1 /* 2 Simple DirectMedia Layer 3 Copyright (C) 1997-2020 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_SAMPLES_CHANGE 0x00000008 144 #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) 145 /* @} */ 146 147 /* @} *//* Audio flags */ 148 149 /** 150 * This function is called when the audio device needs more data. 151 * 152 * \param userdata An application-specific parameter saved in 153 * the SDL_AudioSpec structure 154 * \param stream A pointer to the audio data buffer. 155 * \param len The length of that buffer in bytes. 156 * 157 * Once the callback returns, the buffer will no longer be valid. 158 * Stereo samples are stored in a LRLRLR ordering. 159 * 160 * You can choose to avoid callbacks and use SDL_QueueAudio() instead, if 161 * you like. Just open your audio device with a NULL callback. 162 */ 163 typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, 164 int len); 165 166 /** 167 * The calculated values in this structure are calculated by SDL_OpenAudio(). 168 * 169 * For multi-channel audio, the default SDL channel mapping is: 170 * 2: FL FR (stereo) 171 * 3: FL FR LFE (2.1 surround) 172 * 4: FL FR BL BR (quad) 173 * 5: FL FR FC BL BR (quad + center) 174 * 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR) 175 * 7: FL FR FC LFE BC SL SR (6.1 surround) 176 * 8: FL FR FC LFE BL BR SL SR (7.1 surround) 177 */ 178 typedef struct SDL_AudioSpec 179 { 180 int freq; /**< DSP frequency -- samples per second */ 181 SDL_AudioFormat format; /**< Audio data format */ 182 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ 183 Uint8 silence; /**< Audio buffer silence value (calculated) */ 184 Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */ 185 Uint16 padding; /**< Necessary for some compile environments */ 186 Uint32 size; /**< Audio buffer size in bytes (calculated) */ 187 SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */ 188 void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */ 189 } SDL_AudioSpec; 190 191 192 struct SDL_AudioCVT; 193 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, 194 SDL_AudioFormat format); 195 196 /** 197 * \brief Upper limit of filters in SDL_AudioCVT 198 * 199 * The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is 200 * currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers, 201 * one of which is the terminating NULL pointer. 202 */ 203 #define SDL_AUDIOCVT_MAX_FILTERS 9 204 205 /** 206 * \struct SDL_AudioCVT 207 * \brief A structure to hold a set of audio conversion filters and buffers. 208 * 209 * Note that various parts of the conversion pipeline can take advantage 210 * of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require 211 * you to pass it aligned data, but can possibly run much faster if you 212 * set both its (buf) field to a pointer that is aligned to 16 bytes, and its 213 * (len) field to something that's a multiple of 16, if possible. 214 */ 215 #ifdef __GNUC__ 216 /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't 217 pad it out to 88 bytes to guarantee ABI compatibility between compilers. 218 vvv 219 The next time we rev the ABI, make sure to size the ints and add padding. 220 */ 221 #define SDL_AUDIOCVT_PACKED __attribute__((packed)) 222 #else 223 #define SDL_AUDIOCVT_PACKED 224 #endif 225 /* */ 226 typedef struct SDL_AudioCVT 227 { 228 int needed; /**< Set to 1 if conversion possible */ 229 SDL_AudioFormat src_format; /**< Source audio format */ 230 SDL_AudioFormat dst_format; /**< Target audio format */ 231 double rate_incr; /**< Rate conversion increment */ 232 Uint8 *buf; /**< Buffer to hold entire audio data */ 233 int len; /**< Length of original audio buffer */ 234 int len_cvt; /**< Length of converted audio buffer */ 235 int len_mult; /**< buffer must be len*len_mult big */ 236 double len_ratio; /**< Given len, final size is len*len_ratio */ 237 SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */ 238 int filter_index; /**< Current audio conversion function */ 239 } SDL_AUDIOCVT_PACKED SDL_AudioCVT; 240 241 242 /* Function prototypes */ 243 244 /** 245 * \name Driver discovery functions 246 * 247 * These functions return the list of built in audio drivers, in the 248 * order that they are normally initialized by default. 249 */ 250 /* @{ */ 251 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); 252 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); 253 /* @} */ 254 255 /** 256 * \name Initialization and cleanup 257 * 258 * \internal These functions are used internally, and should not be used unless 259 * you have a specific need to specify the audio driver you want to 260 * use. You should normally use SDL_Init() or SDL_InitSubSystem(). 261 */ 262 /* @{ */ 263 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); 264 extern DECLSPEC void SDLCALL SDL_AudioQuit(void); 265 /* @} */ 266 267 /** 268 * This function returns the name of the current audio driver, or NULL 269 * if no driver has been initialized. 