Mercurial > almixer_isolated
annotate ALmixer.h @ 43:a55e1f3d8f10
merged
author | Eric Wing <ewing@anscamobile.com> |
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date | Tue, 30 Aug 2011 19:43:49 -0700 |
parents | 05e5dc4817a4 |
children | 56855942fdc6 |
rev | line source |
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0 | 1 /* |
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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 */ | |
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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 * Home Page: http://playcontrol.net/opensource/ALmixer |
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70 */ |
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71 |
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72 /** |
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73 * @file |
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74 * ALmixer (which I sometimes call "SDL-OpenAL-Mixer" or "SDL_ALmixer") is a cross-platform audio library built |
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75 * on top of OpenAL to make playing and managing sounds easier. |
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76 * ALmixer provides a simple API inspired by SDL_mixer to make playing sounds easy |
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77 * with having to worry about directly dealing with OpenAL sources, buffers, |
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78 * and buffer queuing directly. |
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79 * ALmixer currently utilizes SDL_sound behind the scenes to decode |
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80 * various audio formats such as WAV, MP3, AAC, MP4, OGG, etc. |
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81 * |
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82 * This library is targeted towards two major groups: |
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83 * - People who just want an easy, high performance, way to play audio (don't care if its OpenAL or not) |
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84 * - People who want to an easy way to play audio in OpenAL but still want access to OpenAL directly. |
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85 * |
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86 * ALmixer exposes OpenAL sources in the API so you can freely use ALmixer |
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87 * in larger OpenAL applications that need to apply OpenAL 3D effects and features |
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88 * to playing sounds. |
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89 * |
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90 * The API is heavily influenced and inspired by SDL_mixer, though there is one major |
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91 * conceptual design difference. ALmixer doesn't divide sound and music playback into two |
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92 * separate play APIs. Instead, there is one unified play API and you specify via the |
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93 * load API whether you want the audio resource loaded as a stream or completely preloaded. |
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94 * This allows you to have any arbitrary number of streaming sources playing simultaneously |
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95 * (such as music and speech) unlike SDL_mixer where you are limited to only one "music" |
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96 * channel. |
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97 * |
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98 * A less major conceptual design difference is every "Channel" API has a corresponding "Source" API. |
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99 * Every "channel" (in the SDL_mixer definition context) maps to a corresponding OpenAL source id. You can use |
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100 * this source ID directly with OpenAL API commands to utilize OpenAL effects such as position, Doppler, etc. |
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101 * Convenience APIs are provided to let you convert channel numbers to source ids and vice-versa. |
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102 * |
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103 * 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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104 * ALmixer callbacks allow you to pass user_data (aka context) pointers through the callback functions. |
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105 * |
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106 * @note There are some #defines you can set to change the behavior at compile time. Most you shouldn't touch. |
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107 * The one worth noting is ENABLE_ALMIXER_THREADS. If enabled, ALmixer_Update() is automatically called on a |
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108 * 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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109 * code won't break.) Having Update run in a separate thread has some advantages, particularly for streaming |
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110 * audio as all the OpenAL buffer queuing happens in this function. It is less likely the background thread will |
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111 * be blocked for long periods and thus less likely your buffer queues will be starved. However, this means you |
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112 * need to be extra careful about what you do in callback functions as they are invoked from the background thread. |
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113 * I still consider this feature a experimental (though I am starting to use it more myself) and there |
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114 * may still be bugs. |
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115 * |
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116 * @author Eric Wing |
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117 * |
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118 * Home Page: http://playcontrol.net/opensource/ALmixer |
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119 */ |
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120 |
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121 |
0 | 122 #ifndef _SDL_ALMIXER_H_ |
123 #define _SDL_ALMIXER_H_ | |
124 | |
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125 |
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126 #ifndef DOXYGEN_SHOULD_IGNORE_THIS |
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127 /** @cond DOXYGEN_SHOULD_IGNORE_THIS */ |
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128 |
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129 /* Note: For Doxygen to produce clean output, you should set the |
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130 * PREDEFINED option to remove ALMIXER_DECLSPEC, ALMIXER_CALL, and |
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131 * the DOXYGEN_SHOULD_IGNORE_THIS blocks. |
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132 * PREDEFINED = DOXYGEN_SHOULD_IGNORE_THIS=1 ALMIXER_DECLSPEC= ALMIXER_CALL= |
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133 */ |
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135 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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136 #if defined(_WIN32) |
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137 #if defined(ALMIXER_BUILD_LIBRARY) |
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138 #define ALMIXER_DECLSPEC __declspec(dllexport) |
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139 #else |
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140 #define ALMIXER_DECLSPEC __declspec(dllimport) |
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141 #endif |
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142 #else |
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143 #if defined(ALMIXER_BUILD_LIBRARY) |
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144 #if defined (__GNUC__) && __GNUC__ >= 4 |
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145 #define ALMIXER_DECLSPEC __attribute__((visibility("default"))) |
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146 #else |
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147 #define ALMIXER_DECLSPEC |
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148 #endif |
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149 #else |
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150 #define ALMIXER_DECLSPEC |
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151 #endif |
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152 #endif |
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154 #if defined(_WIN32) |
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155 #define ALMIXER_CALL __cdecl |
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156 #else |
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157 #define ALMIXER_CALL |
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158 #endif |
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159 #else |
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160 #include "SDL_types.h" /* will include begin_code.h which is what I really want */ |
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161 #define ALMIXER_DECLSPEC DECLSPEC |
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162 #define ALMIXER_CALL SDLCALL |
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163 #endif |
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164 |
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165 /** @endcond DOXYGEN_SHOULD_IGNORE_THIS */ |
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166 #endif /* DOXYGEN_SHOULD_IGNORE_THIS */ |
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167 |
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168 |
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169 |
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170 /* Needed for OpenAL types since altypes.h was removed in 1.1 */ |
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171 #include "al.h" |
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173 /* Set up for C function definitions, even when using C++ */ | |
174 #ifdef __cplusplus | |
175 extern "C" { | |
176 #endif | |
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177 |
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178 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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179 /** |
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180 * Struct that contains the version information of this library. |
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181 * This represents the library's version as three levels: major revision |
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182 * (increments with massive changes, additions, and enhancements), |
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183 * minor revision (increments with backwards-compatible changes to the |
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184 * major revision), and patchlevel (increments with fixes to the minor |
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185 * revision). |
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186 * @see ALMIXER_VERSION, ALmixer_GetLinkedVersion |
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187 */ |
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188 typedef struct ALmixer_version |
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189 { |
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190 ALubyte major; |
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191 ALubyte minor; |
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192 ALubyte patch; |
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193 } ALmixer_version; |
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194 #else |
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195 #include "SDL_version.h" |
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196 #define ALmixer_version SDL_version |
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197 #endif |
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198 |
0 | 199 /* Printable format: "%d.%d.%d", MAJOR, MINOR, PATCHLEVEL |
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200 */ |
0 | 201 #define ALMIXER_MAJOR_VERSION 0 |
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202 #define ALMIXER_MINOR_VERSION 2 |
0 | 203 #define ALMIXER_PATCHLEVEL 0 |
204 | |
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205 |
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206 /** |
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207 * @defgroup CoreOperation Initialization, Tear-down, and Core Operational Commands |
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208 * @{ |
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209 * Functions for setting up and using ALmixer. |
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210 */ |
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211 |
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212 |
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213 /** |
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214 * This macro fills in a version structure with the version of the |
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215 * library you compiled against. This is determined by what header the |
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216 * compiler uses. Note that if you dynamically linked the library, you might |
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217 * have a slightly newer or older version at runtime. That version can be |
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218 * determined with ALmixer_GetLinkedVersion(), which, unlike |
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219 * ALMIXER_GET_COMPILED_VERSION, is not a macro. |
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220 * |
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221 * @note When compiled with SDL, this macro can be used to fill a version structure |
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222 * compatible with SDL_version. |
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223 * |
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224 * @param X A pointer to a ALmixer_version struct to initialize. |
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225 * |
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226 * @see ALmixer_version, ALmixer_GetLinkedVersion |
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227 */ |
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228 #define ALMIXER_GET_COMPILED_VERSION(X) \ |
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229 { \ |
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230 (X)->major = ALMIXER_MAJOR_VERSION; \ |
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231 (X)->minor = ALMIXER_MINOR_VERSION; \ |
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232 (X)->patch = ALMIXER_PATCHLEVEL; \ |
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233 } |
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234 |
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235 /** |
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236 * Gets the library version of the dynamically linked ALmixer you are using. |
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237 * This gets the version of ALmixer that is linked against your program. |
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238 * If you are using a shared library (DLL) version of ALmixer, then it is |
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239 * possible that it will be different than the version you compiled against. |
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240 * |
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241 * This is a real function; the macro ALMIXER_GET_COMPILED_VERSION |
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242 * tells you what version of tErrorLib you compiled against: |
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243 * |
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244 * @code |
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245 * ALmixer_version compiled; |
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246 * ALmixer_version linked; |
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247 * |
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248 * ALMIXER_GET_COMPILED_VERSION(&compiled); |
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249 * ALmixer_GetLinkedVersion(&linked); |
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250 * printf("We compiled against tError version %d.%d.%d ...\n", |
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251 * compiled.major, compiled.minor, compiled.patch); |
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252 * printf("But we linked against tError version %d.%d.%d.\n", |
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253 * linked.major, linked.minor, linked.patch); |
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254 * @endcode |
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255 * |
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256 * @see ALmixer_version, ALMIXER_GET_COMPILED_VERSION |
0 | 257 */ |
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258 extern ALMIXER_DECLSPEC const ALmixer_version* ALMIXER_CALL ALmixer_GetLinkedVersion(void); |
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259 |
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260 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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261 /** |
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262 * Gets the last error string that was set by the system and clears the error. |
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263 * |
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264 * @note When compiled with SDL, this directly uses SDL_GetError. |
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265 * |
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266 * @return Returns a string containing the last error or "" when no error is set. |
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267 */ |
