * @param pcm_data This is a pointer to the data buffer containing ALmixer's |
1 | 851 * version of the decoded data. Consider this data as read-only. In the |
852 * non-threaded backend, this data will persist until potentially the next call | |
853 * to Update(). Currently, data buffers are preallocated and not destroyed | |
854 * until FreeData() is called (though this behavior is subject to change), | |
855 * but the contents will change when the buffer needs to be reused for a | |
856 * future callback. The buffer reuse is tied to the amount of buffers that | |
857 * may be queued. | |
858 * But assuming I don't change this, this may allow for some optimization | |
859 * so you can try referencing data from these buffers without worrying | |
860 * about crashing. (You still need to be aware that the data could be | |
861 * modified behind the scenes on an Update().) | |
2
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862 * The data type listed is an signed 8-bit format, but the real data may |
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863 * not actually be this. ALbyte was chosen as a convenience. If you have |
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864 * a 16 bit format, you will want to cast the data and divide the num_bytes by 2. |
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865 * Typically, data is either Sint16. This seems to be a |
1 | 866 * convention audio people seem to follow though I'm not sure what the |
867 * underlying reasons (if any) are for this. I suspect that there may be | |
2
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868 * some nice alignment/conversion property if you need to cast between ALbyte |
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869 * and ALubyte. |
1 | 870 * |
871 * @param num_bytes This is the total length of the data buffer. It presumes | |
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872 * that this length is measured for ALbyte. So if you have Sint16 data, you |
1 | 873 * should divide num_bytes by two if you access the data as Sint16. |
874 * | |
875 * @param frequency The frequency the data was decoded at. | |
876 * | |
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877 * @param num_channels_in_sample 1 for mono, 2 for stereo. Not to be confused with the ALmixer which_channel. |
1 | 878 * |
879 * @param bit_depth Bits per sample. This is expected to be 8 or 16. This | |
880 * number will tell you if you if you need to treat the data buffer as | |
881 * 16 bit or not. | |
882 * | |
883 * @param is_unsigned 1 if the data is unsigned, 0 if signed. Using this | |
884 * combined with bit_depth will tell you if you need to treat the data | |
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885 * as ALubyte, ALbyte, ALuint, or ALint. |
1 | 886 * |
887 * @param decode_mode_is_predecoded This is here to tell you if the data was totally | |
888 * predecoded or loaded as a stream. If predecoded, you will only get | |
889 * one data callback per playback instance. (This might also be true for | |
890 * looping the same sample...I don't remember how it was implemented. | |
891 * Maybe this should be fixed.) | |
892 * 0 (ALMIXER_DECODE_STREAM) for streamed. | |
893 * 1 (ALMIXER_DECODE_ALL) for predecoded. | |
894 * | |
895 * @param length_in_msec This returns the total length (time) of the data | |
896 * buffer in milliseconds. This could be computed yourself, but is provided | |
897 * as a convenince. | |
898 * | |
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899 * @param user_data The user data you pass in will be passed back to you in the callback. |
1 | 900 */ |
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901 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetPlaybackDataCallback(void (*playback_data_callback)(ALint which_channel, ALuint al_source, ALbyte* pcm_data, ALuint num_bytes, ALuint frequency, ALubyte num_channels_in_sample, ALubyte bit_depth, ALboolean is_unsigned, ALboolean decode_mode_is_predecoded, ALuint length_in_msec, void* user_data), void* user_data); |
0 | 902 |
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903 /** |
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904 * @} |
0 | 905 */ |
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906 |
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907 /** |
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908 * @defgroup PlayAPI Functions useful for playback. |
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909 * @{ |
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910 * These are core functions that are useful for controlling playback. |
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911 * Also see the Volume functions for additional playback functions and Query functions for additional information. |
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912 */ |
0 | 913 |
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914 /** |
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915 * Returns the total time in milliseconds of the audio resource. |
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916 * Returns the total time in milliseconds of the audio resource. |
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917 * If the total length cannot be determined, -1 will be returned. |
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918 * @param almixer_data The audio sample you want to know the total time of. |
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919 * @return The total time in milliseconds or -1 if some kind of failure. |
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920 */ |
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921 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_GetTotalTime(ALmixer_Data* almixer_data); |
0 | 922 |
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923 /** |
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924 * This function will look up the OpenAL source id for the corresponding channel number. |
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925 * @param which_channel The channel which you want to find the corresponding OpenAL source id for. |
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926 * If -1 was specified, an available source for playback will be returned. |
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927 * @return The OpenAL source id corresponding to the channel. 0 if you specified an illegal channel value. |
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928 * Or 0 if you specified -1 and no sources were currently available. |
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929 * @note ALmixer assumes your OpenAL implementation does not use 0 as a valid source ID. While the OpenAL spec |
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930 * does not disallow 0 for valid source ids, as of now, there are no known OpenAL implementations in use that |
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931 * use 0 as a valid source id (partly due to problems this has caused developers in the past). |
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932 */ |
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933 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_GetSource(ALint which_channel); |
0 | 934 |
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935 /** |
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936 * This function will look up the channel for the corresponding source. |
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937 * @param al_source The source id you want to find the corresponding channel number for. |
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938 * If -1 is supplied, it will try to return the first channel not in use. |
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939 * Returns -1 on error, or the channel. |
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940 */ |
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941 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_GetChannel(ALuint al_source); |
0 | 942 |
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943 /** |
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944 * Will look for a channel available for playback. |
