1/*
2 Simple DirectMedia Layer
3 Copyright (C) 1997-2021 Sam Lantinga <slouken@libsdl.org>
4
5 This software is provided 'as-is', without any express or implied
6 warranty. In no event will the authors be held liable for any damages
7 arising from the use of this software.
8
9 Permission is granted to anyone to use this software for any purpose,
10 including commercial applications, and to alter it and redistribute it
11 freely, subject to the following restrictions:
12
13 1. The origin of this software must not be misrepresented; you must not
14 claim that you wrote the original software. If you use this software
15 in a product, an acknowledgment in the product documentation would be
16 appreciated but is not required.
17 2. Altered source versions must be plainly marked as such, and must not be
18 misrepresented as being the original software.
19 3. This notice may not be removed or altered from any source distribution.
20*/
21
22/* !!! FIXME: several functions in here need Doxygen comments. */
23
24/**
25 * \file SDL_audio.h
26 *
27 * Access to the raw audio mixing buffer for the SDL library.
28 */
29
30#ifndef SDL_audio_h_
31#define SDL_audio_h_
32
33#include "SDL_stdinc.h"
34#include "SDL_error.h"
35#include "SDL_endian.h"
36#include "SDL_mutex.h"
37#include "SDL_thread.h"
38#include "SDL_rwops.h"
39
40#include "begin_code.h"
41/* Set up for C function definitions, even when using C++ */
42#ifdef __cplusplus
43extern "C" {
44#endif
45
46/**
47 * \brief Audio format flags.
48 *
49 * These are what the 16 bits in SDL_AudioFormat currently mean...
50 * (Unspecified bits are always zero).
51 *
52 * \verbatim
53 ++-----------------------sample is signed if set
54 ||
55 || ++-----------sample is bigendian if set
56 || ||
57 || || ++---sample is float if set
58 || || ||
59 || || || +---sample bit size---+
60 || || || | |
61 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
62 \endverbatim
63 *
64 * There are macros in SDL 2.0 and later to query these bits.
65 */
66typedef Uint16 SDL_AudioFormat;
67
68/**
69 * \name Audio flags
70 */
71/* @{ */
72
73#define SDL_AUDIO_MASK_BITSIZE (0xFF)
74#define SDL_AUDIO_MASK_DATATYPE (1<<8)
75#define SDL_AUDIO_MASK_ENDIAN (1<<12)
76#define SDL_AUDIO_MASK_SIGNED (1<<15)
77#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)
78#define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE)
79#define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN)
80#define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED)
81#define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x))
82#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x))
83#define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x))
84
85/**
86 * \name Audio format flags
87 *
88 * Defaults to LSB byte order.
89 */
90/* @{ */
91#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
92#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
93#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
94#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
95#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
96#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
97#define AUDIO_U16 AUDIO_U16LSB
98#define AUDIO_S16 AUDIO_S16LSB
99/* @} */
100
101/**
102 * \name int32 support
103 */
104/* @{ */
105#define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */
106#define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */
107#define AUDIO_S32 AUDIO_S32LSB
108/* @} */
109
110/**
111 * \name float32 support
112 */
113/* @{ */
114#define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */
115#define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */
116#define AUDIO_F32 AUDIO_F32LSB
117/* @} */
118
119/**
120 * \name Native audio byte ordering
121 */
122/* @{ */
123#if SDL_BYTEORDER == SDL_LIL_ENDIAN
124#define AUDIO_U16SYS AUDIO_U16LSB
125#define AUDIO_S16SYS AUDIO_S16LSB
126#define AUDIO_S32SYS AUDIO_S32LSB
127#define AUDIO_F32SYS AUDIO_F32LSB
128#else
129#define AUDIO_U16SYS AUDIO_U16MSB
130#define AUDIO_S16SYS AUDIO_S16MSB
131#define AUDIO_S32SYS AUDIO_S32MSB
132#define AUDIO_F32SYS AUDIO_F32MSB
133#endif
134/* @} */
135
136/**
137 * \name Allow change flags
138 *
139 * Which audio format changes are allowed when opening a device.
140 */
141/* @{ */
142#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001
143#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002
144#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004
145#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008
146#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)
147/* @} */
148
149/* @} *//* Audio flags */
150
151/**
152 * This function is called when the audio device needs more data.
153 *
154 * \param userdata An application-specific parameter saved in
155 * the SDL_AudioSpec structure
156 * \param stream A pointer to the audio data buffer.
157 * \param len The length of that buffer in bytes.
158 *
159 * Once the callback returns, the buffer will no longer be valid.
160 * Stereo samples are stored in a LRLRLR ordering.
161 *
162 * You can choose to avoid callbacks and use SDL_QueueAudio() instead, if
163 * you like. Just open your audio device with a NULL callback.
164 */
165typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
166 int len);
167
168/**
169 * The calculated values in this structure are calculated by SDL_OpenAudio().