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268 extern ALMIXER_DECLSPEC const char* ALMIXER_CALL ALmixer_GetError(void); |
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269 /** |
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270 * Sets an error string that can be retrieved by ALmixer_GetError. |
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271 * |
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272 * @note When compiled with SDL, this directly uses SDL_SetError. |
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273 * |
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274 * param The error string to set. |
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275 */ |
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276 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_SetError(const char *fmt, ...); |
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277 #else |
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278 #include "SDL_error.h" |
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279 /** |
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280 * Gets the last error string that was set by the system and clears the error. |
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281 * |
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282 * @note When compiled with SDL, this directly uses SDL_GetError. |
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283 * |
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284 * @return Returns a string containing the last error or "" when no error is set. |
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285 */ |
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286 #define ALmixer_GetError SDL_GetError |
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287 /** |
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288 * Sets an error string that can be retrieved by ALmixer_GetError. |
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289 * |
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290 * @note When compiled with SDL, this directly uses SDL_SetError. |
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291 * |
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292 * param The error string to set. |
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293 */ |
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294 #define ALmixer_SetError SDL_SetError |
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295 #endif |
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296 |
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297 |
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298 #ifdef ALMIXER_COMPILE_WITHOUT_SDL |
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299 #include "ALmixer_RWops.h" |
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300 #else |
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301 #include "SDL_rwops.h" |
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302 /** |
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303 * A struct that mimicks the SDL_RWops structure. |
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304 * |
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305 * @note When compiled with SDL, this directly uses SDL_RWops. |
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306 */ |
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307 #define ALmixer_RWops SDL_RWops |
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308 #endif |
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309 |
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310 |
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311 #define ALMIXER_DEFAULT_FREQUENCY 0 |
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312 #define ALMIXER_DEFAULT_REFRESH 0 |
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313 #define ALMIXER_DEFAULT_NUM_CHANNELS 16 |
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314 #define ALMIXER_DEFAULT_NUM_SOURCES ALMIXER_DEFAULT_NUM_CHANNELS |
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315 |
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316 /** |
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317 * This is the recommended Init function. This will initialize the context, SDL_sound, |
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318 * and the mixer system. You should call this in the setup of your code, after SDL_Init. |
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319 * If you attempt to bypass this function, you do so at your own risk. |
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320 * |
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321 * @note ALmixer expects the SDL audio subsystem to be disabled. In some cases, an enabled |
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322 * SDL audio subsystem will interfere and cause problems in your app. This Init method explicitly |
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323 * disables the SDL subsystem if SDL is compiled in. |
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324 * |
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325 * @note The maximum number of sources is OpenAL implementation dependent. |
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326 * Currently 16 is lowest common denominator for all OpenAL implementations in current use. |
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327 * 32 is currently the second lowest common denominator. |
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328 * If you try to allocate more sources than are actually available, this function may return false depending |
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329 * if the OpenAL implementation returns an error or not. It is possible for OpenAL to silently fail |
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330 * so be very careful about picking too many sources. |
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331 * |
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332 * @param playback_frequency The sample rate you want OpenAL to play at, e.g. 44100 |
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333 * Note that OpenAL is not required to actually respect this value. |
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334 * Pass in 0 or ALMIXER_DEFAULT_FREQUENCY to specify you want to use your implementation's default value. |
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335 * @param num_sources The number of OpenAL sources (also can be thought of as |
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336 * SDL_mixer channels) you wish to allocate. |
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337 * Pass in 0 or ALMIXER_DEFAULT_NUM_SOURCES to use ALmixer's default value. |
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338 * @param refresh_rate The refresh rate you want OpenAL to operate at. |
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339 * Note that OpenAL is not required to respect this value. |
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340 * Pass in 0 or ALMIXER_DEFAULT_REFRESH to use OpenAL default behaviors. |
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341 * @return Returns AL_FALSE on a failure or AL_TRUE if successfully initialized. |
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342 */ |
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343 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_Init(ALuint playback_frequency, ALuint num_sources, ALuint refresh_rate); |
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345 /** |
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346 * InitContext will only initialize the OpenAL context (and not the mixer part). |
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347 * Note that SDL_Sound is also initialized here because load order matters |
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348 * because SDL audio will conflict with OpenAL when using SMPEG. This is only |
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349 * provided as a backdoor and is not recommended. |
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350 * |
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351 * @note This is a backdoor in case you need to initialize the AL context and |
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352 * the mixer system separately. I strongly recommend avoiding these two functions |
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353 * and use the normal Init() function. |
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354 */ |
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355 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_InitContext(ALuint playback_frequency, ALuint refresh_rate); |
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356 |
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357 /** |
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358 * InitMixer will only initialize the Mixer system. This is provided in the case |
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359 * that you need control over the loading of the context. You may load the context |
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360 * yourself, and then call this function. This is not recommended practice, but is |
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361 * provided as a backdoor in case you have good reason to |
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362 * do this. Be warned that if ALmixer_InitMixer() fails, |
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363 * it will not clean up the AL context. Also be warned that Quit() still does try to |
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364 * clean up everything. |
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365 * |
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366 * @note This is a backdoor in case you need to initialize the AL context and |
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367 * the mixer system separately. I strongly recommend avoiding these two functions |
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368 * and use the normal Init() function. |
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369 */ |
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370 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_InitMixer(ALuint num_sources); |
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371 |
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372 /** |
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373 * (EXPERIMENTAL) Call to notify ALmixer that your device needs to handle an interruption. |
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374 * (EXPERIMENTAL) For devices like iOS that need special handling for interruption events like phone calls and alarms, |
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375 * this function will do the correct platform correct thing to handle the interruption w.r.t. OpenAL. |
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376 */ |
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377 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_BeginInterruption(void); |
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378 |
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379 /** |
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380 * (EXPERIMENTAL) Call to notify ALmixer that your device needs to resume from an interruption. |
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381 * (EXPERIMENTAL) For devices like iOS that need special handling for interruption events like phone calls and alarms, |
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382 * this function will do the correct platform correct thing to resume from the interruption w.r.t. OpenAL. |
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383 */ |
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384 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_EndInterruption(void); |
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385 |
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386 /** |
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387 * (EXPERIMENTAL) Call to determine if in an interruption. |
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388 * (EXPERIMENTAL) For devices like iOS that need special handling for interruption events like phone calls and alarms, |
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389 * this function will do the correct platform correct thing to determine if in an interruption. |
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390 */ |
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391 extern ALMIXER_DECLSPEC ALboolean ALmixer_IsInInterruption(void); |
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392 |
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393 |
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394 /** |
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395 * This shuts down ALmixer. Please remember to free your ALmixer_Data* instances |
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396 * before calling this method. |
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397 */ |
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398 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_Quit(void); |
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399 /** |
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400 * Returns whether ALmixer has been initializatized (via Init) or not. |
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401 * @return Returns true for initialized and false for not initialized. |
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402 */ |
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403 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_IsInitialized(void); |
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404 |
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405 /** |
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406 * Returns the frequency that OpenAL is set to. |
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407 * @note This function is not guaranteed to give correct information and is OpenAL implementation dependent. |
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408 * @return Returns the frequency, e.g. 44100. |
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409 */ |
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410 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_GetFrequency(void); |
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411 |
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412 /** |
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413 * Let's you change the maximum number of channels/sources available. |
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414 * This function is not heavily tested. It is probably better to simply initialize |
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415 * ALmixer with the number of sources you want when you initialize it instead of |
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416 * dynamically changing it later. |
0 | 417 */ |
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418 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_AllocateChannels(ALint num_chans); |
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419 |
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420 /** |
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421 * Allows you to reserve a certain number of channels so they won't be automatically |
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422 * allocated to play on. |
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423 * This function will effectively block off a certain number of channels so they won't |
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424 * be automatically assigned to be played on when you call various play functions |
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425 * (applies to both play-channel and play-source functions since they are the same under the hood). |
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426 * The lowest number channels will always be blocked off first. |
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427 * For example, if there are 16 channels available, and you pass 2 into this function, |
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428 * channels 0 and 1 will be reserved so they won't be played on automatically when you specify |
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429 * you want to play a sound on any available channel/source. You can |
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430 * still play on channels 0 and 1 if you explicitly designate you want to play on their channel |
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431 * number or source id. |
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432 * Setting back to 0 will clear all the reserved channels so all will be available again for |
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433 * auto-assignment. |
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434 * As an example, this feature can be useful if you always want your music to be on channel 0 and |
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435 * speech on channel 1 and you don't want sound effects to ever occupy those channels. This allows |
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436 * you to build in certain assumptions about your code, perhaps for deciding which data you want |
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437 * to analyze in a data callback. |
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438 * Specifying the number of reserve channels to the maximum number of channels will effectively |
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439 * disable auto-assignment. |
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440 * @param number_of_reserve_channels The number of channels/sources to reserve. |
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441 * Or pass -1 to find out how many channels are currently reserved. |
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442 * @return Returns the number of currently reserved channels. |
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443 */ |
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444 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ReserveChannels(ALint number_of_reserve_channels); |
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445 |
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446 |
