1 | /* |
2 | Simple DirectMedia Layer |
3 | Copyright (C) 1997-2022 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 | /* !!! FIXME: several functions in here need Doxygen comments. */ |
23 | |
24 | /** |
25 | * \file SDL_audio.h |
26 | * |
27 | * Access to the raw audio mixing buffer for the SDL library. |
28 | */ |
29 | |
30 | #ifndef SDL_audio_h_ |
31 | #define SDL_audio_h_ |
32 | |
33 | #include "SDL_stdinc.h" |
34 | #include "SDL_error.h" |
35 | #include "SDL_endian.h" |
36 | #include "SDL_mutex.h" |
37 | #include "SDL_thread.h" |
38 | #include "SDL_rwops.h" |
39 | |
40 | #include "begin_code.h" |
41 | /* Set up for C function definitions, even when using C++ */ |
42 | #ifdef __cplusplus |
43 | extern "C" { |
44 | #endif |
45 | |
46 | /** |
47 | * \brief Audio format flags. |
48 | * |
49 | * These are what the 16 bits in SDL_AudioFormat currently mean... |
50 | * (Unspecified bits are always zero). |
51 | * |
52 | * \verbatim |
53 | ++-----------------------sample is signed if set |
54 | || |
55 | || ++-----------sample is bigendian if set |
56 | || || |
57 | || || ++---sample is float if set |
58 | || || || |
59 | || || || +---sample bit size---+ |
60 | || || || | | |
61 | 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 |
62 | \endverbatim |
63 | * |
64 | * There are macros in SDL 2.0 and later to query these bits. |
65 | */ |
66 | typedef Uint16 SDL_AudioFormat; |
67 | |
68 | /** |
69 | * \name Audio flags |
70 | */ |
71 | /* @{ */ |
72 | |
73 | #define SDL_AUDIO_MASK_BITSIZE (0xFF) |
74 | #define SDL_AUDIO_MASK_DATATYPE (1<<8) |
75 | #define SDL_AUDIO_MASK_ENDIAN (1<<12) |
76 | #define SDL_AUDIO_MASK_SIGNED (1<<15) |
77 | #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) |
78 | #define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) |
79 | #define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) |
80 | #define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) |
81 | #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) |
82 | #define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) |
83 | #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) |
84 | |
85 | /** |
86 | * \name Audio format flags |
87 | * |
88 | * Defaults to LSB byte order. |
89 | */ |
90 | /* @{ */ |
91 | #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ |
92 | #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ |
93 | #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ |
94 | #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ |
95 | #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ |
96 | #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ |
97 | #define AUDIO_U16 AUDIO_U16LSB |
98 | #define AUDIO_S16 AUDIO_S16LSB |
99 | /* @} */ |
100 | |
101 | /** |
102 | * \name int32 support |
103 | */ |
104 | /* @{ */ |
105 | #define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ |
106 | #define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ |
107 | #define AUDIO_S32 AUDIO_S32LSB |
108 | /* @} */ |
109 | |
110 | /** |
111 | * \name float32 support |
112 | */ |
113 | /* @{ */ |
114 | #define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ |
115 | #define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ |
116 | #define AUDIO_F32 AUDIO_F32LSB |
117 | /* @} */ |
118 | |
119 | /** |
120 | * \name Native audio byte ordering |
121 | */ |
122 | /* @{ */ |
123 | #if SDL_BYTEORDER == SDL_LIL_ENDIAN |
124 | #define AUDIO_U16SYS AUDIO_U16LSB |
125 | #define AUDIO_S16SYS AUDIO_S16LSB |
126 | #define AUDIO_S32SYS AUDIO_S32LSB |
127 | #define AUDIO_F32SYS AUDIO_F32LSB |
128 | #else |
129 | #define AUDIO_U16SYS AUDIO_U16MSB |
130 | #define AUDIO_S16SYS AUDIO_S16MSB |
131 | #define AUDIO_S32SYS AUDIO_S32MSB |
132 | #define AUDIO_F32SYS AUDIO_F32MSB |
133 | #endif |
134 | /* @} */ |
135 | |
136 | /** |
137 | * \name Allow change flags |
138 | * |
139 | * Which audio format changes are allowed when opening a device. |
140 | */ |
141 | /* @{ */ |
142 | #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 |
143 | #define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 |
144 | #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 |
145 | #define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008 |
146 | #define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE) |
147 | /* @} */ |
148 | |
149 | /* @} *//* Audio flags */ |
150 | |
151 | /** |
152 | * This function is called when the audio device needs more data. |
153 | * |
154 | * \param userdata An application-specific parameter saved in |
155 | * the SDL_AudioSpec structure |
156 | * \param stream A pointer to the audio data buffer. |
157 | * \param len The length of that buffer in bytes. |
158 | * |
159 | * Once the callback returns, the buffer will no longer be valid. |
160 | * Stereo samples are stored in a LRLRLR ordering. |
161 | * |
162 | * You can choose to avoid callbacks and use SDL_QueueAudio() instead, if |
163 | * you like. Just open your audio device with a NULL callback. |
164 | */ |
165 | typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, |
166 | int len); |
167 | |
168 | /** |
169 | * The calculated values in this structure are calculated by SDL_OpenAudio(). |
170 | * |
171 | * For multi-channel audio, the default SDL channel mapping is: |
172 | * 2: FL FR (stereo) |
173 | * 3: FL FR LFE (2.1 surround) |
174 | * 4: FL FR BL BR (quad) |
175 | * 5: FL FR FC BL BR (quad + center) |
176 | * 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR) |
177 | * 7: FL FR FC LFE BC SL SR (6.1 surround) |
178 | * 8: FL FR FC LFE BL BR SL SR (7.1 surround) |
179 | */ |
180 | typedef struct SDL_AudioSpec |
181 | { |
182 | int freq; /**< DSP frequency -- samples per second */ |
183 | SDL_AudioFormat format; /**< Audio data format */ |
184 | Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ |
185 | Uint8 silence; /**< Audio buffer silence value (calculated) */ |
186 | Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */ |
187 | Uint16 padding; /**< Necessary for some compile environments */ |
188 | Uint32 size; /**< Audio buffer size in bytes (calculated) */ |
189 | SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */ |
190 | void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */ |
191 | } SDL_AudioSpec; |
192 | |
193 | |
194 | struct SDL_AudioCVT; |
195 | typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, |
196 | SDL_AudioFormat format); |
197 | |
198 | /** |
199 | * \brief Upper limit of filters in SDL_AudioCVT |
200 | * |
201 | * The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is |
202 | * currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers, |
203 | * one of which is the terminating NULL pointer. |
204 | */ |
205 | #define SDL_AUDIOCVT_MAX_FILTERS 9 |
206 | |
207 | /** |
208 | * \struct SDL_AudioCVT |
209 | * \brief A structure to hold a set of audio conversion filters and buffers. |
210 | * |
211 | * Note that various parts of the conversion pipeline can take advantage |
212 | * of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require |
213 | * you to pass it aligned data, but can possibly run much faster if you |
214 | * set both its (buf) field to a pointer that is aligned to 16 bytes, and its |
215 | * (len) field to something that's a multiple of 16, if possible. |
216 | */ |
217 | #if defined(__GNUC__) && !defined(__CHERI_PURE_CAPABILITY__) |
218 | /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't |
219 | pad it out to 88 bytes to guarantee ABI compatibility between compilers. |
220 | This is not a concern on CHERI architectures, where pointers must be stored |
221 | at aligned locations otherwise they will become invalid, and thus structs |
222 | containing pointers cannot be packed without giving a warning or error. |
223 | vvv |
224 | The next time we rev the ABI, make sure to size the ints and add padding. |
225 | */ |
226 | #define SDL_AUDIOCVT_PACKED __attribute__((packed)) |
227 | #else |
228 | #define SDL_AUDIOCVT_PACKED |
229 | #endif |
230 | /* */ |
231 | typedef struct SDL_AudioCVT |
