1/*
2 Simple DirectMedia Layer
3 Copyright (C) 1997-2018 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
41extern "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 */
64typedef 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 */
163typedef 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 */
178typedef 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
192struct SDL_AudioCVT;
193typedef 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/* */
226typedef 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/* @{ */
251extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
252extern 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/* @{ */
263extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
264extern 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 */
271extern 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 */
318extern 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 */
330typedef 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 */
344extern 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 */
359extern 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 */
376extern 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/* @{ */
395typedef enum
396{
397 SDL_AUDIO_STOPPED = 0,
398 SDL_AUDIO_PLAYING,
399 SDL_AUDIO_PAUSED
400} SDL_AudioStatus;
401extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
402
403extern DECLSPEC SDL_AudioStatus SDLCALL
404SDL_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/* @{ */
417extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
418extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
419 int pause_on);
420/* @} *//* Pause audio functions */
421
422/**
423 * This function loads a WAVE from the data source, automatically freeing
424 * that source if \c freesrc is non-zero. For example, to load a WAVE file,
425 * you could do:
426 * \code
427 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
428 * \endcode
429 *
430 * If this function succeeds, it returns the given SDL_AudioSpec,
431 * filled with the audio data format of the wave data, and sets
432 * \c *audio_buf to a malloc()'d buffer containing the audio data,
433 * and sets \c *audio_len to the length of that audio buffer, in bytes.
434 * You need to free the audio buffer with SDL_FreeWAV() when you are
435 * done with it.
436 *
437 * This function returns NULL and sets the SDL error message if the
438 * wave file cannot be opened, uses an unknown data format, or is
439 * corrupt. Currently raw and MS-ADPCM WAVE files are supported.
440 */
441extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
442 int freesrc,
443 SDL_AudioSpec * spec,
444 Uint8 ** audio_buf,
445 Uint32 * audio_len);
446
447/**
448 * Loads a WAV from a file.
449 * Compatibility convenience function.
450 */
451#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
452 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
453
454/**
455 * This function frees data previously allocated with SDL_LoadWAV_RW()
456 */
457extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
458
459/**
460 * This function takes a source format and rate and a destination format
461 * and rate, and initializes the \c cvt structure with information needed
462 * by SDL_ConvertAudio() to convert a buffer of audio data from one format
463 * to the other. An unsupported format causes an error and -1 will be returned.
464 *
465 * \return 0 if no conversion is needed, 1 if the audio filter is set up,
466 * or -1 on error.
467 */
468extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
469 SDL_AudioFormat src_format,
470 Uint8 src_channels,
471 int src_rate,
472 SDL_AudioFormat dst_format,
473 Uint8 dst_channels,
474 int dst_rate);
475
476/**
477 * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
478 * created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
479 * audio data in the source format, this function will convert it in-place
480 * to the desired format.
481 *
482 * The data conversion may expand the size of the audio data, so the buffer
483 * \c cvt->buf should be allocated after the \c cvt structure is initialized by
484 * SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
485 *
486 * \return 0 on success or -1 if \c cvt->buf is NULL.
487 */
488extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
489
490/* SDL_AudioStream is a new audio conversion interface.
491 The benefits vs SDL_AudioCVT:
492 - it can handle resampling data in chunks without generating
493 artifacts, when it doesn't have the complete buffer available.
494 - it can handle incoming data in any variable size.
495 - You push data as you have it, and pull it when you need it
496 */
497/* this is opaque to the outside world. */
498struct _SDL_AudioStream;
499typedef struct _SDL_AudioStream SDL_AudioStream;
500
501/**
502 * Create a new audio stream
503 *
504 * \param src_format The format of the source audio
505 * \param src_channels The number of channels of the source audio
506 * \param src_rate The sampling rate of the source audio
507 * \param dst_format The format of the desired audio output
508 * \param dst_channels The number of channels of the desired audio output
509 * \param dst_rate The sampling rate of the desired audio output
510 * \return 0 on success, or -1 on error.
511 *
512 * \sa SDL_AudioStreamPut
513 * \sa SDL_AudioStreamGet
514 * \sa SDL_AudioStreamAvailable
515 * \sa SDL_AudioStreamFlush
516 * \sa SDL_AudioStreamClear
517 * \sa SDL_FreeAudioStream
518 */
519extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
520 const Uint8 src_channels,
521 const int src_rate,
522 const SDL_AudioFormat dst_format,
523 const Uint8 dst_channels,
524 const int dst_rate);
525
526/**
527 * Add data to be converted/resampled to the stream
528 *
529 * \param stream The stream the audio data is being added to
530 * \param buf A pointer to the audio data to add
531 * \param len The number of bytes to write to the stream
532 * \return 0 on success, or -1 on error.