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447 /** |
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448 * The update function that allows ALmixer to update its internal state. |
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449 * If not compiled with/using threads, this function must be periodically called |
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450 * to poll ALmixer to force streamed music and other events to |
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451 * take place. |
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452 * The typical place to put this function is in your main-loop. |
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453 * If threads are enabled, then this function just |
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454 * returns 0 and is effectively a no-op. With threads, it is not necessary to call this function |
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455 * because updates are handled internally on another thread. However, because threads are still considered |
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456 * experimental, it is recommended you call this function in a proper place in your code in case |
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457 * future versions of this library need to abandon threads. |
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458 * @return Returns 0 if using threads. If not using threads, for debugging purposes, it returns |
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459 * the number of buffers queued during the loop, or a negative value indicating the numer of errors encountered. |
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460 * This is subject to change and should not be relied on. |
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461 */ |
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462 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_Update(void); |
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463 |
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464 /** |
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465 * @} |
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466 */ |
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467 |
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468 /** |
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469 * @defgroup LoadAPI Load Audio Functions |
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470 * @{ |
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471 * Functions for loading and unloading audio data. |
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472 */ |
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473 |
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474 |
0 | 475 |
476 /* | |
477 #define ALmixer_AudioInfo Sound_AudioInfo | |
478 */ | |
479 | |
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480 /* |
0 | 481 #define ALMIXER_DEFAULT_BUFFERSIZE 32768 |
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482 #define ALMIXER_DEFAULT_BUFFERSIZE 16384 |
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483 */ |
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484 #define ALMIXER_DEFAULT_BUFFERSIZE 4096 |
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486 /* You probably never need to use these macros directly. */ |
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487 #ifndef ALMIXER_DISABLE_PREDECODED_PRECOMPUTE_BUFFER_SIZE_OPTIMIZATION |
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488 #define ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE ALMIXER_DEFAULT_BUFFERSIZE * 4 |
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489 #else |
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490 /* I'm picking a smaller buffer because ALmixer will try to create a new larger buffer |
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491 * based on the length of the audio. So creating a large block up-front might just be a waste. |
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492 * However, if my attempts fail for some reason, this buffer size becomes a fallback. |
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493 * Having too small of a buffer might cause performance bottlenecks. |
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494 */ |
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495 #define ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE 1024 |
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496 #endif |
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497 |
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498 /** |
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499 * Specifies the maximum number of queue buffers to use for a sound stream. |
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500 * Default Queue Buffers must be at least 2. |
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501 */ |
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502 /* |
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503 #define ALMIXER_DEFAULT_QUEUE_BUFFERS 5 |
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504 */ |
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505 #define ALMIXER_DEFAULT_QUEUE_BUFFERS 25 |
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506 /** |
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507 * Specifies the number of queue buffers initially filled when first loading a stream. |
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508 * Default startup buffers should be at least 1. */ |
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509 #define ALMIXER_DEFAULT_STARTUP_BUFFERS 8 |
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510 /* |
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511 #define ALMIXER_DEFAULT_STARTUP_BUFFERS 2 |
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512 */ |
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513 #define ALMIXER_DEFAULT_BUFFERS_TO_QUEUE_PER_UPDATE_PASS 4 |
0 | 514 |
1 | 515 /* |
0 | 516 #define ALMIXER_DECODE_STREAM 0 |
517 #define ALMIXER_DECODE_ALL 1 | |
1 | 518 */ |
0 | 519 |
1 | 520 /* This is a trick I picked up from Lua. Doing the typedef separately |
521 * (and I guess before the definition) instead of a single | |
522 * entry: typedef struct {...} YourName; seems to allow me | |
523 * to use forward declarations. Doing it the other way (like SDL) | |
524 * seems to prevent me from using forward declarions as I get conflicting | |
525 * definition errors. I don't really understand why though. | |
526 */ | |
527 typedef struct ALmixer_Data ALmixer_Data; | |
528 typedef struct ALmixer_AudioInfo ALmixer_AudioInfo; | |
0 | 529 |
1 | 530 /** |
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531 * Roughly the equvialent to the Sound_AudioInfo struct in SDL_sound. |
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532 * Types have been changed to use AL types because I know those are available. |
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533 * This is different than SDL which uses fixed types so there might be subtle |
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534 * things you need to pay attention to.. |
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535 * @note Originally, I just used the Sound_AudioInfo directly, but |
1 | 536 * I've been trying to reduce the header dependencies for this file. |
537 * But more to the point, I've been interested in dealing with the | |
538 * WinMain override problem Josh faced when trying to use SDL components | |
539 * in an MFC app which didn't like losing control of WinMain. | |
540 * My theory is that if I can purge the header of any thing that | |
541 * #include's SDL_main.h, then this might work. | |
542 * So I am now introducing my own AudioInfo struct. | |
543 */ | |
544 struct ALmixer_AudioInfo | |
545 { | |
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546 ALushort format; /**< Equivalent of SDL_AudioSpec.format. */ |
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547 ALubyte channels; /**< Number of sound channels. 1 == mono, 2 == stereo. */ |
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548 ALuint rate; /**< Sample rate; frequency of sample points per second. */ |
1 | 549 }; |
0 | 550 |
551 | |
552 | |
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553 /** |
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554 * This is a general loader function to load an audio resource from an RWops. |
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555 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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556 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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557 * @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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558 * decoder to use. |
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559 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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560 * 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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561 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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562 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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563 * @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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564 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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565 * Buffer underrun conditions will also attempt to restart using this value. |
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566 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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567 * of buffers that will be queued. |
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568 * @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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569 * 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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570 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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571 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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572 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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573 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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574 */ |
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575 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, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
0 | 576 |
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577 #ifdef DOXYGEN_ONLY |
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578 /** |
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579 * This is the loader function to load an audio resource from an RWops as a stream. |
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580 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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581 * @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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582 * decoder to use. |
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583 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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584 * 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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585 * @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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586 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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587 * Buffer underrun conditions will also attempt to restart using this value. |
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588 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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589 * of buffers that will be queued. |
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590 * @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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591 * 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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592 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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593 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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594 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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595 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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596 */ |
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597 ALmixer_Data* ALmixer_LoadStream_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
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598 #else |
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599 #define ALmixer_LoadStream_RW(rw_ops, file_ext, buffer_size, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) ALmixer_LoadSample_RW(rw_ops,file_ext, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) |
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600 #endif |
0 | 601 |
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602 #ifdef DOXYGEN_ONLY |
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603 /** |
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604 * This is the loader function to completely preload an audio resource from an RWops into RAM. |
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605 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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606 * @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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607 * decoder to use. |
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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_LoadAll_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALuint access_data); |
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616 #else |
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617 #define ALmixer_LoadAll_RW(rw_ops, file_ext, access_data) ALmixer_LoadSample_RW(rw_ops, fileext, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, 0, access_data) |
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618 #endif |
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620 /** |
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621 * This is a general loader function to load an audio resource from a file. |
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622 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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623 * @param file_name The file of the audio resource you want to load. |
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624 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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625 * 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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626 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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627 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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628 * @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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629 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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630 * Buffer underrun conditions will also attempt to restart using this value. |
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631 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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632 * of buffers that will be queued. |
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633 * @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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634 * 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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635 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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636 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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637 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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638 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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639 */ |
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640 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, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
0 | 641 |
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642 #ifdef DOXYGEN_ONLY |
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643 /** |
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644 * This is the loader function to load an audio resource from a file. |
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645 * @param file_name The file to the audio resource you want to load. |
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646 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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647 * 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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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 * Buffer underrun conditions will also attempt to restart using this value. |
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651 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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652 * of buffers that will be queued. * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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653 * Buffer underrun conditions will also attempt to restart using this value. |
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654 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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655 * of buffers that will be queued. |
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656 * @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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657 * 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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658 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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659 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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660 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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661 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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662 */ |