232 | { |
233 | int needed; /**< Set to 1 if conversion possible */ |
234 | SDL_AudioFormat src_format; /**< Source audio format */ |
235 | SDL_AudioFormat dst_format; /**< Target audio format */ |
236 | double rate_incr; /**< Rate conversion increment */ |
237 | Uint8 *buf; /**< Buffer to hold entire audio data */ |
238 | int len; /**< Length of original audio buffer */ |
239 | int len_cvt; /**< Length of converted audio buffer */ |
240 | int len_mult; /**< buffer must be len*len_mult big */ |
241 | double len_ratio; /**< Given len, final size is len*len_ratio */ |
242 | SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */ |
243 | int filter_index; /**< Current audio conversion function */ |
244 | } SDL_AUDIOCVT_PACKED SDL_AudioCVT; |
245 | |
246 | |
247 | /* Function prototypes */ |
248 | |
249 | /** |
250 | * \name Driver discovery functions |
251 | * |
252 | * These functions return the list of built in audio drivers, in the |
253 | * order that they are normally initialized by default. |
254 | */ |
255 | /* @{ */ |
256 | |
257 | /** |
258 | * Use this function to get the number of built-in audio drivers. |
259 | * |
260 | * This function returns a hardcoded number. This never returns a negative |
261 | * value; if there are no drivers compiled into this build of SDL, this |
262 | * function returns zero. The presence of a driver in this list does not mean |
263 | * it will function, it just means SDL is capable of interacting with that |
264 | * interface. For example, a build of SDL might have esound support, but if |
265 | * there's no esound server available, SDL's esound driver would fail if used. |
266 | * |
267 | * By default, SDL tries all drivers, in its preferred order, until one is |
268 | * found to be usable. |
269 | * |
270 | * \returns the number of built-in audio drivers. |
271 | * |
272 | * \since This function is available since SDL 2.0.0. |
273 | * |
274 | * \sa SDL_GetAudioDriver |
275 | */ |
276 | extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); |
277 | |
278 | /** |
279 | * Use this function to get the name of a built in audio driver. |
280 | * |
281 | * The list of audio drivers is given in the order that they are normally |
282 | * initialized by default; the drivers that seem more reasonable to choose |
283 | * first (as far as the SDL developers believe) are earlier in the list. |
284 | * |
285 | * The names of drivers are all simple, low-ASCII identifiers, like "alsa", |
286 | * "coreaudio" or "xaudio2". These never have Unicode characters, and are not |
287 | * meant to be proper names. |
288 | * |
289 | * \param index the index of the audio driver; the value ranges from 0 to |
290 | * SDL_GetNumAudioDrivers() - 1 |
291 | * \returns the name of the audio driver at the requested index, or NULL if an |
292 | * invalid index was specified. |
293 | * |
294 | * \since This function is available since SDL 2.0.0. |
295 | * |
296 | * \sa SDL_GetNumAudioDrivers |
297 | */ |
298 | extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); |
299 | /* @} */ |
300 | |
301 | /** |
302 | * \name Initialization and cleanup |
303 | * |
304 | * \internal These functions are used internally, and should not be used unless |
305 | * you have a specific need to specify the audio driver you want to |
306 | * use. You should normally use SDL_Init() or SDL_InitSubSystem(). |
307 | */ |
308 | /* @{ */ |
309 | |
310 | /** |
311 | * Use this function to initialize a particular audio driver. |
312 | * |
313 | * This function is used internally, and should not be used unless you have a |
314 | * specific need to designate the audio driver you want to use. You should |
315 | * normally use SDL_Init() or SDL_InitSubSystem(). |
316 | * |
317 | * \param driver_name the name of the desired audio driver |
318 | * \returns 0 on success or a negative error code on failure; call |
319 | * SDL_GetError() for more information. |
320 | * |
321 | * \since This function is available since SDL 2.0.0. |
322 | * |
323 | * \sa SDL_AudioQuit |
324 | */ |
325 | extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); |
326 | |
327 | /** |
328 | * Use this function to shut down audio if you initialized it with |
329 | * SDL_AudioInit(). |
330 | * |
331 | * This function is used internally, and should not be used unless you have a |
332 | * specific need to specify the audio driver you want to use. You should |
333 | * normally use SDL_Quit() or SDL_QuitSubSystem(). |
334 | * |
335 | * \since This function is available since SDL 2.0.0. |
336 | * |
337 | * \sa SDL_AudioInit |
338 | */ |
339 | extern DECLSPEC void SDLCALL SDL_AudioQuit(void); |
340 | /* @} */ |
341 | |
342 | /** |
343 | * Get the name of the current audio driver. |
344 | * |
345 | * The returned string points to internal static memory and thus never becomes |
346 | * invalid, even if you quit the audio subsystem and initialize a new driver |
347 | * (although such a case would return a different static string from another |
348 | * call to this function, of course). As such, you should not modify or free |
349 | * the returned string. |
350 | * |
351 | * \returns the name of the current audio driver or NULL if no driver has been |
352 | * initialized. |
353 | * |
354 | * \since This function is available since SDL 2.0.0. |
355 | * |
356 | * \sa SDL_AudioInit |
357 | */ |
358 | extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); |
359 | |
360 | /** |
361 | * This function is a legacy means of opening the audio device. |
362 | * |
363 | * This function remains for compatibility with SDL 1.2, but also because it's |
364 | * slightly easier to use than the new functions in SDL 2.0. The new, more |
365 | * powerful, and preferred way to do this is SDL_OpenAudioDevice(). |
366 | * |
367 | * This function is roughly equivalent to: |
368 | * |
369 | * ```c |
370 | * SDL_OpenAudioDevice(NULL, 0, desired, obtained, SDL_AUDIO_ALLOW_ANY_CHANGE); |
371 | * ``` |
372 | * |
373 | * With two notable exceptions: |
374 | * |
375 | * - If `obtained` is NULL, we use `desired` (and allow no changes), which |
376 | * means desired will be modified to have the correct values for silence, |
377 | * etc, and SDL will convert any differences between your app's specific |
378 | * request and the hardware behind the scenes. |
379 | * - The return value is always success or failure, and not a device ID, which |
380 | * means you can only have one device open at a time with this function. |
381 | * |
382 | * \param desired an SDL_AudioSpec structure representing the desired output |
383 | * format. Please refer to the SDL_OpenAudioDevice |
384 | * documentation for details on how to prepare this structure. |
385 | * \param obtained an SDL_AudioSpec structure filled in with the actual |
386 | * parameters, or NULL. |
387 | * \returns 0 if successful, placing the actual hardware parameters in the |
388 | * structure pointed to by `obtained`. |
389 | * |
390 | * If `obtained` is NULL, the audio data passed to the callback |
391 | * function will be guaranteed to be in the requested format, and |
392 | * will be automatically converted to the actual hardware audio |
393 | * format if necessary. If `obtained` is NULL, `desired` will have |
394 | * fields modified. |
395 | * |
396 | * This function returns a negative error code on failure to open the |
397 | * audio device or failure to set up the audio thread; call |
398 | * SDL_GetError() for more information. |
399 | * |
400 | * \since This function is available since SDL 2.0.0. |
401 | * |
402 | * \sa SDL_CloseAudio |
403 | * \sa SDL_LockAudio |
404 | * \sa SDL_PauseAudio |
405 | * \sa SDL_UnlockAudio |
406 | */ |
407 | extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, |
408 | SDL_AudioSpec * obtained); |
409 | |
410 | /** |
411 | * SDL Audio Device IDs. |
412 | * |
413 | * A successful call to SDL_OpenAudio() is always device id 1, and legacy |
414 | * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls |
415 | * always returns devices >= 2 on success. The legacy calls are good both |
