2 Simple DirectMedia Layer
3 Copyright (C) 1997-2017 Sam Lantinga <slouken@libsdl.org>
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.
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:
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.
25 * Access to the raw audio mixing buffer for the SDL library.
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"
38 #include "begin_code.h"
39 /* Set up for C function definitions, even when using C++ */
45 * \brief Audio format flags.
47 * These are what the 16 bits in SDL_AudioFormat currently mean...
48 * (Unspecified bits are always zero).
51 ++-----------------------sample is signed if set
53 || ++-----------sample is bigendian if set
55 || || ++---sample is float if set
57 || || || +---sample bit size---+
59 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
62 * There are macros in SDL 2.0 and later to query these bits.
64 typedef Uint16 SDL_AudioFormat;
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))
84 * \name Audio format flags
86 * Defaults to LSB byte order.
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
100 * \name int32 support
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
109 * \name float32 support
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
118 * \name Native audio byte ordering
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
127 #define AUDIO_U16SYS AUDIO_U16MSB
128 #define AUDIO_S16SYS AUDIO_S16MSB
129 #define AUDIO_S32SYS AUDIO_S32MSB
130 #define AUDIO_F32SYS AUDIO_F32MSB
135 * \name Allow change flags
137 * Which audio format changes are allowed when opening a device.
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)
146 /* @} *//* Audio flags */
149 * This function is called when the audio device needs more data.
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.
156 * Once the callback returns, the buffer will no longer be valid.
157 * Stereo samples are stored in a LRLRLR ordering.
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.
162 typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream,
166 * The calculated values in this structure are calculated by SDL_OpenAudio().
168 typedef struct SDL_AudioSpec
170 int freq; /**< DSP frequency -- samples per second */
171 SDL_AudioFormat format; /**< Audio data format */
172 Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
173 Uint8 silence; /**< Audio buffer silence value (calculated) */
174 Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */
175 Uint16 padding; /**< Necessary for some compile environments */
176 Uint32 size; /**< Audio buffer size in bytes (calculated) */
177 SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */
178 void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */
183 typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt,
184 SDL_AudioFormat format);
187 * \brief Upper limit of filters in SDL_AudioCVT
189 * The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is
190 * currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers,
191 * one of which is the terminating NULL pointer.
193 #define SDL_AUDIOCVT_MAX_FILTERS 9
196 * \struct SDL_AudioCVT
197 * \brief A structure to hold a set of audio conversion filters and buffers.
199 * Note that various parts of the conversion pipeline can take advantage
200 * of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
201 * you to pass it aligned data, but can possibly run much faster if you
202 * set both its (buf) field to a pointer that is aligned to 16 bytes, and its
203 * (len) field to something that's a multiple of 16, if possible.
206 /* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't
207 pad it out to 88 bytes to guarantee ABI compatibility between compilers.
209 The next time we rev the ABI, make sure to size the ints and add padding.
211 #define SDL_AUDIOCVT_PACKED __attribute__((packed))
213 #define SDL_AUDIOCVT_PACKED
216 typedef struct SDL_AudioCVT
218 int needed; /**< Set to 1 if conversion possible */
219 SDL_AudioFormat src_format; /**< Source audio format */
220 SDL_AudioFormat dst_format; /**< Target audio format */
221 double rate_incr; /**< Rate conversion increment */
222 Uint8 *buf; /**< Buffer to hold entire audio data */
223 int len; /**< Length of original audio buffer */
224 int len_cvt; /**< Length of converted audio buffer */
225 int len_mult; /**< buffer must be len*len_mult big */
226 double len_ratio; /**< Given len, final size is len*len_ratio */
227 SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */
228 int filter_index; /**< Current audio conversion function */
229 } SDL_AUDIOCVT_PACKED SDL_AudioCVT;
232 /* Function prototypes */
235 * \name Driver discovery functions
237 * These functions return the list of built in audio drivers, in the
238 * order that they are normally initialized by default.
241 extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void);
242 extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
246 * \name Initialization and cleanup
248 * \internal These functions are used internally, and should not be used unless
249 * you have a specific need to specify the audio driver you want to
250 * use. You should normally use SDL_Init() or SDL_InitSubSystem().
253 extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
254 extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
258 * This function returns the name of the current audio driver, or NULL
259 * if no driver has been initialized.