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945 * Given a start channel number, the search will increase to the highest channel until it finds one available. |
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946 * @param start_channel The channel number you want to start looking at. |
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947 * @return A channel available or -1 if none could be found. |
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948 */ |
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949 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FindFreeChannel(ALint start_channel); |
1 | 950 |
0 | 951 |
952 | |
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953 /** |
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954 * Play a sound on a channel with a time limit. |
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955 * Plays a sound on a channel and will auto-stop after a specified number of milliseconds. |
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956 * @param which_channel Allows you to specify the specific channel you want to play on. |
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957 * Channels range from 0 to the (Number of allocated channels - 1). If you specify -1, |
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958 * an available channel will be chosen automatically for you. |
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959 * @note While paused, the auto-stop clock will also be paused. This makes it easy to always stop |
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960 * a sample by a predesignated length without worrying about whether the user paused playback which would |
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961 * throw off your calculations. |
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962 * @param almixer_data The audio resource you want to play. |
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963 * @param number_of_loops The number of times to loop (repeat) playing the data. |
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964 * 0 means the data will play exactly once without repeat. -1 means infinitely loop. |
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965 * @param number_of_milliseconds The number of milliseconds that should be played until the sample is auto-stopped. |
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966 * -1 means don't auto-stop playing and let the sample finish playing normally (or if looping is set to infinite, |
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967 * the sample will never stop playing). |
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968 * @return Returns the channel that was selected for playback or -1 if no channels were available. |
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969 */ |
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970 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PlayChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALint number_of_milliseconds); |
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971 |
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972 #ifdef DOXYGEN_ONLY |
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973 /** |
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974 * The same as ALmixer_PlayChannelTimed, but the sound is played without time limits. |
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975 * @see ALmixer_PlayChannelTimed. |
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976 */ |
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977 ALint ALmixer_PlayChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops); |
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978 #else |
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979 #define ALmixer_PlayChannel(channel,data,loops) ALmixer_PlayChannelTimed(channel,data,loops,-1) |
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980 #endif |
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981 |
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982 |
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983 /** |
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984 * Play a sound on an OpenAL source with a time limit. |
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985 * Plays a sound on an OpenAL source and will auto-stop after a specified number of milliseconds. |
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986 * @param al_source Allows you to specify the OpenAL source you want to play on. |
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987 * If you specify 0, an available source will be chosen automatically for you. |
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988 * @note Source values are not necessarily continguous and their values are implementation dependent. |
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989 * Always use ALmixer functions to determine source values. |
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990 * @note While paused, the auto-stop clock will also be paused. This makes it easy to always stop |
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991 * a sample by a predesignated length without worrying about whether the user paused playback which would |
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992 * throw off your calculations. |
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993 * @param almixer_data The audio resource you want to play. |
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994 * @param number_of_loops The number of times to loop (repeat) playing the data. |
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995 * 0 means the data will play exactly once without repeat. -1 means infinitely loop. |
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996 * @param number_of_milliseconds The number of milliseconds that should be played until the sample is auto-stopped. |
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997 * -1 means don't auto-stop playing and let the sample finish playing normally (or if looping is set to infinite, |
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998 * the sample will never stop playing). |
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999 * @return Returns the OpenAL source that was selected for playback or 0 if no sources were available. |
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1000 */ |
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1001 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_PlaySourceTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALint number_of_milliseconds); |
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1002 |
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1003 #ifdef DOXYGEN_ONLY |
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1004 /** |
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1005 * The same as ALmixer_PlaySourceTimed, but the sound is played without time limits. |
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1006 * @see ALmixer_PlaySourceTimed. |
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1007 */ |
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1008 ALint ALmixer_PlayChannelTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops); |
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1009 #else |
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1010 #define ALmixer_PlaySource(al_source, almixer_data, number_of_loops) ALmixer_PlaySourceTimed(al_source, almixer_data, number_of_loops, -1) |
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1011 #endif |
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1012 |
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1013 /** |
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1014 * Stops playback on a channel. |
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1015 * Stops playback on a channel and clears the channel so it can be played on again. |
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1016 * @note Callbacks will still be invoked, but the finished_naturally flag will be set to AL_FALSE. |
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1017 * @param which_channel The channel to halt or -1 to halt all channels. |