170 *
171 * For multi-channel audio, the default SDL channel mapping is:
172 * 2: FL FR (stereo)
173 * 3: FL FR LFE (2.1 surround)
174 * 4: FL FR BL BR (quad)
175 * 5: FL FR FC BL BR (quad + center)
176 * 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR)
177 * 7: FL FR FC LFE BC SL SR (6.1 surround)
178 * 8: FL FR FC LFE BL BR SL SR (7.1 surround)
179 */
180typedef struct SDL_AudioSpec
181{
182 int freq; /**< DSP frequency -- samples per second */
183 SDL_AudioFormat format; /**< Audio data format */
184 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
185 Uint8 silence; /**< Audio buffer silence value (calculated) */
186 Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
187 Uint16 padding; /**< Necessary for some compile environments */
188 Uint32 size; /**< Audio buffer size in bytes (calculated) */
189 SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
190 void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */
191} SDL_AudioSpec;
192
193
194struct SDL_AudioCVT;
195typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
196 SDL_AudioFormat format);
197
198/**
199 * \brief Upper limit of filters in SDL_AudioCVT
200 *
201 * The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
202 * currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
203 * one of which is the terminating NULL pointer.
204 */
205#define SDL_AUDIOCVT_MAX_FILTERS 9
206
207/**
208 * \struct SDL_AudioCVT
209 * \brief A structure to hold a set of audio conversion filters and buffers.
210 *
211 * Note that various parts of the conversion pipeline can take advantage
212 * of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
213 * you to pass it aligned data, but can possibly run much faster if you
214 * set both its (buf) field to a pointer that is aligned to 16 bytes, and its
215 * (len) field to something that's a multiple of 16, if possible.
216 */
217#ifdef __GNUC__
218/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
219 pad it out to 88 bytes to guarantee ABI compatibility between compilers.
220 vvv
221 The next time we rev the ABI, make sure to size the ints and add padding.
222*/
223#define SDL_AUDIOCVT_PACKED __attribute__((packed))
224#else
225#define SDL_AUDIOCVT_PACKED
226#endif
227/* */
228typedef struct SDL_AudioCVT
229{
230 int needed; /**< Set to 1 if conversion possible */
231 SDL_AudioFormat src_format; /**< Source audio format */
232 SDL_AudioFormat dst_format; /**< Target audio format */
233 double rate_incr; /**< Rate conversion increment */
234 Uint8 *buf; /**< Buffer to hold entire audio data */
235 int len; /**< Length of original audio buffer */
236 int len_cvt; /**< Length of converted audio buffer */
237 int len_mult; /**< buffer must be len*len_mult big */
238 double len_ratio; /**< Given len, final size is len*len_ratio */
239 SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
240 int filter_index; /**< Current audio conversion function */
241} SDL_AUDIOCVT_PACKED SDL_AudioCVT;
242
243
244/* Function prototypes */
245
246/**
247 * \name Driver discovery functions
248 *
249 * These functions return the list of built in audio drivers, in the
250 * order that they are normally initialized by default.
251 */
252/* @{ */
253extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
254extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
255/* @} */
256
257/**
258 * \name Initialization and cleanup
259 *
260 * \internal These functions are used internally, and should not be used unless
261 * you have a specific need to specify the audio driver you want to
262 * use. You should normally use SDL_Init() or SDL_InitSubSystem().
263 */
264/* @{ */
265extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
266extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
267/* @} */
268
269/**
270 * Get the name of the current audio driver.
271 *
272 * The returned string points to internal static memory and thus never becomes
273 * invalid, even if you quit the audio subsystem and initialize a new driver
274 * (although such a case would return a different static string from another
275 * call to this function, of course). As such, you should not modify or free
276 * the returned string.
277 *
278 * \returns the name of the current audio driver or NULL if no driver has been
279 * initialized.
280 *
281 * \since This function is available since SDL 2.0.0.
282 *
283 * \sa SDL_AudioInit
284 */
285extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
286
287/**
288 * This function is a legacy means of opening the audio device.
289 *
290 * This function remains for compatibility with SDL 1.2, but also because it's
291 * slightly easier to use than the new functions in SDL 2.0. The new, more
292 * powerful, and preferred way to do this is SDL_OpenAudioDevice().
293 *
294 * This function is roughly equivalent to:
295 *
296 * ```c++
297 * SDL_OpenAudioDevice(NULL, 0, desired, obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
298 * ```
299 *
300 * With two notable exceptions:
301 *
302 * - If `obtained` is NULL, we use `desired` (and allow no changes), which
303 * means desired will be modified to have the correct values for silence,
304 * etc, and SDL will convert any differences between your app's specific
305 * request and the hardware behind the scenes.
306 * - The return value is always success or failure, and not a device ID, which
307 * means you can only have one device open at a time with this function.
308 *
309 * \param desired an SDL_AudioSpec structure representing the desired output
310 * format. Please refer to the SDL_OpenAudioDevice documentation
311 * for details on how to prepare this structure.
312 * \param obtained an SDL_AudioSpec structure filled in with the actual
313 * parameters, or NULL.
314 * \returns This function opens the audio device with the desired parameters,
315 * and returns 0 if successful, placing the actual hardware
316 * parameters in the structure pointed to by `obtained`.
317 *
318 * If `obtained` is NULL, the audio data passed to the callback
319 * function will be guaranteed to be in the requested format, and
320 * will be automatically converted to the actual hardware audio
321 * format if necessary. If `obtained` is NULL, `desired` will
322 * have fields modified.
323 *
324 * This function returns a negative error code on failure to open the
325 * audio device or failure to set up the audio thread; call
326 * SDL_GetError() for more information.
327 *
328 * \sa SDL_CloseAudio
329 * \sa SDL_LockAudio
330 * \sa SDL_PauseAudio
331 * \sa SDL_UnlockAudio
332 */
333extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
334 SDL_AudioSpec * obtained);
335
336/**
337 * SDL Audio Device IDs.
