1 /* 2 Simple DirectMedia Layer 3 Copyright (C) 1997-2016 Sam Lantinga <slouken@libsdl.org> 4 5 This software is provided 'as-is', without any express or implied 6 warranty. In no event will the authors be held liable for any damages 7 arising from the use of this software. 8 9 Permission is granted to anyone to use this software for any purpose, 10 including commercial applications, and to alter it and redistribute it 11 freely, subject to the following restrictions: 12 13 1. The origin of this software must not be misrepresented; you must not 14 claim that you wrote the original software. If you use this software 15 in a product, an acknowledgment in the product documentation would be 16 appreciated but is not required. 17 2. Altered source versions must be plainly marked as such, and must not be 18 misrepresented as being the original software. 19 3. This notice may not be removed or altered from any source distribution. 20 */ 21 22 /** 23 * \file SDL_audio.h 24 * 25 * Access to the raw audio mixing buffer for the SDL library. 26 */ 27 28 #ifndef _SDL_audio_h 29 #define _SDL_audio_h 30 31 #include "SDL_stdinc.h" 32 #include "SDL_error.h" 33 #include "SDL_endian.h" 34 #include "SDL_mutex.h" 35 #include "SDL_thread.h" 36 #include "SDL_rwops.h" 37 38 #include "begin_code.h" 39 /* Set up for C function definitions, even when using C++ */ 40 #ifdef __cplusplus 41 extern "C" { 42 #endif 43 44 /** 45 * \brief Audio format flags. 46 * 47 * These are what the 16 bits in SDL_AudioFormat currently mean... 48 * (Unspecified bits are always zero). 49 * 50 * \verbatim 51 ++-----------------------sample is signed if set 52 || 53 || ++-----------sample is bigendian if set 54 || || 55 || || ++---sample is float if set 56 || || || 57 || || || +---sample bit size---+ 58 || || || | | 59 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 60 \endverbatim 61 * 62 * There are macros in SDL 2.0 and later to query these bits. 63 */ 64 typedef Uint16 SDL_AudioFormat; 65 66 /** 67 * \name Audio flags 68 */ 69 /* @{ */ 70 71 #define SDL_AUDIO_MASK_BITSIZE (0xFF) 72 #define SDL_AUDIO_MASK_DATATYPE (1<<8) 73 #define SDL_AUDIO_MASK_ENDIAN (1<<12) 74 #define SDL_AUDIO_MASK_SIGNED (1<<15) 75 #define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) 76 #define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) 77 #define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) 78 #define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) 79 #define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) 80 #define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) 81 #define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) 82 83 /** 84 * \name Audio format flags 85 * 86 * Defaults to LSB byte order. 87 */ 88 /* @{ */ 89 #define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ 90 #define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ 91 #define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ 92 #define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ 93 #define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ 94 #define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ 95 #define AUDIO_U16 AUDIO_U16LSB 96 #define AUDIO_S16 AUDIO_S16LSB 97 /* @} */ 98 99 /** 100 * \name int32 support 101 */ 102 /* @{ */ 103 #define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ 104 #define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ 105 #define AUDIO_S32 AUDIO_S32LSB 106 /* @} */ 107 108 /** 109 * \name float32 support 110 */ 111 /* @{ */ 112 #define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ 113 #define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ 114 #define AUDIO_F32 AUDIO_F32LSB 115 /* @} */ 116 117 /** 118 * \name Native audio byte ordering 119 */ 120 /* @{ */ 121 #if SDL_BYTEORDER == SDL_LIL_ENDIAN 122 #define AUDIO_U16SYS AUDIO_U16LSB 123 #define AUDIO_S16SYS AUDIO_S16LSB 124 #define AUDIO_S32SYS AUDIO_S32LSB 125 #define AUDIO_F32SYS AUDIO_F32LSB 126 #else 127 #define AUDIO_U16SYS AUDIO_U16MSB 128 #define AUDIO_S16SYS AUDIO_S16MSB 129 #define AUDIO_S32SYS AUDIO_S32MSB 130 #define AUDIO_F32SYS AUDIO_F32MSB 131 #endif 132 /* @} */ 133 134 /** 135 * \name Allow change flags 136 * 137 * Which audio format changes are allowed when opening a device. 