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1018 * @return The actual number of channels halted on success or -1 on error. |
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1019 */ |
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1020 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_HaltChannel(ALint which_channel); |
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1021 |
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1022 /** |
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1023 * Stops playback on a channel. |
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1024 * Stops playback on a channel and clears the channel so it can be played on again. |
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1025 * @note Callbacks will still be invoked, but the finished_naturally flag will be set to AL_FALSE. |
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1026 * @param al_source The source to halt or 0 to halt all sources. |
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1027 * @return The actual number of sources halted on success or -1 on error. |
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1028 */ |
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1029 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_HaltSource(ALuint al_source); |
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1030 |
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1031 /** |
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1032 * Rewinds the sound to the beginning for a given data. |
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1033 * Rewinds the actual data, but the effect |
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1034 * may not be noticed until the currently buffered data is played. |
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1035 * @param almixer_data The data to rewind. |
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1036 * @returns 0 on success or -1 on error. |
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1037 */ |
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1038 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindData(ALmixer_Data* almixer_data); |
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1039 |
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1040 /** |
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1041 * Rewinds the sound to the beginning that is playing on a specific channel. |
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1042 * If decoded all, rewind will instantly rewind it. Data is not |
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1043 * affected, so it will start at the "Seek"'ed positiond. |
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1044 * Streamed data will rewind the actual data, but the effect |
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1045 * may not be noticed until the currently buffered data is played. |
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1046 * @param which_channel The channel to rewind or -1 to rewind all channels. |
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1047 * @returns 0 on success or -1 on error. |
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1048 */ |
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1049 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindChannel(ALint which_channel); |
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1050 /** |
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1051 * Rewinds the sound to the beginning that is playing on a specific source. |
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1052 * If decoded all, rewind will instantly rewind it. Data is not |
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1053 * affected, so it will start at the "Seek"'ed positiond. |
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1054 * Streamed data will rewind the actual data, but the effect |
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1055 * may not be noticed until the currently buffered data is played. |
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1056 * @param al_source The source to rewind or 0 to rewind all sources. |
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1057 * @returns 1 on success or 0 on error. |
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1058 */ |
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1059 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindSource(ALuint al_source); |
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1060 |
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1061 /** |
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1062 * Seek the sound for a given data. |
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1063 * Seeks the actual data to the given millisecond. It |
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1064 * may not be noticed until the currently buffered data is played. |
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1065 * @param almixer_data |
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1066 * @param msec_pos The time position to seek to in the audio in milliseconds. |
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1067 * @returns 0 on success or -1 on error. |
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1068 */ |
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1069 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_SeekData(ALmixer_Data* almixer_data, ALuint msec_pos); |
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1070 |
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1071 /** |
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1072 * Pauses playback on a channel. |
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1073 * Pauses playback on a channel. Should have no effect on channels that aren't playing. |
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1074 * @param which_channel The channel to pause or -1 to pause all channels. |
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1075 * @return The actual number of channels paused on success or -1 on error. |
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1076 */ |
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1077 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PauseChannel(ALint which_channel); |
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1078 /** |
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1079 * Pauses playback on a source. |
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1080 * Pauses playback on a source. Should have no effect on source that aren't playing. |
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1081 * @param al_source The source to pause or -1 to pause all source. |
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1082 * @return The actual number of source paused on success or -1 on error. |
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1083 */ |
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1084 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PauseSource(ALuint al_source); |
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1085 |
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1086 /** |
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1087 * Resumes playback on a channel that is paused. |
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1088 * Resumes playback on a channel that is paused. Should have no effect on channels that aren't paused. |
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1089 * @param which_channel The channel to resume or -1 to resume all channels. |
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1090 * @return The actual number of channels resumed on success or -1 on error. |
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1091 */ |
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1092 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ResumeChannel(ALint which_channel); |
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1093 |
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1094 /** |
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1095 * Resumes playback on a source that is paused. |
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1096 * Resumes playback on a source that is paused. Should have no effect on sources that aren't paused. |