261 extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);
264 * This function opens the audio device with the desired parameters, and
265 * returns 0 if successful, placing the actual hardware parameters in the
266 * structure pointed to by \c obtained. If \c obtained is NULL, the audio
267 * data passed to the callback function will be guaranteed to be in the
268 * requested format, and will be automatically converted to the hardware
269 * audio format if necessary. This function returns -1 if it failed
270 * to open the audio device, or couldn't set up the audio thread.
272 * When filling in the desired audio spec structure,
273 * - \c desired->freq should be the desired audio frequency in samples-per-
275 * - \c desired->format should be the desired audio format.
276 * - \c desired->samples is the desired size of the audio buffer, in
277 * samples. This number should be a power of two, and may be adjusted by
278 * the audio driver to a value more suitable for the hardware. Good values
279 * seem to range between 512 and 8096 inclusive, depending on the
280 * application and CPU speed. Smaller values yield faster response time,
281 * but can lead to underflow if the application is doing heavy processing
282 * and cannot fill the audio buffer in time. A stereo sample consists of
283 * both right and left channels in LR ordering.
284 * Note that the number of samples is directly related to time by the
285 * following formula: \code ms = (samples*1000)/freq \endcode
286 * - \c desired->size is the size in bytes of the audio buffer, and is
287 * calculated by SDL_OpenAudio().
288 * - \c desired->silence is the value used to set the buffer to silence,
289 * and is calculated by SDL_OpenAudio().
290 * - \c desired->callback should be set to a function that will be called
291 * when the audio device is ready for more data. It is passed a pointer
292 * to the audio buffer, and the length in bytes of the audio buffer.
293 * This function usually runs in a separate thread, and so you should
294 * protect data structures that it accesses by calling SDL_LockAudio()
295 * and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL
296 * pointer here, and call SDL_QueueAudio() with some frequency, to queue
297 * more audio samples to be played (or for capture devices, call
298 * SDL_DequeueAudio() with some frequency, to obtain audio samples).
299 * - \c desired->userdata is passed as the first parameter to your callback
300 * function. If you passed a NULL callback, this value is ignored.
302 * The audio device starts out playing silence when it's opened, and should
303 * be enabled for playing by calling \c SDL_PauseAudio(0) when you are ready
304 * for your audio callback function to be called. Since the audio driver
305 * may modify the requested size of the audio buffer, you should allocate
306 * any local mixing buffers after you open the audio device.
308 extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired,
309 SDL_AudioSpec * obtained);
312 * SDL Audio Device IDs.
314 * A successful call to SDL_OpenAudio() is always device id 1, and legacy
315 * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls
316 * always returns devices >= 2 on success. The legacy calls are good both
317 * for backwards compatibility and when you don't care about multiple,
318 * specific, or capture devices.
320 typedef Uint32 SDL_AudioDeviceID;
323 * Get the number of available devices exposed by the current driver.
324 * Only valid after a successfully initializing the audio subsystem.
325 * Returns -1 if an explicit list of devices can't be determined; this is
326 * not an error. For example, if SDL is set up to talk to a remote audio
327 * server, it can't list every one available on the Internet, but it will
328 * still allow a specific host to be specified to SDL_OpenAudioDevice().
330 * In many common cases, when this function returns a value <= 0, it can still
331 * successfully open the default device (NULL for first argument of
332 * SDL_OpenAudioDevice()).
334 extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture);
337 * Get the human-readable name of a specific audio device.
338 * Must be a value between 0 and (number of audio devices-1).
339 * Only valid after a successfully initializing the audio subsystem.
340 * The values returned by this function reflect the latest call to
341 * SDL_GetNumAudioDevices(); recall that function to redetect available
344 * The string returned by this function is UTF-8 encoded, read-only, and
345 * managed internally. You are not to free it. If you need to keep the
346 * string for any length of time, you should make your own copy of it, as it
347 * will be invalid next time any of several other SDL functions is called.
349 extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index,
354 * Open a specific audio device. Passing in a device name of NULL requests
355 * the most reasonable default (and is equivalent to calling SDL_OpenAudio()).
357 * The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but
358 * some drivers allow arbitrary and driver-specific strings, such as a
359 * hostname/IP address for a remote audio server, or a filename in the
362 * \return 0 on error, a valid device ID that is >= 2 on success.
364 * SDL_OpenAudio(), unlike this function, always acts on device ID 1.
366 extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(const char
382 * Get the current audio state.
387 SDL_AUDIO_STOPPED = 0,
391 extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void);
393 extern DECLSPEC SDL_AudioStatus SDLCALL
394 SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev);
395 /* @} *//* Audio State */
398 * \name Pause audio functions
400 * These functions pause and unpause the audio callback processing.
