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Diffstat (limited to 'dependencies/include/SDL/SDL_audio.h')
| -rw-r--r-- | dependencies/include/SDL/SDL_audio.h | 1500 |
1 files changed, 1500 insertions, 0 deletions
diff --git a/dependencies/include/SDL/SDL_audio.h b/dependencies/include/SDL/SDL_audio.h new file mode 100644 index 0000000..c42de3e --- /dev/null +++ b/dependencies/include/SDL/SDL_audio.h @@ -0,0 +1,1500 @@ +/* + Simple DirectMedia Layer + Copyright (C) 1997-2022 Sam Lantinga <slouken@libsdl.org> + + This software is provided 'as-is', without any express or implied + warranty. In no event will the authors be held liable for any damages + arising from the use of this software. + + Permission is granted to anyone to use this software for any purpose, + including commercial applications, and to alter it and redistribute it + freely, subject to the following restrictions: + + 1. The origin of this software must not be misrepresented; you must not + claim that you wrote the original software. If you use this software + in a product, an acknowledgment in the product documentation would be + appreciated but is not required. + 2. Altered source versions must be plainly marked as such, and must not be + misrepresented as being the original software. + 3. This notice may not be removed or altered from any source distribution. +*/ + +/* !!! FIXME: several functions in here need Doxygen comments. */ + +/** + * \file SDL_audio.h + * + * Access to the raw audio mixing buffer for the SDL library. + */ + +#ifndef SDL_audio_h_ +#define SDL_audio_h_ + +#include "SDL_stdinc.h" +#include "SDL_error.h" +#include "SDL_endian.h" +#include "SDL_mutex.h" +#include "SDL_thread.h" +#include "SDL_rwops.h" + +#include "begin_code.h" +/* Set up for C function definitions, even when using C++ */ +#ifdef __cplusplus +extern "C" { +#endif + +/** + * \brief Audio format flags. + * + * These are what the 16 bits in SDL_AudioFormat currently mean... + * (Unspecified bits are always zero). + * + * \verbatim + ++-----------------------sample is signed if set + || + || ++-----------sample is bigendian if set + || || + || || ++---sample is float if set + || || || + || || || +---sample bit size---+ + || || || | | + 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00 + \endverbatim + * + * There are macros in SDL 2.0 and later to query these bits. + */ +typedef Uint16 SDL_AudioFormat; + +/** + * \name Audio flags + */ +/* @{ */ + +#define SDL_AUDIO_MASK_BITSIZE (0xFF) +#define SDL_AUDIO_MASK_DATATYPE (1<<8) +#define SDL_AUDIO_MASK_ENDIAN (1<<12) +#define SDL_AUDIO_MASK_SIGNED (1<<15) +#define SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE) +#define SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE) +#define SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN) +#define SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED) +#define SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x)) +#define SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x)) +#define SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x)) + +/** + * \name Audio format flags + * + * Defaults to LSB byte order. + */ +/* @{ */ +#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */ +#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */ +#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */ +#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */ +#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */ +#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */ +#define AUDIO_U16 AUDIO_U16LSB +#define AUDIO_S16 AUDIO_S16LSB +/* @} */ + +/** + * \name int32 support + */ +/* @{ */ +#define AUDIO_S32LSB 0x8020 /**< 32-bit integer samples */ +#define AUDIO_S32MSB 0x9020 /**< As above, but big-endian byte order */ +#define AUDIO_S32 AUDIO_S32LSB +/* @} */ + +/** + * \name float32 support + */ +/* @{ */ +#define AUDIO_F32LSB 0x8120 /**< 32-bit floating point samples */ +#define AUDIO_F32MSB 0x9120 /**< As above, but big-endian byte order */ +#define AUDIO_F32 AUDIO_F32LSB +/* @} */ + +/** + * \name Native audio byte ordering + */ +/* @{ */ +#if SDL_BYTEORDER == SDL_LIL_ENDIAN +#define AUDIO_U16SYS AUDIO_U16LSB +#define AUDIO_S16SYS AUDIO_S16LSB +#define AUDIO_S32SYS AUDIO_S32LSB +#define AUDIO_F32SYS AUDIO_F32LSB +#else +#define AUDIO_U16SYS AUDIO_U16MSB +#define AUDIO_S16SYS AUDIO_S16MSB +#define AUDIO_S32SYS AUDIO_S32MSB +#define AUDIO_F32SYS AUDIO_F32MSB +#endif +/* @} */ + +/** + * \name Allow change flags + * + * Which audio format changes are allowed when opening a device. + */ +/* @{ */ +#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001 +#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002 +#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004 +#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008 +#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) +/* @} */ + +/* @} *//* Audio flags */ + +/** + * This function is called when the audio device needs more data. + * + * \param userdata An application-specific parameter saved in + * the SDL_AudioSpec structure + * \param stream A pointer to the audio data buffer. + * \param len The length of that buffer in bytes. + * + * Once the callback returns, the buffer will no longer be valid. + * Stereo samples are stored in a LRLRLR ordering. + * + * You can choose to avoid callbacks and use SDL_QueueAudio() instead, if + * you like. Just open your audio device with a NULL callback. + */ +typedef void (SDLCALL * SDL_AudioCallback) (void *userdata, Uint8 * stream, + int len); + +/** + * The calculated values in this structure are calculated by SDL_OpenAudio(). + * + * For multi-channel audio, the default SDL channel mapping is: + * 2: FL FR (stereo) + * 3: FL FR LFE (2.1 surround) + * 4: FL FR BL BR (quad) + * 5: FL FR LFE BL BR (4.1 surround) + * 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR) + * 7: FL FR FC LFE BC SL SR (6.1 surround) + * 8: FL FR FC LFE BL BR SL SR (7.1 surround) + */ +typedef struct SDL_AudioSpec +{ + int freq; /**< DSP frequency -- samples per second */ + SDL_AudioFormat format; /**< Audio data format */ + Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */ + Uint8 silence; /**< Audio buffer silence value (calculated) */ + Uint16 samples; /**< Audio buffer size in sample FRAMES (total samples divided by channel count) */ + Uint16 padding; /**< Necessary for some compile environments */ + Uint32 size; /**< Audio buffer size in bytes (calculated) */ + SDL_AudioCallback callback; /**< Callback that feeds the audio device (NULL to use SDL_QueueAudio()). */ + void *userdata; /**< Userdata passed to callback (ignored for NULL callbacks). */ +} SDL_AudioSpec; + + +struct SDL_AudioCVT; +typedef void (SDLCALL * SDL_AudioFilter) (struct SDL_AudioCVT * cvt, + SDL_AudioFormat format); + +/** + * \brief Upper limit of filters in SDL_AudioCVT + * + * The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is + * currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers, + * one of which is the terminating NULL pointer. + */ +#define SDL_AUDIOCVT_MAX_FILTERS 9 + +/** + * \struct SDL_AudioCVT + * \brief A structure to hold a set of audio conversion filters and buffers. + * + * Note that various parts of the conversion pipeline can take advantage + * of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require + * you to pass it aligned data, but can possibly run much faster if you + * set both its (buf) field to a pointer that is aligned to 16 bytes, and its + * (len) field to something that's a multiple of 16, if possible. + */ +#if defined(__GNUC__) && !defined(__CHERI_PURE_CAPABILITY__) +/* This structure is 84 bytes on 32-bit architectures, make sure GCC doesn't + pad it out to 88 bytes to guarantee ABI compatibility between compilers. + This is not a concern on CHERI architectures, where pointers must be stored + at aligned locations otherwise they will become invalid, and thus structs + containing pointers cannot be packed without giving a warning or error. + vvv + The next time we rev the ABI, make sure to size the ints and add padding. +*/ +#define SDL_AUDIOCVT_PACKED __attribute__((packed)) +#else +#define SDL_AUDIOCVT_PACKED +#endif +/* */ +typedef struct SDL_AudioCVT +{ + int needed; /**< Set to 1 if conversion possible */ + SDL_AudioFormat src_format; /**< Source audio format */ + SDL_AudioFormat dst_format; /**< Target audio format */ + double rate_incr; /**< Rate conversion increment */ + Uint8 *buf; /**< Buffer to hold entire audio data */ + int len; /**< Length of original audio buffer */ + int len_cvt; /**< Length of converted audio buffer */ + int len_mult; /**< buffer must be len*len_mult big */ + double len_ratio; /**< Given len, final size is len*len_ratio */ + SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS + 1]; /**< NULL-terminated list of filter functions */ + int filter_index; /**< Current audio conversion function */ +} SDL_AUDIOCVT_PACKED SDL_AudioCVT; + + +/* Function prototypes */ + +/** + * \name Driver discovery functions + * + * These functions return the list of built in audio drivers, in the + * order that they are normally initialized by default. + */ +/* @{ */ + +/** + * Use this function to get the number of built-in audio drivers. + * + * This function returns a hardcoded number. This never returns a negative + * value; if there are no drivers compiled into this build of SDL, this + * function returns zero. The presence of a driver in this list does not mean + * it will function, it just means SDL is capable of interacting with that + * interface. For example, a build of SDL might have esound support, but if + * there's no esound server available, SDL's esound driver would fail if used. + * + * By default, SDL tries all drivers, in its preferred order, until one is + * found to be usable. + * + * \returns the number of built-in audio drivers. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_GetAudioDriver + */ +extern DECLSPEC int SDLCALL SDL_GetNumAudioDrivers(void); + +/** + * Use this function to get the name of a built in audio driver. + * + * The list of audio drivers is given in the order that they are normally + * initialized by default; the drivers that seem more reasonable to choose + * first (as far as the SDL developers believe) are earlier in the list. + * + * The names of drivers are all simple, low-ASCII identifiers, like "alsa", + * "coreaudio" or "xaudio2". These never have Unicode characters, and are not + * meant to be proper names. + * + * \param index the index of the audio driver; the value ranges from 0 to + * SDL_GetNumAudioDrivers() - 1 + * \returns the name of the audio driver at the requested index, or NULL if an + * invalid index was specified. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_GetNumAudioDrivers + */ +extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index); +/* @} */ + +/** + * \name Initialization and cleanup + * + * \internal These functions are used internally, and should not be used unless + * you have a specific need to specify the audio driver you want to + * use. You should normally use SDL_Init() or SDL_InitSubSystem(). + */ +/* @{ */ + +/** + * Use this function to initialize a particular audio driver. + * + * This function is used internally, and should not be used unless you have a + * specific need to designate the audio driver you want to use. You should + * normally use SDL_Init() or SDL_InitSubSystem(). + * + * \param driver_name the name of the desired audio driver + * \returns 0 on success or a negative error code on failure; call + * SDL_GetError() for more information. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_AudioQuit + */ +extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name); + +/** + * Use this function to shut down audio if you initialized it with + * SDL_AudioInit(). + * + * This function is used internally, and should not be used unless you have a + * specific need to specify the audio driver you want to use. You should + * normally use SDL_Quit() or SDL_QuitSubSystem(). + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_AudioInit + */ +extern DECLSPEC void SDLCALL SDL_AudioQuit(void); +/* @} */ + +/** + * Get the name of the current audio driver. + * + * The returned string points to internal static memory and thus never becomes + * invalid, even if you quit the audio subsystem and initialize a new driver + * (although such a case would return a different static string from another + * call to this function, of course). As such, you should not modify or free + * the returned string. + * + * \returns the name of the current audio driver or NULL if no driver has been + * initialized. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_AudioInit + */ +extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void); + +/** + * This function is a legacy means of opening the audio device. + * + * This function remains for compatibility with SDL 1.2, but also because it's + * slightly easier to use than the new functions in SDL 2.0. The new, more + * powerful, and preferred way to do this is SDL_OpenAudioDevice(). + * + * This function is roughly equivalent to: + * + * ```c + * SDL_OpenAudioDevice(NULL, 0, desired, obtained, SDL_AUDIO_ALLOW_ANY_CHANGE); + * ``` + * + * With two notable exceptions: + * + * - If `obtained` is NULL, we use `desired` (and allow no changes), which + * means desired will be modified to have the correct values for silence, + * etc, and SDL will