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1097 * @param al_source The source to resume or -1 to resume all sources. |
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1098 * @return The actual number of sources resumed on success or -1 on error. |
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1099 */ |
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1100 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ResumeSource(ALuint al_source); |
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1101 |
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1102 |
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1103 /** |
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1104 * Will cause a currently playing channel to stop playing in the specified number of milliseconds. |
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1105 * Will cause a currently playing channel to stop playing in the specified number of milliseconds. |
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1106 * This will override the value that was set when PlayChannelTimed or PlaySourceTimed was called |
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1107 * or override any previous calls to ExpireChannel or ExpireSource. |
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1108 * @param which_channel The channel to expire or -1 to apply to all channels. |
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1109 * @param number_of_milliseconds How many milliseconds from now until the expire triggers. |
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1110 * @return The actual number of channels this action is applied to on success or -1 on error. |
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1111 */ |
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1112 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ExpireChannel(ALint which_channel, ALint number_of_milliseconds); |
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1113 /** |
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1114 * Will cause a currently playing source to stop playing in the specified number of milliseconds. |
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1115 * Will cause a currently playing source to stop playing in the specified number of milliseconds. |
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1116 * This will override the value that was set when PlayChannelTimed or PlaySourceTimed was called |
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1117 * or override any previous calls to ExpireChannel or ExpireSource. |
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1118 * @param al_source The source to expire or 0 to apply to all sources. |
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1119 * @param number_of_milliseconds How many milliseconds from now until the expire triggers. |
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1120 * @return The actual number of sources this action is applied to on success or -1 on error. |
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1121 */ |
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1122 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ExpireSource(ALuint al_source, ALint number_of_milliseconds); |
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1123 |
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1124 /** |
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1125 * @} |
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1126 */ |
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1127 |
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1128 /** |
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1129 * @defgroup VolumeAPI Volume and Fading |
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1130 * @{ |
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1131 * Fade and volume functions directly call OpenAL functions related to AL_GAIN. |
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1132 * These functions are provided mostly for those who just want to play audio but are not planning |
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1133 * to use OpenAL features directly. |
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1134 * If you are using OpenAL directly (e.g. for 3D effects), you may want to be careful about using these as |
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1135 * they may fight/override values you directly set yourself. |
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1136 */ |
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1137 |
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1138 /** |
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1139 * Similar to ALmixer_PlayChannelTimed except that sound volume fades in from the minimum volume to the current AL_GAIN over the specified amount of time. |
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1140 * @see ALmixer_PlayChannelTimed. |
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1141 */ |
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1142 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeInChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks, ALint expire_ticks); |
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1143 |
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1144 #ifdef DOXYGEN_ONLY |
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1145 /** |
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1146 * The same as ALmixer_FadeInChannelTimed, but the sound is played without time limits. |
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1147 * @see ALmixer_FadeInChannelTimed, ALmixer_PlayChannel. |
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1148 */ |
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1149 ALint ALmixer_FadeInChannel(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks); |
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1150 #else |
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1151 #define ALmixer_FadeInChannel(which_channel, almixer_data, number_of_loops, fade_ticks) ALmixer_FadeInChannelTimed(which_channel, almixer_data, number_of_loops, fade_ticks, -1) |
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1152 #endif |
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1153 |
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1154 /** |
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1155 * Similar to ALmixer_PlaySourceTimed except that sound volume fades in from the minimum volume to the max volume over the specified amount of time. |
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1156 * @see ALmixer_PlaySourceTimed. |
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1157 */ |
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1158 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_FadeInSourceTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks, ALint expire_ticks); |
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1159 |
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1160 #ifdef DOXYGEN_ONLY |
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1161 /** |
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1162 * The same as ALmixer_FadeInSourceTimed, but the sound is played without time limits. |
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1163 * @see ALmixer_FadeInSourceTimed, ALmixer_PlaySource. |
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1164 */ |
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1165 extern ALuint ALmixer_FadeInSource(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks); |
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1166 #else |
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1167 #define ALmixer_FadeInSource(al_source, almixer_data, number_of_loops, fade_ticks) ALmixer_FadeInSourceTimed(al_source, almixer_data, number_of_loops, fade_ticks, -1) |
0 | 1168 #endif |
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1169 |
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1170 /** |
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1171 * Fade out a current playing channel. |