138 */ 139 /* @{ */ 140 #define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 141 #define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 142 #define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 143 #define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE) 144 /* @} */ 145 146 /* @} *//* Audio flags */ 147 148 /** 149 * This function is called when the audio device needs more data. 150 * 151 * \param userdata An application-specific parameter saved in 152 * the SDL_AudioSpec structure 153 * \param stream A pointer to the audio data buffer. 154 * \param len The length of that buffer in bytes. 155 * 156 * Once the callback returns, the buffer will no longer be valid. 157 * Stereo samples are stored in a LRLRLR ordering. 158 * 159 * You can choose to avoid callbacks and use SDL_QueueAudio() instead, if 160 * you like. Just open your audio device with a NULL callback. 161 */ 162 typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, 163 int len); 164 165 /** 166 * The calculated values in this structure are calculated by SDL_OpenAudio(). 167 */ 168 typedef struct SDL_AudioSpec 169 { 170 int freq; /**< DSP frequency -- samples per second */ 171 SDL_AudioFormat format; /**< Audio data format */ 172 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ 173 Uint8 silence; /**< Audio buffer silence value (calculated) */ 174 Uint16 samples; /**< Audio buffer size in samples (power of 2) */ 175 Uint16 padding; /**< Necessary for some compile environments */ 176 Uint32 size; /**< Audio buffer size in bytes (calculated) */ 177 SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */ 178 void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */ 179 } SDL_AudioSpec; 180 181 182 struct SDL_AudioCVT; 183 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, 184 SDL_AudioFormat format); 185 186 /** 187 * A structure to hold a set of audio conversion filters and buffers. 188 */ 189 #ifdef __GNUC__ 190 /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't 191 pad it out to 88 bytes to guarantee ABI compatibility between compilers. 192 vvv 193 The next time we rev the ABI, make sure to size the ints and add padding. 194 */ 195 #define SDL_AUDIOCVT_PACKED __attribute__((packed)) 196 #else 197 #define SDL_AUDIOCVT_PACKED 198 #endif 199 /* */ 200 typedef struct SDL_AudioCVT 201 { 202 int needed; /**< Set to 1 if conversion possible */ 203 SDL_AudioFormat src_format; /**< Source audio format */ 204 SDL_AudioFormat dst_format; /**< Target audio format */ 205 double rate_incr; /**< Rate conversion increment */ 206 Uint8 *buf; /**< Buffer to hold entire audio data */ 207 int len; /**< Length of original audio buffer */ 208 int len_cvt; /**< Length of converted audio buffer */ 209 int len_mult; /**< buffer must be len*len_mult big */ 210 double len_ratio; /**< Given len, final size is len*len_ratio */ 211 SDL_AudioFilter filters[10]; /**< Filter list */ 212 int filter_index; /**< Current audio conversion function */ 213 } SDL_AUDIOCVT_PACKED SDL_AudioCVT; 214 215 216 /* Function prototypes */ 217 218 /** 219 * \name Driver discovery functions 220 * 221 * These functions return the list of built in audio drivers, in the 222 * order that they are normally initialized by default. 223 */ 224 /* @{ */ 225 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); 226 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); 227 /* @} */ 228 229 /** 230 * \name Initialization and cleanup 231 * 232 * \internal These functions are used internally, and should not be used unless 233 * you have a specific need to specify the audio driver you want to 234 * use. You should normally use SDL_Init() or SDL_InitSubSystem(). 235 */ 236 /* @{ */ 237 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); 238 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); 239 /* @} */ 240 241 /** 242 * This function returns the name of the current audio driver, or NULL 243 * if no driver has been initialized. 244 */ 245 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); 246 247 /** 248 * This function opens the audio device with the desired parameters, and 249 * returns 0 if successful, placing the actual hardware parameters in the 250 * structure pointed to by \c obtained. If \c obtained is NULL, the audio 251 * data passed to the callback function will be guaranteed to be in the 252 * requested format, and will be automatically converted to the hardware 253 * audio format if necessary. This function returns -1 if it failed 254 * to open the audio device, or couldn't set up the audio thread. 255 * 256 * When filling in the desired audio spec structure, 257 * - \c desired->freq should be the desired audio frequency in samples-per- 258 * second. 259 * - \c desired->format should be the desired audio format. 