| 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 | |