Mercurial > sdl-ios-xcode
annotate include/SDL_audio.h @ 2772:22c908f96d6e
Added the SDL_haptic files to the project
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Thu, 16 Oct 2008 12:35:28 +0000 |
| parents | 4eb759edddf5 |
| children | 99210400e8b9 |
| rev | line source |
|---|---|
| 0 | 1 /* |
| 2 SDL - Simple DirectMedia Layer | |
|
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3 Copyright (C) 1997-2006 Sam Lantinga |
| 0 | 4 |
| 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 |
| 0 | 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. |
| 0 | 9 |
| 10 This library is distributed in the hope that it will be useful, | |
| 11 but WITHOUT ANY WARRANTY; without even the implied warranty of | |
| 12 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
|
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13 Lesser General Public License for more details. |
| 0 | 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 |
| 0 | 18 |
| 19 Sam Lantinga | |
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20 slouken@libsdl.org |
| 0 | 21 */ |
| 22 | |
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23 /** |
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24 * \file SDL_audio.h |
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25 * |
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26 * Access to the raw audio mixing buffer for the SDL library |
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27 */ |
| 0 | 28 |
| 29 #ifndef _SDL_audio_h | |
| 30 #define _SDL_audio_h | |
| 31 | |
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32 #include "SDL_stdinc.h" |
| 0 | 33 #include "SDL_error.h" |
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34 #include "SDL_endian.h" |
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35 #include "SDL_mutex.h" |
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36 #include "SDL_thread.h" |
| 0 | 37 #include "SDL_rwops.h" |
| 38 | |
| 39 #include "begin_code.h" | |
| 40 /* Set up for C function definitions, even when using C++ */ | |
| 41 #ifdef __cplusplus | |
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42 /* *INDENT-OFF* */ |
| 0 | 43 extern "C" { |
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44 /* *INDENT-ON* */ |
| 0 | 45 #endif |
| 46 | |
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47 typedef Uint16 SDL_AudioFormat; |
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48 |
| 0 | 49 /* The calculated values in this structure are calculated by SDL_OpenAudio() */ |
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50 typedef struct SDL_AudioSpec |
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51 { |
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52 int freq; /* DSP frequency -- samples per second */ |
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53 SDL_AudioFormat format; /* Audio data format */ |
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54 Uint8 channels; /* Number of channels: 1 mono, 2 stereo */ |
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55 Uint8 silence; /* Audio buffer silence value (calculated) */ |
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56 Uint16 samples; /* Audio buffer size in samples (power of 2) */ |
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57 Uint16 padding; /* Necessary for some compile environments */ |
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58 Uint32 size; /* Audio buffer size in bytes (calculated) */ |
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59 /* This function is called when the audio device needs more data. |
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60 'stream' is a pointer to the audio data buffer |
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61 'len' is the length of that buffer in bytes. |
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62 Once the callback returns, the buffer will no longer be valid. |
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63 Stereo samples are stored in a LRLRLR ordering. |
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64 */ |
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65 void (SDLCALL * callback) (void *userdata, Uint8 * stream, int len); |
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66 void *userdata; |
| 0 | 67 } SDL_AudioSpec; |
| 68 | |
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69 |
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70 /* |
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71 These are what the 16 bits in SDL_AudioFormat currently mean... |
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72 (Unspecified bits are always zero.) |
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73 |
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74 ++-----------------------sample is signed if set |
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75 || |
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76 || ++-----------sample is bigendian if set |
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77 || || |
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78 || || ++---sample is float if set |
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79 || || || |
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80 || || || +---sample bit size---+ |
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81 || || || | | |
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82 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 |
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83 |
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84 There are macros in SDL 1.3 and later to query these bits. |
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85 */ |
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86 |
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87 #define SDL_AUDIO_MASK_BITSIZE (0xFF) |
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88 #define SDL_AUDIO_MASK_DATATYPE (1<<8) |
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89 #define SDL_AUDIO_MASK_ENDIAN (1<<12) |
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90 #define SDL_AUDIO_MASK_SIGNED (1<<15) |
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91 #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) |
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92 #define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) |
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93 #define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) |
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94 #define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) |
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95 #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) |
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96 #define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) |
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97 #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) |
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98 |
| 0 | 99 /* Audio format flags (defaults to LSB byte order) */ |
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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 */ |
| 0 | 106 #define AUDIO_U16 AUDIO_U16LSB |
| 107 #define AUDIO_S16 AUDIO_S16LSB | |
| 108 | |
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109 /* int32 support new to SDL 1.3 */ |
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110 #define AUDIO_S32LSB 0x8020 /* 32-bit integer samples */ |
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111 #define AUDIO_S32MSB 0x9020 /* As above, but big-endian byte order */ |
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112 #define AUDIO_S32 AUDIO_S32LSB |