416 | * for backwards compatibility and when you don't care about multiple, |
417 | * specific, or capture devices. |
418 | */ |
419 | typedef Uint32 SDL_AudioDeviceID; |
420 | |
421 | /** |
422 | * Get the number of built-in audio devices. |
423 | * |
424 | * This function is only valid after successfully initializing the audio |
425 | * subsystem. |
426 | * |
427 | * Note that audio capture support is not implemented as of SDL 2.0.4, so the |
428 | * `iscapture` parameter is for future expansion and should always be zero for |
429 | * now. |
430 | * |
431 | * This function will return -1 if an explicit list of devices can't be |
432 | * determined. Returning -1 is not an error. For example, if SDL is set up to |
433 | * talk to a remote audio server, it can't list every one available on the |
434 | * Internet, but it will still allow a specific host to be specified in |
435 | * SDL_OpenAudioDevice(). |
436 | * |
437 | * In many common cases, when this function returns a value <= 0, it can still |
438 | * successfully open the default device (NULL for first argument of |
439 | * SDL_OpenAudioDevice()). |
440 | * |
441 | * This function may trigger a complete redetect of available hardware. It |
442 | * should not be called for each iteration of a loop, but rather once at the |
443 | * start of a loop: |
444 | * |
445 | * ```c |
446 | * // Don't do this: |
447 | * for (int i = 0; i < SDL_GetNumAudioDevices(0); i++) |
448 | * |
449 | * // do this instead: |
450 | * const int count = SDL_GetNumAudioDevices(0); |
451 | * for (int i = 0; i < count; ++i) { do_something_here(); } |
452 | * ``` |
453 | * |
454 | * \param iscapture zero to request playback devices, non-zero to request |
455 | * recording devices |
456 | * \returns the number of available devices exposed by the current driver or |
457 | * -1 if an explicit list of devices can't be determined. A return |
458 | * value of -1 does not necessarily mean an error condition. |
459 | * |
460 | * \since This function is available since SDL 2.0.0. |
461 | * |
462 | * \sa SDL_GetAudioDeviceName |
463 | * \sa SDL_OpenAudioDevice |
464 | */ |
465 | extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); |
466 | |
467 | /** |
468 | * Get the human-readable name of a specific audio device. |
469 | * |
470 | * This function is only valid after successfully initializing the audio |
471 | * subsystem. The values returned by this function reflect the latest call to |
472 | * SDL_GetNumAudioDevices(); re-call that function to redetect available |
473 | * hardware. |
474 | * |
475 | * The string returned by this function is UTF-8 encoded, read-only, and |
476 | * managed internally. You are not to free it. If you need to keep the string |
477 | * for any length of time, you should make your own copy of it, as it will be |
478 | * invalid next time any of several other SDL functions are called. |
479 | * |
480 | * \param index the index of the audio device; valid values range from 0 to |
481 | * SDL_GetNumAudioDevices() - 1 |
482 | * \param iscapture non-zero to query the list of recording devices, zero to |
483 | * query the list of output devices. |
484 | * \returns the name of the audio device at the requested index, or NULL on |
485 | * error. |
486 | * |
487 | * \since This function is available since SDL 2.0.0. |
488 | * |
489 | * \sa SDL_GetNumAudioDevices |
490 | */ |
491 | extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, |
492 | int iscapture); |
493 | |
494 | /** |
495 | * Get the preferred audio format of a specific audio device. |
496 | * |
497 | * This function is only valid after a successfully initializing the audio |
498 | * subsystem. The values returned by this function reflect the latest call to |
499 | * SDL_GetNumAudioDevices(); re-call that function to redetect available |
500 | * hardware. |
501 | * |
502 | * `spec` will be filled with the sample rate, sample format, and channel |
503 | * count. All other values in the structure are filled with 0. When the |
504 | * supported struct members are 0, SDL was unable to get the property from the |
505 | * backend. |
506 | * |
507 | * \param index the index of the audio device; valid values range from 0 to |
508 | * SDL_GetNumAudioDevices() - 1 |
509 | * \param iscapture non-zero to query the list of recording devices, zero to |
510 | * query the list of output devices. |
511 | * \param spec The SDL_AudioSpec to be initialized by this function. |
512 | * \returns 0 on success, nonzero on error |
513 | * |
514 | * \since This function is available since SDL 2.0.16. |
515 | * |
516 | * \sa SDL_GetNumAudioDevices |
517 | */ |
518 | extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index, |
519 | int iscapture, |
520 | SDL_AudioSpec *spec); |
521 | |
522 | |
523 | /** |
524 | * Open a specific audio device. |
525 | * |
526 | * SDL_OpenAudio(), unlike this function, always acts on device ID 1. As such, |
527 | * this function will never return a 1 so as not to conflict with the legacy |
528 | * function. |
529 | * |
530 | * Please note that SDL 2.0 before 2.0.5 did not support recording; as such, |
531 | * this function would fail if `iscapture` was not zero. Starting with SDL |
532 | * 2.0.5, recording is implemented and this value can be non-zero. |
533 | * |
534 | * Passing in a `device` name of NULL requests the most reasonable default |
535 | * (and is equivalent to what SDL_OpenAudio() does to choose a device). The |
536 | * `device` name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but |
537 | * some drivers allow arbitrary and driver-specific strings, such as a |
538 | * hostname/IP address for a remote audio server, or a filename in the |
539 | * diskaudio driver. |
540 | * |
541 | * An opened audio device starts out paused, and should be enabled for playing |
542 | * by calling SDL_PauseAudioDevice(devid, 0) when you are ready for your audio |
543 | * callback function to be called. Since the audio driver may modify the |
544 | * requested size of the audio buffer, you should allocate any local mixing |
545 | * buffers after you open the audio device. |
546 | * |
547 | * The audio callback runs in a separate thread in most cases; you can prevent |
548 | * race conditions between your callback and other threads without fully |
549 | * pausing playback with SDL_LockAudioDevice(). For more information about the |
550 | * callback, see SDL_AudioSpec. |
551 | * |
552 | * Managing the audio spec via 'desired' and 'obtained': |
553 | * |
554 | * When filling in the desired audio spec structure: |
555 | * |
556 | * - `desired->freq` should be the frequency in sample-frames-per-second (Hz). |
557 | * - `desired->format` should be the audio format (`AUDIO_S16SYS`, etc). |
558 | * - `desired->samples` is the desired size of the audio buffer, in _sample |
559 | * frames_ (with stereo output, two samples--left and right--would make a |
560 | * single sample frame). This number should be a power of two, and may be |
561 | * adjusted by the audio driver to a value more suitable for the hardware. |
562 | * Good values seem to range between 512 and 8096 inclusive, depending on |
563 | * the application and CPU speed. Smaller values reduce latency, but can |
564 | * lead to underflow if the application is doing heavy processing and cannot |
565 | * fill the audio buffer in time. Note that the number of sample frames is |
566 | * directly related to time by the following formula: `ms = |
567 | * (sampleframes*1000)/freq` |
568 | * - `desired->size` is the size in _bytes_ of the audio buffer, and is |
569 | * calculated by SDL_OpenAudioDevice(). You don't initialize this. |
570 | * - `desired->silence` is the value used to set the buffer to silence, and is |
571 | * calculated by SDL_OpenAudioDevice(). You don't initialize this. |
572 | * - `desired->callback` should be set to a function that will be called when |
573 | * the audio device is ready for more data. It is passed a pointer to the |