401 * They should be called with a parameter of 0 after opening the audio
402 * device to start playing sound. This is so you can safely initialize
403 * data for your callback function after opening the audio device.
404 * Silence will be written to the audio device during the pause.
407 extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
408 extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev,
410 /* @} *//* Pause audio functions */
413 * This function loads a WAVE from the data source, automatically freeing
414 * that source if \c freesrc is non-zero. For example, to load a WAVE file,
417 * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
420 * If this function succeeds, it returns the given SDL_AudioSpec,
421 * filled with the audio data format of the wave data, and sets
422 * \c *audio_buf to a malloc()'d buffer containing the audio data,
423 * and sets \c *audio_len to the length of that audio buffer, in bytes.
424 * You need to free the audio buffer with SDL_FreeWAV() when you are
427 * This function returns NULL and sets the SDL error message if the
428 * wave file cannot be opened, uses an unknown data format, or is
429 * corrupt. Currently raw and MS-ADPCM WAVE files are supported.
431 extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src,
433 SDL_AudioSpec * spec,
438 * Loads a WAV from a file.
439 * Compatibility convenience function.
441 #define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
442 SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
445 * This function frees data previously allocated with SDL_LoadWAV_RW()
447 extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf);
450 * This function takes a source format and rate and a destination format
451 * and rate, and initializes the \c cvt structure with information needed
452 * by SDL_ConvertAudio() to convert a buffer of audio data from one format
453 * to the other. An unsupported format causes an error and -1 will be returned.
455 * \return 0 if no conversion is needed, 1 if the audio filter is set up,
458 extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt,
459 SDL_AudioFormat src_format,
462 SDL_AudioFormat dst_format,
467 * Once you have initialized the \c cvt structure using SDL_BuildAudioCVT(),
468 * created an audio buffer \c cvt->buf, and filled it with \c cvt->len bytes of
469 * audio data in the source format, this function will convert it in-place
470 * to the desired format.
472 * The data conversion may expand the size of the audio data, so the buffer
473 * \c cvt->buf should be allocated after the \c cvt structure is initialized by
474 * SDL_BuildAudioCVT(), and should be \c cvt->len*cvt->len_mult bytes long.
476 * \return 0 on success or -1 if \c cvt->buf is NULL.
478 extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt);
480 /* SDL_AudioStream is a new audio conversion interface.
481 The benefits vs SDL_AudioCVT:
482 - it can handle resampling data in chunks without generating
483 artifacts, when it doesn't have the complete buffer available.
484 - it can handle incoming data in any variable size.
485 - You push data as you have it, and pull it when you need it
487 /* this is opaque to the outside world. */
488 struct _SDL_AudioStream;
489 typedef struct _SDL_AudioStream SDL_AudioStream;
492 * Create a new audio stream
494 * \param src_format The format of the source audio
495 * \param src_channels The number of channels of the source audio
496 * \param src_rate The sampling rate of the source audio
497 * \param dst_format The format of the desired audio output
498 * \param dst_channels The number of channels of the desired audio output
499 * \param dst_rate The sampling rate of the desired audio output
500 * \return 0 on success, or -1 on error.
502 * \sa SDL_AudioStreamPut
503 * \sa SDL_AudioStreamGet
504 * \sa SDL_AudioStreamAvailable
505 * \sa SDL_AudioStreamFlush
506 * \sa SDL_AudioStreamClear
507 * \sa SDL_FreeAudioStream
509 extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format,
510 const Uint8 src_channels,
512 const SDL_AudioFormat dst_format,
513 const Uint8 dst_channels,
517 * Add data to be converted/resampled to the stream
519 * \param stream The stream the audio data is being added to
520 * \param buf A pointer to the audio data to add
521 * \param int The number of bytes to write to the stream
522 * \return 0 on success, or -1 on error.
524 * \sa SDL_NewAudioStream
525 * \sa SDL_AudioStreamGet
526 * \sa SDL_AudioStreamAvailable
527 * \sa SDL_AudioStreamFlush
528 * \sa SDL_AudioStreamClear
529 * \sa SDL_FreeAudioStream
531 extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len);
534 * Get converted/resampled data from the stream
536 * \param stream The stream the audio is being requested from
537 * \param buf A buffer to fill with audio data
538 * \param len The maximum number of bytes to fill
539 * \return The number of bytes read from the stream, or -1 on error
541 * \sa SDL_NewAudioStream
542 * \sa SDL_AudioStreamPut
543 * \sa SDL_AudioStreamAvailable
544 * \sa SDL_AudioStreamFlush
545 * \sa SDL_AudioStreamClear
546 * \sa SDL_FreeAudioStream
548 extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len);
551 * Get the number of converted/resampled bytes available. The stream may be
552 * buffering data behind the scenes until it has enough to resample
553 * correctly, so this number might be lower than what you expect, or even
554 * be zero. Add more data or flush the stream if you need the data now.