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113 |
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114 /* float32 support new to SDL 1.3 */ |
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115 #define AUDIO_F32LSB 0x8120 /* 32-bit floating point samples */ |
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116 #define AUDIO_F32MSB 0x9120 /* As above, but big-endian byte order */ |
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117 #define AUDIO_F32 AUDIO_F32LSB |
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118 |
| 0 | 119 /* Native audio byte ordering */ |
| 120 #if SDL_BYTEORDER == SDL_LIL_ENDIAN | |
| 121 #define AUDIO_U16SYS AUDIO_U16LSB | |
| 122 #define AUDIO_S16SYS AUDIO_S16LSB | |
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123 #define AUDIO_S32SYS AUDIO_S32LSB |
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124 #define AUDIO_F32SYS AUDIO_F32LSB |
| 0 | 125 #else |
| 126 #define AUDIO_U16SYS AUDIO_U16MSB | |
| 127 #define AUDIO_S16SYS AUDIO_S16MSB | |
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128 #define AUDIO_S32SYS AUDIO_S32MSB |
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129 #define AUDIO_F32SYS AUDIO_F32MSB |
| 0 | 130 #endif |
| 131 | |
| 132 | |
| 133 /* A structure to hold a set of audio conversion filters and buffers */ | |
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134 struct SDL_AudioCVT; |
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135 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, |
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136 SDL_AudioFormat format); |
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137 |
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138 typedef struct SDL_AudioCVT |
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139 { |
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140 int needed; /* Set to 1 if conversion possible */ |
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141 SDL_AudioFormat src_format; /* Source audio format */ |
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142 SDL_AudioFormat dst_format; /* Target audio format */ |
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143 double rate_incr; /* Rate conversion increment */ |
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144 Uint8 *buf; /* Buffer to hold entire audio data */ |
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145 Uint8 *coeff; /* Filter coefficients: either big windowed sinc filter, or 6 IIR lowpass coefficients */ |
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146 Uint8 *state_buf; /* Sample history for either the FIR or IIR filter. For IIR filter, first two elements are X, second two are Y, and state_pos toggles the order */ |
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147 int state_pos; /* Position in the state */ |
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148 int len_sinc; /* Length of windowed sinc filter, in appropriate units (not necessarily bytes) */ |
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149 int len; /* Length of original audio buffer */ |
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150 int len_cvt; /* Length of converted audio buffer */ |
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151 int len_mult; /* buffer must be len*len_mult big */ |
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152 int len_div; /* destination length = len_mult / len_div * src length */ |
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153 double len_ratio; /* Given len, final size is len*len_ratio ( len_ratio = len_mult / len_div ) */ |
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154 SDL_AudioFilter filters[10]; /* Filter list */ |
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155 int filter_index; /* Current audio conversion function */ |
| 0 | 156 } SDL_AudioCVT; |
| 157 | |
| 158 | |
| 159 /* Function prototypes */ | |
| 160 | |
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161 /* These functions return the list of built in audio drivers, in the |
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162 * order that they are normally initialized by default. |
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163 */ |
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164 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); |
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165 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); |
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166 |
| 0 | 167 /* These functions are used internally, and should not be used unless you |
| 168 * have a specific need to specify the audio driver you want to use. | |
| 169 * You should normally use SDL_Init() or SDL_InitSubSystem(). | |
| 170 */ | |
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171 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
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172 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
| 0 | 173 |
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174 /* This function returns the name of the current audio driver, or NULL |
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175 * if no driver has been initialized. |
| 0 | 176 */ |
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177 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); |
| 0 | 178 |
| 179 /* | |
| 180 * This function opens the audio device with the desired parameters, and | |
| 181 * returns 0 if successful, placing the actual hardware parameters in the | |
| 182 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio | |
| 183 * data passed to the callback function will be guaranteed to be in the | |
| 184 * requested format, and will be automatically converted to the hardware | |
| 185 * audio format if necessary. This function returns -1 if it failed | |
| 186 * to open the audio device, or couldn't set up the audio thread. | |
| 187 * | |
| 188 * When filling in the desired audio spec structure, | |
| 189 * 'desired->freq' should be the desired audio frequency in samples-per-second. | |
| 190 * 'desired->format' should be the desired audio format. | |
| 191 * 'desired->samples' is the desired size of the audio buffer, in samples. | |
| 192 * This number should be a power of two, and may be adjusted by the audio | |
| 193 * driver to a value more suitable for the hardware. Good values seem to | |
| 194 * range between 512 and 8096 inclusive, depending on the application and | |
| 195 * CPU speed. Smaller values yield faster response time, but can lead | |
| 196 * to underflow if the application is doing heavy processing and cannot | |
| 197 * fill the audio buffer in time. A stereo sample consists of both right | |
| 198 * and left channels in LR ordering. | |
| 199 * Note that the number of samples is directly related to time by the | |
| 200 * following formula: ms = (samples*1000)/freq | |
| 201 * 'desired->size' is the size in bytes of the audio buffer, and is | |
| 202 * calculated by SDL_OpenAudio(). | |
| 203 * 'desired->silence' is the value used to set the buffer to silence, | |
| 204 * and is calculated by SDL_OpenAudio(). | |
| 205 * 'desired->callback' should be set to a function that will be called | |
| 206 * when the audio device is ready for more data. It is passed a pointer | |
| 207 * to the audio buffer, and the length in bytes of the audio buffer. | |
| 208 * This function usually runs in a separate thread, and so you should | |
| 209 * protect data structures that it accesses by calling SDL_LockAudio() | |
| 210 * and SDL_UnlockAudio() in your code. | |
| 211 * 'desired->userdata' is passed as the first parameter to your callback | |
| 212 * function. | |
