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1 | /* | |
2 | Simple DirectMedia Layer | |
3 | Copyright (C) 1997-2013 Sam Lantinga <slouken@libsdl.org> | |
4 | ||
5 | This software is provided 'as-is', without any express or implied | |
6 | warranty. In no event will the authors be held liable for any damages | |
7 | arising from the use of this software. | |
8 | ||
9 | Permission is granted to anyone to use this software for any purpose, | |
10 | including commercial applications, and to alter it and redistribute it | |
11 | freely, subject to the following restrictions: | |
12 | ||
13 | 1. The origin of this software must not be misrepresented; you must not | |
14 | claim that you wrote the original software. If you use this software | |
15 | in a product, an acknowledgment in the product documentation would be | |
16 | appreciated but is not required. | |
17 | 2. Altered source versions must be plainly marked as such, and must not be | |
18 | misrepresented as being the original software. | |
19 | 3. This notice may not be removed or altered from any source distribution. | |
20 | */ | |
21 | ||
22 | /** | |
23 | * \file SDL_audio.h | |
24 | * | |
25 | * Access to the raw audio mixing buffer for the SDL library. | |
26 | */ | |
27 | ||
28 | #ifndef _SDL_audio_h | |
29 | #define _SDL_audio_h | |
30 | ||
31 | #include "SDL_stdinc.h" | |
32 | #include "SDL_error.h" | |
33 | #include "SDL_endian.h" | |
34 | #include "SDL_mutex.h" | |
35 | #include "SDL_thread.h" | |
36 | #include "SDL_rwops.h" | |
37 | ||
38 | #include "begin_code.h" | |
39 | /* Set up for C function definitions, even when using C++ */ | |
40 | #ifdef __cplusplus | |
41 | extern "C" { | |
42 | #endif | |
43 | ||
44 | /** | |
45 | * \brief Audio format flags. | |
46 | * | |
47 | * These are what the 16 bits in SDL_AudioFormat currently mean... | |
48 | * (Unspecified bits are always zero). | |
49 | * | |
50 | * \verbatim | |
51 | ++-----------------------sample is signed if set | |
52 | || | |
53 | || ++-----------sample is bigendian if set | |
54 | || || | |
55 | || || ++---sample is float if set | |
56 | || || || | |
57 | || || || +---sample bit size---+ | |
58 | || || || | | | |
59 | 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 | |
60 | \endverbatim | |
61 | * | |
62 | * There are macros in SDL 2.0 and later to query these bits. | |
63 | */ | |
64 | typedef Uint16 SDL_AudioFormat; | |
65 | ||
66 | /** | |
67 | * \name Audio flags | |
68 | */ | |
69 | /*@{*/ | |
70 | ||
71 | #define SDL_AUDIO_MASK_BITSIZE (0xFF) | |
72 | #define SDL_AUDIO_MASK_DATATYPE (1<<8) | |
73 | #define SDL_AUDIO_MASK_ENDIAN (1<<12) | |
74 | #define SDL_AUDIO_MASK_SIGNED (1<<15) | |
75 | #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) | |
76 | #define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) | |
77 | #define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) | |
78 | #define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) | |
79 | #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) | |
80 | #define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) | |
81 | #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) | |
82 | ||
83 | /** | |
84 | * \name Audio format flags | |
85 | * | |
86 | * Defaults to LSB byte order. | |
87 | */ | |
88 | /*@{*/ | |
89 | #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ | |
90 | #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ | |
91 | #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ | |
92 | #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ | |
93 | #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ | |
94 | #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ | |
95 | #define AUDIO_U16 AUDIO_U16LSB | |
96 | #define AUDIO_S16 AUDIO_S16LSB | |
97 | /*@}*/ | |
98 | ||
99 | /** | |
100 | * \name int32 support | |
101 | */ | |
102 | /*@{*/ | |
103 | #define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ | |
104 | #define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ | |
105 | #define AUDIO_S32 AUDIO_S32LSB | |
106 | /*@}*/ | |
107 | ||
108 | /** | |
109 | * \name float32 support | |
110 | */ | |
111 | /*@{*/ | |
112 | #define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ | |
113 | #define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ | |
114 | #define AUDIO_F32 AUDIO_F32LSB | |
115 | /*@}*/ | |
116 | ||
117 | /** | |