556 * \sa SDL_NewAudioStream
557 * \sa SDL_AudioStreamPut
558 * \sa SDL_AudioStreamGet
559 * \sa SDL_AudioStreamFlush
560 * \sa SDL_AudioStreamClear
561 * \sa SDL_FreeAudioStream
563 extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream);
566 * Tell the stream that you're done sending data, and anything being buffered
567 * should be converted/resampled and made available immediately.
569 * It is legal to add more data to a stream after flushing, but there will
570 * be audio gaps in the output. Generally this is intended to signal the
571 * end of input, so the complete output becomes available.
573 * \sa SDL_NewAudioStream
574 * \sa SDL_AudioStreamPut
575 * \sa SDL_AudioStreamGet
576 * \sa SDL_AudioStreamAvailable
577 * \sa SDL_AudioStreamClear
578 * \sa SDL_FreeAudioStream
580 extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream);
583 * Clear any pending data in the stream without converting it
585 * \sa SDL_NewAudioStream
586 * \sa SDL_AudioStreamPut
587 * \sa SDL_AudioStreamGet
588 * \sa SDL_AudioStreamAvailable
589 * \sa SDL_AudioStreamFlush
590 * \sa SDL_FreeAudioStream
592 extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream);
595 * Free an audio stream
597 * \sa SDL_NewAudioStream
598 * \sa SDL_AudioStreamPut
599 * \sa SDL_AudioStreamGet
600 * \sa SDL_AudioStreamAvailable
601 * \sa SDL_AudioStreamFlush
602 * \sa SDL_AudioStreamClear
604 extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream);
606 #define SDL_MIX_MAXVOLUME 128
608 * This takes two audio buffers of the playing audio format and mixes
609 * them, performing addition, volume adjustment, and overflow clipping.
610 * The volume ranges from 0 - 128, and should be set to ::SDL_MIX_MAXVOLUME
611 * for full audio volume. Note this does not change hardware volume.
612 * This is provided for convenience -- you can mix your own audio data.
614 extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src,
615 Uint32 len, int volume);
618 * This works like SDL_MixAudio(), but you specify the audio format instead of
619 * using the format of audio device 1. Thus it can be used when no audio
620 * device is open at all.
622 extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst,
624 SDL_AudioFormat format,
625 Uint32 len, int volume);
628 * Queue more audio on non-callback devices.
630 * (If you are looking to retrieve queued audio from a non-callback capture
631 * device, you want SDL_DequeueAudio() instead. This will return -1 to
632 * signify an error if you use it with capture devices.)
634 * SDL offers two ways to feed audio to the device: you can either supply a
635 * callback that SDL triggers with some frequency to obtain more audio
636 * (pull method), or you can supply no callback, and then SDL will expect
637 * you to supply data at regular intervals (push method) with this function.
639 * There are no limits on the amount of data you can queue, short of
640 * exhaustion of address space. Queued data will drain to the device as
641 * necessary without further intervention from you. If the device needs
642 * audio but there is not enough queued, it will play silence to make up
643 * the difference. This means you will have skips in your audio playback
644 * if you aren't routinely queueing sufficient data.
646 * This function copies the supplied data, so you are safe to free it when
647 * the function returns. This function is thread-safe, but queueing to the
648 * same device from two threads at once does not promise which buffer will
651 * You may not queue audio on a device that is using an application-supplied
652 * callback; doing so returns an error. You have to use the audio callback
653 * or queue audio with this function, but not both.
655 * You should not call SDL_LockAudio() on the device before queueing; SDL
656 * handles locking internally for this function.
658 * \param dev The device ID to which we will queue audio.
659 * \param data The data to queue to the device for later playback.
660 * \param len The number of bytes (not samples!) to which (data) points.
661 * \return 0 on success, or -1 on error.
663 * \sa SDL_GetQueuedAudioSize
664 * \sa SDL_ClearQueuedAudio
666 extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len);
669 * Dequeue more audio on non-callback devices.
671 * (If you are looking to queue audio for output on a non-callback playback
672 * device, you want SDL_QueueAudio() instead. This will always return 0
673 * if you use it with playback devices.)
