Mercurial > sdl-ios-xcode
annotate include/SDL_audio.h @ 1691:c901fd2a42da SDL-1.3
I think this fixes bug #244
| author | Sam Lantinga <slouken@libsdl.org> |
|---|---|
| date | Tue, 20 Jun 2006 05:40:57 +0000 |
| parents | 4da1ee79c9af |
| children |
| 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 | |
| 47 /* The calculated values in this structure are calculated by SDL_OpenAudio() */ | |
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48 typedef struct SDL_AudioSpec |
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49 { |
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50 int freq; /* DSP frequency -- samples per second */ |
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51 Uint16 format; /* Audio data format */ |
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52 Uint8 channels; /* Number of channels: 1 mono, 2 stereo */ |
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53 Uint8 silence; /* Audio buffer silence value (calculated) */ |
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54 Uint16 samples; /* Audio buffer size in samples (power of 2) */ |
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55 Uint16 padding; /* Necessary for some compile environments */ |
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56 Uint32 size; /* Audio buffer size in bytes (calculated) */ |
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57 /* This function is called when the audio device needs more data. |
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58 'stream' is a pointer to the audio data buffer |
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59 'len' is the length of that buffer in bytes. |
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60 Once the callback returns, the buffer will no longer be valid. |
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61 Stereo samples are stored in a LRLRLR ordering. |
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62 */ |
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63 void (SDLCALL * callback) (void *userdata, Uint8 * stream, int len); |
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64 void *userdata; |
| 0 | 65 } SDL_AudioSpec; |
| 66 | |
| 67 /* Audio format flags (defaults to LSB byte order) */ | |
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68 #define AUDIO_U8 0x0008 /* Unsigned 8-bit samples */ |
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69 #define AUDIO_S8 0x8008 /* Signed 8-bit samples */ |
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70 #define AUDIO_U16LSB 0x0010 /* Unsigned 16-bit samples */ |
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71 #define AUDIO_S16LSB 0x8010 /* Signed 16-bit samples */ |
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72 #define AUDIO_U16MSB 0x1010 /* As above, but big-endian byte order */ |
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73 #define AUDIO_S16MSB 0x9010 /* As above, but big-endian byte order */ |
| 0 | 74 #define AUDIO_U16 AUDIO_U16LSB |
| 75 #define AUDIO_S16 AUDIO_S16LSB | |
| 76 | |
| 77 /* Native audio byte ordering */ | |
| 78 #if SDL_BYTEORDER == SDL_LIL_ENDIAN | |
| 79 #define AUDIO_U16SYS AUDIO_U16LSB | |
| 80 #define AUDIO_S16SYS AUDIO_S16LSB | |
| 81 #else | |
| 82 #define AUDIO_U16SYS AUDIO_U16MSB | |
| 83 #define AUDIO_S16SYS AUDIO_S16MSB | |
| 84 #endif | |
| 85 | |
| 86 | |
| 87 /* A structure to hold a set of audio conversion filters and buffers */ | |
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88 typedef struct SDL_AudioCVT |
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89 { |
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90 int needed; /* Set to 1 if conversion possible */ |
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91 Uint16 src_format; /* Source audio format */ |
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92 Uint16 dst_format; /* Target audio format */ |
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93 double rate_incr; /* Rate conversion increment */ |
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94 Uint8 *buf; /* Buffer to hold entire audio data */ |
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95 int len; /* Length of original audio buffer */ |
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96 int len_cvt; /* Length of converted audio buffer */ |
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97 int len_mult; /* buffer must be len*len_mult big */ |
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98 double len_ratio; /* Given len, final size is len*len_ratio */ |
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99 void (SDLCALL * filters[10]) (struct SDL_AudioCVT * cvt, Uint16 format); |
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100 int filter_index; /* Current audio conversion function */ |
| 0 | 101 } SDL_AudioCVT; |
| 102 | |
| 103 | |
| 104 /* Function prototypes */ | |
| 105 | |
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106 /* These functions return the list of built in video drivers, in the |
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107 * order that they are normally initialized by default. |
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108 */ |
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109 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); |