270 */ 271 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); 272 273 /** 274 * This function opens the audio device with the desired parameters, and 275 * returns 0 if successful, placing the actual hardware parameters in the 276 * structure pointed to by \c obtained. If \c obtained is NULL, the audio 277 * data passed to the callback function will be guaranteed to be in the 278 * requested format, and will be automatically converted to the hardware 279 * audio format if necessary. This function returns -1 if it failed 280 * to open the audio device, or couldn't set up the audio thread. 281 * 282 * When filling in the desired audio spec structure, 283 * - \c desired->freq should be the desired audio frequency in samples-per- 284 * second. 285 * - \c desired->format should be the desired audio format. 286 * - \c desired->samples is the desired size of the audio buffer, in 287 * samples. This number should be a power of two, and may be adjusted by 288 * the audio driver to a value more suitable for the hardware. Good values 289 * seem to range between 512 and 8096 inclusive, depending on the 290 * application and CPU speed. Smaller values yield faster response time, 291 * but can lead to underflow if the application is doing heavy processing 292 * and cannot fill the audio buffer in time. A stereo sample consists of 293 * both right and left channels in LR ordering. 294 * Note that the number of samples is directly related to time by the 295 * following formula: \code ms = (samples*1000)/freq \endcode 296 * - \c desired->size is the size in bytes of the audio buffer, and is 297 * calculated by SDL_OpenAudio(). 298 * - \c desired->silence is the value used to set the buffer to silence, 299 * and is calculated by SDL_OpenAudio(). 300 * - \c desired->callback should be set to a function that will be called 301 * when the audio device is ready for more data. It is passed a pointer 302 * to the audio buffer, and the length in bytes of the audio buffer. 303 * This function usually runs in a separate thread, and so you should 304 * protect data structures that it accesses by calling SDL_LockAudio() 305 * and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL 306 * pointer here, and call SDL_QueueAudio() with some frequency, to queue 307 * more audio samples to be played (or for capture devices, call 308 * SDL_DequeueAudio() with some frequency, to obtain audio samples). 309 * - \c desired->userdata is passed as the first parameter to your callback 310 * function. If you passed a NULL callback, this value is ignored. 311 * 312 * The audio device starts out playing silence when it's opened, and should 313 * be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready 314 * for your audio callback function to be called. Since the audio driver 315 * may modify the requested size of the audio buffer, you should allocate 316 * any local mixing buffers after you open the audio device. 317 */ 318 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, 319 SDL_AudioSpec * obtained); 320 321 /** 322 * SDL Audio Device IDs. 323 * 324 * A successful call to SDL_OpenAudio() is always device id 1, and legacy 325 * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls 326 * always returns devices >= 2 on success. The legacy calls are good both 327 * for backwards compatibility and when you don't care about multiple, 328 * specific, or capture devices. 329 */ 330 typedef Uint32 SDL_AudioDeviceID; 331 332 /** 333 * Get the number of available devices exposed by the current driver. 334 * Only valid after a successfully initializing the audio subsystem. 335 * Returns -1 if an explicit list of devices can't be determined; this is 336 * not an error. For example, if SDL is set up to talk to a remote audio 337 * server, it can't list every one available on the Internet, but it will 338 * still allow a specific host to be specified to SDL_OpenAudioDevice(). 339 * 340 * In many common cases, when this function returns a value <= 0, it can still 341 * successfully open the default device (NULL for first argument of 342 * SDL_OpenAudioDevice()). 343 */ 344 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); 345 346 /** 347 * Get the human-readable name of a specific audio device. 348 * Must be a value between 0 and (number of audio devices-1). 349 * Only valid after a successfully initializing the audio subsystem. 350 * The values returned by this function reflect the latest call to 351 * SDL_GetNumAudioDevices(); recall that function to redetect available 352 * hardware. 353 * 354 * The string returned by this function is UTF-8 encoded, read-only, and 355 * managed internally. You are not to free it. If you need to keep the 356 * string for any length of time, you should make your own copy of it, as it 357 * will be invalid next time any of several other SDL functions is called. 358 */ 359 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, 360 int iscapture); 361 362 363 /** 364 * Open a specific audio device. Passing in a device name of NULL requests 365 * the most reasonable default (and is equivalent to calling SDL_OpenAudio()). 