574 | * audio buffer, and the length in bytes of the audio buffer. This function |
575 | * usually runs in a separate thread, and so you should protect data |
576 | * structures that it accesses by calling SDL_LockAudioDevice() and |
577 | * SDL_UnlockAudioDevice() in your code. Alternately, you may pass a NULL |
578 | * pointer here, and call SDL_QueueAudio() with some frequency, to queue |
579 | * more audio samples to be played (or for capture devices, call |
580 | * SDL_DequeueAudio() with some frequency, to obtain audio samples). |
581 | * - `desired->userdata` is passed as the first parameter to your callback |
582 | * function. If you passed a NULL callback, this value is ignored. |
583 | * |
584 | * `allowed_changes` can have the following flags OR'd together: |
585 | * |
586 | * - `SDL_AUDIO_ALLOW_FREQUENCY_CHANGE` |
587 | * - `SDL_AUDIO_ALLOW_FORMAT_CHANGE` |
588 | * - `SDL_AUDIO_ALLOW_CHANNELS_CHANGE` |
589 | * - `SDL_AUDIO_ALLOW_ANY_CHANGE` |
590 | * |
591 | * These flags specify how SDL should behave when a device cannot offer a |
592 | * specific feature. If the application requests a feature that the hardware |
593 | * doesn't offer, SDL will always try to get the closest equivalent. |
594 | * |
595 | * For example, if you ask for float32 audio format, but the sound card only |
596 | * supports int16, SDL will set the hardware to int16. If you had set |
597 | * SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the `obtained` |
598 | * structure. If that flag was *not* set, SDL will prepare to convert your |
599 | * callback's float32 audio to int16 before feeding it to the hardware and |
600 | * will keep the originally requested format in the `obtained` structure. |
601 | * |
602 | * The resulting audio specs, varying depending on hardware and on what |
603 | * changes were allowed, will then be written back to `obtained`. |
604 | * |
605 | * If your application can only handle one specific data format, pass a zero |
606 | * for `allowed_changes` and let SDL transparently handle any differences. |
607 | * |
608 | * \param device a UTF-8 string reported by SDL_GetAudioDeviceName() or a |
609 | * driver-specific name as appropriate. NULL requests the most |
610 | * reasonable default device. |
611 | * \param iscapture non-zero to specify a device should be opened for |
612 | * recording, not playback |
613 | * \param desired an SDL_AudioSpec structure representing the desired output |
614 | * format; see SDL_OpenAudio() for more information |
615 | * \param obtained an SDL_AudioSpec structure filled in with the actual output |
616 | * format; see SDL_OpenAudio() for more information |
617 | * \param allowed_changes 0, or one or more flags OR'd together |
618 | * \returns a valid device ID that is > 0 on success or 0 on failure; call |
619 | * SDL_GetError() for more information. |
620 | * |
621 | * For compatibility with SDL 1.2, this will never return 1, since |
622 | * SDL reserves that ID for the legacy SDL_OpenAudio() function. |
623 | * |
624 | * \since This function is available since SDL 2.0.0. |
625 | * |
626 | * \sa SDL_CloseAudioDevice |
627 | * \sa SDL_GetAudioDeviceName |
628 | * \sa SDL_LockAudioDevice |
629 | * \sa SDL_OpenAudio |
630 | * \sa SDL_PauseAudioDevice |
631 | * \sa SDL_UnlockAudioDevice |
632 | */ |
633 | extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice( |
634 | const char *device, |
635 | int iscapture, |
636 | const SDL_AudioSpec *desired, |
637 | SDL_AudioSpec *obtained, |
638 | int allowed_changes); |
639 | |
640 | |
641 | |
642 | /** |
643 | * \name Audio state |
644 | * |
645 | * Get the current audio state. |
646 | */ |
647 | /* @{ */ |
648 | typedef enum |
649 | { |
650 | SDL_AUDIO_STOPPED = 0, |
651 | SDL_AUDIO_PLAYING, |
652 | SDL_AUDIO_PAUSED |
653 | } SDL_AudioStatus; |
654 | |
655 | /** |
656 | * This function is a legacy means of querying the audio device. |
657 | * |
658 | * New programs might want to use SDL_GetAudioDeviceStatus() instead. This |
659 | * function is equivalent to calling... |
660 | * |
661 | * ```c |
662 | * SDL_GetAudioDeviceStatus(1); |
663 | * ``` |
664 | * |
665 | * ...and is only useful if you used the legacy SDL_OpenAudio() function. |
666 | * |
667 | * \returns the SDL_AudioStatus of the audio device opened by SDL_OpenAudio(). |
668 | * |
669 | * \since This function is available since SDL 2.0.0. |
670 | * |
671 | * \sa SDL_GetAudioDeviceStatus |
672 | */ |
673 | extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); |
674 | |
675 | /** |
676 | * Use this function to get the current audio state of an audio device. |
677 | * |
678 | * \param dev the ID of an audio device previously opened with |
679 | * SDL_OpenAudioDevice() |
680 | * \returns the SDL_AudioStatus of the specified audio device. |
681 | * |
682 | * \since This function is available since SDL 2.0.0. |
683 | * |
684 | * \sa SDL_PauseAudioDevice |
685 | */ |
686 | extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); |
687 | /* @} *//* Audio State */ |
688 | |
689 | /** |
690 | * \name Pause audio functions |
691 | * |
692 | * These functions pause and unpause the audio callback processing. |
693 | * They should be called with a parameter of 0 after opening the audio |
694 | * device to start playing sound. This is so you can safely initialize |
695 | * data for your callback function after opening the audio device. |
696 | * Silence will be written to the audio device during the pause. |
697 | */ |
698 | /* @{ */ |
699 | |
700 | /** |
701 | * This function is a legacy means of pausing the audio device. |
702 | * |
703 | * New programs might want to use SDL_PauseAudioDevice() instead. This |
704 | * function is equivalent to calling... |
705 | * |
706 | * ```c |
707 | * SDL_PauseAudioDevice(1, pause_on); |
708 | * ``` |
709 | * |
710 | * ...and is only useful if you used the legacy SDL_OpenAudio() function. |
711 | * |
712 | * \param pause_on non-zero to pause, 0 to unpause |
713 | * |
714 | * \since This function is available since SDL 2.0.0. |
715 | * |
716 | * \sa SDL_GetAudioStatus |
717 | * \sa SDL_PauseAudioDevice |
718 | */ |
719 | extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); |
720 | |
721 | /** |
722 | * Use this function to pause and unpause audio playback on a specified |
723 | * device. |
724 | * |
725 | * This function pauses and unpauses the audio callback processing for a given |
726 | * device. Newly-opened audio devices start in the paused state, so you must |
727 | * call this function with **pause_on**=0 after opening the specified audio |
728 | * device to start playing sound. This allows you to safely initialize data |
729 | * for your callback function after opening the audio device. Silence will be |
730 | * written to the audio device while paused, and the audio callback is |
731 | * guaranteed to not be called. Pausing one device does not prevent other |
732 | * unpaused devices from running their callbacks. |
733 | * |
734 | * Pausing state does not stack; even if you pause a device several times, a |
735 | * single unpause will start the device playing again, and vice versa. This is |
736 | * different from how SDL_LockAudioDevice() works. |
737 | * |
738 | * If you just need to protect a few variables from race conditions vs your |
739 | * callback, you shouldn't pause the audio device, as it will lead to dropouts |
740 | * in the audio playback. Instead, you should use SDL_LockAudioDevice(). |
741 | * |
742 | * \param dev a device opened by SDL_OpenAudioDevice() |
743 | * \param pause_on non-zero to pause, 0 to unpause |
744 | * |
745 | * \since This function is available since SDL 2.0.0. |
746 | * |
747 | * \sa SDL_LockAudioDevice |
748 | */ |
749 | extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, |
750 | int pause_on); |
751 | /* @} *//* Pause audio functions */ |
752 | |
753 | /** |
754 | * Load the audio data of a WAVE file into memory. |
755 | * |
756 | * Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len` to |
757 | * be valid pointers. The entire data portion of the file is then loaded into |