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663 ALmixer_Data* ALmixer_LoadStream(const char* file_name, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
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664 #else |
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665 #define ALmixer_LoadStream(file_name, buffer_size, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) ALmixer_LoadSample(file_name, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) |
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666 #endif |
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667 |
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668 #ifdef DOXYGEN_ONLY |
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669 /** |
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670 * This is the loader function to completely preload an audio resource from a file into RAM. |
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671 * @param file_name The file to the audio resource you want to load. |
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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_LoadAll(const char* file_name, ALuint access_data); |
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680 #else |
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681 #define ALmixer_LoadAll(file_name, access_data) ALmixer_LoadSample(file_name, ALMIXER_DEFAULT_PREDECODED_BUFFERSIZE, AL_TRUE, 0, 0, 0, access_data) |
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682 #endif |
0 | 683 |
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684 /** |
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685 * 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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686 * Use at your own risk. |
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687 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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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 buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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693 * 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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694 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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695 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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696 * @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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697 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
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698 * Buffer underrun conditions will also attempt to restart using this value. |
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699 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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700 * of buffers that will be queued. |
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701 * @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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702 * 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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703 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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704 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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705 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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706 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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707 */ |
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708 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, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
0 | 709 |
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710 #ifdef DOXYGEN_ONLY |
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711 /** |
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712 * 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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713 * Use at your own risk. |
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714 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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|
715 * @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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716 * decoder to use. Pass "raw" for raw formats. |
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717 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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718 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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719 * 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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720 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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721 * @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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722 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
42
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41
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723 * Buffer underrun conditions will also attempt to restart using this value. |
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724 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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725 * of buffers that will be queued. |
2
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726 * @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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727 * 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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728 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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729 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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730 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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|
731 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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732 */ |
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41
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733 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, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
2
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734 #else |
42
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735 #define ALmixer_LoadStream_RAW_RW(rw_ops, file_ext, desired_format, buffer_size, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) ALmixer_LoadSample_RAW_RW(rw_ops, file_ext, desired_format, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) |
2
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736 #endif |
0 | 737 |
2
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738 #ifdef DOXYGEN_ONLY |
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739 /** |
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740 * 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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741 * Use at your own risk. |
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742 * @param rw_ops The rwops pointing to the audio resource you want to load. |
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743 * @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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744 * decoder to use. Pass "raw" for raw formats. |
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745 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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746 * @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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747 * 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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748 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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749 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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|
750 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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751 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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752 */ |
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Eric Wing <ewing@anscamobile.com>
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41
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753 ALmixer_Data* ALmixer_LoadAll_RAW_RW(ALmixer_RWops* rw_ops, const char* file_ext, ALmixer_AudioInfo* desired_format, ALuint access_data); |
2
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754 #else |
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Eric Wing <ewing@anscamobile.com>
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41
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|
755 #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, 0, access_data) |
2
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756 #endif |
0 | 757 |
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|
758 /** |
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759 * 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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760 * Use at your own risk. |
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761 * Generally, you should use the LoadStream and LoadAll specializations of this function instead which call this. |
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|
762 * @param file_name The file to the audio resource you want to load. Extension should be "raw" for raw formats. |
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763 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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764 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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1
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765 * 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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766 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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767 * @param decode_mode_is_predecoded Specifies whether you want to completely preload the data or stream the data in chunks. |
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768 * @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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769 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
42
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|
770 * Buffer underrun conditions will also attempt to restart using this value. |
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|
771 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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772 * of buffers that will be queued. |
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773 * @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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|
774 * 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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|
775 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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|
776 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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|
777 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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778 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
0 | 779 */ |
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780 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, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
2
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781 |
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782 #ifdef DOXYGEN_ONLY |
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783 /** |
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|
784 * 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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785 * Use at your own risk. |
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|
786 * @param file_name The file to the audio resource you want to load.Extension should be "raw" for raw formats. |
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787 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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788 * @param buffer_size The size of a buffer to allocate for read chunks. This number should be in quantized with |
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|
789 * 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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790 * If the file is to be predecoded, optimizations may occur and this value might be ignored. |
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791 * @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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792 * @param num_startup_buffers For streamed data, specifies the number of buffers to fill before playback starts. |
42
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|
793 * Buffer underrun conditions will also attempt to restart using this value. |
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|
794 * @param suggested_number_of_buffers_to_queue_per_update_pass For each ALmixer_Update() pass, this is the targeted number |
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|
795 * of buffers that will be queued. |
2
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|
796 * @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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|
797 * 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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|
798 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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|
799 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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|
800 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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|
801 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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|
802 */ |
42
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41
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|
803 ALmixer_Data* ALmixer_LoadStream_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALuint buffer_size, ALuint max_queue_buffers, ALuint num_startup_buffers, ALuint suggested_number_of_buffers_to_queue_per_update_pass, ALuint access_data); |
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804 #else |
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805 #define ALmixer_LoadStream_RAW(file_name, desired_format, buffer_size, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) ALmixer_LoadSample_RAW(file_name, desired_format, buffer_size, AL_FALSE, max_queue_buffers, num_startup_buffers, suggested_number_of_buffers_to_queue_per_update_pass, access_data) |
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806 #endif |
0 | 807 |
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808 #ifdef DOXYGEN_ONLY |
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809 /** |
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810 * 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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811 * Use at your own risk. |
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812 * @param file_name The file to the audio resource you want to load. Extension should be "raw" for raw formats. |
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813 * @param desired_format The format you want audio decoded to. NULL will pick a default for you. |
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814 * @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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815 * 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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816 * audio first starts playing. With streamed data, you get the data in buffer_size chunks. Callbacks are not guarnanteed |
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817 * to be perfectly in-sync as this is a best-effort implementaiton. There are memory and performance considerations for |
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818 * using this feature, so if you don't need data callbacks, you should pass false to this function. |
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819 * @return Returns an ALmixer_Data* of the loaded sample or NULL if failed. |
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820 */ |
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821 ALmixer_Data* ALmixer_LoadAll_RAW(const char* file_name, ALmixer_AudioInfo* desired_format, ALuint access_data); |
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822 #else |
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823 #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, 0, access_data) |
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824 #endif |
0 | 825 |
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826 /** |
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827 * Frees an ALmixer_Data. |
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828 * Releases the memory associated with a ALmixer_Data. Use this when you are done playing the audio sample |
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829 * and wish to release the memory. |
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830 * @warning Do not try releasing data that is currently in use (e.g. playing, paused). |
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831 * @warning Make sure to free your data before calling ALmixer_Quit. Do not free data aftter ALmixer_Quit(). |
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832 * @param almixer_data The ALmixer_Data* you want to free. |
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833 */ |
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834 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_FreeData(ALmixer_Data* almixer_data); |
