annotate include/SDL_audio.h @ 4277:803eeeeaa8a0 SDL-1.2

Updated libtool to version 2.2.6a
author Sam Lantinga <slouken@libsdl.org>
date Sun, 04 Oct 2009 22:31:25 +0000
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1 /*
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2 SDL - Simple DirectMedia Layer
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3 Copyright (C) 1997-2009 Sam Lantinga
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4
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5 This library is free software; you can redistribute it and/or
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6 modify it under the terms of the GNU Lesser General Public
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7 License as published by the Free Software Foundation; either
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8 version 2.1 of the License, or (at your option) any later version.
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9
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10 This library is distributed in the hope that it will be useful,
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11 but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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13 Lesser General Public License for more details.
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14
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15 You should have received a copy of the GNU Lesser General Public
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16 License along with this library; if not, write to the Free Software
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17 Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
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19 Sam Lantinga
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20 slouken@libsdl.org
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21 */
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22
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23 /**
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24 * @file SDL_audio.h
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25 * Access to the raw audio mixing buffer for the SDL library
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26 */
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27
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28 #ifndef _SDL_audio_h
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29 #define _SDL_audio_h
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30
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31 #include "SDL_stdinc.h"
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32 #include "SDL_error.h"
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33 #include "SDL_endian.h"
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34 #include "SDL_mutex.h"
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35 #include "SDL_thread.h"
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36 #include "SDL_rwops.h"
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37
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38 #include "begin_code.h"
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39 /* Set up for C function definitions, even when using C++ */
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40 #ifdef __cplusplus
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41 extern "C" {
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42 #endif
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43
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44 /**
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45 * When filling in the desired audio spec structure,
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46 * - 'desired->freq' should be the desired audio frequency in samples-per-second.
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47 * - 'desired->format' should be the desired audio format.
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48 * - 'desired->samples' is the desired size of the audio buffer, in samples.
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49 * This number should be a power of two, and may be adjusted by the audio
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50 * driver to a value more suitable for the hardware. Good values seem to
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51 * range between 512 and 8096 inclusive, depending on the application and
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52 * CPU speed. Smaller values yield faster response time, but can lead
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53 * to underflow if the application is doing heavy processing and cannot
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54 * fill the audio buffer in time. A stereo sample consists of both right
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55 * and left channels in LR ordering.
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56 * Note that the number of samples is directly related to time by the
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57 * following formula: ms = (samples*1000)/freq
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58 * - 'desired->size' is the size in bytes of the audio buffer, and is
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59 * calculated by SDL_OpenAudio().
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60 * - 'desired->silence' is the value used to set the buffer to silence,
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61 * and is calculated by SDL_OpenAudio().
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62 * - 'desired->callback' should be set to a function that will be called
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63 * when the audio device is ready for more data. It is passed a pointer
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64 * to the audio buffer, and the length in bytes of the audio buffer.
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65 * This function usually runs in a separate thread, and so you should
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66 * protect data structures that it accesses by calling SDL_LockAudio()
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67 * and SDL_UnlockAudio() in your code.
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68 * - 'desired->userdata' is passed as the first parameter to your callback
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69 * function.
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70 *
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71 * @note The calculated values in this structure are calculated by SDL_OpenAudio()
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72 *
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73 */
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74 typedef struct SDL_AudioSpec {
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75 int freq; /**< DSP frequency -- samples per second */
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76 Uint16 format; /**< Audio data format */
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77 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
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78 Uint8 silence; /**< Audio buffer silence value (calculated) */
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79 Uint16 samples; /**< Audio buffer size in samples (power of 2) */
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80 Uint16 padding; /**< Necessary for some compile environments */
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81 Uint32 size; /**< Audio buffer size in bytes (calculated) */
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82 /**
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83 * This function is called when the audio device needs more data.
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84 *
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85 * @param[out] stream A pointer to the audio data buffer
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86 * @param[in] len The length of the audio buffer in bytes.
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87 *
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88 * Once the callback returns, the buffer will no longer be valid.
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89 * Stereo samples are stored in a LRLRLR ordering.
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90 */
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91 void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
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92 void *userdata;
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93 } SDL_AudioSpec;
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94
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95 /**
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96 * @name Audio format flags
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97 * defaults to LSB byte order
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98 */
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99 /*@{*/
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100 #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
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101 #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
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102 #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
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103 #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
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104 #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
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105 #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
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106 #define AUDIO_U16 AUDIO_U16LSB
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107 #define AUDIO_S16 AUDIO_S16LSB
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108
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109 /**
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110 * @name Native audio byte ordering
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111 */
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112 /*@{*/
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113 #if SDL_BYTEORDER == SDL_LIL_ENDIAN
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114 #define AUDIO_U16SYS AUDIO_U16LSB
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115 #define AUDIO_S16SYS AUDIO_S16LSB
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116 #else
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117 #define AUDIO_U16SYS AUDIO_U16MSB
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118 #define AUDIO_S16SYS AUDIO_S16MSB
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119 #endif
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120 /*@}*/
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121
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122 /*@}*/
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123
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124
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125 /** A structure to hold a set of audio conversion filters and buffers */
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126 typedef struct SDL_AudioCVT {
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127 int needed; /**< Set to 1 if conversion possible */
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128 Uint16 src_format; /**< Source audio format */
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129 Uint16 dst_format; /**< Target audio format */
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130 double rate_incr; /**< Rate conversion increment */
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131 Uint8 *buf; /**< Buffer to hold entire audio data */
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132 int len; /**< Length of original audio buffer */
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133 int len_cvt; /**< Length of converted audio buffer */
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134 int len_mult; /**< buffer must be len*len_mult big */
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135 double len_ratio; /**< Given len, final size is len*len_ratio */
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136 void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
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137 int filter_index; /**< Current audio conversion function */
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138 } SDL_AudioCVT;
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139
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140
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141 /* Function prototypes */
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142
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143 /**
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144 * @name Audio Init and Quit
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145 * These functions are used internally, and should not be used unless you
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146 * have a specific need to specify the audio driver you want to use.