| 213 * | |
| 214 * The audio device starts out playing silence when it's opened, and should | |
| 215 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready | |
| 216 * for your audio callback function to be called. Since the audio driver | |
| 217 * may modify the requested size of the audio buffer, you should allocate | |
| 218 * any local mixing buffers after you open the audio device. | |
| 219 */ | |
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220 extern DECLSPEC int SDLCALL SDL_OpenAudio(const SDL_AudioSpec * desired, |
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221 SDL_AudioSpec * obtained); |
| 0 | 222 |
| 223 /* | |
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224 * SDL Audio Device IDs. |
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225 * A successful call to SDL_OpenAudio() is always device id 1, and legacy |
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226 * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls |
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227 * always returns devices >= 2 on success. The legacy calls are good both |
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228 * for backwards compatibility and when you don't care about multiple, |
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229 * specific, or capture devices. |
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230 */ |
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231 typedef Uint32 SDL_AudioDeviceID; |
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232 |
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233 /* |
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234 * Get the number of available devices exposed by the current driver. |
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235 * Only valid after a successfully initializing the audio subsystem. |
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236 * Returns -1 if an explicit list of devices can't be determined; this is |
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237 * not an error. For example, if SDL is set up to talk to a remote audio |
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238 * server, it can't list every one available on the Internet, but it will |
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239 * still allow a specific host to be specified to SDL_OpenAudioDevice(). |
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240 * In many common cases, when this function returns a value <= 0, it can still |
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241 * successfully open the default device (NULL for first argument of |
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242 * SDL_OpenAudioDevice()). |
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243 */ |
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244 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); |
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245 |
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246 /* |
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247 * Get the human-readable name of a specific audio device. |
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248 * Must be a value between 0 and (number of audio devices-1). |
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249 * Only valid after a successfully initializing the audio subsystem. |
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250 * The values returned by this function reflect the latest call to |
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251 * SDL_GetNumAudioDevices(); recall that function to redetect available |
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252 * hardware. |
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253 * |
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254 * The string returned by this function is UTF-8 encoded, read-only, and |
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255 * managed internally. You are not to free it. If you need to keep the |
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256 * string for any length of time, you should make your own copy of it, as it |
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257 * will be invalid next time any of several other SDL functions is called. |
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258 */ |
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259 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, |
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260 int iscapture); |
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261 |
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262 |
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263 /* |
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264 * Open a specific audio device. Passing in a device name of NULL requests |
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265 * the most reasonable default (and is equivalent to calling SDL_OpenAudio()). |
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266 * The device name is a UTF-8 string reported by SDL_GetAudioDevice(), but |
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267 * some drivers allow arbitrary and driver-specific strings, such as a |
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268 * hostname/IP address for a remote audio server, or a filename in the |
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269 * diskaudio driver. |
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270 * Returns 0 on error, a valid device ID that is >= 2 on success. |
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271 * SDL_OpenAudio(), unlike this function, always acts on device ID 1. |
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272 */ |
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273 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char |
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274 *device, |
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275 int iscapture, |
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276 const |
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277 SDL_AudioSpec * |
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278 desired, |
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279 SDL_AudioSpec * |
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280 obtained); |
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281 |
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282 |
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283 |
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284 /* |
| 0 | 285 * Get the current audio state: |
| 286 */ | |
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287 typedef enum |
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288 { |
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289 SDL_AUDIO_STOPPED = 0, |
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290 SDL_AUDIO_PLAYING, |
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291 SDL_AUDIO_PAUSED |
| 0 | 292 } SDL_audiostatus; |
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293 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
| 0 | 294 |
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295 extern DECLSPEC SDL_audiostatus SDLCALL |
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296 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); |
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297 |
| 0 | 298 /* |
| 299 * This function pauses and unpauses the audio callback processing. | |
| 300 * It should be called with a parameter of 0 after opening the audio | |
| 301 * device to start playing sound. This is so you can safely initialize | |