758 | * memory and decoded if necessary. |
759 | * |
760 | * If `freesrc` is non-zero, the data source gets automatically closed and |
761 | * freed before the function returns. |
762 | * |
763 | * Supported formats are RIFF WAVE files with the formats PCM (8, 16, 24, and |
764 | * 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and |
765 | * A-law and mu-law (8 bits). Other formats are currently unsupported and |
766 | * cause an error. |
767 | * |
768 | * If this function succeeds, the pointer returned by it is equal to `spec` |
769 | * and the pointer to the audio data allocated by the function is written to |
770 | * `audio_buf` and its length in bytes to `audio_len`. The SDL_AudioSpec |
771 | * members `freq`, `channels`, and `format` are set to the values of the audio |
772 | * data in the buffer. The `samples` member is set to a sane default and all |
773 | * others are set to zero. |
774 | * |
775 | * It's necessary to use SDL_FreeWAV() to free the audio data returned in |
776 | * `audio_buf` when it is no longer used. |
777 | * |
778 | * Because of the underspecification of the .WAV format, there are many |
779 | * problematic files in the wild that cause issues with strict decoders. To |
780 | * provide compatibility with these files, this decoder is lenient in regards |
781 | * to the truncation of the file, the fact chunk, and the size of the RIFF |
782 | * chunk. The hints `SDL_HINT_WAVE_RIFF_CHUNK_SIZE`, |
783 | * `SDL_HINT_WAVE_TRUNCATION`, and `SDL_HINT_WAVE_FACT_CHUNK` can be used to |
784 | * tune the behavior of the loading process. |
785 | * |
786 | * Any file that is invalid (due to truncation, corruption, or wrong values in |
787 | * the headers), too big, or unsupported causes an error. Additionally, any |
788 | * critical I/O error from the data source will terminate the loading process |
789 | * with an error. The function returns NULL on error and in all cases (with |
790 | * the exception of `src` being NULL), an appropriate error message will be |
791 | * set. |
792 | * |
793 | * It is required that the data source supports seeking. |
794 | * |
795 | * Example: |
796 | * |
797 | * ```c |
798 | * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, &spec, &buf, &len); |
799 | * ``` |
800 | * |
801 | * Note that the SDL_LoadWAV macro does this same thing for you, but in a less |
802 | * messy way: |
803 | * |
804 | * ```c |
805 | * SDL_LoadWAV("sample.wav", &spec, &buf, &len); |
806 | * ``` |
807 | * |
808 | * \param src The data source for the WAVE data |
809 | * \param freesrc If non-zero, SDL will _always_ free the data source |
810 | * \param spec An SDL_AudioSpec that will be filled in with the wave file's |
811 | * format details |
812 | * \param audio_buf A pointer filled with the audio data, allocated by the |
813 | * function. |
814 | * \param audio_len A pointer filled with the length of the audio data buffer |
815 | * in bytes |
816 | * \returns This function, if successfully called, returns `spec`, which will |
817 | * be filled with the audio data format of the wave source data. |
818 | * `audio_buf` will be filled with a pointer to an allocated buffer |
819 | * containing the audio data, and `audio_len` is filled with the |
820 | * length of that audio buffer in bytes. |
821 | * |
822 | * This function returns NULL if the .WAV file cannot be opened, uses |
823 | * an unknown data format, or is corrupt; call SDL_GetError() for |
824 | * more information. |
825 | * |
826 | * When the application is done with the data returned in |
827 | * `audio_buf`, it should call SDL_FreeWAV() to dispose of it. |
828 | * |
829 | * \since This function is available since SDL 2.0.0. |
830 | * |
831 | * \sa SDL_FreeWAV |
832 | * \sa SDL_LoadWAV |
833 | */ |
834 | extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, |
835 | int freesrc, |
836 | SDL_AudioSpec * spec, |
837 | Uint8 ** audio_buf, |
838 | Uint32 * audio_len); |
839 | |
840 | /** |
841 | * Loads a WAV from a file. |
842 | * Compatibility convenience function. |
843 | */ |
844 | #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ |
845 | SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) |
846 | |
847 | /** |
848 | * Free data previously allocated with SDL_LoadWAV() or SDL_LoadWAV_RW(). |
849 | * |
850 | * After a WAVE file has been opened with SDL_LoadWAV() or SDL_LoadWAV_RW() |
851 | * its data can eventually be freed with SDL_FreeWAV(). It is safe to call |
852 | * this function with a NULL pointer. |
853 | * |
854 | * \param audio_buf a pointer to the buffer created by SDL_LoadWAV() or |
855 | * SDL_LoadWAV_RW() |
856 | * |
857 | * \since This function is available since SDL 2.0.0. |
858 | * |
859 | * \sa SDL_LoadWAV |
860 | * \sa SDL_LoadWAV_RW |
861 | */ |
862 | extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); |
863 | |
864 | /** |
865 | * Initialize an SDL_AudioCVT structure for conversion. |
866 | * |
867 | * Before an SDL_AudioCVT structure can be used to convert audio data it must |
868 | * be initialized with source and destination information. |
869 | * |
870 | * This function will zero out every field of the SDL_AudioCVT, so it must be |
871 | * called before the application fills in the final buffer information. |
872 | * |
873 | * Once this function has returned successfully, and reported that a |
874 | * conversion is necessary, the application fills in the rest of the fields in |
875 | * SDL_AudioCVT, now that it knows how large a buffer it needs to allocate, |
876 | * and then can call SDL_ConvertAudio() to complete the conversion. |
877 | * |
878 | * \param cvt an SDL_AudioCVT structure filled in with audio conversion |
879 | * information |
880 | * \param src_format the source format of the audio data; for more info see |
881 | * SDL_AudioFormat |
882 | * \param src_channels the number of channels in the source |
883 | * \param src_rate the frequency (sample-frames-per-second) of the source |
884 | * \param dst_format the destination format of the audio data; for more info |
885 | * see SDL_AudioFormat |
886 | * \param dst_channels the number of channels in the destination |
887 | * \param dst_rate the frequency (sample-frames-per-second) of the destination |
888 | * \returns 1 if the audio filter is prepared, 0 if no conversion is needed, |
889 | * or a negative error code on failure; call SDL_GetError() for more |
890 | * information. |
891 | * |
892 | * \since This function is available since SDL 2.0.0. |
893 | * |
894 | * \sa SDL_ConvertAudio |
895 | */ |
896 | extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, |
897 | SDL_AudioFormat src_format, |
898 | Uint8 src_channels, |
899 | int src_rate, |
900 | SDL_AudioFormat dst_format, |
901 | Uint8 dst_channels, |
902 | int dst_rate); |
903 | |
904 | /** |
905 | * Convert audio data to a desired audio format. |
906 | * |
907 | * This function does the actual audio data conversion, after the application |
908 | * has called SDL_BuildAudioCVT() to prepare the conversion information and |
909 | * then filled in the buffer details. |
910 | * |
911 | * Once the application has initialized the `cvt` structure using |
912 | * SDL_BuildAudioCVT(), allocated an audio buffer and filled it with audio |
913 | * data in the source format, this function will convert the buffer, in-place, |
914 | * to the desired format. |
915 | * |
916 | * The data conversion may go through several passes; any given pass may |
917 | * possibly temporarily increase the size of the data. For example, SDL might |
918 | * expand 16-bit data to 32 bits before resampling to a lower frequency, |
919 | * shrinking the data size after having grown it briefly. Since the supplied |
920 | * buffer will be both the source and destination, converting as necessary |
921 | * in-place, the application must allocate a buffer that will fully contain |
922 | * the data during its largest conversion pass. After SDL_BuildAudioCVT() |
923 | * returns, the application should set the `cvt->len` field to the size, in |