0 | 835 |
836 | |
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837 /** |
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838 * Returns true if the almixer_data was completely loaded into memory or false if it was loaded |
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839 * as a stream. |
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840 * @param almixer_data The audio resource you want to know about. |
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841 * @return AL_TRUE is predecoded, or AL_FALSE if streamed. |
0 | 842 */ |
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843 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_IsPredecoded(ALmixer_Data* almixer_data); |
0 | 844 |
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845 /** |
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846 * @} |
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847 */ |
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848 |
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849 /** |
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850 * @defgroup CallbackAPI Callbacks |
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851 * @{ |
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852 * Functions for callbacks |
0 | 853 */ |
854 | |
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855 /** |
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856 * Allows you to set a callback for when a sound has finished playing on a channel/source. |
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857 * @param playback_finished_callback The function you want to be invoked when a sound finishes. |
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858 * The callback function will pass you back the channel number which just finished playing, |
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859 * the OpenAL source id associated with the channel, the ALmixer_Data* that was played, |
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860 * a boolean telling you whether a sound finished playing because it ended normally or because |
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861 * something interrupted the playback (such as the user calling ALmixer_Halt*), and the |
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862 * user_data supplied as the second parameter to this function. |
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863 * @param which_chan The ALmixer channel that the data is currently playing on. |
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864 * @param al_source The OpenAL source that the data is currently playing on. |
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865 * @param almixer_data The ALmixer_Data that was played. |
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866 * @param finished_naturally AL_TRUE if the sound finished playing because it ended normally |
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867 * or AL_FALSE because something interrupted playback (such as the user calling ALmixer_Halt*). |
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868 * @param user_data This will be passed back to you in the callback. |
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869 * |
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870 * @warning You should not call other ALmixer functions in this callback. |
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871 * Particularly in the case of when compiled with threads, recursive locking |
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872 * will occur which will lead to deadlocks. Also be aware that particularly in the |
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873 * threaded case, the callbacks may (and currently do) occur on a background thread. |
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874 * One typical thread safe strategy is to set flags or schedule events to occur on the |
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875 * main thread. |
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876 * One possible exception to the no-calling ALmixer functions rule is ALmixer_Free. ALmixer_Free |
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877 * currently does not lock so it might okay to call this to free your data. However, this is not |
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878 * tested and not the expected pattern to be used. |
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879 */ |
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880 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 | 881 |
882 /** | |
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883 * Allows you to set a callback for getting audio data. |
1 | 884 * This is a callback function pointer that when set, will trigger a function |
885 * anytime there is new data loaded for a sample. The appropriate load | |
886 * parameter must be set in order for a sample to appear here. | |
887 * Keep in mind the the current backend implementation must do an end run | |
888 * around OpenAL because OpenAL lacks support for this kind of thing. | |
889 * As such, buffers are copied at decode time, and there is no attempt to do | |
890 * fine grained timing syncronization. You will be provided the entire buffer | |
891 * that is decoded regardless of length. So if you predecoded the entire | |
892 * audio file, the entire data buffer will be provided in a single callback. | |
893 * If you stream the data, you will be getting chunk sizes that are the same as | |
894 * what you specified the decode size to be. Unfortunely, this means if you | |
895 * pick smaller buffers, you get finer detail at the expense/risk of buffer | |
896 * underruns. If you decode more data, you have to deal with the syncronization | |
897 * issues if you want to display the data during playback in something like an | |
898 * oscilloscope. | |
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899 * |
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900 * @warning You should not call other ALmixer functions in this callback. |
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901 * Particularly in the case of when compiled with threads, recursive locking |
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902 * will occur which will lead to deadlocks. Also be aware that particularly in the |
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903 * threaded case, the callbacks may (and currently do) occur on a background thread. |
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904 * One typical thread safe strategy is to set flags or schedule events to occur on the |
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905 * main thread. |
1 | 906 * |
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907 * @param playback_data_callback The function you want called back. |
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908 * @param which_channel The ALmixer channel that the data is currently playing on. |
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909 * @param al_source The OpenAL source that the data is currently playing on. |
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910 * @param pcm_data This is a pointer to the data buffer containing ALmixer's |
1 | 911 * version of the decoded data. Consider this data as read-only. In the |
912 * non-threaded backend, this data will persist until potentially the next call | |
913 * to Update(). Currently, data buffers are preallocated and not destroyed | |
914 * until FreeData() is called (though this behavior is subject to change), | |
915 * but the contents will change when the buffer needs to be reused for a | |
916 * future callback. The buffer reuse is tied to the amount of buffers that | |
917 * may be queued. | |
918 * But assuming I don't change this, this may allow for some optimization | |
919 * so you can try referencing data from these buffers without worrying | |
920 * about crashing. (You still need to be aware that the data could be | |
921 * modified behind the scenes on an Update().) | |
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922 * The data type listed is an signed 8-bit format, but the real data may |
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923 * not actually be this. ALbyte was chosen as a convenience. If you have |
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924 * a 16 bit format, you will want to cast the data and divide the num_bytes by 2. |
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925 * Typically, data is either Sint16. This seems to be a |
1 | 926 * convention audio people seem to follow though I'm not sure what the |
927 * underlying reasons (if any) are for this. I suspect that there may be | |
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928 * some nice alignment/conversion property if you need to cast between ALbyte |
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929 * and ALubyte. |
1 | 930 * |
931 * @param num_bytes This is the total length of the data buffer. It presumes | |
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932 * that this length is measured for ALbyte. So if you have Sint16 data, you |
1 | 933 * should divide num_bytes by two if you access the data as Sint16. |
934 * | |
935 * @param frequency The frequency the data was decoded at. | |
936 * | |
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937 * @param num_channels_in_sample 1 for mono, 2 for stereo. Not to be confused with the ALmixer which_channel. |
1 | 938 * |
939 * @param bit_depth Bits per sample. This is expected to be 8 or 16. This | |
940 * number will tell you if you if you need to treat the data buffer as | |
941 * 16 bit or not. | |
942 * | |
943 * @param is_unsigned 1 if the data is unsigned, 0 if signed. Using this | |
944 * combined with bit_depth will tell you if you need to treat the data | |
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945 * as ALubyte, ALbyte, ALuint, or ALint. |
1 | 946 * |
947 * @param decode_mode_is_predecoded This is here to tell you if the data was totally | |
948 * predecoded or loaded as a stream. If predecoded, you will only get | |
949 * one data callback per playback instance. (This might also be true for | |
950 * looping the same sample...I don't remember how it was implemented. | |
951 * Maybe this should be fixed.) | |
952 * 0 (ALMIXER_DECODE_STREAM) for streamed. | |
953 * 1 (ALMIXER_DECODE_ALL) for predecoded. | |
954 * | |
955 * @param length_in_msec This returns the total length (time) of the data | |
956 * buffer in milliseconds. This could be computed yourself, but is provided | |
957 * as a convenince. | |
958 * | |
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959 * @param user_data The user data you pass in will be passed back to you in the callback. |
1 | 960 */ |
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961 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 | 962 |
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963 /** |
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964 * @} |
0 | 965 */ |
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966 |
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967 /** |
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968 * @defgroup PlayAPI Functions useful for playback. |
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969 * @{ |
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970 * These are core functions that are useful for controlling playback. |
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971 * Also see the Volume functions for additional playback functions and Query functions for additional information. |
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972 */ |
0 | 973 |
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974 /** |
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975 * Returns the total time in milliseconds of the audio resource. |
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976 * Returns the total time in milliseconds of the audio resource. |
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977 * If the total length cannot be determined, -1 will be returned. |
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978 * @param almixer_data The audio sample you want to know the total time of. |
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979 * @return The total time in milliseconds or -1 if some kind of failure. |
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980 */ |
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981 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_GetTotalTime(ALmixer_Data* almixer_data); |
0 | 982 |
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983 /** |
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984 * This function will look up the OpenAL source id for the corresponding channel number. |
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985 * @param which_channel The channel which you want to find the corresponding OpenAL source id for. |
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986 * If -1 was specified, an available source for playback will be returned. |
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987 * @return The OpenAL source id corresponding to the channel. 0 if you specified an illegal channel value. |
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988 * Or 0 if you specified -1 and no sources were currently available. |
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989 * @note ALmixer assumes your OpenAL implementation does not use 0 as a valid source ID. While the OpenAL spec |
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990 * does not disallow 0 for valid source ids, as of now, there are no known OpenAL implementations in use that |
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991 * use 0 as a valid source id (partly due to problems this has caused developers in the past). |
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992 */ |
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993 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_GetSource(ALint which_channel); |
0 | 994 |
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995 /** |
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996 * This function will look up the channel for the corresponding source. |
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997 * @param al_source The source id you want to find the corresponding channel number for. |
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998 * If 0 is supplied, it will try to return the first channel not in use. |
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999 * Returns -1 on error, or the channel. |
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1000 */ |
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1001 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_GetChannel(ALuint al_source); |
0 | 1002 |
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1003 /** |
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1004 * Will look for a channel available for playback. |
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1005 * Given a start channel number, the search will increase to the highest channel until it finds one available. |
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1006 * @param start_channel The channel number you want to start looking at. |
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1007 * @return A channel available or -1 if none could be found. |
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1008 */ |
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1009 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FindFreeChannel(ALint start_channel); |
1 | 1010 |
0 | 1011 |
1012 | |
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1013 /** |
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1014 * Play a sound on a channel with a time limit. |
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1015 * Plays a sound on a channel and will auto-stop after a specified number of milliseconds. |
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1016 * @param which_channel Allows you to specify the specific channel you want to play on. |
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1017 * Channels range from 0 to the (Number of allocated channels - 1). If you specify -1, |
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1018 * an available channel will be chosen automatically for you. |
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1019 * @note While paused, the auto-stop clock will also be paused. This makes it easy to always stop |
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1020 * a sample by a predesignated length without worrying about whether the user paused playback which would |
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1021 * throw off your calculations. |
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1022 * @param almixer_data The audio resource you want to play. |
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1023 * @param number_of_loops The number of times to loop (repeat) playing the data. |
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1024 * 0 means the data will play exactly once without repeat. -1 means infinitely loop. |
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1025 * @param number_of_milliseconds The number of milliseconds that should be played until the sample is auto-stopped. |
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1026 * -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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1027 * the sample will never stop playing). |
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1028 * @return Returns the channel that was selected for playback or -1 if no channels were available. |
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1029 */ |
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1030 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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1031 |
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1032 #ifdef DOXYGEN_ONLY |
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1033 /** |
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1034 * The same as ALmixer_PlayChannelTimed, but the sound is played without time limits. |
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1035 * @see ALmixer_PlayChannelTimed. |
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1036 */ |
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1037 ALint ALmixer_PlayChannelTimed(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops); |