118 | * \name Native audio byte ordering | |
119 | */ | |
120 | /*@{*/ | |
121 | #if SDL_BYTEORDER == SDL_LIL_ENDIAN | |
122 | #define AUDIO_U16SYS AUDIO_U16LSB | |
123 | #define AUDIO_S16SYS AUDIO_S16LSB | |
124 | #define AUDIO_S32SYS AUDIO_S32LSB | |
125 | #define AUDIO_F32SYS AUDIO_F32LSB | |
126 | #else | |
127 | #define AUDIO_U16SYS AUDIO_U16MSB | |
128 | #define AUDIO_S16SYS AUDIO_S16MSB | |
129 | #define AUDIO_S32SYS AUDIO_S32MSB | |
130 | #define AUDIO_F32SYS AUDIO_F32MSB | |
131 | #endif | |
132 | /*@}*/ | |
133 | ||
134 | /** | |
135 | * \name Allow change flags | |
136 | * | |
137 | * Which audio format changes are allowed when opening a device. | |
138 | */ | |
139 | /*@{*/ | |
140 | #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 | |
141 | #define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 | |
142 | #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 | |
143 | #define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE) | |
144 | /*@}*/ | |
145 | ||
146 | /*@}*//*Audio flags*/ | |
147 | ||
148 | /** | |
149 | * This function is called when the audio device needs more data. | |
150 | * | |
151 | * \param userdata An application-specific parameter saved in | |
152 | * the SDL_AudioSpec structure | |
153 | * \param stream A pointer to the audio data buffer. | |
154 | * \param len The length of that buffer in bytes. | |
155 | * | |
156 | * Once the callback returns, the buffer will no longer be valid. | |
157 | * Stereo samples are stored in a LRLRLR ordering. | |
158 | */ | |
159 | typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, | |
160 | int len); | |
161 | ||
162 | /** | |
163 | * The calculated values in this structure are calculated by SDL_OpenAudio(). | |
164 | */ | |
165 | typedef struct SDL_AudioSpec | |
166 | { | |
167 | int freq; /**< DSP frequency -- samples per second */ | |
168 | SDL_AudioFormat format; /**< Audio data format */ | |
169 | Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ | |
170 | Uint8 silence; /**< Audio buffer silence value (calculated) */ | |
171 | Uint16 samples; /**< Audio buffer size in samples (power of 2) */ | |
172 | Uint16 padding; /**< Necessary for some compile environments */ | |
173 | Uint32 size; /**< Audio buffer size in bytes (calculated) */ | |
174 | SDL_AudioCallback callback; | |
175 | void *userdata; | |
176 | } SDL_AudioSpec; | |
177 | ||
178 | ||
179 | struct SDL_AudioCVT; | |
180 | typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, | |
181 | SDL_AudioFormat format); | |
182 | ||
183 | /** | |
184 | * A structure to hold a set of audio conversion filters and buffers. | |
185 | */ | |
186 | #ifdef __GNUC__ | |
187 | /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't | |
188 | pad it out to 88 bytes to guarantee ABI compatibility between compilers. | |
189 | vvv | |
190 | The next time we rev the ABI, make sure to size the ints and add padding. | |
191 | */ | |
192 | #define SDL_AUDIOCVT_PACKED __attribute__((packed)) | |
193 | #else | |
194 | #define SDL_AUDIOCVT_PACKED | |
195 | #endif | |
196 | /* */ | |
197 | typedef struct SDL_AudioCVT | |
198 | { | |
199 | int needed; /**< Set to 1 if conversion possible */ | |
200 | SDL_AudioFormat src_format; /**< Source audio format */ | |
201 | SDL_AudioFormat dst_format; /**< Target audio format */ | |
202 | double rate_incr; /**< Rate conversion increment */ | |
203 | Uint8 *buf; /**< Buffer to hold entire audio data */ | |
204 | int len; /**< Length of original audio buffer */ | |
205 | int len_cvt; /**< Length of converted audio buffer */ | |
206 | int len_mult; /**< buffer must be len*len_mult big */ | |
207 | double len_ratio; /**< Given len, final size is len*len_ratio */ | |
208 | SDL_AudioFilter filters[10]; /**< Filter list */ | |
209 | int filter_index; /**< Current audio conversion function */ | |
210 | } SDL_AUDIOCVT_PACKED SDL_AudioCVT; | |
211 | ||
212 | ||
213 | /* Function prototypes */ | |
214 | ||
215 | /** | |
216 | * \name Driver discovery functions | |
217 | * | |
218 | * These functions return the list of built in audio drivers, in the | |
219 | * order that they are normally initialized by default. | |
220 | */ | |
221 | /*@{*/ | |
222 | extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); | |
223 | extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); | |
224 | /*@}*/ | |
225 | ||
226 | /** | |