924 | * bytes, of the source data, and allocate a buffer that is `cvt->len * |
925 | * cvt->len_mult` bytes long for the `buf` field. |
926 | * |
927 | * The source data should be copied into this buffer before the call to |
928 | * SDL_ConvertAudio(). Upon successful return, this buffer will contain the |
929 | * converted audio, and `cvt->len_cvt` will be the size of the converted data, |
930 | * in bytes. Any bytes in the buffer past `cvt->len_cvt` are undefined once |
931 | * this function returns. |
932 | * |
933 | * \param cvt an SDL_AudioCVT structure that was previously set up by |
934 | * SDL_BuildAudioCVT(). |
935 | * \returns 0 if the conversion was completed successfully or a negative error |
936 | * code on failure; call SDL_GetError() for more information. |
937 | * |
938 | * \since This function is available since SDL 2.0.0. |
939 | * |
940 | * \sa SDL_BuildAudioCVT |
941 | */ |
942 | extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); |
943 | |
944 | /* SDL_AudioStream is a new audio conversion interface. |
945 | The benefits vs SDL_AudioCVT: |
946 | - it can handle resampling data in chunks without generating |
947 | artifacts, when it doesn't have the complete buffer available. |
948 | - it can handle incoming data in any variable size. |
949 | - You push data as you have it, and pull it when you need it |
950 | */ |
951 | /* this is opaque to the outside world. */ |
952 | struct _SDL_AudioStream; |
953 | typedef struct _SDL_AudioStream SDL_AudioStream; |
954 | |
955 | /** |
956 | * Create a new audio stream. |
957 | * |
958 | * \param src_format The format of the source audio |
959 | * \param src_channels The number of channels of the source audio |
960 | * \param src_rate The sampling rate of the source audio |
961 | * \param dst_format The format of the desired audio output |
962 | * \param dst_channels The number of channels of the desired audio output |
963 | * \param dst_rate The sampling rate of the desired audio output |
964 | * \returns 0 on success, or -1 on error. |
965 | * |
966 | * \since This function is available since SDL 2.0.7. |
967 | * |
968 | * \sa SDL_AudioStreamPut |
969 | * \sa SDL_AudioStreamGet |
970 | * \sa SDL_AudioStreamAvailable |
971 | * \sa SDL_AudioStreamFlush |
972 | * \sa SDL_AudioStreamClear |
973 | * \sa SDL_FreeAudioStream |
974 | */ |
975 | extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format, |
976 | const Uint8 src_channels, |
977 | const int src_rate, |
978 | const SDL_AudioFormat dst_format, |
979 | const Uint8 dst_channels, |
980 | const int dst_rate); |
981 | |
982 | /** |
983 | * Add data to be converted/resampled to the stream. |
984 | * |
985 | * \param stream The stream the audio data is being added to |
986 | * \param buf A pointer to the audio data to add |
987 | * \param len The number of bytes to write to the stream |
988 | * \returns 0 on success, or -1 on error. |
989 | * |
990 | * \since This function is available since SDL 2.0.7. |
991 | * |
992 | * \sa SDL_NewAudioStream |
993 | * \sa SDL_AudioStreamGet |
994 | * \sa SDL_AudioStreamAvailable |
995 | * \sa SDL_AudioStreamFlush |
996 | * \sa SDL_AudioStreamClear |
997 | * \sa SDL_FreeAudioStream |
998 | */ |
999 | extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len); |
1000 | |
1001 | /** |
1002 | * Get converted/resampled data from the stream |
1003 | * |
1004 | * \param stream The stream the audio is being requested from |
1005 | * \param buf A buffer to fill with audio data |
1006 | * \param len The maximum number of bytes to fill |
1007 | * \returns the number of bytes read from the stream, or -1 on error |
1008 | * |
1009 | * \since This function is available since SDL 2.0.7. |
1010 | * |
1011 | * \sa SDL_NewAudioStream |
1012 | * \sa SDL_AudioStreamPut |
1013 | * \sa SDL_AudioStreamAvailable |
1014 | * \sa SDL_AudioStreamFlush |
1015 | * \sa SDL_AudioStreamClear |
1016 | * \sa SDL_FreeAudioStream |
1017 | */ |
1018 | extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len); |
1019 | |
1020 | /** |
1021 | * Get the number of converted/resampled bytes available. |
1022 | * |
1023 | * The stream may be buffering data behind the scenes until it has enough to |
1024 | * resample correctly, so this number might be lower than what you expect, or |
1025 | * even be zero. Add more data or flush the stream if you need the data now. |
1026 | * |
1027 | * \since This function is available since SDL 2.0.7. |
1028 | * |
1029 | * \sa SDL_NewAudioStream |
1030 | * \sa SDL_AudioStreamPut |
1031 | * \sa SDL_AudioStreamGet |
1032 | * \sa SDL_AudioStreamFlush |
1033 | * \sa SDL_AudioStreamClear |
1034 | * \sa SDL_FreeAudioStream |
1035 | */ |
1036 | extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream); |
1037 | |
1038 | /** |
1039 | * Tell the stream that you're done sending data, and anything being buffered |
1040 | * should be converted/resampled and made available immediately. |
1041 | * |
1042 | * It is legal to add more data to a stream after flushing, but there will be |
1043 | * audio gaps in the output. Generally this is intended to signal the end of |
1044 | * input, so the complete output becomes available. |
1045 | * |
1046 | * \since This function is available since SDL 2.0.7. |
1047 | * |
1048 | * \sa SDL_NewAudioStream |
1049 | * \sa SDL_AudioStreamPut |
1050 | * \sa SDL_AudioStreamGet |
1051 | * \sa SDL_AudioStreamAvailable |
1052 | * \sa SDL_AudioStreamClear |
1053 | * \sa SDL_FreeAudioStream |
1054 | */ |
1055 | extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream); |
1056 | |
1057 | /** |
1058 | * Clear any pending data in the stream without converting it |
1059 | * |
1060 | * \since This function is available since SDL 2.0.7. |
1061 | * |
1062 | * \sa SDL_NewAudioStream |
1063 | * \sa SDL_AudioStreamPut |
1064 | * \sa SDL_AudioStreamGet |
1065 | * \sa SDL_AudioStreamAvailable |
1066 | * \sa SDL_AudioStreamFlush |
1067 | * \sa SDL_FreeAudioStream |
1068 | */ |
1069 | extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream); |
1070 | |
1071 | /** |
1072 | * Free an audio stream |
1073 | * |
1074 | * \since This function is available since SDL 2.0.7. |
1075 | * |
1076 | * \sa SDL_NewAudioStream |
1077 | * \sa SDL_AudioStreamPut |
1078 | * \sa SDL_AudioStreamGet |
1079 | * \sa SDL_AudioStreamAvailable |
1080 | * \sa SDL_AudioStreamFlush |
1081 | * \sa SDL_AudioStreamClear |
1082 | */ |
1083 | extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream); |
1084 | |
1085 | #define SDL_MIX_MAXVOLUME 128 |
1086 | |
1087 | /** |
1088 | * This function is a legacy means of mixing audio. |
1089 | * |
1090 | * This function is equivalent to calling... |
1091 | * |
1092 | * ```c |
1093 | * SDL_MixAudioFormat(dst, src, format, len, volume); |
1094 | * ``` |
1095 | * |
1096 | * ...where `format` is the obtained format of the audio device from the |
1097 | * legacy SDL_OpenAudio() function. |
1098 | * |
1099 | * \param dst the destination for the mixed audio |
1100 | * \param src the source audio buffer to be mixed |
1101 | * \param len the length of the audio buffer in bytes |
1102 | * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME |
1103 | * for full audio volume |
1104 | * |
1105 | * \since This function is available since SDL 2.0.0. |
1106 | * |
1107 | * \sa SDL_MixAudioFormat |
1108 | */ |
1109 | extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, |
1110 | Uint32 len, int volume); |
1111 | |
1112 | /** |
1113 | * Mix audio data in a specified format. |
1114 | * |
1115 | * This takes an audio buffer `src` of `len` bytes of `format` data and mixes |
1116 | * it into `dst`, performing addition, volume adjustment, and overflow |
1117 | * clipping. The buffer pointed to by `dst` must also be `len` bytes of |
1118 | * `format` data. |
1119 | * |
1120 | * This is provided for convenience -- you can mix your own audio data. |
1121 | * |
1122 | * Do not use this function for mixing together more than two streams of |
