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