convert any differences between your app's specific + * request and the hardware behind the scenes. + * - The return value is always success or failure, and not a device ID, which + * means you can only have one device open at a time with this function. + * + * \param desired an SDL_AudioSpec structure representing the desired output + * format. Please refer to the SDL_OpenAudioDevice + * documentation for details on how to prepare this structure. + * \param obtained an SDL_AudioSpec structure filled in with the actual + * parameters, or NULL. + * \returns 0 if successful, placing the actual hardware parameters in the + * structure pointed to by `obtained`. + * + * If `obtained` is NULL, the audio data passed to the callback + * function will be guaranteed to be in the requested format, and + * will be automatically converted to the actual hardware audio + * format if necessary. If `obtained` is NULL, `desired` will have + * fields modified. + * + * This function returns a negative error code on failure to open the + * audio device or failure to set up the audio thread; call + * SDL_GetError() for more information. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_CloseAudio + * \sa SDL_LockAudio + * \sa SDL_PauseAudio + * \sa SDL_UnlockAudio + */ +extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec * desired, + SDL_AudioSpec * obtained); + +/** + * SDL Audio Device IDs. + * + * A successful call to SDL_OpenAudio() is always device id 1, and legacy + * SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls + * always returns devices >= 2 on success. The legacy calls are good both + * for backwards compatibility and when you don't care about multiple, + * specific, or capture devices. + */ +typedef Uint32 SDL_AudioDeviceID; + +/** + * Get the number of built-in audio devices. + * + * This function is only valid after successfully initializing the audio + * subsystem. + * + * Note that audio capture support is not implemented as of SDL 2.0.4, so the + * `iscapture` parameter is for future expansion and should always be zero for + * now. + * + * This function will return -1 if an explicit list of devices can't be + * determined. Returning -1 is not an error. For example, if SDL is set up to + * talk to a remote audio server, it can't list every one available on the + * Internet, but it will still allow a specific host to be specified in + * SDL_OpenAudioDevice(). + * + * In many common cases, when this function returns a value <= 0, it can still + * successfully open the default device (NULL for first argument of + * SDL_OpenAudioDevice()). + * + * This function may trigger a complete redetect of available hardware. It + * should not be called for each iteration of a loop, but rather once at the + * start of a loop: + * + * ```c + * // Don't do this: + * for (int i = 0; i < SDL_GetNumAudioDevices(0); i++) + * + * // do this instead: + * const int count = SDL_GetNumAudioDevices(0); + * for (int i = 0; i < count; ++i) { do_something_here(); } + * ``` + * + * \param iscapture zero to request playback devices, non-zero to request + * recording devices + * \returns the number of available devices exposed by the current driver or + * -1 if an explicit list of devices can't be determined. A return + * value of -1 does not necessarily mean an error condition. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_GetAudioDeviceName + * \sa SDL_OpenAudioDevice + */ +extern DECLSPEC int SDLCALL SDL_GetNumAudioDevices(int iscapture); + +/** + * Get the human-readable name of a specific audio device. + * + * This function is only valid after successfully initializing the audio + * subsystem. The values returned by this function reflect the latest call to + * SDL_GetNumAudioDevices(); re-call that function to redetect available + * hardware. + * + * The string returned by this function is UTF-8 encoded, read-only, and + * managed internally. You are not to free it. If you need to keep the string + * for any length of time, you should make your own copy of it, as it will be + * invalid next time any of several other SDL functions are called. + * + * \param index the index of the audio device; valid values range from 0 to + * SDL_GetNumAudioDevices() - 1 + * \param iscapture non-zero to query the list of recording devices, zero to + * query the list of output devices. + * \returns the name of the audio device at the requested index, or NULL on + * error. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_GetNumAudioDevices + * \sa SDL_GetDefaultAudioInfo + */ +extern DECLSPEC const char *SDLCALL SDL_GetAudioDeviceName(int index, + int iscapture); + +/** + * Get the preferred audio format of a specific audio device. + * + * This function is only valid after a successfully initializing the audio + * subsystem. The values returned by this function reflect the latest call to + * SDL_GetNumAudioDevices(); re-call that function to redetect available + * hardware. + * + * `spec` will be filled with the sample rate, sample format, and channel + * count. + * + * \param index the index of the audio device; valid values range from 0 to + * SDL_GetNumAudioDevices() - 1 + * \param iscapture non-zero to query the list of recording devices, zero to + * query the list of output devices. + * \param spec The SDL_AudioSpec to be initialized by this function. + * \returns 0 on success, nonzero on error + * + * \since This function is available since SDL 2.0.16. + * + * \sa SDL_GetNumAudioDevices + * \sa SDL_GetDefaultAudioInfo + */ +extern DECLSPEC int SDLCALL SDL_GetAudioDeviceSpec(int index, + int iscapture, + SDL_AudioSpec *spec); + + +/** + * Get the name and preferred format of the default audio device. + * + * Some (but not all!) platforms have an isolated mechanism to get information + * about the "default" device. This can actually be a completely different + * device that's not in the list you get from SDL_GetAudioDeviceSpec(). It can + * even be a network address! (This is discussed in SDL_OpenAudioDevice().) + * + * As a result, this call is not guaranteed to be performant, as it can query + * the sound server directly every time, unlike the other query functions. You + * should call this function sparingly! + * + * `spec` will be filled with the sample rate, sample format, and channel + * count, if a default device exists on the system. If `name` is provided, + * will be filled with either a dynamically-allocated UTF-8 string or NULL. + * + * \param name A pointer to be filled with the name of the default device (can + * be NULL). Please call SDL_free() when you are done with this + * pointer! + * \param spec The SDL_AudioSpec to be initialized by this function. + * \param iscapture non-zero to query the default recording device, zero to + * query the default output device. + * \returns 0 on success, nonzero on error + * + * \since This function is available since SDL 2.24.0. + * + * \sa SDL_GetAudioDeviceName + * \sa SDL_GetAudioDeviceSpec + * \sa SDL_OpenAudioDevice + */ +extern DECLSPEC int SDLCALL SDL_GetDefaultAudioInfo(char **name, + SDL_AudioSpec *spec, + int iscapture); + + +/** + * Open a specific audio device. + * + * SDL_OpenAudio(), unlike this function, always acts on device ID 1. As such, + * this function will never return a 1 so as not to conflict with the legacy + * function. + * + * Please note that SDL 2.0 before 2.0.5 did not support recording; as such, + * this function would fail if `iscapture` was not zero. Starting with SDL + * 2.0.5, recording is implemented and this value can be non-zero. + * + * Passing in a `device` name of NULL requests the most reasonable default + * (and is equivalent to what SDL_OpenAudio() does to choose a device). The + * `device` name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but + * some drivers allow arbitrary and driver-specific strings, such as a + * hostname/IP address for a remote audio server, or a filename in the + * diskaudio driver. + * + * An opened audio device starts out paused, and should be enabled for playing + * by calling SDL_PauseAudioDevice(devid, 0) when you are ready for your audio + * callback function to be called. Since the audio driver may modify the + * requested size of the audio buffer, you should allocate any local mixing + * buffers after you open the audio device. + * + * The audio callback runs in a separate thread in most cases; you can prevent + * race conditions between your callback and other threads without fully + * pausing playback with SDL_LockAudioDevice(). For more information about the + * callback, see SDL_AudioSpec. + * + * Managing the audio spec via 'desired' and 'obtained': + * + * When filling in the desired audio spec structure: + * + * - `desired->freq` should be the frequency in sample-frames-per-second (Hz). + * - `desired->format` should be the audio format (`AUDIO_S16SYS`, etc). + * - `desired->samples` is the desired size of the audio buffer, in _sample + * frames_ (with stereo output, two