| 302 * data for your callback function after opening the audio device. | |
| 303 * Silence will be written to the audio device during the pause. | |
| 304 */ | |
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305 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
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306 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, |
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307 int pause_on); |
| 0 | 308 |
| 309 /* | |
| 310 * This function loads a WAVE from the data source, automatically freeing | |
| 311 * that source if 'freesrc' is non-zero. For example, to load a WAVE file, | |
| 312 * you could do: | |
| 313 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); | |
| 314 * | |
| 315 * If this function succeeds, it returns the given SDL_AudioSpec, | |
| 316 * filled with the audio data format of the wave data, and sets | |
| 317 * 'audio_buf' to a malloc()'d buffer containing the audio data, | |
| 318 * and sets 'audio_len' to the length of that audio buffer, in bytes. | |
| 319 * You need to free the audio buffer with SDL_FreeWAV() when you are | |
| 320 * done with it. | |
| 321 * | |
| 322 * This function returns NULL and sets the SDL error message if the | |
| 323 * wave file cannot be opened, uses an unknown data format, or is | |
| 324 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. | |
| 325 */ | |
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326 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, |
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327 int freesrc, |
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328 SDL_AudioSpec * spec, |
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329 Uint8 ** audio_buf, |
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330 Uint32 * audio_len); |
| 0 | 331 |
| 332 /* Compatibility convenience function -- loads a WAV from a file */ | |
| 333 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ | |
| 334 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) | |
| 335 | |
| 336 /* | |
| 337 * This function frees data previously allocated with SDL_LoadWAV_RW() | |
| 338 */ | |
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339 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); |
| 0 | 340 |
| 341 /* | |
| 342 * This function takes a source format and rate and a destination format | |
| 343 * and rate, and initializes the 'cvt' structure with information needed | |
| 344 * by SDL_ConvertAudio() to convert a buffer of audio data from one format | |
| 345 * to the other. | |
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346 * Returns -1 if the format conversion is not supported, 0 if there's |
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347 * no conversion needed, or 1 if the audio filter is set up. |
| 0 | 348 */ |
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349 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, |
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350 SDL_AudioFormat src_format, |
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351 Uint8 src_channels, |
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352 int src_rate, |
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353 SDL_AudioFormat dst_format, |
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354 Uint8 dst_channels, |
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355 int dst_rate); |
| 0 | 356 |
| 357 /* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), | |
| 358 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of | |
| 359 * audio data in the source format, this function will convert it in-place | |
| 360 * to the desired format. | |
| 361 * The data conversion may expand the size of the audio data, so the buffer | |
| 362 * cvt->buf should be allocated after the cvt structure is initialized by | |
| 363 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. | |
| 364 */ | |
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365 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); |
| 0 | 366 |
| 367 /* | |
| 368 * This takes two audio buffers of the playing audio format and mixes | |
| 369 * them, performing addition, volume adjustment, and overflow clipping. | |
| 370 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME | |
| 371 * for full audio volume. Note this does not change hardware volume. | |
| 372 * This is provided for convenience -- you can mix your own audio data. | |
| 373 */ | |
| 374 #define SDL_MIX_MAXVOLUME 128 | |
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375 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, |
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376 Uint32 len, int volume); |
| 0 | 377 |
| 378 /* | |
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379 * This works like SDL_MixAudio, but you specify the audio format instead of |
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380 * using the format of audio device 1. Thus it can be used when no audio |
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381 * device is open at all. |
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382 */ |
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383 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, |
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384 const Uint8 * src, |
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385 SDL_AudioFormat format, |
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386 Uint32 len, int volume); |
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387 |
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388 /* |
| 0 | 389 * The lock manipulated by these functions protects the callback function. |
| 390 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the | |
| 391 * callback function is not running. Do not call these from the callback | |
| 392 * function or you will cause deadlock. | |
| 393 */ | |
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394 extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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395 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); |
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396 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
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397 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); |
| 0 | 398 |
| 399 /* | |
| 400 * This function shuts down audio processing and closes the audio device. | |
| 401 */ | |
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402 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
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403 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); |
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404 |
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405 /* |
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406 * Returns 1 if audio device is still functioning, zero if not, -1 on error. |
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407 */ |
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408 extern DECLSPEC int SDLCALL SDL_AudioDeviceConnected(SDL_AudioDeviceID dev); |
| 0 | 409 |
| 410 | |
| 411 /* Ends C function definitions when using C++ */ | |
| 412 #ifdef __cplusplus | |
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413 /* *INDENT-OFF* */ |
| 0 | 414 } |
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415 /* *INDENT-ON* */ |
| 0 | 416 #endif |
| 417 #include "close_code.h" | |
| 418 | |
| 419 #endif /* _SDL_audio_h */ | |
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420 |
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421 /* vi: set ts=4 sw=4 expandtab: */ |