260 * - \c desired->samples is the desired size of the audio buffer, in 261 * samples. This number should be a power of two, and may be adjusted by 262 * the audio driver to a value more suitable for the hardware. Good values 263 * seem to range between 512 and 8096 inclusive, depending on the 264 * application and CPU speed. Smaller values yield faster response time, 265 * but can lead to underflow if the application is doing heavy processing 266 * and cannot fill the audio buffer in time. A stereo sample consists of 267 * both right and left channels in LR ordering. 268 * Note that the number of samples is directly related to time by the 269 * following formula: \code ms = (samples*1000)/freq \endcode 270 * - \c desired->size is the size in bytes of the audio buffer, and is 271 * calculated by SDL_OpenAudio(). 272 * - \c desired->silence is the value used to set the buffer to silence, 273 * and is calculated by SDL_OpenAudio(). 274 * - \c desired->callback should be set to a function that will be called 275 * when the audio device is ready for more data. It is passed a pointer 276 * to the audio buffer, and the length in bytes of the audio buffer. 277 * This function usually runs in a separate thread, and so you should 278 * protect data structures that it accesses by calling SDL_LockAudio() 279 * and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL 280 * pointer here, and call SDL_QueueAudio() with some frequency, to queue 281 * more audio samples to be played (or for capture devices, call 282 * SDL_DequeueAudio() with some frequency, to obtain audio samples). 283 * - \c desired->userdata is passed as the first parameter to your callback 284 * function. If you passed a NULL callback, this value is ignored. 285 * 286 * The audio device starts out playing silence when it's opened, and should 287 * be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready 288 * for your audio callback function to be called. Since the audio driver 289 * may modify the requested size of the audio buffer, you should allocate 290 * any local mixing buffers after you open the audio device. 291 */ 292 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, 293 SDL_AudioSpec * obtained); 294 295 /** 296 * SDL Audio Device IDs. 297 * 298 * A successful call to SDL_OpenAudio() is always device id 1, and legacy 299 * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls 300 * always returns devices >= 2 on success. The legacy calls are good both 301 * for backwards compatibility and when you don't care about multiple, 302 * specific, or capture devices. 303 */ 304 typedef Uint32 SDL_AudioDeviceID; 305 306 /** 307 * Get the number of available devices exposed by the current driver. 308 * Only valid after a successfully initializing the audio subsystem. 309 * Returns -1 if an explicit list of devices can't be determined; this is 310 * not an error. For example, if SDL is set up to talk to a remote audio 311 * server, it can't list every one available on the Internet, but it will 312 * still allow a specific host to be specified to SDL_OpenAudioDevice(). 313 * 314 * In many common cases, when this function returns a value <= 0, it can still 315 * successfully open the default device (NULL for first argument of 316 * SDL_OpenAudioDevice()). 317 */ 318 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); 319 320 /** 321 * Get the human-readable name of a specific audio device. 322 * Must be a value between 0 and (number of audio devices-1). 323 * Only valid after a successfully initializing the audio subsystem. 324 * The values returned by this function reflect the latest call to 325 * SDL_GetNumAudioDevices(); recall that function to redetect available 326 * hardware. 327 * 328 * The string returned by this function is UTF-8 encoded, read-only, and 329 * managed internally. You are not to free it. If you need to keep the 330 * string for any length of time, you should make your own copy of it, as it 331 * will be invalid next time any of several other SDL functions is called. 332 */ 333 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, 334 int iscapture); 335 336 337 /** 338 * Open a specific audio device. Passing in a device name of NULL requests 339 * the most reasonable default (and is equivalent to calling SDL_OpenAudio()). 340 * 341 * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but 342 * some drivers allow arbitrary and driver-specific strings, such as a 343 * hostname/IP address for a remote audio server, or a filename in the 344 * diskaudio driver. 