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110 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); |
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111 |
| 0 | 112 /* These functions are used internally, and should not be used unless you |
| 113 * have a specific need to specify the audio driver you want to use. | |
| 114 * You should normally use SDL_Init() or SDL_InitSubSystem(). | |
| 115 */ | |
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116 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
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117 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
| 0 | 118 |
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119 /* This function returns the name of the current audio driver, or NULL |
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120 * if no driver has been initialized. |
| 0 | 121 */ |
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122 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); |
| 0 | 123 |
| 124 /* | |
| 125 * This function opens the audio device with the desired parameters, and | |
| 126 * returns 0 if successful, placing the actual hardware parameters in the | |
| 127 * structure pointed to by 'obtained'. If 'obtained' is NULL, the audio | |
| 128 * data passed to the callback function will be guaranteed to be in the | |
| 129 * requested format, and will be automatically converted to the hardware | |
| 130 * audio format if necessary. This function returns -1 if it failed | |
| 131 * to open the audio device, or couldn't set up the audio thread. | |
| 132 * | |
| 133 * When filling in the desired audio spec structure, | |
| 134 * 'desired->freq' should be the desired audio frequency in samples-per-second. | |
| 135 * 'desired->format' should be the desired audio format. | |
| 136 * 'desired->samples' is the desired size of the audio buffer, in samples. | |
| 137 * This number should be a power of two, and may be adjusted by the audio | |
| 138 * driver to a value more suitable for the hardware. Good values seem to | |
| 139 * range between 512 and 8096 inclusive, depending on the application and | |
| 140 * CPU speed. Smaller values yield faster response time, but can lead | |
| 141 * to underflow if the application is doing heavy processing and cannot | |
| 142 * fill the audio buffer in time. A stereo sample consists of both right | |
| 143 * and left channels in LR ordering. | |
| 144 * Note that the number of samples is directly related to time by the | |
| 145 * following formula: ms = (samples*1000)/freq | |
| 146 * 'desired->size' is the size in bytes of the audio buffer, and is | |
| 147 * calculated by SDL_OpenAudio(). | |
| 148 * 'desired->silence' is the value used to set the buffer to silence, | |
| 149 * and is calculated by SDL_OpenAudio(). | |
| 150 * 'desired->callback' should be set to a function that will be called | |
| 151 * when the audio device is ready for more data. It is passed a pointer | |
| 152 * to the audio buffer, and the length in bytes of the audio buffer. | |
| 153 * This function usually runs in a separate thread, and so you should | |
| 154 * protect data structures that it accesses by calling SDL_LockAudio() | |
| 155 * and SDL_UnlockAudio() in your code. | |
| 156 * 'desired->userdata' is passed as the first parameter to your callback | |
| 157 * function. | |
| 158 * | |
| 159 * The audio device starts out playing silence when it's opened, and should | |
| 160 * be enabled for playing by calling SDL_PauseAudio(0) when you are ready | |
| 161 * for your audio callback function to be called. Since the audio driver | |
| 162 * may modify the requested size of the audio buffer, you should allocate | |
| 163 * any local mixing buffers after you open the audio device. | |
| 164 */ | |
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165 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, |
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166 SDL_AudioSpec * obtained); |
| 0 | 167 |
| 168 /* | |
| 169 * Get the current audio state: | |
| 170 */ | |
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171 typedef enum |
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172 { |
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173 SDL_AUDIO_STOPPED = 0, |
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174 SDL_AUDIO_PLAYING, |
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175 SDL_AUDIO_PAUSED |
| 0 | 176 } SDL_audiostatus; |
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177 extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); |
| 0 | 178 |
| 179 /* | |
| 180 * This function pauses and unpauses the audio callback processing. | |
| 181 * It should be called with a parameter of 0 after opening the audio | |
| 182 * device to start playing sound. This is so you can safely initialize | |
| 183 * data for your callback function after opening the audio device. | |
| 184 * Silence will be written to the audio device during the pause. | |
| 185 */ | |
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186 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
| 0 | 187 |
| 188 /* | |
| 189 * This function loads a WAVE from the data source, automatically freeing | |