366 * 367 * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but 368 * some drivers allow arbitrary and driver-specific strings, such as a 369 * hostname/IP address for a remote audio server, or a filename in the 370 * diskaudio driver. 371 * 372 * \return 0 on error, a valid device ID that is >= 2 on success. 373 * 374 * SDL_OpenAudio(), unlike this function, always acts on device ID 1. 375 */ 376 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char 377 *device, 378 int iscapture, 379 const 380 SDL_AudioSpec * 381 desired, 382 SDL_AudioSpec * 383 obtained, 384 int 385 allowed_changes); 386 387 388 389 /** 390 * \name Audio state 391 * 392 * Get the current audio state. 393 */ 394 /* @{ */ 395 typedef enum 396 { 397 SDL_AUDIO_STOPPED = 0, 398 SDL_AUDIO_PLAYING, 399 SDL_AUDIO_PAUSED 400 } SDL_AudioStatus; 401 extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); 402 403 extern DECLSPEC SDL_AudioStatus SDLCALL 404 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); 405 /* @} *//* Audio State */ 406 407 /** 408 * \name Pause audio functions 409 * 410 * These functions pause and unpause the audio callback processing. 411 * They should be called with a parameter of 0 after opening the audio 412 * device to start playing sound. This is so you can safely initialize 413 * data for your callback function after opening the audio device. 414 * Silence will be written to the audio device during the pause. 415 */ 416 /* @{ */ 417 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); 418 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, 419 int pause_on); 420 /* @} *//* Pause audio functions */ 421 422 /** 423 * \brief Load the audio data of a WAVE file into memory 424 * 425 * Loading a WAVE file requires \c src, \c spec, \c audio_buf and \c audio_len 426 * to be valid pointers. The entire data portion of the file is then loaded 427 * into memory and decoded if necessary. 428 * 429 * If \c freesrc is non-zero, the data source gets automatically closed and 430 * freed before the function returns. 431 * 432 * Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits), 433 * IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and 434 * µ-law (8 bits). Other formats are currently unsupported and cause an error. 435 * 436 * If this function succeeds, the pointer returned by it is equal to \c spec 437 * and the pointer to the audio data allocated by the function is written to 438 * \c audio_buf and its length in bytes to \c audio_len. The \ref SDL_AudioSpec 439 * members \c freq, \c channels, and \c format are set to the values of the 440 * audio data in the buffer. The \c samples member is set to a sane default and 441 * all others are set to zero. 442 * 443 * It's necessary to use SDL_FreeWAV() to free the audio data returned in 444 * \c audio_buf when it is no longer used. 445 * 446 * Because of the underspecification of the Waveform format, there are many 447 * problematic files in the wild that cause issues with strict decoders. To 448 * provide compatibility with these files, this decoder is lenient in regards 449 * to the truncation of the file, the fact chunk, and the size of the RIFF 450 * chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION, 451 * and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the 452 * loading process. 453 * 454 * Any file that is invalid (due to truncation, corruption, or wrong values in 455 * the headers), too big, or unsupported causes an error. Additionally, any 456 * critical I/O error from the data source will terminate the loading process 457 * with an error. The function returns NULL on error and in all cases (with the 458 * exception of \c src being NULL), an appropriate error message will be set. 459 * 460 * It is required that the data source supports seeking. 461 * 462 * Example: 463 * \code 464 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); 465 * \endcode 466 * 467 * \param src The data source with the WAVE data 468 * \param freesrc A integer value that makes the function close the data source if non-zero 469 * \param spec A pointer filled with the audio format of the audio data 470 * \param audio_buf A pointer filled with the audio data allocated by the function 471 * \param audio_len A pointer filled with the length of the audio data buffer in bytes 472 * \return NULL on error, or non-NULL on success. 473 */ 474 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, 475 int freesrc, 476 SDL_AudioSpec * spec, 477 Uint8 ** audio_buf, 478 Uint32 * audio_len); 479 480 /** 481 * Loads a WAV from a file. 482 * Compatibility convenience function. 483 */ 484 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ 485 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) 486 487 /** 488 * This function frees data previously allocated with SDL_LoadWAV_RW() 489 */ 490 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); 491 492 /** 493 * This function takes a source format and rate and a destination format 494 * and rate, and initializes the \c cvt structure with information needed 495 * by SDL_ConvertAudio() to convert a buffer of audio data from one format 496 * to the other. An unsupported format causes an error and -1 will be returned. 