227 | * \name Initialization and cleanup | |
228 | * | |
229 | * \internal These functions are used internally, and should not be used unless | |
230 | * you have a specific need to specify the audio driver you want to | |
231 | * use. You should normally use SDL_Init() or SDL_InitSubSystem(). | |
232 | */ | |
233 | /*@{*/ | |
234 | extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); | |
235 | extern DECLSPEC void SDLCALL SDL_AudioQuit(void); | |
236 | /*@}*/ | |
237 | ||
238 | /** | |
239 | * This function returns the name of the current audio driver, or NULL | |
240 | * if no driver has been initialized. | |
241 | */ | |
242 | extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); | |
243 | ||
244 | /** | |
245 | * This function opens the audio device with the desired parameters, and | |
246 | * returns 0 if successful, placing the actual hardware parameters in the | |
247 | * structure pointed to by \c obtained. If \c obtained is NULL, the audio | |
248 | * data passed to the callback function will be guaranteed to be in the | |
249 | * requested format, and will be automatically converted to the hardware | |
250 | * audio format if necessary. This function returns -1 if it failed | |
251 | * to open the audio device, or couldn't set up the audio thread. | |
252 | * | |
253 | * When filling in the desired audio spec structure, | |
254 | * - \c desired->freq should be the desired audio frequency in samples-per- | |
255 | * second. | |
256 | * - \c desired->format should be the desired audio format. | |
257 | * - \c desired->samples is the desired size of the audio buffer, in | |
258 | * samples. This number should be a power of two, and may be adjusted by | |
259 | * the audio driver to a value more suitable for the hardware. Good values | |
260 | * seem to range between 512 and 8096 inclusive, depending on the | |
261 | * application and CPU speed. Smaller values yield faster response time, | |
262 | * but can lead to underflow if the application is doing heavy processing | |
263 | * and cannot fill the audio buffer in time. A stereo sample consists of | |
264 | * both right and left channels in LR ordering. | |
265 | * Note that the number of samples is directly related to time by the | |
266 | * following formula: \code ms = (samples*1000)/freq \endcode | |
267 | * - \c desired->size is the size in bytes of the audio buffer, and is | |
268 | * calculated by SDL_OpenAudio(). | |
269 | * - \c desired->silence is the value used to set the buffer to silence, | |
270 | * and is calculated by SDL_OpenAudio(). | |
271 | * - \c desired->callback should be set to a function that will be called | |
272 | * when the audio device is ready for more data. It is passed a pointer | |
273 | * to the audio buffer, and the length in bytes of the audio buffer. | |
274 | * This function usually runs in a separate thread, and so you should | |
275 | * protect data structures that it accesses by calling SDL_LockAudio() | |
276 | * and SDL_UnlockAudio() in your code. | |
277 | * - \c desired->userdata is passed as the first parameter to your callback | |
278 | * function. | |
279 | * | |
280 | * The audio device starts out playing silence when it's opened, and should | |
281 | * be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready | |
282 | * for your audio callback function to be called. Since the audio driver | |
283 | * may modify the requested size of the audio buffer, you should allocate | |
284 | * any local mixing buffers after you open the audio device. | |
285 | */ | |
286 | extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, | |
287 | SDL_AudioSpec * obtained); | |
288 | ||
289 | /** | |
290 | * SDL Audio Device IDs. | |
291 | * | |
292 | * A successful call to SDL_OpenAudio() is always device id 1, and legacy | |
293 | * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls | |
294 | * always returns devices >= 2 on success. The legacy calls are good both | |
295 | * for backwards compatibility and when you don't care about multiple, | |
296 | * specific, or capture devices. | |
297 | */ | |
298 | typedef Uint32 SDL_AudioDeviceID; | |
299 | ||
300 | /** | |
301 | * Get the number of available devices exposed by the current driver. | |
302 | * Only valid after a successfully initializing the audio subsystem. | |
303 | * Returns -1 if an explicit list of devices can't be determined; this is | |