1123 | * sample data. The output from repeated application of this function may be |
1124 | * distorted by clipping, because there is no accumulator with greater range |
1125 | * than the input (not to mention this being an inefficient way of doing it). |
1126 | * |
1127 | * It is a common misconception that this function is required to write audio |
1128 | * data to an output stream in an audio callback. While you can do that, |
1129 | * SDL_MixAudioFormat() is really only needed when you're mixing a single |
1130 | * audio stream with a volume adjustment. |
1131 | * |
1132 | * \param dst the destination for the mixed audio |
1133 | * \param src the source audio buffer to be mixed |
1134 | * \param format the SDL_AudioFormat structure representing the desired audio |
1135 | * format |
1136 | * \param len the length of the audio buffer in bytes |
1137 | * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME |
1138 | * for full audio volume |
1139 | * |
1140 | * \since This function is available since SDL 2.0.0. |
1141 | */ |
1142 | extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, |
1143 | const Uint8 * src, |
1144 | SDL_AudioFormat format, |
1145 | Uint32 len, int volume); |
1146 | |
1147 | /** |
1148 | * Queue more audio on non-callback devices. |
1149 | * |
1150 | * If you are looking to retrieve queued audio from a non-callback capture |
1151 | * device, you want SDL_DequeueAudio() instead. SDL_QueueAudio() will return |
1152 | * -1 to signify an error if you use it with capture devices. |
1153 | * |
1154 | * SDL offers two ways to feed audio to the device: you can either supply a |
1155 | * callback that SDL triggers with some frequency to obtain more audio (pull |
1156 | * method), or you can supply no callback, and then SDL will expect you to |
1157 | * supply data at regular intervals (push method) with this function. |
1158 | * |
1159 | * There are no limits on the amount of data you can queue, short of |
1160 | * exhaustion of address space. Queued data will drain to the device as |
1161 | * necessary without further intervention from you. If the device needs audio |
1162 | * but there is not enough queued, it will play silence to make up the |
1163 | * difference. This means you will have skips in your audio playback if you |
1164 | * aren't routinely queueing sufficient data. |
1165 | * |
1166 | * This function copies the supplied data, so you are safe to free it when the |
1167 | * function returns. This function is thread-safe, but queueing to the same |
1168 | * device from two threads at once does not promise which buffer will be |
1169 | * queued first. |
1170 | * |
1171 | * You may not queue audio on a device that is using an application-supplied |
1172 | * callback; doing so returns an error. You have to use the audio callback or |
1173 | * queue audio with this function, but not both. |
1174 | * |
1175 | * You should not call SDL_LockAudio() on the device before queueing; SDL |
1176 | * handles locking internally for this function. |
1177 | * |
1178 | * Note that SDL2 does not support planar audio. You will need to resample |
1179 | * from planar audio formats into a non-planar one (see SDL_AudioFormat) |
1180 | * before queuing audio. |
1181 | * |
1182 | * \param dev the device ID to which we will queue audio |
1183 | * \param data the data to queue to the device for later playback |
1184 | * \param len the number of bytes (not samples!) to which `data` points |
1185 | * \returns 0 on success or a negative error code on failure; call |
1186 | * SDL_GetError() for more information. |
1187 | * |
1188 | * \since This function is available since SDL 2.0.4. |
1189 | * |
1190 | * \sa SDL_ClearQueuedAudio |
1191 | * \sa SDL_GetQueuedAudioSize |
1192 | */ |
1193 | extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len); |
1194 | |
1195 | /** |
1196 | * Dequeue more audio on non-callback devices. |
1197 | * |
1198 | * If you are looking to queue audio for output on a non-callback playback |
1199 | * device, you want SDL_QueueAudio() instead. SDL_DequeueAudio() will always |
1200 | * return 0 if you use it with playback devices. |
1201 | * |
1202 | * SDL offers two ways to retrieve audio from a capture device: you can either |
1203 | * supply a callback that SDL triggers with some frequency as the device |
1204 | * records more audio data, (push method), or you can supply no callback, and |
1205 | * then SDL will expect you to retrieve data at regular intervals (pull |
1206 | * method) with this function. |
1207 | * |
1208 | * There are no limits on the amount of data you can queue, short of |
1209 | * exhaustion of address space. Data from the device will keep queuing as |
1210 | * necessary without further intervention from you. This means you will |
1211 | * eventually run out of memory if you aren't routinely dequeueing data. |
1212 | * |
1213 | * Capture devices will not queue data when paused; if you are expecting to |
1214 | * not need captured audio for some length of time, use SDL_PauseAudioDevice() |
1215 | * to stop the capture device from queueing more data. This can be useful |
1216 | * during, say, level loading times. When unpaused, capture devices will start |
1217 | * queueing data from that point, having flushed any capturable data available |
1218 | * while paused. |
1219 | * |
1220 | * This function is thread-safe, but dequeueing from the same device from two |
1221 | * threads at once does not promise which thread will dequeue data first. |
1222 | * |
1223 | * You may not dequeue audio from a device that is using an |
1224 | * application-supplied callback; doing so returns an error. You have to use |
1225 | * the audio callback, or dequeue audio with this function, but not both. |
1226 | * |
1227 | * You should not call SDL_LockAudio() on the device before dequeueing; SDL |
1228 | * handles locking internally for this function. |
1229 | * |
1230 | * \param dev the device ID from which we will dequeue audio |
1231 | * \param data a pointer into where audio data should be copied |
1232 | * \param len the number of bytes (not samples!) to which (data) points |
1233 | * \returns the number of bytes dequeued, which could be less than requested; |
1234 | * call SDL_GetError() for more information. |
1235 | * |
1236 | * \since This function is available since SDL 2.0.5. |
1237 | * |
1238 | * \sa SDL_ClearQueuedAudio |
1239 | * \sa SDL_GetQueuedAudioSize |
1240 | */ |
1241 | extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len); |
1242 | |
1243 | /** |
1244 | * Get the number of bytes of still-queued audio. |
1245 | * |
1246 | * For playback devices: this is the number of bytes that have been queued for |
1247 | * playback with SDL_QueueAudio(), but have not yet been sent to the hardware. |
1248 | * |
1249 | * Once we've sent it to the hardware, this function can not decide the exact |
1250 | * byte boundary of what has been played. It's possible that we just gave the |
1251 | * hardware several kilobytes right before you called this function, but it |
1252 | * hasn't played any of it yet, or maybe half of it, etc. |
1253 | * |
1254 | * For capture devices, this is the number of bytes that have been captured by |
1255 | * the device and are waiting for you to dequeue. This number may grow at any |
1256 | * time, so this only informs of the lower-bound of available data. |
1257 | * |
1258 | * You may not queue or dequeue audio on a device that is using an |
1259 | * application-supplied callback; calling this function on such a device |
1260 | * always returns 0. You have to use the audio callback or queue audio, but |
1261 | * not both. |
1262 | * |
1263 | * You should not call SDL_LockAudio() on the device before querying; SDL |
1264 | * handles locking internally for this function. |
1265 | * |
1266 | * \param dev the device ID of which we will query queued audio size |
1267 | * \returns the number of bytes (not samples!) of queued audio. |
1268 | * |
1269 | * \since This function is available since SDL 2.0.4. |
1270 | * |