345 * 346 * \return 0 on error, a valid device ID that is >= 2 on success. 347 * 348 * SDL_OpenAudio(), unlike this function, always acts on device ID 1. 349 */ 350 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char 351 *device, 352 int iscapture, 353 const 354 SDL_AudioSpec * 355 desired, 356 SDL_AudioSpec * 357 obtained, 358 int 359 allowed_changes); 360 361 362 363 /** 364 * \name Audio state 365 * 366 * Get the current audio state. 367 */ 368 /* @{ */ 369 typedef enum 370 { 371 SDL_AUDIO_STOPPED = 0, 372 SDL_AUDIO_PLAYING, 373 SDL_AUDIO_PAUSED 374 } SDL_AudioStatus; 375 extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); 376 377 extern DECLSPEC SDL_AudioStatus SDLCALL 378 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); 379 /* @} *//* Audio State */ 380 381 /** 382 * \name Pause audio functions 383 * 384 * These functions pause and unpause the audio callback processing. 385 * They should be called with a parameter of 0 after opening the audio 386 * device to start playing sound. This is so you can safely initialize 387 * data for your callback function after opening the audio device. 388 * Silence will be written to the audio device during the pause. 389 */ 390 /* @{ */ 391 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); 392 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, 393 int pause_on); 394 /* @} *//* Pause audio functions */ 395 396 /** 397 * This function loads a WAVE from the data source, automatically freeing 398 * that source if \c freesrc is non-zero. For example, to load a WAVE file, 399 * you could do: 400 * \code 401 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); 402 * \endcode 403 * 404 * If this function succeeds, it returns the given SDL_AudioSpec, 405 * filled with the audio data format of the wave data, and sets 406 * \c *audio_buf to a malloc()'d buffer containing the audio data, 407 * and sets \c *audio_len to the length of that audio buffer, in bytes. 408 * You need to free the audio buffer with SDL_FreeWAV() when you are 409 * done with it. 410 * 411 * This function returns NULL and sets the SDL error message if the 412 * wave file cannot be opened, uses an unknown data format, or is 413 * corrupt. Currently raw and MS-ADPCM WAVE files are supported. 414 */ 415 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, 416 int freesrc, 417 SDL_AudioSpec * spec, 418 Uint8 ** audio_buf, 419 Uint32 * audio_len); 420 421 /** 422 * Loads a WAV from a file. 423 * Compatibility convenience function. 424 */ 425 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ 426 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) 427 428 /** 429 * This function frees data previously allocated with SDL_LoadWAV_RW() 430 */ 431 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); 432 433 /** 434 * This function takes a source format and rate and a destination format 435 * and rate, and initializes the \c cvt structure with information needed 436 * by SDL_ConvertAudio() to convert a buffer of audio data from one format 437 * to the other. 438 * 439 * \return -1 if the format conversion is not supported, 0 if there's 440 * no conversion needed, or 1 if the audio filter is set up. 441 */ 442 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, 443 SDL_AudioFormat src_format, 444 Uint8 src_channels, 445 int src_rate, 446 SDL_AudioFormat dst_format, 447 Uint8 dst_channels, 448 int dst_rate); 449 450 /** 451 * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(), 452 * created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of 453 * audio data in the source format, this function will convert it in-place 454 * to the desired format. 455 * 456 * The data conversion may expand the size of the audio data, so the buffer 457 * \c cvt->buf should be allocated after the \c cvt structure is initialized by 458 * SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long. 459 */ 460 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); 461 462 #define SDL_MIX_MAXVOLUME 128 463 /** 464 * This takes two audio buffers of the playing audio format and mixes 465 * them, performing addition, volume adjustment, and overflow clipping. 466 * The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME 467 * for full audio volume. Note this does not change hardware volume. 