533 *
534 * \sa SDL_NewAudioStream
535 * \sa SDL_AudioStreamGet
536 * \sa SDL_AudioStreamAvailable
537 * \sa SDL_AudioStreamFlush
538 * \sa SDL_AudioStreamClear
539 * \sa SDL_FreeAudioStream
540 */
541extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
542
543/**
544 * Get converted/resampled data from the stream
545 *
546 * \param stream The stream the audio is being requested from
547 * \param buf A buffer to fill with audio data
548 * \param len The maximum number of bytes to fill
549 * \return The number of bytes read from the stream, or -1 on error
550 *
551 * \sa SDL_NewAudioStream
552 * \sa SDL_AudioStreamPut
553 * \sa SDL_AudioStreamAvailable
554 * \sa SDL_AudioStreamFlush
555 * \sa SDL_AudioStreamClear
556 * \sa SDL_FreeAudioStream
557 */
558extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
559
560/**
561 * Get the number of converted/resampled bytes available. The stream may be
562 * buffering data behind the scenes until it has enough to resample
563 * correctly, so this number might be lower than what you expect, or even
564 * be zero. Add more data or flush the stream if you need the data now.
565 *
566 * \sa SDL_NewAudioStream
567 * \sa SDL_AudioStreamPut
568 * \sa SDL_AudioStreamGet
569 * \sa SDL_AudioStreamFlush
570 * \sa SDL_AudioStreamClear
571 * \sa SDL_FreeAudioStream
572 */
573extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
574
575/**
576 * Tell the stream that you're done sending data, and anything being buffered
577 * should be converted/resampled and made available immediately.
578 *
579 * It is legal to add more data to a stream after flushing, but there will
580 * be audio gaps in the output. Generally this is intended to signal the
581 * end of input, so the complete output becomes available.
582 *
583 * \sa SDL_NewAudioStream
584 * \sa SDL_AudioStreamPut
585 * \sa SDL_AudioStreamGet
586 * \sa SDL_AudioStreamAvailable
587 * \sa SDL_AudioStreamClear
588 * \sa SDL_FreeAudioStream
589 */
590extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
591
592/**
593 * Clear any pending data in the stream without converting it
594 *
595 * \sa SDL_NewAudioStream
596 * \sa SDL_AudioStreamPut
597 * \sa SDL_AudioStreamGet
598 * \sa SDL_AudioStreamAvailable
599 * \sa SDL_AudioStreamFlush
600 * \sa SDL_FreeAudioStream
601 */
602extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
603
604/**
605 * Free an audio stream
606 *
607 * \sa SDL_NewAudioStream
608 * \sa SDL_AudioStreamPut
609 * \sa SDL_AudioStreamGet
610 * \sa SDL_AudioStreamAvailable
611 * \sa SDL_AudioStreamFlush
612 * \sa SDL_AudioStreamClear
613 */
614extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
615
616#define SDL_MIX_MAXVOLUME 128
617/**
618 * This takes two audio buffers of the playing audio format and mixes
619 * them, performing addition, volume adjustment, and overflow clipping.
620 * The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
621 * for full audio volume. Note this does not change hardware volume.
622 * This is provided for convenience -- you can mix your own audio data.
623 */
624extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
625 Uint32 len, int volume);
626
627/**
628 * This works like SDL_MixAudio(), but you specify the audio format instead of
629 * using the format of audio device 1. Thus it can be used when no audio
630 * device is open at all.
631 */
632extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
633 const Uint8 * src,
634 SDL_AudioFormat format,
635 Uint32 len, int volume);
636
637/**
638 * Queue more audio on non-callback devices.
639 *
640 * (If you are looking to retrieve queued audio from a non-callback capture
641 * device, you want SDL_DequeueAudio() instead. This will return -1 to
642 * signify an error if you use it with capture devices.)
643 *
644 * SDL offers two ways to feed audio to the device: you can either supply a
645 * callback that SDL triggers with some frequency to obtain more audio
646 * (pull method), or you can supply no callback, and then SDL will expect
647 * you to supply data at regular intervals (push method) with this function.
648 *
649 * There are no limits on the amount of data you can queue, short of
650 * exhaustion of address space. Queued data will drain to the device as
651 * necessary without further intervention from you. If the device needs
652 * audio but there is not enough queued, it will play silence to make up
653 * the difference. This means you will have skips in your audio playback
654 * if you aren't routinely queueing sufficient data.
655 *
656 * This function copies the supplied data, so you are safe to free it when
657 * the function returns. This function is thread-safe, but queueing to the
658 * same device from two threads at once does not promise which buffer will
659 * be queued first.
660 *
661 * You may not queue audio on a device that is using an application-supplied
662 * callback; doing so returns an error. You have to use the audio callback
663 * or queue audio with this function, but not both.
664 *
665 * You should not call SDL_LockAudio() on the device before queueing; SDL
666 * handles locking internally for this function.
667 *
668 * \param dev The device ID to which we will queue audio.
669 * \param data The data to queue to the device for later playback.
670 * \param len The number of bytes (not samples!) to which (data) points.
671 * \return 0 on success, or -1 on error.
672 *
673 * \sa SDL_GetQueuedAudioSize
674 * \sa SDL_ClearQueuedAudio
675 */
676extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
677
678/**
679 * Dequeue more audio on non-callback devices.
680 *
681 * (If you are looking to queue audio for output on a non-callback playback
682 * device, you want SDL_QueueAudio() instead. This will always return 0
683 * if you use it with playback devices.)