675 * SDL offers two ways to retrieve audio from a capture device: you can
676 * either supply a callback that SDL triggers with some frequency as the
677 * device records more audio data, (push method), or you can supply no
678 * callback, and then SDL will expect you to retrieve data at regular
679 * intervals (pull method) with this function.
681 * There are no limits on the amount of data you can queue, short of
682 * exhaustion of address space. Data from the device will keep queuing as
683 * necessary without further intervention from you. This means you will
684 * eventually run out of memory if you aren't routinely dequeueing data.
686 * Capture devices will not queue data when paused; if you are expecting
687 * to not need captured audio for some length of time, use
688 * SDL_PauseAudioDevice() to stop the capture device from queueing more
689 * data. This can be useful during, say, level loading times. When
690 * unpaused, capture devices will start queueing data from that point,
691 * having flushed any capturable data available while paused.
693 * This function is thread-safe, but dequeueing from the same device from
694 * two threads at once does not promise which thread will dequeued data
697 * You may not dequeue audio from a device that is using an
698 * application-supplied callback; doing so returns an error. You have to use
699 * the audio callback, or dequeue audio with this function, but not both.
701 * You should not call SDL_LockAudio() on the device before queueing; SDL
702 * handles locking internally for this function.
704 * \param dev The device ID from which we will dequeue audio.
705 * \param data A pointer into where audio data should be copied.
706 * \param len The number of bytes (not samples!) to which (data) points.
707 * \return number of bytes dequeued, which could be less than requested.
709 * \sa SDL_GetQueuedAudioSize
710 * \sa SDL_ClearQueuedAudio
712 extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len);
715 * Get the number of bytes of still-queued audio.
717 * For playback device:
719 * This is the number of bytes that have been queued for playback with
720 * SDL_QueueAudio(), but have not yet been sent to the hardware. This
721 * number may shrink at any time, so this only informs of pending data.
723 * Once we've sent it to the hardware, this function can not decide the
724 * exact byte boundary of what has been played. It's possible that we just
725 * gave the hardware several kilobytes right before you called this
726 * function, but it hasn't played any of it yet, or maybe half of it, etc.
728 * For capture devices:
730 * This is the number of bytes that have been captured by the device and
731 * are waiting for you to dequeue. This number may grow at any time, so
732 * this only informs of the lower-bound of available data.
734 * You may not queue audio on a device that is using an application-supplied
735 * callback; calling this function on such a device always returns 0.
736 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
737 * the audio callback, but not both.
739 * You should not call SDL_LockAudio() on the device before querying; SDL
740 * handles locking internally for this function.
742 * \param dev The device ID of which we will query queued audio size.
743 * \return Number of bytes (not samples!) of queued audio.
746 * \sa SDL_ClearQueuedAudio
748 extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev);
751 * Drop any queued audio data. For playback devices, this is any queued data
752 * still waiting to be submitted to the hardware. For capture devices, this
753 * is any data that was queued by the device that hasn't yet been dequeued by
756 * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For
757 * playback devices, the hardware will start playing silence if more audio
758 * isn't queued. Unpaused capture devices will start filling the queue again
759 * as soon as they have more data available (which, depending on the state
760 * of the hardware and the thread, could be before this function call
763 * This will not prevent playback of queued audio that's already been sent
764 * to the hardware, as we can not undo that, so expect there to be some
765 * fraction of a second of audio that might still be heard. This can be
766 * useful if you want to, say, drop any pending music during a level change
769 * You may not queue audio on a device that is using an application-supplied
770 * callback; calling this function on such a device is always a no-op.
771 * You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use
772 * the audio callback, but not both.
774 * You should not call SDL_LockAudio() on the device before clearing the
775 * queue; SDL handles locking internally for this function.
777 * This function always succeeds and thus returns void.
779 * \param dev The device ID of which to clear the audio queue.
782 * \sa SDL_GetQueuedAudioSize
784 extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev);
788 * \name Audio lock functions
790 * The lock manipulated by these functions protects the callback function.
791 * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that
792 * the callback function is not running. Do not call these from the callback
793 * function or you will cause deadlock.
796 extern DECLSPEC void SDLCALL SDL_LockAudio(void);
797 extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev);
798 extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
799 extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev);
800 /* @} *//* Audio lock functions */
803 * This function shuts down audio processing and closes the audio device.
805 extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
806 extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev);
808 /* Ends C function definitions when using C++ */
812 #include "close_code.h"
814 #endif /* SDL_audio_h_ */
816 /* vi: set ts=4 sw=4 expandtab: */