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1172 * Will fade out a currently playing channel over the specified period of time starting from now. |
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1173 * The volume will be changed from the current AL_GAIN level to the AL_MIN_GAIN. |
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1174 * The volume fade will interpolate over the specified period of time. |
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1175 * The playback will halt at the end of the time period. |
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1176 * @param which_channel The channel to fade or -1 to fade all playing channels. |
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1177 * @param fade_ticks In milliseconds, the amount of time the fade out should take to complete. |
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1178 * @return Returns -1 on error or the number of channels faded. |
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1179 */ |
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1180 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeOutChannel(ALint which_channel, ALuint fade_ticks); |
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1181 |
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1182 /** |
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1183 * Fade out a current playing source. |
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1184 * Will fade out a currently playing source over the specified period of time starting from now. |
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1185 * The volume will be changed from the current AL_GAIN level to the AL_MIN_GAIN. |
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1186 * The volume fade will interpolate over the specified period of time. |
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1187 * The playback will halt at the end of the time period. |
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1188 * @param al_source The source to fade or -1 to fade all playing sources. |
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1189 * @param fade_ticks In milliseconds, the amount of time the fade out should take to complete. |
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1190 * @return Returns -1 on error or the number of sources faded. |
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1191 */ |
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1192 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeOutSource(ALuint al_source, ALuint fade_ticks); |
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1193 |
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1194 /** |
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1195 * Gradually changes the volume from the current AL_GAIN to the specified volume. |
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1196 * Gradually changes the volume from the current AL_GAIN to the specified volume over the specified period of time. |
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1197 * This is some times referred to as volume ducking. |
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1198 * Note that this function works for setting the volume higher as well as lower. |
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1199 * @param which_channel The channel to fade or -1 to fade all playing channels. |
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1200 * @param fade_ticks In milliseconds, the amount of time the volume change should take to complete. |
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1201 * @param volume The volume to change to. Valid values are 0.0 to 1.0. |
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1202 * @return Returns -1 on error or the number of channels faded. |
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1203 */ |
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1204 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeChannel(ALint which_channel, ALuint fade_ticks, ALfloat volume); |
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1205 |
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1206 /** |
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1207 * Gradually changes the volume from the current AL_GAIN to the specified volume. |
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1208 * Gradually changes the volume from the current AL_GAIN to the specified volume over the specified period of time. |
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1209 * This is some times referred to as volume ducking. |
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1210 * Note that this function works for setting the volume higher as well as lower. |
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1211 * @param al_source The source to fade or -1 to fade all playing sources. |
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1212 * @param fade_ticks In milliseconds, the amount of time the volume change should take to complete. |
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1213 * @param volume The volume to change to. Valid values are 0.0 to 1.0. |
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1214 * @return Returns -1 on error or the number of sources faded. |
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1215 */ |
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1216 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeSource(ALuint al_source, ALuint fade_ticks, ALfloat volume); |
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1217 |
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1218 /** |
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1219 * Sets the volume via the AL_GAIN source property. |
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1220 * Sets the volume for a given channel via the AL_GAIN source property. |
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1221 * @param which_channel The channel to set the volume to or -1 to set on all channels. |
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1222 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1223 * @return AL_TRUE on success or AL_FALSE on error. |
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1224 */ |
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1225 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetVolumeChannel(ALint which_channel, ALfloat volume); |
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1226 |
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1227 /** |
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1228 * Sets the volume via the AL_GAIN source property. |
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1229 * Sets the volume for a given source via the AL_GAIN source property. |
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1230 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1231 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1232 * @return AL_TRUE on success or AL_FALSE on error. |
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1233 */ |
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1234 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetVolumeSource(ALuint al_source, ALfloat volume); |
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1235 |
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1236 /** |
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1237 * Gets the volume via the AL_GAIN source property. |
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1238 * Gets the volume for a given channel via the AL_GAIN source property. |
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1239 * @param which_channel The channel to get the volume from. |
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1240 * -1 will return the average volume set across all channels. |
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1241 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error. |
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1242 */ |
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1243 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetVolumeChannel(ALint which_channel); |
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1244 |
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1245 /** |