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1038 #else |
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1039 #define ALmixer_PlayChannel(channel,data,loops) ALmixer_PlayChannelTimed(channel,data,loops,-1) |
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1040 #endif |
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1041 |
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1042 |
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1043 /** |
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1044 * Play a sound on an OpenAL source with a time limit. |
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1045 * Plays a sound on an OpenAL source and will auto-stop after a specified number of milliseconds. |
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1046 * @param al_source Allows you to specify the OpenAL source you want to play on. |
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1047 * If you specify 0, an available source will be chosen automatically for you. |
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1048 * @note Source values are not necessarily continguous and their values are implementation dependent. |
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1049 * Always use ALmixer functions to determine source values. |
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1050 * @note While paused, the auto-stop clock will also be paused. This makes it easy to always stop |
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1051 * a sample by a predesignated length without worrying about whether the user paused playback which would |
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1052 * throw off your calculations. |
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1053 * @param almixer_data The audio resource you want to play. |
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1054 * @param number_of_loops The number of times to loop (repeat) playing the data. |
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1055 * 0 means the data will play exactly once without repeat. -1 means infinitely loop. |
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1056 * @param number_of_milliseconds The number of milliseconds that should be played until the sample is auto-stopped. |
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1057 * -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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1058 * the sample will never stop playing). |
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1059 * @return Returns the OpenAL source that was selected for playback or 0 if no sources were available. |
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1060 */ |
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1061 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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1062 |
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1063 #ifdef DOXYGEN_ONLY |
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1064 /** |
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1065 * The same as ALmixer_PlaySourceTimed, but the sound is played without time limits. |
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1066 * @see ALmixer_PlaySourceTimed. |
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1067 */ |
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1068 ALint ALmixer_PlayChannelTimed(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops); |
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1069 #else |
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1070 #define ALmixer_PlaySource(al_source, almixer_data, number_of_loops) ALmixer_PlaySourceTimed(al_source, almixer_data, number_of_loops, -1) |
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1071 #endif |
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1072 |
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1073 /** |
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1074 * Stops playback on a channel. |
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1075 * Stops playback on a channel and clears the channel so it can be played on again. |
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1076 * @note Callbacks will still be invoked, but the finished_naturally flag will be set to AL_FALSE. |
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1077 * @param which_channel The channel to halt or -1 to halt all channels. |
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1078 * @return The actual number of channels halted on success or -1 on error. |
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1079 */ |
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1080 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_HaltChannel(ALint which_channel); |
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1081 |
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1082 /** |
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1083 * Stops playback on a channel. |
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1084 * Stops playback on a channel and clears the channel so it can be played on again. |
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1085 * @note Callbacks will still be invoked, but the finished_naturally flag will be set to AL_FALSE. |
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1086 * @param al_source The source to halt or 0 to halt all sources. |
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1087 * @return The actual number of sources halted on success or -1 on error. |
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1088 */ |
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1089 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_HaltSource(ALuint al_source); |
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1090 |
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1091 /** |
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1092 * Rewinds the sound to the beginning for a given data. |
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1093 * Rewinds the actual data, but the effect |
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1094 * may not be noticed until the currently buffered data is played. |
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1095 * @param almixer_data The data to rewind. |
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1096 * @returns true on success or false on error. |
2
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1097 */ |
12
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1098 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_RewindData(ALmixer_Data* almixer_data); |
2
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1099 |
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1100 /** |
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1101 * Rewinds the sound to the beginning that is playing on a specific channel. |
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1102 * If decoded all, rewind will instantly rewind it. Data is not |
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1103 * affected, so it will start at the "Seek"'ed positiond. |
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1104 * Streamed data will rewind the actual data, but the effect |
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1105 * may not be noticed until the currently buffered data is played. |
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1106 * @param which_channel The channel to rewind or -1 to rewind all channels. |
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1107 * @return The actual number of channels rewound on success or -1 on error. |
2
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1108 */ |
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1109 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindChannel(ALint which_channel); |
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1110 /** |
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1111 * Rewinds the sound to the beginning that is playing on a specific source. |
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1112 * If decoded all, rewind will instantly rewind it. Data is not |
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1113 * affected, so it will start at the "Seek"'ed positiond. |
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1114 * Streamed data will rewind the actual data, but the effect |
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1115 * may not be noticed until the currently buffered data is played. |
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1116 * @param al_source The source to rewind or 0 to rewind all sources. |
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1117 * @return The actual number of sources rewound on success or -1 on error. |
2
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1118 */ |
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1119 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_RewindSource(ALuint al_source); |
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1120 |
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1121 /** |
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1122 * Seek the sound for a given data. |
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1123 * Seeks the actual data to the given millisecond. It |
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1124 * may not be noticed until the currently buffered data is played. |
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1125 * @param almixer_data The data to seek on. |
2
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1126 * @param msec_pos The time position to seek to in the audio in milliseconds. |
12
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1127 * @returns true on success or false on error. |
2
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1128 */ |
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1129 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SeekData(ALmixer_Data* almixer_data, ALuint msec_pos); |
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1130 |
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1131 /** |
20
9365e714fc4b
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1132 * Seeks the sound to the beginning that is playing on a specific channel. |
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1133 * If decoded all, seek will instantly seek it. Data is not |
9365e714fc4b
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1134 * affected, so it will start at the "Seek"'ed positiond. |
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1135 * Streamed data will seek the actual data, but the effect |
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1136 * may not be noticed until the currently buffered data is played. |
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1137 * @param which_channel The channel to seek or -1 to seek all channels. |
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1138 * @return The actual number of channels rewound on success or -1 on error. |
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1139 */ |
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1140 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_SeekChannel(ALint which_channel, ALuint msec_pos); |
9365e714fc4b
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1141 /** |
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1142 * Seeks the sound to the beginning that is playing on a specific source. |
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1143 * If decoded all, seek will instantly seek it. Data is not |
9365e714fc4b
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1144 * affected, so it will start at the "Seek"'ed positiond. |
9365e714fc4b
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1145 * Streamed data will seek the actual data, but the effect |
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1146 * may not be noticed until the currently buffered data is played. |
9365e714fc4b
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1147 * @param al_source The source to seek or 0 to seek all sources. |
9365e714fc4b
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1148 * @return The actual number of sources rewound on success or -1 on error. |
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1149 */ |
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1150 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_SeekSource(ALuint al_source, ALuint msec_pos); |
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1151 |
9365e714fc4b
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1152 /** |
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1153 * Pauses playback on a channel. |
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1154 * Pauses playback on a channel. Should have no effect on channels that aren't playing. |
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1155 * @param which_channel The channel to pause or -1 to pause all channels. |
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1156 * @return The actual number of channels paused on success or -1 on error. |
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1157 */ |
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1158 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PauseChannel(ALint which_channel); |
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1159 /** |
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1160 * Pauses playback on a source. |
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1161 * Pauses playback on a source. Should have no effect on source that aren't playing. |
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1162 * @param al_source The source to pause or -1 to pause all source. |
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1163 * @return The actual number of source paused on success or -1 on error. |
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1164 */ |
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1165 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_PauseSource(ALuint al_source); |
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1166 |
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1167 /** |
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1168 * Resumes playback on a channel that is paused. |
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1169 * Resumes playback on a channel that is paused. Should have no effect on channels that aren't paused. |
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1170 * @param which_channel The channel to resume or -1 to resume all channels. |
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1171 * @return The actual number of channels resumed on success or -1 on error. |
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1172 */ |
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1173 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ResumeChannel(ALint which_channel); |
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1174 |
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1175 /** |
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1176 * Resumes playback on a source that is paused. |
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1177 * Resumes playback on a source that is paused. Should have no effect on sources that aren't paused. |
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1178 * @param al_source The source to resume or -1 to resume all sources. |
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1179 * @return The actual number of sources resumed on success or -1 on error. |
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1180 */ |
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1181 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ResumeSource(ALuint al_source); |
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1182 |
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1183 |
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1184 /** |
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|
1185 * Will cause a currently playing channel to stop playing in the specified number of milliseconds. |
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|
1186 * Will cause a currently playing channel to stop playing in the specified number of milliseconds. |
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1187 * This will override the value that was set when PlayChannelTimed or PlaySourceTimed was called |
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1188 * or override any previous calls to ExpireChannel or ExpireSource. |
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1189 * @param which_channel The channel to expire or -1 to apply to all channels. |
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1190 * @param number_of_milliseconds How many milliseconds from now until the expire triggers. |
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1191 * @return The actual number of channels this action is applied to on success or -1 on error. |
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1192 */ |
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1193 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ExpireChannel(ALint which_channel, ALint number_of_milliseconds); |
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1194 /** |
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1195 * Will cause a currently playing source to stop playing in the specified number of milliseconds. |
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1196 * Will cause a currently playing source to stop playing in the specified number of milliseconds. |
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1197 * This will override the value that was set when PlayChannelTimed or PlaySourceTimed was called |
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1198 * or override any previous calls to ExpireChannel or ExpireSource. |
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1199 * @param al_source The source to expire or 0 to apply to all sources. |
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1200 * @param number_of_milliseconds How many milliseconds from now until the expire triggers. |
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1201 * @return The actual number of sources this action is applied to on success or -1 on error. |
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1202 */ |
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1203 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_ExpireSource(ALuint al_source, ALint number_of_milliseconds); |