| 190 * that source if 'freesrc' is non-zero. For example, to load a WAVE file, | |
| 191 * you could do: | |
| 192 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); | |
| 193 * | |
| 194 * If this function succeeds, it returns the given SDL_AudioSpec, | |
| 195 * filled with the audio data format of the wave data, and sets | |
| 196 * 'audio_buf' to a malloc()'d buffer containing the audio data, | |
| 197 * and sets 'audio_len' to the length of that audio buffer, in bytes. | |
| 198 * You need to free the audio buffer with SDL_FreeWAV() when you are | |
| 199 * done with it. | |
| 200 * | |
| 201 * This function returns NULL and sets the SDL error message if the | |
| 202 * wave file cannot be opened, uses an unknown data format, or is | |
| 203 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. | |
| 204 */ | |
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205 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, |
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206 int freesrc, |
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207 SDL_AudioSpec * spec, |
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208 Uint8 ** audio_buf, |
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209 Uint32 * audio_len); |
| 0 | 210 |
| 211 /* Compatibility convenience function -- loads a WAV from a file */ | |
| 212 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ | |
| 213 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) | |
| 214 | |
| 215 /* | |
| 216 * This function frees data previously allocated with SDL_LoadWAV_RW() | |
| 217 */ | |
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218 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); |
| 0 | 219 |
| 220 /* | |
| 221 * This function takes a source format and rate and a destination format | |
| 222 * and rate, and initializes the 'cvt' structure with information needed | |
| 223 * by SDL_ConvertAudio() to convert a buffer of audio data from one format | |
| 224 * to the other. | |
| 225 * This function returns 0, or -1 if there was an error. | |
| 226 */ | |
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227 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, |
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228 Uint16 src_format, |
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229 Uint8 src_channels, |
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230 int src_rate, |
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231 Uint16 dst_format, |
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232 Uint8 dst_channels, |
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233 int dst_rate); |
| 0 | 234 |
| 235 /* Once you have initialized the 'cvt' structure using SDL_BuildAudioCVT(), | |
| 236 * created an audio buffer cvt->buf, and filled it with cvt->len bytes of | |
| 237 * audio data in the source format, this function will convert it in-place | |
| 238 * to the desired format. | |
| 239 * The data conversion may expand the size of the audio data, so the buffer | |
| 240 * cvt->buf should be allocated after the cvt structure is initialized by | |
| 241 * SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long. | |
| 242 */ | |
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243 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); |
| 0 | 244 |
| 245 /* | |
| 246 * This takes two audio buffers of the playing audio format and mixes | |
| 247 * them, performing addition, volume adjustment, and overflow clipping. | |
| 248 * The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME | |
| 249 * for full audio volume. Note this does not change hardware volume. | |
| 250 * This is provided for convenience -- you can mix your own audio data. | |
| 251 */ | |
| 252 #define SDL_MIX_MAXVOLUME 128 | |
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253 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, |
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254 Uint32 len, int volume); |
| 0 | 255 |
| 256 /* | |
| 257 * The lock manipulated by these functions protects the callback function. | |
| 258 * During a LockAudio/UnlockAudio pair, you can be guaranteed that the | |
| 259 * callback function is not running. Do not call these from the callback | |
| 260 * function or you will cause deadlock. | |
| 261 */ | |
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262 extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
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263 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
| 0 | 264 |
| 265 /* | |
| 266 * This function shuts down audio processing and closes the audio device. | |
| 267 */ | |
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268 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
| 0 | 269 |
| 270 | |
| 271 /* Ends C function definitions when using C++ */ | |
| 272 #ifdef __cplusplus | |
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273 /* *INDENT-OFF* */ |
| 0 | 274 } |
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275 /* *INDENT-ON* */ |
| 0 | 276 #endif |
| 277 #include "close_code.h" | |
| 278 | |
| 279 #endif /* _SDL_audio_h */ | |
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280 |
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281 /* vi: set ts=4 sw=4 expandtab: */ |