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1246 * Gets the volume via the AL_GAIN source property. |
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1247 * Gets the volume for a given source via the AL_GAIN source property. |
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1248 * @param al_source The source to get the volume from. |
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1249 * -1 will return the average volume set across all source. |
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1250 * @return Returns the volume for the specified source, or the average set volume for all sources, or -1.0 on error. |
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1251 */ |
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1252 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetVolumeSource(ALuint al_source); |
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1253 |
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1254 /** |
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1255 * Sets the maximum volume via the AL_MAX_GAIN source property. |
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1256 * Sets the maximum volume for a given channel via the AL_MAX_GAIN source property. |
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1257 * Max volumes will be clamped to this value. |
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1258 * @param which_channel The channel to set the volume to or -1 to set on all channels. |
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1259 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1260 * @return AL_TRUE on success or AL_FALSE on error. |
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1261 */ |
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1262 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMaxVolumeChannel(ALint which_channel, ALfloat volume); |
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1263 |
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1264 /** |
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1265 * Sets the maximum volume via the AL_MAX_GAIN source property. |
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1266 * Sets the maximum volume for a given source via the AL_MAX_GAIN source property. |
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1267 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1268 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1269 * @return AL_TRUE on success or AL_FALSE on error. |
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1270 */ |
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1271 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMaxVolumeSource(ALuint al_source, ALfloat volume); |
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1272 |
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1273 /** |
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1274 * Gets the max volume via the AL_MAX_GAIN source property. |
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1275 * Gets the max volume for a given channel via the AL_MAX_GAIN source property. |
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1276 * @param which_channel The channel to get the volume from. |
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1277 * -1 will return the average volume set across all channels. |
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1278 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error. |
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1279 */ |
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1280 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMaxVolumeChannel(ALint which_channel); |
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1281 |
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1282 /** |
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1283 * Gets the maximum volume via the AL_MAX_GAIN source property. |
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1284 * Gets the maximum volume for a given source via the AL_MAX_GAIN source property. |
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1285 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1286 * 0 will return the average volume set across all channels. |
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1287 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error. |
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1288 */ |
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1289 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMaxVolumeSource(ALuint al_source); |
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1290 |
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1291 /** |
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1292 * Sets the minimum volume via the AL_MIN_GAIN source property. |
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1293 * Sets the minimum volume for a given channel via the AL_MIN_GAIN source property. |
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1294 * Min volumes will be clamped to this value. |
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1295 * @param which_channel The channel to set the volume to or -1 to set on all channels. |
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1296 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1297 * @return AL_TRUE on success or AL_FALSE on error. |
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1298 */ |
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1299 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMinVolumeChannel(ALint which_channel, ALfloat volume); |
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1300 |
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1301 /** |
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1302 * Sets the minimum volume via the AL_MIN_GAIN source property. |
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1303 * Sets the minimum volume for a given source via the AL_MIN_GAIN source property. |
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1304 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1305 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1306 * @return AL_TRUE on success or AL_FALSE on error. |
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1307 */ |
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1308 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMinVolumeSource(ALuint al_source, ALfloat volume); |
0 | 1309 |
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1310 /** |
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1311 * Gets the min volume via the AL_MIN_GAIN source property. |
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1312 * Gets the min volume for a given channel via the AL_MIN_GAIN source property. |
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1313 * @param which_channel The channel to get the volume from. |
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1314 * -1 will return the average volume set across all channels. |
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1315 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error. |
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1316 */ |
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1317 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMinVolumeChannel(ALint which_channel); |
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1318 |
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1319 /** |
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1320 * Gets the min volume via the AL_MIN_GAIN source property. |
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1321 * Gets the min volume for a given source via the AL_MIN_GAIN source property. |
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1322 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1323 * 0 will return the average volume set across all channels. |
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1324 * @return Returns the volume for the specified channel, or the average set volume for all channels, or -1.0 on error. |
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1325 */ |
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1326 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMinVolumeSource(ALuint al_source); |