338 *
339 * A successful call to SDL_OpenAudio() is always device id 1, and legacy
340 * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
341 * always returns devices >= 2 on success. The legacy calls are good both
342 * for backwards compatibility and when you don't care about multiple,
343 * specific, or capture devices.
344 */
345typedef Uint32 SDL_AudioDeviceID;
346
347/**
348 * Get the number of built-in audio devices.
349 *
350 * This function is only valid after successfully initializing the audio
351 * subsystem.
352 *
353 * Note that audio capture support is not implemented as of SDL 2.0.4, so the
354 * `iscapture` parameter is for future expansion and should always be zero
355 * for now.
356 *
357 * This function will return -1 if an explicit list of devices can't be
358 * determined. Returning -1 is not an error. For example, if SDL is set up to
359 * talk to a remote audio server, it can't list every one available on the
360 * Internet, but it will still allow a specific host to be specified in
361 * SDL_OpenAudioDevice().
362 *
363 * In many common cases, when this function returns a value <= 0, it can still
364 * successfully open the default device (NULL for first argument of
365 * SDL_OpenAudioDevice()).
366 *
367 * This function may trigger a complete redetect of available hardware. It
368 * should not be called for each iteration of a loop, but rather once at the
369 * start of a loop:
370 *
371 * ```c++
372 * // Don't do this:
373 * for (int i = 0; i < SDL_GetNumAudioDevices(0); i++)
374 *
375 * // do this instead:
376 * const int count = SDL_GetNumAudioDevices(0);
377 * for (int i = 0; i < count; ++i) { do_something_here(); }
378 * ```
379 *
380 * \param iscapture zero to request playback devices, non-zero to request
381 * recording devices
382 * \returns the number of available devices exposed by the current driver or
383 * -1 if an explicit list of devices can't be determined. A return
384 * value of -1 does not necessarily mean an error condition.
385 *
386 * \since This function is available since SDL 2.0.0.
387 *
388 * \sa SDL_GetAudioDeviceName
389 * \sa SDL_OpenAudioDevice
390 */
391extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
392
393/**
394 * Get the human-readable name of a specific audio device.
395 *
396 * This function is only valid after successfully initializing the audio
397 * subsystem. The values returned by this function reflect the latest call to
398 * SDL_GetNumAudioDevices(); re-call that function to redetect available
399 * hardware.
400 *
401 * The string returned by this function is UTF-8 encoded, read-only, and
402 * managed internally. You are not to free it. If you need to keep the string
403 * for any length of time, you should make your own copy of it, as it will be
404 * invalid next time any of several other SDL functions are called.
405 *
406 * \param index the index of the audio device; valid values range from 0 to
407 * SDL_GetNumAudioDevices() - 1
408 * \param iscapture non-zero to query the list of recording devices, zero to
409 * query the list of output devices.
410 * \returns the name of the audio device at the requested index, or NULL on
411 * error.
412 *
413 * \sa SDL_GetNumAudioDevices
414 */
415extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
416 int iscapture);
417
418/**
419 * Get the preferred audio format of a specific audio device.
420 *
421 * This function is only valid after a successfully initializing the audio
422 * subsystem. The values returned by this function reflect the latest call to
423 * SDL_GetNumAudioDevices(); re-call that function to redetect available
424 * hardware.
425 *
426 * `spec` will be filled with the sample rate, sample format, and channel
427 * count. All other values in the structure are filled with 0. When the
428 * supported struct members are 0, SDL was unable to get the property from the
429 * backend.
430 *
431 * \param index the index of the audio device; valid values range from 0 to
432 * SDL_GetNumAudioDevices() - 1
433 * \param iscapture non-zero to query the list of recording devices, zero to
434 * query the list of output devices.
435 * \param spec The SDL_AudioSpec to be initialized by this function.
436 * \returns 0 on success, nonzero on error
437 *
438 * \sa SDL_GetNumAudioDevices
439 */
440extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index,
441 int iscapture,
442 SDL_AudioSpec *spec);
443
444
445/**
446 * Open a specific audio device.
447 *
448 * SDL_OpenAudio(), unlike this function, always acts on device ID 1. As such,
449 * this function will never return a 1 so as not to conflict with the legacy
450 * function.
451 *
452 * Please note that SDL 2.0 before 2.0.5 did not support recording; as such,
453 * this function would fail if `iscapture` was not zero. Starting with SDL
454 * 2.0.5, recording is implemented and this value can be non-zero.
455 *
456 * Passing in a `device` name of NULL requests the most reasonable default
457 * (and is equivalent to what SDL_OpenAudio() does to choose a device). The
458 * `device` name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
459 * some drivers allow arbitrary and driver-specific strings, such as a
460 * hostname/IP address for a remote audio server, or a filename in the
461 * diskaudio driver.
462 *
463 * When filling in the desired audio spec structure:
464 *
465 * - `desired->freq` should be the frequency in sample-frames-per-second (Hz).
466 * - `desired->format` should be the audio format (`AUDIO_S16SYS`, etc).
467 * - `desired->samples` is the desired size of the audio buffer, in
468 * _sample frames_ (with stereo output, two samples--left and right--would
469 * make a single sample frame). This number should be a power of two, and
470 * may be adjusted by the audio driver to a value more suitable for the
471 * hardware. Good values seem to range between 512 and 8096 inclusive,
472 * depending on the application and CPU speed. Smaller values reduce
473 * latency, but can lead to underflow if the application is doing heavy
474 * processing and cannot fill the audio buffer in time. Note that the
475 * number of sample frames is directly related to time by the following
476 * formula: `ms = (sampleframes*1000)/freq`
477 * - `desired->size` is the size in _bytes_ of the audio buffer, and is
478 * calculated by SDL_OpenAudioDevice(). You don't initialize this.
