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1 /* |
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2 SDL - Simple DirectMedia Layer |
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3 Copyright (C) 1997-2006 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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18 |
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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 /* Access to the raw audio mixing buffer for the SDL library */ |
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24 |
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25 #ifndef _SDL_audio_h |
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26 #define _SDL_audio_h |
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27 |
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28 #include "SDL_stdinc.h" |
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29 #include "SDL_error.h" |
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30 #include "SDL_endian.h" |
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31 #include "SDL_mutex.h" |
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32 #include "SDL_thread.h" |
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33 #include "SDL_rwops.h" |
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34 |
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35 #include "begin_code.h" |
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36 /* Set up for C function definitions, even when using C++ */ |
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37 #ifdef __cplusplus |
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38 extern "C" { |
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39 #endif |
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40 |
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41 /* The calculated values in this structure are calculated by SDL_OpenAudio() */ |
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42 typedef struct SDL_AudioSpec { |
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43 int freq; /* DSP frequency -- samples per second */ |
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44 Uint16 format; /* Audio data format */ |
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45 Uint8 channels; /* Number of channels: 1 mono, 2 stereo */ |
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46 Uint8 silence; /* Audio buffer silence value (calculated) */ |
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47 Uint16 samples; /* Audio buffer size in samples (power of 2) */ |
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48 Uint16 padding; /* Necessary for some compile environments */ |
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49 Uint32 size; /* Audio buffer size in bytes (calculated) */ |
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50 /* This function is called when the audio device needs more data. |
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51 'stream' is a pointer to the audio data buffer |
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52 'len' is the length of that buffer in bytes. |
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53 Once the callback returns, the buffer will no longer be valid. |
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54 Stereo samples are stored in a LRLRLR ordering. |
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55 */ |
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56 void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len); |
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57 void *userdata; |
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58 } SDL_AudioSpec; |
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59 |
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60 /* Audio format flags (defaults to LSB byte order) */ |
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61 #define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */ |
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62 #define AUDIO_S8 0x8008 /* Signed 8-bit samples */ |
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63 #define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */ |
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64 #define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */ |
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65 #define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */ |
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66 #define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */ |
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67 #define AUDIO_U16 AUDIO_U16LSB |
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68 #define AUDIO_S16 AUDIO_S16LSB |
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69 |
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70 /* Native audio byte ordering */ |
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71 #if SDL_BYTEORDER == SDL_LIL_ENDIAN |
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72 #define AUDIO_U16SYS AUDIO_U16LSB |
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73 #define AUDIO_S16SYS AUDIO_S16LSB |
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74 #else |
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75 #define AUDIO_U16SYS AUDIO_U16MSB |
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76 #define AUDIO_S16SYS AUDIO_S16MSB |
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77 #endif |
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78 |
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79 |
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80 /* A structure to hold a set of audio conversion filters and buffers */ |
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81 typedef struct SDL_AudioCVT { |
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82 int needed; /* Set to 1 if conversion possible */ |
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83 Uint16 src_format; /* Source audio format */ |
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84 Uint16 dst_format; /* Target audio format */ |
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85 double rate_incr; /* Rate conversion increment */ |
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86 Uint8 *buf; /* Buffer to hold entire audio data */ |
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87 int len; /* Length of original audio buffer */ |
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88 int len_cvt; /* Length of converted audio buffer */ |
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89 int len_mult; /* buffer must be len*len_mult big */ |
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90 double len_ratio; /* Given len, final size is len*len_ratio */ |
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91 void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format); |
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92 int filter_index; /* Current audio conversion function */ |
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93 } SDL_AudioCVT; |
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94 |
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95 |
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96 /* Function prototypes */ |
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97 |
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98 /* These functions are used internally, and should not be used unless you |
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99 * have a specific need to specify the audio driver you want to use. |
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100 * You should normally use SDL_Init() or SDL_InitSubSystem(). |
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101 */ |
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102 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
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103 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
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104 |
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105 /* This function fills the given character buffer with the name of the |
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106 * current audio driver, and returns a pointer to it if the audio driver has |
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107 * been initialized. It returns NULL if no driver has been initialized. |
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108 */ |
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109 extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen); |
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110 |
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111 /* |
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112 * This function opens the audio device with the desired parameters, and |
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113 * returns 0 if successful, placing the actual hardware parameters in the |
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114 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio |
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115 * data passed to the callback function will be guaranteed to be in the |
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116 * requested format, and will be automatically converted to the hardware |
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117 * audio format if necessary. This function returns -1 if it failed |
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118 * to open the audio device, or couldn't set up the audio thread. |
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119 * |
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120 * When filling in the desired audio spec structure, |
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121 * 'desired->freq' should be the desired audio frequency in samples-per-second. |
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122 * 'desired->format' should be the desired audio format. |
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123 * 'desired->samples' is the desired size of the audio buffer, in samples. |
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124 * This number should be a power of two, and may be adjusted by the audio |
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125 * driver to a value more suitable for the hardware. Good values seem to |
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126 * range between 512 and 8096 inclusive, depending on the application and |
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127 * CPU speed. Smaller values yield faster response time, but can lead |
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128 * to underflow if the application is doing heavy processing and cannot |
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129 * fill the audio buffer in time. A stereo sample consists of both right |
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130 * and left channels in LR ordering. |
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131 * Note that the number of samples is directly related to time by the |
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132 * following formula: ms = (samples*1000)/freq |
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133 * 'desired->size' is the size in bytes of the audio buffer, and is |
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134 * calculated by SDL_OpenAudio(). |