304 | * not an error. For example, if SDL is set up to talk to a remote audio | |
305 | * server, it can't list every one available on the Internet, but it will | |
306 | * still allow a specific host to be specified to SDL_OpenAudioDevice(). | |
307 | * | |
308 | * In many common cases, when this function returns a value <= 0, it can still | |
309 | * successfully open the default device (NULL for first argument of | |
310 | * SDL_OpenAudioDevice()). | |
311 | */ | |
312 | extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); | |
313 | ||
314 | /** | |
315 | * Get the human-readable name of a specific audio device. | |
316 | * Must be a value between 0 and (number of audio devices-1). | |
317 | * Only valid after a successfully initializing the audio subsystem. | |
318 | * The values returned by this function reflect the latest call to | |
319 | * SDL_GetNumAudioDevices(); recall that function to redetect available | |
320 | * hardware. | |
321 | * | |
322 | * The string returned by this function is UTF-8 encoded, read-only, and | |
323 | * managed internally. You are not to free it. If you need to keep the | |
324 | * string for any length of time, you should make your own copy of it, as it | |
325 | * will be invalid next time any of several other SDL functions is called. | |
326 | */ | |
327 | extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, | |
328 | int iscapture); | |
329 | ||
330 | ||
331 | /** | |
332 | * Open a specific audio device. Passing in a device name of NULL requests | |
333 | * the most reasonable default (and is equivalent to calling SDL_OpenAudio()). | |
334 | * | |
335 | * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but | |
336 | * some drivers allow arbitrary and driver-specific strings, such as a | |
337 | * hostname/IP address for a remote audio server, or a filename in the | |
338 | * diskaudio driver. | |
339 | * | |
340 | * \return 0 on error, a valid device ID that is >= 2 on success. | |
341 | * | |
342 | * SDL_OpenAudio(), unlike this function, always acts on device ID 1. | |
343 | */ | |
344 | extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char | |
345 | *device, | |
346 | int iscapture, | |
347 | const | |
348 | SDL_AudioSpec * | |
349 | desired, | |
350 | SDL_AudioSpec * | |
351 | obtained, | |
352 | int | |
353 | allowed_changes); | |
354 | ||
355 | ||
356 | ||
357 | /** | |
358 | * \name Audio state | |
359 | * | |
360 | * Get the current audio state. | |
361 | */ | |
362 | /*@{*/ | |
363 | typedef enum | |
364 | { | |
365 | SDL_AUDIO_STOPPED = 0, | |
366 | SDL_AUDIO_PLAYING, | |
367 | SDL_AUDIO_PAUSED | |
368 | } SDL_AudioStatus; | |
369 | extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); | |
370 | ||
371 | extern DECLSPEC SDL_AudioStatus SDLCALL | |
372 | SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); | |
373 | /*@}*//*Audio State*/ | |
374 | ||
375 | /** | |
376 | * \name Pause audio functions | |
377 | * | |
378 | * These functions pause and unpause the audio callback processing. | |
379 | * They should be called with a parameter of 0 after opening the audio | |
380 | * device to start playing sound. This is so you can safely initialize | |
381 | * data for your callback function after opening the audio device. | |
382 | * Silence will be written to the audio device during the pause. | |
383 | */ | |
384 | /*@{*/ | |
385 | extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); | |
386 | extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, | |
387 | int pause_on); | |
388 | /*@}*//*Pause audio functions*/ | |
389 | ||
390 | /** | |
391 | * This function loads a WAVE from the data source, automatically freeing | |
392 | * that source if \c freesrc is non-zero. For example, to load a WAVE file, | |
393 | * you could do: | |
394 | * \code | |
395 | * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); | |
396 | * \endcode | |
397 | * | |
398 | * If this function succeeds, it returns the given SDL_AudioSpec, | |
399 | * filled with the audio data format of the wave data, and sets | |
400 | * \c *audio_buf to a malloc()'d buffer containing the audio data, | |
401 | * and sets \c *audio_len to the length of that audio buffer, in bytes. | |
402 | * You need to free the audio buffer with SDL_FreeWAV() when you are | |
403 | * done with it. | |
404 | * | |