479 * - `desired->silence` is the value used to set the buffer to silence,
480 * and is calculated by SDL_OpenAudioDevice(). You don't initialize this.
481 * - `desired->callback` should be set to a function that will be called
482 * when the audio device is ready for more data. It is passed a pointer
483 * to the audio buffer, and the length in bytes of the audio buffer.
484 * This function usually runs in a separate thread, and so you should
485 * protect data structures that it accesses by calling SDL_LockAudioDevice()
486 * and SDL_UnlockAudioDevice() in your code. Alternately, you may pass a NULL
487 * pointer here, and call SDL_QueueAudio() with some frequency, to queue
488 * more audio samples to be played (or for capture devices, call
489 * SDL_DequeueAudio() with some frequency, to obtain audio samples).
490 * - `desired->userdata` is passed as the first parameter to your callback
491 * function. If you passed a NULL callback, this value is ignored.
492 *
493 * `allowed_changes` can have the following flags OR'd together:
494 *
495 * - `SDL_AUDIO_ALLOW_FREQUENCY_CHANGE`
496 * - `SDL_AUDIO_ALLOW_FORMAT_CHANGE`
497 * - `SDL_AUDIO_ALLOW_CHANNELS_CHANGE`
498 * - `SDL_AUDIO_ALLOW_ANY_CHANGE`
499 *
500 * These flags specify how SDL should behave when a device cannot offer a
501 * specific feature. If the application requests a feature that the hardware
502 * doesn't offer, SDL will always try to get the closest equivalent.
503 *
504 * For example, if you ask for float32 audio format, but the sound card only
505 * supports int16, SDL will set the hardware to int16. If you had set
506 * SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the
507 * `obtained` structure. If that flag was *not* set, SDL will prepare to
508 * convert your callback's float32 audio to int16 before feeding it to the
509 * hardware and will keep the originally requested format in the `obtained`
510 * structure.
511 *
512 * If your application can only handle one specific data format, pass a zero
513 * for `allowed_changes` and let SDL transparently handle any differences.
514 *
515 * An opened audio device starts out paused, and should be enabled for playing
516 * by calling SDL_PauseAudioDevice(devid, 0) when you are ready for your audio
517 * callback function to be called. Since the audio driver may modify the
518 * requested size of the audio buffer, you should allocate any local mixing
519 * buffers after you open the audio device.
520 *
521 * The audio callback runs in a separate thread in most cases; you can prevent
522 * race conditions between your callback and other threads without fully
523 * pausing playback with SDL_LockAudioDevice(). For more information about the
524 * callback, see SDL_AudioSpec.
525 *
526 * \param device a UTF-8 string reported by SDL_GetAudioDeviceName() or a
527 * driver-specific name as appropriate. NULL requests the most
528 * reasonable default device.
529 * \param iscapture non-zero to specify a device should be opened for
530 * recording, not playback
531 * \param desired an SDL_AudioSpec structure representing the desired output
532 * format; see SDL_OpenAudio() for more information
533 * \param obtained an SDL_AudioSpec structure filled in with the actual output
534 * format; see SDL_OpenAudio() for more information
535 * \param allowed_changes 0, or one or more flags OR'd together
536 * \returns a valid device ID that is > 0 on success or 0 on failure; call
537 * SDL_GetError() for more information.
538 *
539 * For compatibility with SDL 1.2, this will never return 1, since
540 * SDL reserves that ID for the legacy SDL_OpenAudio() function.
541 *
542 * \since This function is available since SDL 2.0.0.
543 *
544 * \sa SDL_CloseAudioDevice
545 * \sa SDL_GetAudioDeviceName
546 * \sa SDL_LockAudioDevice
547 * \sa SDL_OpenAudio
548 * \sa SDL_PauseAudioDevice
549 * \sa SDL_UnlockAudioDevice
550 */
551extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(
552 const char *device,
553 int iscapture,
554 const SDL_AudioSpec *desired,
555 SDL_AudioSpec *obtained,
556 int allowed_changes);
557
558
559
560/**
561 * \name Audio state
562 *
563 * Get the current audio state.
564 */
565/* @{ */
566typedef enum
567{
568 SDL_AUDIO_STOPPED = 0,
569 SDL_AUDIO_PLAYING,
570 SDL_AUDIO_PAUSED
571} SDL_AudioStatus;
572extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
573extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
574/* @} *//* Audio State */
575
576/**
577 * \name Pause audio functions
578 *
579 * These functions pause and unpause the audio callback processing.
580 * They should be called with a parameter of 0 after opening the audio
581 * device to start playing sound. This is so you can safely initialize
582 * data for your callback function after opening the audio device.
583 * Silence will be written to the audio device during the pause.
584 */
585/* @{ */
586extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
587extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
588 int pause_on);
589/* @} *//* Pause audio functions */
590
591/**
592 * Load the audio data of a WAVE file into memory.
593 *
594 * Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len`
595 * to be valid pointers. The entire data portion of the file is then loaded
596 * into memory and decoded if necessary.
597 *
598 * If `freesrc` is non-zero, the data source gets automatically closed and
599 * freed before the function returns.