684 *
685 * SDL offers two ways to retrieve audio from a capture device: you can
686 * either supply a callback that SDL triggers with some frequency as the
687 * device records more audio data, (push method), or you can supply no
688 * callback, and then SDL will expect you to retrieve data at regular
689 * intervals (pull method) with this function.
690 *
691 * There are no limits on the amount of data you can queue, short of
692 * exhaustion of address space. Data from the device will keep queuing as
693 * necessary without further intervention from you. This means you will
694 * eventually run out of memory if you aren't routinely dequeueing data.
695 *
696 * Capture devices will not queue data when paused; if you are expecting
697 * to not need captured audio for some length of time, use
698 * SDL_PauseAudioDevice() to stop the capture device from queueing more
699 * data. This can be useful during, say, level loading times. When
700 * unpaused, capture devices will start queueing data from that point,
701 * having flushed any capturable data available while paused.
702 *
703 * This function is thread-safe, but dequeueing from the same device from
704 * two threads at once does not promise which thread will dequeued data
705 * first.
706 *
707 * You may not dequeue audio from a device that is using an
708 * application-supplied callback; doing so returns an error. You have to use
709 * the audio callback, or dequeue audio with this function, but not both.
710 *
711 * You should not call SDL_LockAudio() on the device before queueing; SDL
712 * handles locking internally for this function.
713 *
714 * \param dev The device ID from which we will dequeue audio.
715 * \param data A pointer into where audio data should be copied.
716 * \param len The number of bytes (not samples!) to which (data) points.
717 * \return number of bytes dequeued, which could be less than requested.
718 *
719 * \sa SDL_GetQueuedAudioSize
720 * \sa SDL_ClearQueuedAudio
721 */
722extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
723
724/**
725 * Get the number of bytes of still-queued audio.
726 *
727 * For playback device:
728 *
729 * This is the number of bytes that have been queued for playback with
730 * SDL_QueueAudio(), but have not yet been sent to the hardware. This
731 * number may shrink at any time, so this only informs of pending data.
732 *
733 * Once we've sent it to the hardware, this function can not decide the
734 * exact byte boundary of what has been played. It's possible that we just
735 * gave the hardware several kilobytes right before you called this
736 * function, but it hasn't played any of it yet, or maybe half of it, etc.
737 *
738 * For capture devices:
739 *
740 * This is the number of bytes that have been captured by the device and
741 * are waiting for you to dequeue. This number may grow at any time, so
742 * this only informs of the lower-bound of available data.
743 *
744 * You may not queue audio on a device that is using an application-supplied
745 * callback; calling this function on such a device always returns 0.
746 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
747 * the audio callback, but not both.
748 *
749 * You should not call SDL_LockAudio() on the device before querying; SDL
750 * handles locking internally for this function.
751 *
752 * \param dev The device ID of which we will query queued audio size.
753 * \return Number of bytes (not samples!) of queued audio.
754 *
755 * \sa SDL_QueueAudio
756 * \sa SDL_ClearQueuedAudio
757 */
758extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
759
760/**
761 * Drop any queued audio data. For playback devices, this is any queued data
762 * still waiting to be submitted to the hardware. For capture devices, this
763 * is any data that was queued by the device that hasn't yet been dequeued by
764 * the application.
765 *
766 * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
767 * playback devices, the hardware will start playing silence if more audio
768 * isn't queued. Unpaused capture devices will start filling the queue again
769 * as soon as they have more data available (which, depending on the state
770 * of the hardware and the thread, could be before this function call
771 * returns!).
772 *
773 * This will not prevent playback of queued audio that's already been sent
774 * to the hardware, as we can not undo that, so expect there to be some
775 * fraction of a second of audio that might still be heard. This can be
776 * useful if you want to, say, drop any pending music during a level change
777 * in your game.
778 *
779 * You may not queue audio on a device that is using an application-supplied
780 * callback; calling this function on such a device is always a no-op.
781 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
782 * the audio callback, but not both.
783 *
784 * You should not call SDL_LockAudio() on the device before clearing the
785 * queue; SDL handles locking internally for this function.
786 *
787 * This function always succeeds and thus returns void.
788 *
789 * \param dev The device ID of which to clear the audio queue.
790 *
791 * \sa SDL_QueueAudio
792 * \sa SDL_GetQueuedAudioSize
793 */
794extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
795
796
797/**
798 * \name Audio lock functions
799 *
800 * The lock manipulated by these functions protects the callback function.
801 * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
802 * the callback function is not running. Do not call these from the callback
803 * function or you will cause deadlock.
804 */
805/* @{ */
806extern DECLSPEC void SDLCALL SDL_LockAudio(void);
807extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
808extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
809extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
810/* @} *//* Audio lock functions */
811
812/**
813 * This function shuts down audio processing and closes the audio device.
814 */
815extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
816extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
817
818/* Ends C function definitions when using C++ */
819#ifdef __cplusplus
820}
821#endif
822#include "close_code.h"
823
824#endif /* SDL_audio_h_ */
825
826/* vi: set ts=4 sw=4 expandtab: */
827