497 * 498 * \return 0 if no conversion is needed, 1 if the audio filter is set up, 499 * or -1 on error. 500 */ 501 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, 502 SDL_AudioFormat src_format, 503 Uint8 src_channels, 504 int src_rate, 505 SDL_AudioFormat dst_format, 506 Uint8 dst_channels, 507 int dst_rate); 508 509 /** 510 * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(), 511 * created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of 512 * audio data in the source format, this function will convert it in-place 513 * to the desired format. 514 * 515 * The data conversion may expand the size of the audio data, so the buffer 516 * \c cvt->buf should be allocated after the \c cvt structure is initialized by 517 * SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long. 518 * 519 * \return 0 on success or -1 if \c cvt->buf is NULL. 520 */ 521 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); 522 523 /* SDL_AudioStream is a new audio conversion interface. 524 The benefits vs SDL_AudioCVT: 525 - it can handle resampling data in chunks without generating 526 artifacts, when it doesn't have the complete buffer available. 527 - it can handle incoming data in any variable size. 528 - You push data as you have it, and pull it when you need it 529 */ 530 /* this is opaque to the outside world. */ 531 struct _SDL_AudioStream; 532 typedef struct _SDL_AudioStream SDL_AudioStream; 533 534 /** 535 * Create a new audio stream 536 * 537 * \param src_format The format of the source audio 538 * \param src_channels The number of channels of the source audio 539 * \param src_rate The sampling rate of the source audio 540 * \param dst_format The format of the desired audio output 541 * \param dst_channels The number of channels of the desired audio output 542 * \param dst_rate The sampling rate of the desired audio output 543 * \return 0 on success, or -1 on error. 544 * 545 * \sa SDL_AudioStreamPut 546 * \sa SDL_AudioStreamGet 547 * \sa SDL_AudioStreamAvailable 548 * \sa SDL_AudioStreamFlush 549 * \sa SDL_AudioStreamClear 550 * \sa SDL_FreeAudioStream 551 */ 552 extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format, 553 const Uint8 src_channels, 554 const int src_rate, 555 const SDL_AudioFormat dst_format, 556 const Uint8 dst_channels, 557 const int dst_rate); 558 559 /** 560 * Add data to be converted/resampled to the stream 561 * 562 * \param stream The stream the audio data is being added to 563 * \param buf A pointer to the audio data to add 564 * \param len The number of bytes to write to the stream 565 * \return 0 on success, or -1 on error. 566 * 567 * \sa SDL_NewAudioStream 568 * \sa SDL_AudioStreamGet 569 * \sa SDL_AudioStreamAvailable 570 * \sa SDL_AudioStreamFlush 571 * \sa SDL_AudioStreamClear 572 * \sa SDL_FreeAudioStream 573 */ 574 extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len); 575 576 /** 577 * Get converted/resampled data from the stream 578 * 579 * \param stream The stream the audio is being requested from 580 * \param buf A buffer to fill with audio data 581 * \param len The maximum number of bytes to fill 582 * \return The number of bytes read from the stream, or -1 on error 583 * 584 * \sa SDL_NewAudioStream 585 * \sa SDL_AudioStreamPut 586 * \sa SDL_AudioStreamAvailable 587 * \sa SDL_AudioStreamFlush 588 * \sa SDL_AudioStreamClear 589 * \sa SDL_FreeAudioStream 590 */ 591 extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len); 592 593 /** 594 * Get the number of converted/resampled bytes available. The stream may be 595 * buffering data behind the scenes until it has enough to resample 596 * correctly, so this number might be lower than what you expect, or even 597 * be zero. Add more data or flush the stream if you need the data now. 598 * 599 * \sa SDL_NewAudioStream 600 * \sa SDL_AudioStreamPut 601 * \sa SDL_AudioStreamGet 602 * \sa SDL_AudioStreamFlush 603 * \sa SDL_AudioStreamClear 604 * \sa SDL_FreeAudioStream 605 */ 606 extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream); 607 608 /** 609 * Tell the stream that you're done sending data, and anything being buffered 610 * should be converted/resampled and made available immediately. 611 * 612 * It is legal to add more data to a stream after flushing, but there will 613 * be audio gaps in the output. Generally this is intended to signal the 614 * end of input, so the complete output becomes available. 