600 *
601 * Supported formats are RIFF WAVE files with the formats PCM (8, 16, 24, and
602 * 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits),
603 * and A-law and mu-law (8 bits). Other formats are currently unsupported and
604 * cause an error.
605 *
606 * If this function succeeds, the pointer returned by it is equal to `spec`
607 * and the pointer to the audio data allocated by the function is written to
608 * `audio_buf` and its length in bytes to `audio_len`. The SDL_AudioSpec
609 * members `freq`, `channels`, and `format` are set to the values of the
610 * audio data in the buffer. The `samples` member is set to a sane default
611 * and all others are set to zero.
612 *
613 * It's necessary to use SDL_FreeWAV() to free the audio data returned in
614 * `audio_buf` when it is no longer used.
615 *
616 * Because of the underspecification of the .WAV format, there are many
617 * problematic files in the wild that cause issues with strict decoders. To
618 * provide compatibility with these files, this decoder is lenient in regards
619 * to the truncation of the file, the fact chunk, and the size of the RIFF
620 * chunk. The hints `SDL_HINT_WAVE_RIFF_CHUNK_SIZE`, `SDL_HINT_WAVE_TRUNCATION`,
621 * and `SDL_HINT_WAVE_FACT_CHUNK` can be used to tune the behavior of the
622 * loading process.
623 *
624 * Any file that is invalid (due to truncation, corruption, or wrong values in
625 * the headers), too big, or unsupported causes an error. Additionally, any
626 * critical I/O error from the data source will terminate the loading process
627 * with an error. The function returns NULL on error and in all cases (with the
628 * exception of `src` being NULL), an appropriate error message will be set.
629 *
630 * It is required that the data source supports seeking.
631 *
632 * Example:
633 * ```c++
634 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, &spec, &buf, &len);
635 * ```
636 *
637 * Note that the SDL_LoadWAV macro does this same thing for you, but in a less
638 * messy way:
639 *
640 * ```c++
641 * SDL_LoadWAV("sample.wav", &spec, &buf, &len);
642 * ```
643 *
644 * \param src The data source for the WAVE data
645 * \param freesrc If non-zero, SDL will _always_ free the data source
646 * \param spec An SDL_AudioSpec that will be filled in with the wave file's
647 * format details
648 * \param audio_buf A pointer filled with the audio data, allocated by the function.
649 * \param audio_len A pointer filled with the length of the audio data buffer in bytes
650 * \returns This function, if successfully called, returns `spec`, which will
651 * be filled with the audio data format of the wave source data.
652 * `audio_buf` will be filled with a pointer to an allocated buffer
653 * containing the audio data, and `audio_len` is filled with the
654 * length of that audio buffer in bytes.
655 *
656 * This function returns NULL if the .WAV file cannot be opened, uses
657 * an unknown data format, or is corrupt; call SDL_GetError() for
658 * more information.
659 *
660 * When the application is done with the data returned in
661 * `audio_buf`, it should call SDL_FreeWAV() to dispose of it.
662 *
663 * \sa SDL_FreeWAV
664 * \sa SDL_LoadWAV
665 */
666extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
667 int freesrc,
668 SDL_AudioSpec * spec,
669 Uint8 ** audio_buf,
670 Uint32 * audio_len);
671
672/**
673 * Loads a WAV from a file.
674 * Compatibility convenience function.
675 */
676#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
677 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
678
679/**
680 * Free data previously allocated with SDL_LoadWAV() or SDL_LoadWAV_RW().
681 *
682 * After a WAVE file has been opened with SDL_LoadWAV() or SDL_LoadWAV_RW()
683 * its data can eventually be freed with SDL_FreeWAV(). It is safe to call
684 * this function with a NULL pointer.
685 *
686 * \param audio_buf a pointer to the buffer created by SDL_LoadWAV() or
687 * SDL_LoadWAV_RW()
688 *
689 * \sa SDL_LoadWAV
690 * \sa SDL_LoadWAV_RW
691 */
692extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
693
694/**
695 * Initialize an SDL_AudioCVT structure for conversion.
696 *
697 * Before an SDL_AudioCVT structure can be used to convert audio data it must
698 * be initialized with source and destination information.
699 *
700 * This function will zero out every field of the SDL_AudioCVT, so it must be
701 * called before the application fills in the final buffer information.
702 *
703 * Once this function has returned successfully, and reported that a
704 * conversion is necessary, the application fills in the rest of the fields in
705 * SDL_AudioCVT, now that it knows how large a buffer it needs to allocate,
706 * and then can call SDL_ConvertAudio() to complete the conversion.
707 *
708 * \param cvt an SDL_AudioCVT structure filled in with audio conversion
709 * information
710 * \param src_format the source format of the audio data; for more info see
711 * SDL_AudioFormat
712 * \param src_channels the number of channels in the source
713 * \param src_rate the frequency (sample-frames-per-second) of the source
714 * \param dst_format the destination format of the audio data; for more info
715 * see SDL_AudioFormat
716 * \param dst_channels the number of channels in the destination
717 * \param dst_rate the frequency (sample-frames-per-second) of the
718 * destination
719 * \returns 1 if the audio filter is prepared, 0 if no conversion is needed,
720 * or a negative error code on failure; call SDL_GetError() for more
721 * information.
722 *
723 * \sa SDL_ConvertAudio
724 */
725extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
726 SDL_AudioFormat src_format,
727 Uint8 src_channels,
728 int src_rate,
729 SDL_AudioFormat dst_format,
730 Uint8 dst_channels,
731 int dst_rate);
732
733/**
734 * Convert audio data to a desired audio format.