468 * This is provided for convenience -- you can mix your own audio data. 469 */ 470 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, 471 Uint32 len, int volume); 472 473 /** 474 * This works like SDL_MixAudio(), but you specify the audio format instead of 475 * using the format of audio device 1. Thus it can be used when no audio 476 * device is open at all. 477 */ 478 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, 479 const Uint8 * src, 480 SDL_AudioFormat format, 481 Uint32 len, int volume); 482 483 /** 484 * Queue more audio on non-callback devices. 485 * 486 * (If you are looking to retrieve queued audio from a non-callback capture 487 * device, you want SDL_DequeueAudio() instead. This will return -1 to 488 * signify an error if you use it with capture devices.) 489 * 490 * SDL offers two ways to feed audio to the device: you can either supply a 491 * callback that SDL triggers with some frequency to obtain more audio 492 * (pull method), or you can supply no callback, and then SDL will expect 493 * you to supply data at regular intervals (push method) with this function. 494 * 495 * There are no limits on the amount of data you can queue, short of 496 * exhaustion of address space. Queued data will drain to the device as 497 * necessary without further intervention from you. If the device needs 498 * audio but there is not enough queued, it will play silence to make up 499 * the difference. This means you will have skips in your audio playback 500 * if you aren't routinely queueing sufficient data. 501 * 502 * This function copies the supplied data, so you are safe to free it when 503 * the function returns. This function is thread-safe, but queueing to the 504 * same device from two threads at once does not promise which buffer will 505 * be queued first. 506 * 507 * You may not queue audio on a device that is using an application-supplied 508 * callback; doing so returns an error. You have to use the audio callback 509 * or queue audio with this function, but not both. 510 * 511 * You should not call SDL_LockAudio() on the device before queueing; SDL 512 * handles locking internally for this function. 513 * 514 * \param dev The device ID to which we will queue audio. 515 * \param data The data to queue to the device for later playback. 516 * \param len The number of bytes (not samples!) to which (data) points. 517 * \return zero on success, -1 on error. 518 * 519 * \sa SDL_GetQueuedAudioSize 520 * \sa SDL_ClearQueuedAudio 521 */ 522 extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len); 523 524 /** 525 * Dequeue more audio on non-callback devices. 526 * 527 * (If you are looking to queue audio for output on a non-callback playback 528 * device, you want SDL_QueueAudio() instead. This will always return 0 529 * if you use it with playback devices.) 530 * 531 * SDL offers two ways to retrieve audio from a capture device: you can 532 * either supply a callback that SDL triggers with some frequency as the 533 * device records more audio data, (push method), or you can supply no 534 * callback, and then SDL will expect you to retrieve data at regular 535 * intervals (pull method) with this function. 536 * 537 * There are no limits on the amount of data you can queue, short of 538 * exhaustion of address space. Data from the device will keep queuing as 539 * necessary without further intervention from you. This means you will 540 * eventually run out of memory if you aren't routinely dequeueing data. 541 * 542 * Capture devices will not queue data when paused; if you are expecting 543 * to not need captured audio for some length of time, use 544 * SDL_PauseAudioDevice() to stop the capture device from queueing more 545 * data. This can be useful during, say, level loading times. When 546 * unpaused, capture devices will start queueing data from that point, 547 * having flushed any capturable data available while paused. 548 * 549 * This function is thread-safe, but dequeueing from the same device from 550 * two threads at once does not promise which thread will dequeued data 551 * first. 552 * 553 * You may not dequeue audio from a device that is using an 554 * application-supplied callback; doing so returns an error. You have to use 555 * the audio callback, or dequeue audio with this function, but not both. 556 * 557 * You should not call SDL_LockAudio() on the device before queueing; SDL 558 * handles locking internally for this function. 559 * 560 * \param dev The device ID from which we will dequeue audio. 561 * \param data A pointer into where audio data should be copied. 562 * \param len The number of bytes (not samples!) to which (data) points. 