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135 * 'desired->silence' is the value used to set the buffer to silence, |
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136 * and is calculated by SDL_OpenAudio(). |
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137 * 'desired->callback' should be set to a function that will be called |
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138 * when the audio device is ready for more data. It is passed a pointer |
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139 * to the audio buffer, and the length in bytes of the audio buffer. |
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140 * This function usually runs in a separate thread, and so you should |
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141 * protect data structures that it accesses by calling SDL_LockAudio() |
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142 * and SDL_UnlockAudio() in your code. |
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143 * 'desired->userdata' is passed as the first parameter to your callback |
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144 * function. |
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145 * |
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146 * The audio device starts out playing silence when it's opened, and should |
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147 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready |
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148 * for your audio callback function to be called. Since the audio driver |
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149 * may modify the requested size of the audio buffer, you should allocate |
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150 * any local mixing buffers after you open the audio device. |
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151 */ |
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152 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained); |
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153 |
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154 /* |
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155 * Get the current audio state: |
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156 */ |
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157 typedef enum { |
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158 SDL_AUDIO_STOPPED = 0, |
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159 SDL_AUDIO_PLAYING, |
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160 SDL_AUDIO_PAUSED |
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161 } SDL_audiostatus; |
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162 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
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163 |
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164 /* |
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165 * This function pauses and unpauses the audio callback processing. |
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166 * It should be called with a parameter of 0 after opening the audio |
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167 * device to start playing sound. This is so you can safely initialize |
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168 * data for your callback function after opening the audio device. |
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169 * Silence will be written to the audio device during the pause. |
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170 */ |
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171 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
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172 |
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173 /* |
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174 * This function loads a WAVE from the data source, automatically freeing |
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175 * that source if 'freesrc' is non-zero. For example, to load a WAVE file, |
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176 * you could do: |
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177 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); |
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178 * |
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179 * If this function succeeds, it returns the given SDL_AudioSpec, |
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180 * filled with the audio data format of the wave data, and sets |
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181 * 'audio_buf' to a malloc()'d buffer containing the audio data, |
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182 * and sets 'audio_len' to the length of that audio buffer, in bytes. |
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183 * You need to free the audio buffer with SDL_FreeWAV() when you are |
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184 * done with it. |
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185 * |
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186 * This function returns NULL and sets the SDL error message if the |
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187 * wave file cannot be opened, uses an unknown data format, or is |
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188 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. |
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189 */ |
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190 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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191 |
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192 /* Compatibility convenience function -- loads a WAV from a file */ |
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193 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ |
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194 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) |
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195 |
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196 /* |
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197 * This function frees data previously allocated with SDL_LoadWAV_RW() |
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198 */ |
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199 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf); |
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200 |
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201 /* |
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202 * This function takes a source format and rate and a destination format |
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203 * and rate, and initializes the 'cvt' structure with information needed |
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204 * by SDL_ConvertAudio() to convert a buffer of audio data from one format |
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205 * to the other. |
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206 * This function returns 0, or -1 if there was an error. |
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207 */ |
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208 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt, |
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209 Uint16 src_format, Uint8 src_channels, int src_rate, |
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210 Uint16 dst_format, Uint8 dst_channels, int dst_rate); |
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211 |
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212 /* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), |
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213 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of |
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214 * audio data in the source format, this function will convert it in-place |
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215 * to the desired format. |
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216 * The data conversion may expand the size of the audio data, so the buffer |
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217 * cvt->buf should be allocated after the cvt structure is initialized by |
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218 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. |
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219 */ |
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220 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt); |
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221 |
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222 /* |
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223 * This takes two audio buffers of the playing audio format and mixes |
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224 * them, performing addition, volume adjustment, and overflow clipping. |
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225 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME |
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226 * for full audio volume. Note this does not change hardware volume. |
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227 * This is provided for convenience -- you can mix your own audio data. |
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228 */ |
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229 #define SDL_MIX_MAXVOLUME 128 |
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230 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume); |
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231 |
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232 /* |
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233 * The lock manipulated by these functions protects the callback function. |
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234 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the |
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235 * callback function is not running. Do not call these from the callback |
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236 * function or you will cause deadlock. |
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237 */ |
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238 extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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239 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
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240 |
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241 /* |
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242 * This function shuts down audio processing and closes the audio device. |
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243 */ |
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244 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
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245 |
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246 |
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247 /* Ends C function definitions when using C++ */ |
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248 #ifdef __cplusplus |
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249 } |
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250 #endif |
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251 #include "close_code.h" |
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252 |
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253 #endif /* _SDL_audio_h */ |