735 *
736 * This function does the actual audio data conversion, after the application
737 * has called SDL_BuildAudioCVT() to prepare the conversion information and
738 * then filled in the buffer details.
739 *
740 * Once the application has initialized the `cvt` structure using
741 * SDL_BuildAudioCVT(), allocated an audio buffer and filled it with audio
742 * data in the source format, this function will convert the buffer, in-place,
743 * to the desired format.
744 *
745 * The data conversion may go through several passes; any given pass may
746 * possibly temporarily increase the size of the data. For example, SDL might
747 * expand 16-bit data to 32 bits before resampling to a lower frequency,
748 * shrinking the data size after having grown it briefly. Since the supplied
749 * buffer will be both the source and destination, converting as necessary
750 * in-place, the application must allocate a buffer that will fully contain
751 * the data during its largest conversion pass. After SDL_BuildAudioCVT()
752 * returns, the application should set the `cvt->len` field to the size, in
753 * bytes, of the source data, and allocate a buffer that is
754 * `cvt->len * cvt->len_mult` bytes long for the `buf` field.
755 *
756 * The source data should be copied into this buffer before the call to
757 * SDL_ConvertAudio(). Upon successful return, this buffer will contain the
758 * converted audio, and `cvt->len_cvt` will be the size of the converted data,
759 * in bytes. Any bytes in the buffer past `cvt->len_cvt` are undefined once
760 * this function returns.
761 *
762 * \param cvt an SDL_AudioCVT structure that was previously set up by
763 * SDL_BuildAudioCVT().
764 * \returns 0 if the conversion was completed successfully or a negative error
765 * code on failure; call SDL_GetError() for more information.
766 *
767 * \sa SDL_BuildAudioCVT
768 */
769extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
770
771/* SDL_AudioStream is a new audio conversion interface.
772 The benefits vs SDL_AudioCVT:
773 - it can handle resampling data in chunks without generating
774 artifacts, when it doesn't have the complete buffer available.
775 - it can handle incoming data in any variable size.
776 - You push data as you have it, and pull it when you need it
777 */
778/* this is opaque to the outside world. */
779struct _SDL_AudioStream;
780typedef struct _SDL_AudioStream SDL_AudioStream;
781
782/**
783 * Create a new audio stream.
784 *
785 * \param src_format The format of the source audio
786 * \param src_channels The number of channels of the source audio
787 * \param src_rate The sampling rate of the source audio
788 * \param dst_format The format of the desired audio output
789 * \param dst_channels The number of channels of the desired audio output
790 * \param dst_rate The sampling rate of the desired audio output
791 * \returns 0 on success, or -1 on error.
792 *
793 * \sa SDL_AudioStreamPut
794 * \sa SDL_AudioStreamGet
795 * \sa SDL_AudioStreamAvailable
796 * \sa SDL_AudioStreamFlush
797 * \sa SDL_AudioStreamClear
798 * \sa SDL_FreeAudioStream
799 */
800extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
801 const Uint8 src_channels,
802 const int src_rate,
803 const SDL_AudioFormat dst_format,
804 const Uint8 dst_channels,
805 const int dst_rate);
806
807/**
808 * Add data to be converted/resampled to the stream.
809 *
810 * \param stream The stream the audio data is being added to
811 * \param buf A pointer to the audio data to add
812 * \param len The number of bytes to write to the stream
813 * \returns 0 on success, or -1 on error.
814 *
815 * \sa SDL_NewAudioStream
816 * \sa SDL_AudioStreamGet
817 * \sa SDL_AudioStreamAvailable
818 * \sa SDL_AudioStreamFlush
819 * \sa SDL_AudioStreamClear
820 * \sa SDL_FreeAudioStream
821 */
822extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
823
824/**
825 * Get converted/resampled data from the stream
826 *
827 * \param stream The stream the audio is being requested from
828 * \param buf A buffer to fill with audio data
829 * \param len The maximum number of bytes to fill
830 * \returns the number of bytes read from the stream, or -1 on error
831 *
832 * \sa SDL_NewAudioStream
833 * \sa SDL_AudioStreamPut
834 * \sa SDL_AudioStreamAvailable
835 * \sa SDL_AudioStreamFlush
836 * \sa SDL_AudioStreamClear
837 * \sa SDL_FreeAudioStream
838 */
839extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
840
841/**
842 * Get the number of converted/resampled bytes available. The stream may be
843 * buffering data behind the scenes until it has enough to resample
844 * correctly, so this number might be lower than what you expect, or even
845 * be zero. Add more data or flush the stream if you need the data now.
846 *
847 * \sa SDL_NewAudioStream
848 * \sa SDL_AudioStreamPut
849 * \sa SDL_AudioStreamGet
850 * \sa SDL_AudioStreamFlush
851 * \sa SDL_AudioStreamClear
852 * \sa SDL_FreeAudioStream
853 */
854extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
855
856/**
857 * Tell the stream that you're done sending data, and anything being buffered
858 * should be converted/resampled and made available immediately.
859 *
860 * It is legal to add more data to a stream after flushing, but there will
861 * be audio gaps in the output. Generally this is intended to signal the
862 * end of input, so the complete output becomes available.