405 | * This function returns NULL and sets the SDL error message if the | |
406 | * wave file cannot be opened, uses an unknown data format, or is | |
407 | * corrupt. Currently raw and MS-ADPCM WAVE files are supported. | |
408 | */ | |
409 | extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, | |
410 | int freesrc, | |
411 | SDL_AudioSpec * spec, | |
412 | Uint8 ** audio_buf, | |
413 | Uint32 * audio_len); | |
414 | ||
415 | /** | |
416 | * Loads a WAV from a file. | |
417 | * Compatibility convenience function. | |
418 | */ | |
419 | #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ | |
420 | SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) | |
421 | ||
422 | /** | |
423 | * This function frees data previously allocated with SDL_LoadWAV_RW() | |
424 | */ | |
425 | extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); | |
426 | ||
427 | /** | |
428 | * This function takes a source format and rate and a destination format | |
429 | * and rate, and initializes the \c cvt structure with information needed | |
430 | * by SDL_ConvertAudio() to convert a buffer of audio data from one format | |
431 | * to the other. | |
432 | * | |
433 | * \return -1 if the format conversion is not supported, 0 if there's | |
434 | * no conversion needed, or 1 if the audio filter is set up. | |
435 | */ | |
436 | extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, | |
437 | SDL_AudioFormat src_format, | |
438 | Uint8 src_channels, | |
439 | int src_rate, | |
440 | SDL_AudioFormat dst_format, | |
441 | Uint8 dst_channels, | |
442 | int dst_rate); | |
443 | ||
444 | /** | |
445 | * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(), | |
446 | * created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of | |
447 | * audio data in the source format, this function will convert it in-place | |
448 | * to the desired format. | |
449 | * | |
450 | * The data conversion may expand the size of the audio data, so the buffer | |
451 | * \c cvt->buf should be allocated after the \c cvt structure is initialized by | |
452 | * SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long. | |
453 | */ | |
454 | extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); | |
455 | ||
456 | #define SDL_MIX_MAXVOLUME 128 | |
457 | /** | |
458 | * This takes two audio buffers of the playing audio format and mixes | |
459 | * them, performing addition, volume adjustment, and overflow clipping. | |
460 | * The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME | |
461 | * for full audio volume. Note this does not change hardware volume. | |
462 | * This is provided for convenience -- you can mix your own audio data. | |
463 | */ | |
464 | extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, | |
465 | Uint32 len, int volume); | |
466 | ||
467 | /** | |
468 | * This works like SDL_MixAudio(), but you specify the audio format instead of | |
469 | * using the format of audio device 1. Thus it can be used when no audio | |
470 | * device is open at all. | |
471 | */ | |
472 | extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, | |
473 | const Uint8 * src, | |
474 | SDL_AudioFormat format, | |
475 | Uint32 len, int volume); | |
476 | ||
477 | /** | |
478 | * \name Audio lock functions | |
479 | * | |
480 | * The lock manipulated by these functions protects the callback function. | |
481 | * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that | |
482 | * the callback function is not running. Do not call these from the callback | |
483 | * function or you will cause deadlock. | |
484 | */ | |
485 | /*@{*/ | |
486 | extern DECLSPEC void SDLCALL SDL_LockAudio(void); | |
487 | extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); | |
488 | extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); | |
489 | extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); | |
490 | /*@}*//*Audio lock functions*/ | |
491 | ||
492 | /** | |
493 | * This function shuts down audio processing and closes the audio device. | |
494 | */ | |
495 | extern DECLSPEC void SDLCALL SDL_CloseAudio(void); | |
496 | extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); | |
497 | ||
498 | /* Ends C function definitions when using C++ */ | |
499 | #ifdef __cplusplus | |
500 | } | |
501 | #endif | |
502 | #include "close_code.h" | |
503 | ||
504 | #endif /* _SDL_audio_h */ | |
505 | ||
506 | /* vi: set ts=4 sw=4 expandtab: */ |