563 * \return number of bytes dequeued, which could be less than requested. 564 * 565 * \sa SDL_GetQueuedAudioSize 566 * \sa SDL_ClearQueuedAudio 567 */ 568 extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len); 569 570 /** 571 * Get the number of bytes of still-queued audio. 572 * 573 * For playback device: 574 * 575 * This is the number of bytes that have been queued for playback with 576 * SDL_QueueAudio(), but have not yet been sent to the hardware. This 577 * number may shrink at any time, so this only informs of pending data. 578 * 579 * Once we've sent it to the hardware, this function can not decide the 580 * exact byte boundary of what has been played. It's possible that we just 581 * gave the hardware several kilobytes right before you called this 582 * function, but it hasn't played any of it yet, or maybe half of it, etc. 583 * 584 * For capture devices: 585 * 586 * This is the number of bytes that have been captured by the device and 587 * are waiting for you to dequeue. This number may grow at any time, so 588 * this only informs of the lower-bound of available data. 589 * 590 * You may not queue audio on a device that is using an application-supplied 591 * callback; calling this function on such a device always returns 0. 592 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use 593 * the audio callback, but not both. 594 * 595 * You should not call SDL_LockAudio() on the device before querying; SDL 596 * handles locking internally for this function. 597 * 598 * \param dev The device ID of which we will query queued audio size. 599 * \return Number of bytes (not samples!) of queued audio. 600 * 601 * \sa SDL_QueueAudio 602 * \sa SDL_ClearQueuedAudio 603 */ 604 extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev); 605 606 /** 607 * Drop any queued audio data. For playback devices, this is any queued data 608 * still waiting to be submitted to the hardware. For capture devices, this 609 * is any data that was queued by the device that hasn't yet been dequeued by 610 * the application. 611 * 612 * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For 613 * playback devices, the hardware will start playing silence if more audio 614 * isn't queued. Unpaused capture devices will start filling the queue again 615 * as soon as they have more data available (which, depending on the state 616 * of the hardware and the thread, could be before this function call 617 * returns!). 618 * 619 * This will not prevent playback of queued audio that's already been sent 620 * to the hardware, as we can not undo that, so expect there to be some 621 * fraction of a second of audio that might still be heard. This can be 622 * useful if you want to, say, drop any pending music during a level change 623 * in your game. 624 * 625 * You may not queue audio on a device that is using an application-supplied 626 * callback; calling this function on such a device is always a no-op. 627 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use 628 * the audio callback, but not both. 629 * 630 * You should not call SDL_LockAudio() on the device before clearing the 631 * queue; SDL handles locking internally for this function. 632 * 633 * This function always succeeds and thus returns void. 634 * 635 * \param dev The device ID of which to clear the audio queue. 636 * 637 * \sa SDL_QueueAudio 638 * \sa SDL_GetQueuedAudioSize 639 */ 640 extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev); 641 642 643 /** 644 * \name Audio lock functions 645 * 646 * The lock manipulated by these functions protects the callback function. 647 * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that 648 * the callback function is not running. Do not call these from the callback 649 * function or you will cause deadlock. 650 */ 651 /* @{ */ 652 extern DECLSPEC void SDLCALL SDL_LockAudio(void); 653 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); 654 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); 655 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); 656 /* @} *//* Audio lock functions */ 657 658 /** 659 * This function shuts down audio processing and closes the audio device. 660 */ 661 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); 662 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); 663 664 /* Ends C function definitions when using C++ */ 665 #ifdef __cplusplus 666 } 667 #endif 668 #include "close_code.h" 669 670 #endif /* _SDL_audio_h */ 671 672 /* vi: set ts=4 sw=4 expandtab: */ 673