615 * 616 * \sa SDL_NewAudioStream 617 * \sa SDL_AudioStreamPut 618 * \sa SDL_AudioStreamGet 619 * \sa SDL_AudioStreamAvailable 620 * \sa SDL_AudioStreamClear 621 * \sa SDL_FreeAudioStream 622 */ 623 extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream); 624 625 /** 626 * Clear any pending data in the stream without converting it 627 * 628 * \sa SDL_NewAudioStream 629 * \sa SDL_AudioStreamPut 630 * \sa SDL_AudioStreamGet 631 * \sa SDL_AudioStreamAvailable 632 * \sa SDL_AudioStreamFlush 633 * \sa SDL_FreeAudioStream 634 */ 635 extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream); 636 637 /** 638 * Free an audio stream 639 * 640 * \sa SDL_NewAudioStream 641 * \sa SDL_AudioStreamPut 642 * \sa SDL_AudioStreamGet 643 * \sa SDL_AudioStreamAvailable 644 * \sa SDL_AudioStreamFlush 645 * \sa SDL_AudioStreamClear 646 */ 647 extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream); 648 649 #define SDL_MIX_MAXVOLUME 128 650 /** 651 * This takes two audio buffers of the playing audio format and mixes 652 * them, performing addition, volume adjustment, and overflow clipping. 653 * The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME 654 * for full audio volume. Note this does not change hardware volume. 655 * This is provided for convenience -- you can mix your own audio data. 656 */ 657 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, 658 Uint32 len, int volume); 659 660 /** 661 * This works like SDL_MixAudio(), but you specify the audio format instead of 662 * using the format of audio device 1. Thus it can be used when no audio 663 * device is open at all. 664 */ 665 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, 666 const Uint8 * src, 667 SDL_AudioFormat format, 668 Uint32 len, int volume); 669 670 /** 671 * Queue more audio on non-callback devices. 672 * 673 * (If you are looking to retrieve queued audio from a non-callback capture 674 * device, you want SDL_DequeueAudio() instead. This will return -1 to 675 * signify an error if you use it with capture devices.) 676 * 677 * SDL offers two ways to feed audio to the device: you can either supply a 678 * callback that SDL triggers with some frequency to obtain more audio 679 * (pull method), or you can supply no callback, and then SDL will expect 680 * you to supply data at regular intervals (push method) with this function. 681 * 682 * There are no limits on the amount of data you can queue, short of 683 * exhaustion of address space. Queued data will drain to the device as 684 * necessary without further intervention from you. If the device needs 685 * audio but there is not enough queued, it will play silence to make up 686 * the difference. This means you will have skips in your audio playback 687 * if you aren't routinely queueing sufficient data. 688 * 689 * This function copies the supplied data, so you are safe to free it when 690 * the function returns. This function is thread-safe, but queueing to the 691 * same device from two threads at once does not promise which buffer will 692 * be queued first. 693 * 694 * You may not queue audio on a device that is using an application-supplied 695 * callback; doing so returns an error. You have to use the audio callback 696 * or queue audio with this function, but not both. 697 * 698 * You should not call SDL_LockAudio() on the device before queueing; SDL 699 * handles locking internally for this function. 700 * 701 * \param dev The device ID to which we will queue audio. 702 * \param data The data to queue to the device for later playback. 703 * \param len The number of bytes (not samples!) to which (data) points. 704 * \return 0 on success, or -1 on error. 705 * 706 * \sa SDL_GetQueuedAudioSize 707 * \sa SDL_ClearQueuedAudio 708 */ 709 extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len); 710 711 /** 712 * Dequeue more audio on non-callback devices. 713 * 714 * (If you are looking to queue audio for output on a non-callback playback 715 * device, you want SDL_QueueAudio() instead. This will always return 0 716 * if you use it with playback devices.) 717 * 718 * SDL offers two ways to retrieve audio from a capture device: you can 719 * either supply a callback that SDL triggers with some frequency as the 720 * device records more audio data, (push method), or you can supply no 721 * callback, and then SDL will expect you to retrieve data at regular 722 * intervals (pull method) with this function. 723 * 724 * There are no limits on the amount of data you can queue, short of 725 * exhaustion of address space. Data from the device will keep queuing as 726 * necessary without further intervention from you. This means you will 727 * eventually run out of memory if you aren't routinely dequeueing data. 728 * 729 * Capture devices will not queue data when paused; if you are expecting 730 * to not need captured audio for some length of time, use 731 * SDL_PauseAudioDevice() to stop the capture device from queueing more 732 * data. This can be useful during, say, level loading times. When 733 * unpaused, capture devices will start queueing data from that point, 734 * having flushed any capturable data available while paused. 735 * 736 * This function is thread-safe, but dequeueing from the same device from 737 * two threads at once does not promise which thread will dequeued data 738 * first. 