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147 * You should normally use SDL_Init() or SDL_InitSubSystem().
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148 */
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149 /*@{*/
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150 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
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151 extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
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152 /*@}*/
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153
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154 /**
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155 * This function fills the given character buffer with the name of the
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156 * current audio driver, and returns a pointer to it if the audio driver has
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157 * been initialized. It returns NULL if no driver has been initialized.
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158 */
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159 extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
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160
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161 /**
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162 * This function opens the audio device with the desired parameters, and
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163 * returns 0 if successful, placing the actual hardware parameters in the
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164 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio
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165 * data passed to the callback function will be guaranteed to be in the
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166 * requested format, and will be automatically converted to the hardware
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167 * audio format if necessary. This function returns -1 if it failed
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168 * to open the audio device, or couldn't set up the audio thread.
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169 *
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170 * The audio device starts out playing silence when it's opened, and should
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171 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready
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172 * for your audio callback function to be called. Since the audio driver
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173 * may modify the requested size of the audio buffer, you should allocate
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174 * any local mixing buffers after you open the audio device.
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175 *
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176 * @sa SDL_AudioSpec
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177 */
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178 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
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179
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180 typedef enum {
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181 SDL_AUDIO_STOPPED = 0,
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182 SDL_AUDIO_PLAYING,
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183 SDL_AUDIO_PAUSED
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184 } SDL_audiostatus;
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185
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186 /** Get the current audio state */
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187 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
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188
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189 /**
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190 * This function pauses and unpauses the audio callback processing.
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191 * It should be called with a parameter of 0 after opening the audio
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192 * device to start playing sound. This is so you can safely initialize
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193 * data for your callback function after opening the audio device.
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194 * Silence will be written to the audio device during the pause.
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195 */
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196 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
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197
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198 /**
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199 * This function loads a WAVE from the data source, automatically freeing
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200 * that source if 'freesrc' is non-zero. For example, to load a WAVE file,
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201 * you could do:
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202 * @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
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203 *
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204 * If this function succeeds, it returns the given SDL_AudioSpec,
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205 * filled with the audio data format of the wave data, and sets
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206 * 'audio_buf' to a malloc()'d buffer containing the audio data,
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207 * and sets 'audio_len' to the length of that audio buffer, in bytes.
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208 * You need to free the audio buffer with SDL_FreeWAV() when you are
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209 * done with it.
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210 *
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211 * This function returns NULL and sets the SDL error message if the
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212 * wave file cannot be opened, uses an unknown data format, or is
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213 * corrupt. Currently raw and MS-ADPCM WAVE files are supported.
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214 */
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215 extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
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216
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217 /** Compatibility convenience function -- loads a WAV from a file */
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218 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
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219 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
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220
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221 /**
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222 * This function frees data previously allocated with SDL_LoadWAV_RW()
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223 */
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224 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
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225
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226 /**
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227 * This function takes a source format and rate and a destination format
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228 * and rate, and initializes the 'cvt' structure with information needed
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229 * by SDL_ConvertAudio() to convert a buffer of audio data from one format
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230 * to the other.
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231 *
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232 * @return This function returns 0, or -1 if there was an error.
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233 */
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234 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
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235 Uint16 src_format, Uint8 src_channels, int src_rate,
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236 Uint16 dst_format, Uint8 dst_channels, int dst_rate);
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237
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238 /**
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239 * Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(),
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240 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of
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241 * audio data in the source format, this function will convert it in-place
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242 * to the desired format.
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243 * The data conversion may expand the size of the audio data, so the buffer
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244 * cvt->buf should be allocated after the cvt structure is initialized by
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245 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.
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246 */
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247 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
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248
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249
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250 #define SDL_MIX_MAXVOLUME 128
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251 /**
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252 * This takes two audio buffers of the playing audio format and mixes
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253 * them, performing addition, volume adjustment, and overflow clipping.
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254 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
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255 * for full audio volume. Note this does not change hardware volume.
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256 * This is provided for convenience -- you can mix your own audio data.
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257 */
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258 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
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259
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260 /**
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261 * @name Audio Locks
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262 * The lock manipulated by these functions protects the callback function.
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263 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the
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264 * callback function is not running. Do not call these from the callback
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265 * function or you will cause deadlock.
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266 */
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267 /*@{*/
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268 extern DECLSPEC void SDLCALL SDL_LockAudio(void);
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269 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
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270 /*@}*/
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271
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272 /**
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273 * This function shuts down audio processing and closes the audio device.
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274 */
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275 extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
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276
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277
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278 /* Ends C function definitions when using C++ */
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279 #ifdef __cplusplus
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280 }
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281 #endif
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282 #include "close_code.h"
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283
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284 #endif /* _SDL_audio_h */