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1327 |
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1328 /** |
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1329 * Sets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1330 * Sets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1331 * @param new_volume The new volume level to be set. Range is 0.0 to 1.0 where 1.0 is the max volume. |
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1332 * @return AL_TRUE on success or AL_FALSE on an error. |
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1333 */ |
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1334 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMasterVolume(ALfloat new_volume); |
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1335 |
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1336 /** |
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1337 * Gets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1338 * Gets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1339 * @return The current volume level on the listener. -1.0 will be returned on an error. |
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1340 */ |
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1341 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMasterVolume(void); |
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1342 |
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1343 /** |
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1344 * @} |
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1345 */ |
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1346 |
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1347 /** |
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1348 * @defgroup QueryAPI Query APIs |
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1349 * @{ |
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1350 * Functions to query ALmixer. |
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1351 */ |
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1352 |
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1353 |
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1354 /** |
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1355 * Returns true if the specified channel is currently playing or paused, |
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1356 * or if -1 is passed the number of channels that are currently playing or paused. |
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1357 * @param which_channel The channel you want to know about or -1 to get the count of |
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1358 * currently playing/paused channels. |
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1359 * @return For a specific channel, 1 if the channel is playing or paused, 0 if not. |
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1360 * Or returns the count of currently playing/paused channels. |
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1361 * Or -1 on an error. |
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1362 */ |
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1363 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsActiveChannel(ALint which_channel); |
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1364 |
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1365 /** |
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1366 * Returns true if the specified source is currently playing or paused, |
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1367 * or if -1 is passed the number of sources that are currently playing or paused. |
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1368 * @param al_source The channel you want to know about or -1 to get the count of |
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1369 * currently playing/paused sources. |
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1370 * @return For a specific sources, 1 if the channel is playing or paused, 0 if not. |
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1371 * Or returns the count of currently playing/paused sources. |
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1372 * Or -1 on an error. |
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1373 */ |
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1374 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsActiveSource(ALuint al_source); |
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1375 |
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1376 /** |
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1377 * Returns true if the specified channel is currently playing. |
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1378 * or if -1 is passed the number of channels that are currently playing. |
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1379 * @param which_channel The channel you want to know about or -1 to get the count of |
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1380 * currently playing channels. |
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1381 * @return For a specific channel, 1 if the channel is playing, 0 if not. |
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1382 * Or returns the count of currently playing channels. |
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1383 * Or -1 on an error. |
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1384 */ |
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1385 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPlayingChannel(ALint which_channel); |
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1386 |
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1387 /** |
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1388 * Returns true if the specified sources is currently playing. |
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1389 * or if -1 is passed the number of sources that are currently playing. |
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1390 * @param al_source The sources you want to know about or -1 to get the count of |
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1391 * currently playing sources. |
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1392 * @return For a specific source, 1 if the source is playing, 0 if not. |
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1393 * Or returns the count of currently playing sources. |
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1394 * Or -1 on an error. |
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1395 */ |
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1396 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPlayingSource(ALuint al_source); |
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1397 |
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1398 /** |
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1399 * Returns true if the specified channel is currently paused. |
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1400 * or if -1 is passed the number of channels that are currently paused. |
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1401 * @param which_channel The channel you want to know about or -1 to get the count of |
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1402 * currently paused channels. |
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1403 * @return For a specific channel, 1 if the channel is paused, 0 if not. |
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1404 * Or returns the count of currently paused channels. |
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1405 * Or -1 on an error. |
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1406 */ |
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1407 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPausedChannel(ALint which_channel); |
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1408 |
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1409 /** |
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1410 * Returns true if the specified sources is currently paused. |
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1411 * or if -1 is passed the number of sources that are currently paused. |