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1204 |
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1205 /** |
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1206 * @} |
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1207 */ |
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1208 |
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1209 /** |
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1210 * @defgroup VolumeAPI Volume and Fading |
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1211 * @{ |
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1212 * Fade and volume functions directly call OpenAL functions related to AL_GAIN. |
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1213 * These functions are provided mostly for those who just want to play audio but are not planning |
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1214 * to use OpenAL features directly. |
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1215 * 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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1216 * they may fight/override values you directly set yourself. |
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1217 */ |
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1218 |
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1219 /** |
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1220 * 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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1221 * @see ALmixer_PlayChannelTimed. |
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1222 */ |
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1223 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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1224 |
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1225 #ifdef DOXYGEN_ONLY |
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1226 /** |
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1227 * The same as ALmixer_FadeInChannelTimed, but the sound is played without time limits. |
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1228 * @see ALmixer_FadeInChannelTimed, ALmixer_PlayChannel. |
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1229 */ |
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1230 ALint ALmixer_FadeInChannel(ALint which_channel, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks); |
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1231 #else |
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1232 #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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1233 #endif |
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1234 |
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1235 /** |
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1236 * 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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1237 * @see ALmixer_PlaySourceTimed. |
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1238 */ |
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1239 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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1240 |
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1241 #ifdef DOXYGEN_ONLY |
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1242 /** |
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1243 * The same as ALmixer_FadeInSourceTimed, but the sound is played without time limits. |
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1244 * @see ALmixer_FadeInSourceTimed, ALmixer_PlaySource. |
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1245 */ |
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1246 extern ALuint ALmixer_FadeInSource(ALuint al_source, ALmixer_Data* almixer_data, ALint number_of_loops, ALuint fade_ticks); |
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1247 #else |
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1248 #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 | 1249 #endif |
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1250 |
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1251 /** |
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1252 * Fade out a current playing channel. |
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1253 * Will fade out a currently playing channel over the specified period of time starting from now. |
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1254 * The volume will be changed from the current AL_GAIN level to the AL_MIN_GAIN. |
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1255 * The volume fade will interpolate over the specified period of time. |
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1256 * The playback will halt at the end of the time period. |
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1257 * @param which_channel The channel to fade or -1 to fade all playing channels. |
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1258 * @param fade_ticks In milliseconds, the amount of time the fade out should take to complete. |
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1259 * @return Returns -1 on error or the number of channels faded. |
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1260 */ |
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1261 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeOutChannel(ALint which_channel, ALuint fade_ticks); |
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1262 |
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1263 /** |
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1264 * Fade out a current playing source. |
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1265 * Will fade out a currently playing source over the specified period of time starting from now. |
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1266 * The volume will be changed from the current AL_GAIN level to the AL_MIN_GAIN. |
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1267 * The volume fade will interpolate over the specified period of time. |
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1268 * The playback will halt at the end of the time period. |
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1269 * @param al_source The source to fade or -1 to fade all playing sources. |
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1270 * @param fade_ticks In milliseconds, the amount of time the fade out should take to complete. |
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1271 * @return Returns -1 on error or the number of sources faded. |
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1272 */ |
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1273 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeOutSource(ALuint al_source, ALuint fade_ticks); |
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1274 |
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1275 /** |
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1276 * Gradually changes the volume from the current AL_GAIN to the specified volume. |
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1277 * Gradually changes the volume from the current AL_GAIN to the specified volume over the specified period of time. |
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1278 * This is some times referred to as volume ducking. |
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1279 * Note that this function works for setting the volume higher as well as lower. |
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1280 * @param which_channel The channel to fade or -1 to fade all playing channels. |
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1281 * @param fade_ticks In milliseconds, the amount of time the volume change should take to complete. |
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1282 * @param volume The volume to change to. Valid values are 0.0 to 1.0. |
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1283 * @return Returns -1 on error or the number of channels faded. |
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1284 */ |
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1285 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeChannel(ALint which_channel, ALuint fade_ticks, ALfloat volume); |
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1286 |
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1287 /** |
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1288 * Gradually changes the volume from the current AL_GAIN to the specified volume. |
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1289 * Gradually changes the volume from the current AL_GAIN to the specified volume over the specified period of time. |
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1290 * This is some times referred to as volume ducking. |
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1291 * Note that this function works for setting the volume higher as well as lower. |
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1292 * @param al_source The source to fade or -1 to fade all playing sources. |
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1293 * @param fade_ticks In milliseconds, the amount of time the volume change should take to complete. |
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1294 * @param volume The volume to change to. Valid values are 0.0 to 1.0. |
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1295 * @return Returns -1 on error or the number of sources faded. |
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1296 */ |
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1297 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_FadeSource(ALuint al_source, ALuint fade_ticks, ALfloat volume); |
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1298 |
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1299 /** |
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1300 * Sets the volume via the AL_GAIN source property. |
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1301 * Sets the volume for a given channel via the AL_GAIN source property. |
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1302 * @param which_channel The channel to set the volume to or -1 to set on all channels. |
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1303 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1304 * @return AL_TRUE on success or AL_FALSE on error. |
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1305 */ |
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1306 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetVolumeChannel(ALint which_channel, ALfloat volume); |
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|
1307 |
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1308 /** |
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1309 * Sets the volume via the AL_GAIN source property. |
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1310 * Sets the volume for a given source via the AL_GAIN source property. |
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1311 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1312 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1313 * @return AL_TRUE on success or AL_FALSE on error. |
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1314 */ |
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1315 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetVolumeSource(ALuint al_source, ALfloat volume); |
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|
1316 |
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1317 /** |
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1318 * Gets the volume via the AL_GAIN source property. |
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1319 * Gets the volume for a given channel via the AL_GAIN source property. |
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1320 * @param which_channel The channel to get the volume from. |
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1321 * -1 will return the average volume set across all channels. |
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1322 * @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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1323 */ |
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|
1324 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetVolumeChannel(ALint which_channel); |
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|
1325 |
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1326 /** |
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1327 * Gets the volume via the AL_GAIN source property. |
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1328 * Gets the volume for a given source via the AL_GAIN source property. |
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1329 * @param al_source The source to get the volume from. |
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1330 * -1 will return the average volume set across all source. |
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1331 * @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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1332 */ |
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1333 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetVolumeSource(ALuint al_source); |
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|
1334 |
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|
1335 /** |
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|
1336 * Sets the maximum volume via the AL_MAX_GAIN source property. |
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1337 * Sets the maximum volume for a given channel via the AL_MAX_GAIN source property. |
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1338 * Max volumes will be clamped to this value. |
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1339 * @param which_channel The channel to set the volume to or -1 to set on all channels. |
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1340 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1341 * @return AL_TRUE on success or AL_FALSE on error. |
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1342 */ |
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1343 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMaxVolumeChannel(ALint which_channel, ALfloat volume); |
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1344 |
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1345 /** |
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1346 * Sets the maximum volume via the AL_MAX_GAIN source property. |
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1347 * Sets the maximum volume for a given source via the AL_MAX_GAIN source property. |
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1348 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1349 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1350 * @return AL_TRUE on success or AL_FALSE on error. |
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1351 */ |
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1352 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMaxVolumeSource(ALuint al_source, ALfloat volume); |
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1353 |
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1354 /** |
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1355 * Gets the max volume via the AL_MAX_GAIN source property. |
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1356 * Gets the max volume for a given channel via the AL_MAX_GAIN source property. |
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1357 * @param which_channel The channel to get the volume from. |
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1358 * -1 will return the average volume set across all channels. |
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1359 * @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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1360 */ |
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1361 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMaxVolumeChannel(ALint which_channel); |
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1362 |
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1363 /** |
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1364 * Gets the maximum volume via the AL_MAX_GAIN source property. |
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1365 * Gets the maximum volume for a given source via the AL_MAX_GAIN source property. |
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1366 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1367 * 0 will return the average volume set across all channels. |
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1368 * @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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1369 */ |
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1370 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMaxVolumeSource(ALuint al_source); |
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1371 |
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1372 /** |
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1373 * Sets the minimum volume via the AL_MIN_GAIN source property. |
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1374 * Sets the minimum volume for a given channel via the AL_MIN_GAIN source property. |
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1375 * Min volumes will be clamped to this value. |
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1376 * @param which_channel The channel to set the volume to or -1 to set on all channels. |
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1377 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1378 * @return AL_TRUE on success or AL_FALSE on error. |
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1379 */ |
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1380 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMinVolumeChannel(ALint which_channel, ALfloat volume); |
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1381 |
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1382 /** |
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1383 * Sets the minimum volume via the AL_MIN_GAIN source property. |
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1384 * Sets the minimum volume for a given source via the AL_MIN_GAIN source property. |
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1385 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1386 * @param volume The new volume to use. Valid values are 0.0 to 1.0. |