1271 | * \sa SDL_ClearQueuedAudio |
1272 | * \sa SDL_QueueAudio |
1273 | * \sa SDL_DequeueAudio |
1274 | */ |
1275 | extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev); |
1276 | |
1277 | /** |
1278 | * Drop any queued audio data waiting to be sent to the hardware. |
1279 | * |
1280 | * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For |
1281 | * output devices, the hardware will start playing silence if more audio isn't |
1282 | * queued. For capture devices, the hardware will start filling the empty |
1283 | * queue with new data if the capture device isn't paused. |
1284 | * |
1285 | * This will not prevent playback of queued audio that's already been sent to |
1286 | * the hardware, as we can not undo that, so expect there to be some fraction |
1287 | * of a second of audio that might still be heard. This can be useful if you |
1288 | * want to, say, drop any pending music or any unprocessed microphone input |
1289 | * during a level change in your game. |
1290 | * |
1291 | * You may not queue or dequeue audio on a device that is using an |
1292 | * application-supplied callback; calling this function on such a device |
1293 | * always returns 0. You have to use the audio callback or queue audio, but |
1294 | * not both. |
1295 | * |
1296 | * You should not call SDL_LockAudio() on the device before clearing the |
1297 | * queue; SDL handles locking internally for this function. |
1298 | * |
1299 | * This function always succeeds and thus returns void. |
1300 | * |
1301 | * \param dev the device ID of which to clear the audio queue |
1302 | * |
1303 | * \since This function is available since SDL 2.0.4. |
1304 | * |
1305 | * \sa SDL_GetQueuedAudioSize |
1306 | * \sa SDL_QueueAudio |
1307 | * \sa SDL_DequeueAudio |
1308 | */ |
1309 | extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev); |
1310 | |
1311 | |
1312 | /** |
1313 | * \name Audio lock functions |
1314 | * |
1315 | * The lock manipulated by these functions protects the callback function. |
1316 | * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that |
1317 | * the callback function is not running. Do not call these from the callback |
1318 | * function or you will cause deadlock. |
1319 | */ |
1320 | /* @{ */ |
1321 | |
1322 | /** |
1323 | * This function is a legacy means of locking the audio device. |
1324 | * |
1325 | * New programs might want to use SDL_LockAudioDevice() instead. This function |
1326 | * is equivalent to calling... |
1327 | * |
1328 | * ```c |
1329 | * SDL_LockAudioDevice(1); |
1330 | * ``` |
1331 | * |
1332 | * ...and is only useful if you used the legacy SDL_OpenAudio() function. |
1333 | * |
1334 | * \since This function is available since SDL 2.0.0. |
1335 | * |
1336 | * \sa SDL_LockAudioDevice |
1337 | * \sa SDL_UnlockAudio |
1338 | * \sa SDL_UnlockAudioDevice |
1339 | */ |
1340 | extern DECLSPEC void SDLCALL SDL_LockAudio(void); |
1341 | |
1342 | /** |
1343 | * Use this function to lock out the audio callback function for a specified |
1344 | * device. |
1345 | * |
1346 | * The lock manipulated by these functions protects the audio callback |
1347 | * function specified in SDL_OpenAudioDevice(). During a |
1348 | * SDL_LockAudioDevice()/SDL_UnlockAudioDevice() pair, you can be guaranteed |
1349 | * that the callback function for that device is not running, even if the |
1350 | * device is not paused. While a device is locked, any other unpaused, |
1351 | * unlocked devices may still run their callbacks. |
1352 | * |
1353 | * Calling this function from inside your audio callback is unnecessary. SDL |
1354 | * obtains this lock before calling your function, and releases it when the |
1355 | * function returns. |
1356 | * |
1357 | * You should not hold the lock longer than absolutely necessary. If you hold |
1358 | * it too long, you'll experience dropouts in your audio playback. Ideally, |
1359 | * your application locks the device, sets a few variables and unlocks again. |
1360 | * Do not do heavy work while holding the lock for a device. |
1361 | * |
1362 | * It is safe to lock the audio device multiple times, as long as you unlock |
1363 | * it an equivalent number of times. The callback will not run until the |
1364 | * device has been unlocked completely in this way. If your application fails |
1365 | * to unlock the device appropriately, your callback will never run, you might |
1366 | * hear repeating bursts of audio, and SDL_CloseAudioDevice() will probably |
1367 | * deadlock. |
1368 | * |
1369 | * Internally, the audio device lock is a mutex; if you lock from two threads |
1370 | * at once, not only will you block the audio callback, you'll block the other |
1371 | * thread. |
1372 | * |
1373 | * \param dev the ID of the device to be locked |
1374 | * |
1375 | * \since This function is available since SDL 2.0.0. |
1376 | * |
1377 | * \sa SDL_UnlockAudioDevice |
1378 | */ |
1379 | extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); |
1380 | |
1381 | /** |
1382 | * This function is a legacy means of unlocking the audio device. |
1383 | * |
1384 | * New programs might want to use SDL_UnlockAudioDevice() instead. This |
1385 | * function is equivalent to calling... |
1386 | * |
1387 | * ```c |
1388 | * SDL_UnlockAudioDevice(1); |
1389 | * ``` |
1390 | * |
1391 | * ...and is only useful if you used the legacy SDL_OpenAudio() function. |
1392 | * |
1393 | * \since This function is available since SDL 2.0.0. |
1394 | * |
1395 | * \sa SDL_LockAudio |
1396 | * \sa SDL_UnlockAudioDevice |
1397 | */ |
1398 | extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); |
1399 | |
1400 | /** |
1401 | * Use this function to unlock the audio callback function for a specified |
1402 | * device. |
1403 | * |
1404 | * This function should be paired with a previous SDL_LockAudioDevice() call. |
1405 | * |
1406 | * \param dev the ID of the device to be unlocked |
1407 | * |
1408 | * \since This function is available since SDL 2.0.0. |
1409 | * |
1410 | * \sa SDL_LockAudioDevice |
1411 | */ |
1412 | extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); |
1413 | /* @} *//* Audio lock functions */ |
1414 | |
1415 | /** |
1416 | * This function is a legacy means of closing the audio device. |
1417 | * |
1418 | * This function is equivalent to calling... |
1419 | * |
1420 | * ```c |
1421 | * SDL_CloseAudioDevice(1); |
1422 | * ``` |
1423 | * |
1424 | * ...and is only useful if you used the legacy SDL_OpenAudio() function. |
1425 | * |
1426 | * \since This function is available since SDL 2.0.0. |
1427 | * |
1428 | * \sa SDL_OpenAudio |
1429 | */ |
1430 | extern DECLSPEC void SDLCALL SDL_CloseAudio(void); |
1431 | |
1432 | /** |
1433 | * Use this function to shut down audio processing and close the audio device. |
1434 | * |
1435 | * The application should close open audio devices once they are no longer |
1436 | * needed. Calling this function will wait until the device's audio callback |
1437 | * is not running, release the audio hardware and then clean up internal |
1438 | * state. No further audio will play from this device once this function |
1439 | * returns. |
1440 | * |
1441 | * This function may block briefly while pending audio data is played by the |
1442 | * hardware, so that applications don't drop the last buffer of data they |
1443 | * supplied. |
1444 | * |
1445 | * The device ID is invalid as soon as the device is closed, and is eligible |
1446 | * for reuse in a new SDL_OpenAudioDevice() call immediately. |
1447 | * |
1448 | * \param dev an audio device previously opened with SDL_OpenAudioDevice() |
1449 | * |
1450 | * \since This function is available since SDL 2.0.0. |
1451 | * |
1452 | * \sa SDL_OpenAudioDevice |
1453 | */ |
1454 | extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); |
1455 | |
1456 | /* Ends C function definitions when using C++ */ |
1457 | #ifdef __cplusplus |
1458 | } |
1459 | #endif |
1460 | #include "close_code.h" |
1461 | |
1462 | #endif /* SDL_audio_h_ */ |
1463 | |
1464 | /* vi: set ts=4 sw=4 expandtab: */ |
1465 | |