739 * 740 * You may not dequeue audio from a device that is using an 741 * application-supplied callback; doing so returns an error. You have to use 742 * the audio callback, or dequeue audio with this function, but not both. 743 * 744 * You should not call SDL_LockAudio() on the device before queueing; SDL 745 * handles locking internally for this function. 746 * 747 * \param dev The device ID from which we will dequeue audio. 748 * \param data A pointer into where audio data should be copied. 749 * \param len The number of bytes (not samples!) to which (data) points. 750 * \return number of bytes dequeued, which could be less than requested. 751 * 752 * \sa SDL_GetQueuedAudioSize 753 * \sa SDL_ClearQueuedAudio 754 */ 755 extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len); 756 757 /** 758 * Get the number of bytes of still-queued audio. 759 * 760 * For playback device: 761 * 762 * This is the number of bytes that have been queued for playback with 763 * SDL_QueueAudio(), but have not yet been sent to the hardware. This 764 * number may shrink at any time, so this only informs of pending data. 765 * 766 * Once we've sent it to the hardware, this function can not decide the 767 * exact byte boundary of what has been played. It's possible that we just 768 * gave the hardware several kilobytes right before you called this 769 * function, but it hasn't played any of it yet, or maybe half of it, etc. 770 * 771 * For capture devices: 772 * 773 * This is the number of bytes that have been captured by the device and 774 * are waiting for you to dequeue. This number may grow at any time, so 775 * this only informs of the lower-bound of available data. 776 * 777 * You may not queue audio on a device that is using an application-supplied 778 * callback; calling this function on such a device always returns 0. 779 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use 780 * the audio callback, but not both. 781 * 782 * You should not call SDL_LockAudio() on the device before querying; SDL 783 * handles locking internally for this function. 784 * 785 * \param dev The device ID of which we will query queued audio size. 786 * \return Number of bytes (not samples!) of queued audio. 787 * 788 * \sa SDL_QueueAudio 789 * \sa SDL_ClearQueuedAudio 790 */ 791 extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev); 792 793 /** 794 * Drop any queued audio data. For playback devices, this is any queued data 795 * still waiting to be submitted to the hardware. For capture devices, this 796 * is any data that was queued by the device that hasn't yet been dequeued by 797 * the application. 798 * 799 * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For 800 * playback devices, the hardware will start playing silence if more audio 801 * isn't queued. Unpaused capture devices will start filling the queue again 802 * as soon as they have more data available (which, depending on the state 803 * of the hardware and the thread, could be before this function call 804 * returns!). 805 * 806 * This will not prevent playback of queued audio that's already been sent 807 * to the hardware, as we can not undo that, so expect there to be some 808 * fraction of a second of audio that might still be heard. This can be 809 * useful if you want to, say, drop any pending music during a level change 810 * in your game. 811 * 812 * You may not queue audio on a device that is using an application-supplied 813 * callback; calling this function on such a device is always a no-op. 814 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use 815 * the audio callback, but not both. 816 * 817 * You should not call SDL_LockAudio() on the device before clearing the 818 * queue; SDL handles locking internally for this function. 819 * 820 * This function always succeeds and thus returns void. 821 * 822 * \param dev The device ID of which to clear the audio queue. 823 * 824 * \sa SDL_QueueAudio 825 * \sa SDL_GetQueuedAudioSize 826 */ 827 extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev); 828 829 830 /** 831 * \name Audio lock functions 832 * 833 * The lock manipulated by these functions protects the callback function. 834 * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that 835 * the callback function is not running. Do not call these from the callback 836 * function or you will cause deadlock. 837 */ 838 /* @{ */ 839 extern DECLSPEC void SDLCALL SDL_LockAudio(void); 840 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); 841 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); 842 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); 843 /* @} *//* Audio lock functions */ 844 845 /** 846 * This function shuts down audio processing and closes the audio device. 847 */ 848 extern DECLSPEC void SDLCALL SDL_CloseAudio(void); 849 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); 850 851 /* Ends C function definitions when using C++ */ 852 #ifdef __cplusplus 853 } 854 #endif 855 #include "close_code.h" 856 857 #endif /* SDL_audio_h_ */ 858 859 /* vi: set ts=4 sw=4 expandtab: */ 860