samples--left and right--would make a + * single sample frame). This number should be a power of two, and may be + * adjusted by the audio driver to a value more suitable for the hardware. + * Good values seem to range between 512 and 8096 inclusive, depending on + * the application and CPU speed. Smaller values reduce latency, but can + * lead to underflow if the application is doing heavy processing and cannot + * fill the audio buffer in time. Note that the number of sample frames is + * directly related to time by the following formula: `ms = + * (sampleframes*1000)/freq` + * - `desired->size` is the size in _bytes_ of the audio buffer, and is + * calculated by SDL_OpenAudioDevice(). You don't initialize this. + * - `desired->silence` is the value used to set the buffer to silence, and is + * calculated by SDL_OpenAudioDevice(). You don't initialize this. + * - `desired->callback` should be set to a function that will be called when + * the audio device is ready for more data. It is passed a pointer to the + * audio buffer, and the length in bytes of the audio buffer. This function + * usually runs in a separate thread, and so you should protect data + * structures that it accesses by calling SDL_LockAudioDevice() and + * SDL_UnlockAudioDevice() in your code. Alternately, you may pass a NULL + * pointer here, and call SDL_QueueAudio() with some frequency, to queue + * more audio samples to be played (or for capture devices, call + * SDL_DequeueAudio() with some frequency, to obtain audio samples). + * - `desired->userdata` is passed as the first parameter to your callback + * function. If you passed a NULL callback, this value is ignored. + * + * `allowed_changes` can have the following flags OR'd together: + * + * - `SDL_AUDIO_ALLOW_FREQUENCY_CHANGE` + * - `SDL_AUDIO_ALLOW_FORMAT_CHANGE` + * - `SDL_AUDIO_ALLOW_CHANNELS_CHANGE` + * - `SDL_AUDIO_ALLOW_SAMPLES_CHANGE` + * - `SDL_AUDIO_ALLOW_ANY_CHANGE` + * + * These flags specify how SDL should behave when a device cannot offer a + * specific feature. If the application requests a feature that the hardware + * doesn't offer, SDL will always try to get the closest equivalent. + * + * For example, if you ask for float32 audio format, but the sound card only + * supports int16, SDL will set the hardware to int16. If you had set + * SDL_AUDIO_ALLOW_FORMAT_CHANGE, SDL will change the format in the `obtained` + * structure. If that flag was *not* set, SDL will prepare to convert your + * callback's float32 audio to int16 before feeding it to the hardware and + * will keep the originally requested format in the `obtained` structure. + * + * The resulting audio specs, varying depending on hardware and on what + * changes were allowed, will then be written back to `obtained`. + * + * If your application can only handle one specific data format, pass a zero + * for `allowed_changes` and let SDL transparently handle any differences. + * + * \param device a UTF-8 string reported by SDL_GetAudioDeviceName() or a + * driver-specific name as appropriate. NULL requests the most + * reasonable default device. + * \param iscapture non-zero to specify a device should be opened for + * recording, not playback + * \param desired an SDL_AudioSpec structure representing the desired output + * format; see SDL_OpenAudio() for more information + * \param obtained an SDL_AudioSpec structure filled in with the actual output + * format; see SDL_OpenAudio() for more information + * \param allowed_changes 0, or one or more flags OR'd together + * \returns a valid device ID that is > 0 on success or 0 on failure; call + * SDL_GetError() for more information. + * + * For compatibility with SDL 1.2, this will never return 1, since + * SDL reserves that ID for the legacy SDL_OpenAudio() function. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_CloseAudioDevice + * \sa SDL_GetAudioDeviceName + * \sa SDL_LockAudioDevice + * \sa SDL_OpenAudio + * \sa SDL_PauseAudioDevice + * \sa SDL_UnlockAudioDevice + */ +extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice( + const char *device, + int iscapture, + const SDL_AudioSpec *desired, + SDL_AudioSpec *obtained, + int allowed_changes); + + + +/** + * \name Audio state + * + * Get the current audio state. + */ +/* @{ */ +typedef enum +{ + SDL_AUDIO_STOPPED = 0, + SDL_AUDIO_PLAYING, + SDL_AUDIO_PAUSED +} SDL_AudioStatus; + +/** + * This function is a legacy means of querying the audio device. + * + * New programs might want to