863 *
864 * \sa SDL_NewAudioStream
865 * \sa SDL_AudioStreamPut
866 * \sa SDL_AudioStreamGet
867 * \sa SDL_AudioStreamAvailable
868 * \sa SDL_AudioStreamClear
869 * \sa SDL_FreeAudioStream
870 */
871extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
872
873/**
874 * Clear any pending data in the stream without converting it
875 *
876 * \sa SDL_NewAudioStream
877 * \sa SDL_AudioStreamPut
878 * \sa SDL_AudioStreamGet
879 * \sa SDL_AudioStreamAvailable
880 * \sa SDL_AudioStreamFlush
881 * \sa SDL_FreeAudioStream
882 */
883extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
884
885/**
886 * Free an audio stream
887 *
888 * \sa SDL_NewAudioStream
889 * \sa SDL_AudioStreamPut
890 * \sa SDL_AudioStreamGet
891 * \sa SDL_AudioStreamAvailable
892 * \sa SDL_AudioStreamFlush
893 * \sa SDL_AudioStreamClear
894 */
895extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
896
897#define SDL_MIX_MAXVOLUME 128
898/**
899 * This function is a legacy means of mixing audio.
900 *
901 * This function is equivalent to calling
902 *
903 * ```c++
904 * SDL_MixAudioFormat(dst, src, format, len, volume);
905 * ```
906 *
907 * where `format` is the obtained format of the audio device from the legacy
908 * SDL_OpenAudio() function.
909 *
910 * \param dst the destination for the mixed audio
911 * \param src the source audio buffer to be mixed
912 * \param len the length of the audio buffer in bytes
913 * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
914 * for full audio volume
915 *
916 * \sa SDL_MixAudioFormat
917 */
918extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
919 Uint32 len, int volume);
920
921/**
922 * Mix audio data in a specified format.
923 *
924 * This takes an audio buffer `src` of `len` bytes of `format` data and
925 * mixes it into `dst`, performing addition, volume adjustment, and overflow
926 * clipping. The buffer pointed to by `dst` must also be `len` bytes of
927 * `format` data.
928 *
929 * This is provided for convenience -- you can mix your own audio data.
930 *
931 * Do not use this function for mixing together more than two streams of
932 * sample data. The output from repeated application of this function may be
933 * distorted by clipping, because there is no accumulator with greater range
934 * than the input (not to mention this being an inefficient way of doing it).
935 *
936 * It is a common misconception that this function is required to write audio
937 * data to an output stream in an audio callback. While you can do that,
938 * SDL_MixAudioFormat() is really only needed when you're mixing a single
939 * audio stream with a volume adjustment.
940 *
941 * \param dst the destination for the mixed audio
942 * \param src the source audio buffer to be mixed
943 * \param format the SDL_AudioFormat structure representing the desired audio
944 * format
945 * \param len the length of the audio buffer in bytes
946 * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME
947 * for full audio volume
948 */
949extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
950 const Uint8 * src,
951 SDL_AudioFormat format,
952 Uint32 len, int volume);
953
954/**
955 * Queue more audio on non-callback devices.
956 *
957 * If you are looking to retrieve queued audio from a non-callback capture
958 * device, you want SDL_DequeueAudio() instead. SDL_QueueAudio() will return
959 * -1 to signify an error if you use it with capture devices.
960 *
961 * SDL offers two ways to feed audio to the device: you can either supply a
962 * callback that SDL triggers with some frequency to obtain more audio (pull
963 * method), or you can supply no callback, and then SDL will expect you to
964 * supply data at regular intervals (push method) with this function.
965 *
966 * There are no limits on the amount of data you can queue, short of
967 * exhaustion of address space. Queued data will drain to the device as
968 * necessary without further intervention from you. If the device needs audio
969 * but there is not enough queued, it will play silence to make up the
970 * difference. This means you will have skips in your audio playback if you
971 * aren't routinely queueing sufficient data.
972 *
973 * This function copies the supplied data, so you are safe to free it when the
974 * function returns. This function is thread-safe, but queueing to the same
975 * device from two threads at once does not promise which buffer will be
976 * queued first.
977 *
978 * You may not queue audio on a device that is using an application-supplied
979 * callback; doing so returns an error. You have to use the audio callback or
980 * queue audio with this function, but not both.
981 *
982 * You should not call SDL_LockAudio() on the device before queueing; SDL
983 * handles locking internally for this function.
984 *
985 * \param dev the device ID to which we will queue audio
986 * \param data the data to queue to the device for later playback
987 * \param len the number of bytes (not samples!) to which `data` points
988 * \returns 0 on success or a negative error code on failure; call
989 * SDL_GetError() for more information.
990 *
991 * \since This function is available since SDL 2.0.4.
992 *
993 * \sa SDL_ClearQueuedAudio
994 * \sa SDL_GetQueuedAudioSize
995 */
996extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
997
998/**
999 * Dequeue more audio on non-callback devices.
1000 *
1001 * If you are looking to queue audio for output on a non-callback playback
1002 * device, you want SDL_QueueAudio() instead. SDL_DequeueAudio() will always
1003 * return 0 if you use it with playback devices.
1004 *
1005 * SDL offers two ways to retrieve audio from a capture device: you can either
1006 * supply a callback that SDL triggers with some frequency as the device
1007 * records more audio data, (push method), or you can supply no callback, and
1008 * then SDL will expect you to retrieve data at regular intervals (pull
1009 * method) with this function.