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1412 * @param al_source The source you want to know about or -1 to get the count of |
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1413 * currently paused sources. |
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1414 * @return For a specific source, 1 if the source is paused, 0 if not. |
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1415 * Or returns the count of currently paused sources. |
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1416 * Or -1 on an error. |
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1417 */ |
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1418 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPausedSource(ALuint al_source); |
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1419 |
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1420 /** |
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1421 * Returns the number of channels that are currently available for playback (not playing, not paused). |
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1422 * @return The number of channels that are currently free. |
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1423 */ |
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1424 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountAllFreeChannels(void); |
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1425 |
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1426 /** |
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1427 * Returns the number of channels that are currently available for playback (not playing, not paused), |
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1428 * excluding the channels that have been reserved. |
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1429 * @return The number of channels that are currently in free, excluding the channels that have been reserved. |
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1430 * @see ALmixer_ReserveChannels |
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1431 */ |
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1432 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountUnreservedFreeChannels(void); |
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1433 |
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1434 /** |
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1435 * Returns the number of channels that are currently in use (playing/paused), |
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1436 * excluding the channels that have been reserved. |
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1437 * @return The number of channels that are currently in use. |
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1438 * @see ALmixer_ReserveChannels |
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1439 */ |
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1440 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountAllUsedChannels(void); |
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1441 |
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1442 /** |
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1443 * Returns the number of channels that are currently in use (playing/paused), |
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1444 * excluding the channels that have been reserved. |
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1445 * @return The number of channels that are currently in use excluding the channels that have been reserved. |
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1446 * @see ALmixer_ReserveChannels |
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1447 */ |
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1448 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountUnreservedUsedChannels(void); |
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1449 |
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1450 |
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1451 #ifdef DOXYGEN_ONLY |
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1452 /** |
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1453 * Returns the number of allocated channels. |
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1454 * This is just a convenience alias to ALmixer_AllocateChannels(-1). |
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1455 * @see ALmixer_AllocateChannels |
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1456 */ |
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1457 ALint ALmixer_CountTotalChannels(void); |
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1458 #else |
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1459 #define ALmixer_CountTotalChannels() ALmixer_AllocateChannels(-1) |
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1460 #endif |
0 | 1461 |
1462 | |
1463 | |
2
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1464 |
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1465 #ifdef DOXYGEN_ONLY |
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1466 /** |
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1467 * Returns the number of allocated channels. |
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1468 * This is just a convenience alias to ALmixer_ReserveChannels(-1). |
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1469 * @see ALmixer_ReserveChannels |
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1470 */ |
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1471 ALint ALmixer_CountReservedChannels(void); |
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1472 #else |
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1473 #define ALmixer_CountReservedChannels() ALmixer_ReserveChannels(-1) |
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1474 #endif |
0 | 1475 |
1476 | |
2
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1477 /** |
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1478 * @} |
0 | 1479 */ |
1480 | |
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1481 /** |
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1482 * @defgroup DebugAPI Debug APIs |
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1483 * @{ |
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1484 * Functions for debugging purposes. These may be removed in future versions. |
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1485 */ |
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1486 |
0 | 1487 |
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1488 /* For testing */ |
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1489 #if 0 |
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1490 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputAttributes(void); |
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1491 #endif |
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1492 /** This function may be removed in the future. For debugging. Prints to stderr. Lists the decoders available. */ |
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1493 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputDecoders(void); |
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1494 /** This function may be removed in the future. For debugging. Prints to stderr. */ |
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1495 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputOpenALInfo(void); |
0 | 1496 |
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1497 /** This function may be removed in the future. Returns true if compiled with threads, false if not. */ |
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1498 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_CompiledWithThreadBackend(void); |
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1499 |
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1500 /** |
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1501 * @} |
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1502 */ |
0 | 1503 |
1504 | |
1505 | |
1506 | |
1507 /* Ends C function definitions when using C++ */ | |
1508 #ifdef __cplusplus | |
1509 } | |
1510 #endif | |
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1511 |
0 | 1512 |
1513 #endif /* _SDL_ALMIXER_H_ */ | |
1514 | |
1515 /* end of SDL_ALmixer.h ... */ | |
1516 | |
1517 |