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1387 * @return AL_TRUE on success or AL_FALSE on error. |
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1388 */ |
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1389 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMinVolumeSource(ALuint al_source, ALfloat volume); |
0 | 1390 |
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1391 /** |
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1392 * Gets the min volume via the AL_MIN_GAIN source property. |
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1393 * Gets the min volume for a given channel via the AL_MIN_GAIN source property. |
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1394 * @param which_channel The channel to get the volume from. |
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1395 * -1 will return the average volume set across all channels. |
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1396 * @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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1397 */ |
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1398 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMinVolumeChannel(ALint which_channel); |
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1399 |
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1400 /** |
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1401 * Gets the min volume via the AL_MIN_GAIN source property. |
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1402 * Gets the min volume for a given source via the AL_MIN_GAIN source property. |
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1403 * @param al_source The source to set the volume to or 0 to set on all sources. |
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1404 * 0 will return the average volume set across all channels. |
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1405 * @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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1406 */ |
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1407 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMinVolumeSource(ALuint al_source); |
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1408 |
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1409 /** |
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1410 * Sets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1411 * Sets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1412 * @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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1413 * @return AL_TRUE on success or AL_FALSE on an error. |
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1414 */ |
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1415 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_SetMasterVolume(ALfloat new_volume); |
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1416 |
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1417 /** |
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1418 * Gets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1419 * Gets the OpenAL listener AL_GAIN which can be thought of as the "master volume". |
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1420 * @return The current volume level on the listener. -1.0 will be returned on an error. |
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1421 */ |
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1422 extern ALMIXER_DECLSPEC ALfloat ALMIXER_CALL ALmixer_GetMasterVolume(void); |
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1423 |
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1424 /** |
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1425 * @} |
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1426 */ |
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1427 |
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1428 /** |
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1429 * @defgroup QueryAPI Query APIs |
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1430 * @{ |
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1431 * Functions to query ALmixer. |
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1432 */ |
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1433 |
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1434 |
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1435 /** |
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1436 * Returns true if the specified channel is currently playing or paused, |
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1437 * or if -1 is passed the number of channels that are currently playing or paused. |
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1438 * @param which_channel The channel you want to know about or -1 to get the count of |
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1439 * currently playing/paused channels. |
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1440 * @return For a specific channel, 1 if the channel is playing or paused, 0 if not. |
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1441 * Or returns the count of currently playing/paused channels. |
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1442 * Or -1 on an error. |
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1443 */ |
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1444 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsActiveChannel(ALint which_channel); |
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1445 |
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1446 /** |
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1447 * Returns true if the specified source is currently playing or paused, |
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1448 * or if -1 is passed the number of sources that are currently playing or paused. |
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1449 * @param al_source The channel you want to know about or -1 to get the count of |
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1450 * currently playing/paused sources. |
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1451 * @return For a specific sources, 1 if the channel is playing or paused, 0 if not. |
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1452 * Or returns the count of currently playing/paused sources. |
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1453 * Or -1 on an error. |
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1454 */ |
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1455 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsActiveSource(ALuint al_source); |
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1456 |
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1457 /** |
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1458 * Returns true if the specified channel is currently playing. |
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1459 * or if -1 is passed the number of channels that are currently playing. |
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1460 * @param which_channel The channel you want to know about or -1 to get the count of |
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1461 * currently playing channels. |
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1462 * @return For a specific channel, 1 if the channel is playing, 0 if not. |
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1463 * Or returns the count of currently playing channels. |
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1464 * Or -1 on an error. |
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1465 */ |
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1466 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPlayingChannel(ALint which_channel); |
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1467 |
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1468 /** |
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1469 * Returns true if the specified sources is currently playing. |
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1470 * or if -1 is passed the number of sources that are currently playing. |
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1471 * @param al_source The sources you want to know about or -1 to get the count of |
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1472 * currently playing sources. |
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1473 * @return For a specific source, 1 if the source is playing, 0 if not. |
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1474 * Or returns the count of currently playing sources. |
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1475 * Or -1 on an error. |
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1476 */ |
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1477 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPlayingSource(ALuint al_source); |
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1478 |
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1479 /** |
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1480 * Returns true if the specified channel is currently paused. |
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1481 * or if -1 is passed the number of channels that are currently paused. |
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1482 * @param which_channel The channel you want to know about or -1 to get the count of |
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1483 * currently paused channels. |
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1484 * @return For a specific channel, 1 if the channel is paused, 0 if not. |
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1485 * Or returns the count of currently paused channels. |
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1486 * Or -1 on an error. |
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1487 */ |
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1488 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPausedChannel(ALint which_channel); |
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1489 |
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1490 /** |
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1491 * Returns true if the specified sources is currently paused. |
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1492 * or if -1 is passed the number of sources that are currently paused. |
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1493 * @param al_source The source you want to know about or -1 to get the count of |
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1494 * currently paused sources. |
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1495 * @return For a specific source, 1 if the source is paused, 0 if not. |
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1496 * Or returns the count of currently paused sources. |
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1497 * Or -1 on an error. |
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1498 */ |
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1499 extern ALMIXER_DECLSPEC ALint ALMIXER_CALL ALmixer_IsPausedSource(ALuint al_source); |
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1500 |
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1501 /** |
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1502 * Returns the number of channels that are currently available for playback (not playing, not paused). |
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1503 * @return The number of channels that are currently free. |
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1504 */ |
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1505 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountAllFreeChannels(void); |
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1506 |
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1507 /** |
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1508 * Returns the number of channels that are currently available for playback (not playing, not paused), |
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1509 * excluding the channels that have been reserved. |
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1510 * @return The number of channels that are currently in free, excluding the channels that have been reserved. |
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1511 * @see ALmixer_ReserveChannels |
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1512 */ |
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1513 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountUnreservedFreeChannels(void); |
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1514 |
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1515 /** |
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1516 * Returns the number of channels that are currently in use (playing/paused). |
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1517 * @return The number of channels that are currently in use. |
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1518 * @see ALmixer_ReserveChannels |
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1519 */ |
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1520 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountAllUsedChannels(void); |
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1521 |
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1522 /** |
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1523 * Returns the number of channels that are currently in use (playing/paused), |
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1524 * excluding the channels that have been reserved. |
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1525 * @return The number of channels that are currently in use excluding the channels that have been reserved. |
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1526 * @see ALmixer_ReserveChannels |
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1527 */ |
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1528 extern ALMIXER_DECLSPEC ALuint ALMIXER_CALL ALmixer_CountUnreservedUsedChannels(void); |
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1529 |
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1530 |
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1531 #ifdef DOXYGEN_ONLY |
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1532 /** |
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1533 * Returns the number of allocated channels. |
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1534 * This is just a convenience alias to ALmixer_AllocateChannels(-1). |
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1535 * @see ALmixer_AllocateChannels |
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1536 */ |
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1537 ALuint ALmixer_CountTotalChannels(void); |
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1538 #else |
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1539 #define ALmixer_CountTotalChannels() ALmixer_AllocateChannels(-1) |
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1540 #endif |
0 | 1541 |
1542 | |
1543 | |
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1544 |
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1545 #ifdef DOXYGEN_ONLY |
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1546 /** |
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1547 * Returns the number of reserved channels. |
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1548 * This is just a convenience alias to ALmixer_ReserveChannels(-1). |
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1549 * @see ALmixer_ReserveChannels |
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1550 */ |
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1551 ALuint ALmixer_CountReservedChannels(void); |
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1552 #else |
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1553 #define ALmixer_CountReservedChannels() ALmixer_ReserveChannels(-1) |
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1554 #endif |
0 | 1555 |
1556 | |
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1557 /** |
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1558 * @} |
0 | 1559 */ |
1560 | |
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1561 /** |
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1562 * @defgroup DebugAPI Debug APIs |
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1563 * @{ |
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1564 * Functions for debugging purposes. These may be removed in future versions. |
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1565 */ |
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1566 |
0 | 1567 |
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1568 /* For testing */ |
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1569 #if 0 |
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1570 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputAttributes(void); |
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1571 #endif |
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1572 /** This function may be removed in the future. For debugging. Prints to stderr. Lists the decoders available. */ |
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1573 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputDecoders(void); |
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1574 /** This function may be removed in the future. For debugging. Prints to stderr. */ |
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1575 extern ALMIXER_DECLSPEC void ALMIXER_CALL ALmixer_OutputOpenALInfo(void); |
0 | 1576 |
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1577 /** This function may be removed in the future. Returns true if compiled with threads, false if not. */ |
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1578 extern ALMIXER_DECLSPEC ALboolean ALMIXER_CALL ALmixer_CompiledWithThreadBackend(void); |
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1579 |
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1580 /** |
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1581 * @} |
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1582 */ |
0 | 1583 |
1584 | |
1585 | |
1586 | |
1587 /* Ends C function definitions when using C++ */ | |
1588 #ifdef __cplusplus | |
1589 } | |
1590 #endif | |
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1591 |
0 | 1592 |
1593 #endif /* _SDL_ALMIXER_H_ */ | |
1594 | |
1595 /* end of SDL_ALmixer.h ... */ | |
1596 | |
1597 |