1010 *
1011 * There are no limits on the amount of data you can queue, short of
1012 * exhaustion of address space. Data from the device will keep queuing as
1013 * necessary without further intervention from you. This means you will
1014 * eventually run out of memory if you aren't routinely dequeueing data.
1015 *
1016 * Capture devices will not queue data when paused; if you are expecting to
1017 * not need captured audio for some length of time, use SDL_PauseAudioDevice()
1018 * to stop the capture device from queueing more data. This can be useful
1019 * during, say, level loading times. When unpaused, capture devices will start
1020 * queueing data from that point, having flushed any capturable data available
1021 * while paused.
1022 *
1023 * This function is thread-safe, but dequeueing from the same device from two
1024 * threads at once does not promise which thread will dequeue data first.
1025 *
1026 * You may not dequeue audio from a device that is using an
1027 * application-supplied callback; doing so returns an error. You have to use
1028 * the audio callback, or dequeue audio with this function, but not both.
1029 *
1030 * You should not call SDL_LockAudio() on the device before dequeueing; SDL
1031 * handles locking internally for this function.
1032 *
1033 * \param dev the device ID from which we will dequeue audio
1034 * \param data a pointer into where audio data should be copied
1035 * \param len the number of bytes (not samples!) to which (data) points
1036 * \returns number of bytes dequeued, which could be less than requested; call
1037 * SDL_GetError() for more information.
1038 *
1039 * \since This function is available since SDL 2.0.5.
1040 *
1041 * \sa SDL_ClearQueuedAudio
1042 * \sa SDL_GetQueuedAudioSize
1043 */
1044extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
1045
1046/**
1047 * Get the number of bytes of still-queued audio.
1048 *
1049 * For playback devices: this is the number of bytes that have been queued
1050 * for playback with SDL_QueueAudio(), but have not yet been sent to the
1051 * hardware.
1052 *
1053 * Once we've sent it to the hardware, this function can not decide the exact
1054 * byte boundary of what has been played. It's possible that we just gave the
1055 * hardware several kilobytes right before you called this function, but it
1056 * hasn't played any of it yet, or maybe half of it, etc.
1057 *
1058 * For capture devices, this is the number of bytes that have been captured by
1059 * the device and are waiting for you to dequeue. This number may grow at any
1060 * time, so this only informs of the lower-bound of available data.
1061 *
1062 * You may not queue or dequeue audio on a device that is using an
1063 * application-supplied callback; calling this function on such a device
1064 * always returns 0. You have to use the audio callback or queue audio, but
1065 * not both.
1066 *
1067 * You should not call SDL_LockAudio() on the device before querying; SDL
1068 * handles locking internally for this function.
1069 *
1070 * \param dev the device ID of which we will query queued audio size
1071 * \returns the number of bytes (not samples!) of queued audio.
1072 *
1073 * \since This function is available since SDL 2.0.4.
1074 *
1075 * \sa SDL_ClearQueuedAudio
1076 * \sa SDL_QueueAudio
1077 * \sa SDL_DequeueAudio
1078 */
1079extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
1080
1081/**
1082 * Drop any queued audio data waiting to be sent to the hardware.
1083 *
1084 * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
1085 * output devices, the hardware will start playing silence if more audio isn't
1086 * queued. For capture devices, the hardware will start filling the empty
1087 * queue with new data if the capture device isn't paused.
1088 *
1089 * This will not prevent playback of queued audio that's already been sent to
1090 * the hardware, as we can not undo that, so expect there to be some fraction
1091 * of a second of audio that might still be heard. This can be useful if you
1092 * want to, say, drop any pending music or any unprocessed microphone input
1093 * during a level change in your game.
1094 *
1095 * You may not queue or dequeue audio on a device that is using an
1096 * application-supplied callback; calling this function on such a device
1097 * always returns 0. You have to use the audio callback or queue audio, but
1098 * not both.
1099 *
1100 * You should not call SDL_LockAudio() on the device before clearing the
1101 * queue; SDL handles locking internally for this function.
1102 *
1103 * This function always succeeds and thus returns void.
1104 *
1105 * \param dev the device ID of which to clear the audio queue
1106 *
1107 * \since This function is available since SDL 2.0.4.
1108 *
1109 * \sa SDL_GetQueuedAudioSize
1110 * \sa SDL_QueueAudio
1111 * \sa SDL_DequeueAudio
1112 */
1113extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
1114
1115
1116/**
1117 * \name Audio lock functions
1118 *
1119 * The lock manipulated by these functions protects the callback function.
1120 * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
1121 * the callback function is not running. Do not call these from the callback
1122 * function or you will cause deadlock.
1123 */
1124/* @{ */
1125extern DECLSPEC void SDLCALL SDL_LockAudio(void);
1126extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
1127extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
1128extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
1129/* @} *//* Audio lock functions */
1130
1131/**
1132 * This function is a legacy means of closing the audio device.
1133 *
1134 * This function is equivalent to calling
1135 *
1136 * ```c++
1137 * SDL_CloseAudioDevice(1);
1138 * ```
1139 *
1140 * and is only useful if you used the legacy SDL_OpenAudio() function.
1141 *
1142 * \sa SDL_OpenAudio
1143 */
1144extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
1145extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
1146
1147/* Ends C function definitions when using C++ */
1148#ifdef __cplusplus
1149}
1150#endif
1151#include "close_code.h"
1152
1153#endif /* SDL_audio_h_ */
1154
1155/* vi: set ts=4 sw=4 expandtab: */
1156