use SDL_GetAudioDeviceStatus() instead. This + * function is equivalent to calling... + * + * ```c + * SDL_GetAudioDeviceStatus(1); + * ``` + * + * ...and is only useful if you used the legacy SDL_OpenAudio() function. + * + * \returns the SDL_AudioStatus of the audio device opened by SDL_OpenAudio(). + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_GetAudioDeviceStatus + */ +extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioStatus(void); + +/** + * Use this function to get the current audio state of an audio device. + * + * \param dev the ID of an audio device previously opened with + * SDL_OpenAudioDevice() + * \returns the SDL_AudioStatus of the specified audio device. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_PauseAudioDevice + */ +extern DECLSPEC SDL_AudioStatus SDLCALL SDL_GetAudioDeviceStatus(SDL_AudioDeviceID dev); +/* @} *//* Audio State */ + +/** + * \name Pause audio functions + * + * These functions pause and unpause the audio callback processing. + * They should be called with a parameter of 0 after opening the audio + * device to start playing sound. This is so you can safely initialize + * data for your callback function after opening the audio device. + * Silence will be written to the audio device during the pause. + */ +/* @{ */ + +/** + * This function is a legacy means of pausing the audio device. + * + * New programs might want to use SDL_PauseAudioDevice() instead. This + * function is equivalent to calling... + * + * ```c + * SDL_PauseAudioDevice(1, pause_on); + * ``` + * + * ...and is only useful if you used the legacy SDL_OpenAudio() function. + * + * \param pause_on non-zero to pause, 0 to unpause + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_GetAudioStatus + * \sa SDL_PauseAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); + +/** + * Use this function to pause and unpause audio playback on a specified + * device. + * + * This function pauses and unpauses the audio callback processing for a given + * device. Newly-opened audio devices start in the paused state, so you must + * call this function with **pause_on**=0 after opening the specified audio + * device to start playing sound. This allows you to safely initialize data + * for your callback function after opening the audio device. Silence will be + * written to the audio device while paused, and the audio callback is + * guaranteed to not be called. Pausing one device does not prevent other + * unpaused devices from running their callbacks. + * + * Pausing state does not stack; even if you pause a device several times, a + * single unpause will start the device playing again, and vice versa. This is + * different from how SDL_LockAudioDevice() works. + * + * If you just need to protect a few variables from race conditions vs your + * callback, you shouldn't pause the audio device, as it will lead to dropouts + * in the audio playback. Instead, you should use SDL_LockAudioDevice(). + * + * \param dev a device opened by SDL_OpenAudioDevice() + * \param pause_on non-zero to pause, 0 to unpause + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_LockAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev, + int pause_on); +/* @} *//* Pause audio functions */ + +/** + * Load the audio data of a WAVE file into memory. + * + * Loading a WAVE file requires `src`, `spec`, `audio_buf` and `audio_len` to + * be valid pointers. The entire data portion of the file is then loaded into + * memory and decoded if necessary. + * + * If `freesrc` is non-zero, the data source gets automatically closed and + * freed before the function returns. + * + * Supported formats are RIFF WAVE files with the formats PCM (8, 16, 24, and + * 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and + * A-law and mu-law (8 bits). Other formats are currently unsupported and + * cause an error. + * + * If this function succeeds, the pointer returned by it is equal to `spec` + * and the pointer to the audio data allocated by the function is written to + * `audio_buf` and its length in bytes to `audio_len`. The SDL_AudioSpec + * members `freq`, `channels`, and `format` are set to the values of the audio + * data in the buffer. The `samples` member is set to a sane default and all + * others are set to zero. + * + * It's necessary to use SDL_FreeWAV() to free the audio data returned in + * `audio_buf` when it is no longer used. + * + * Because of the underspecification of the .WAV format, there are many + * problematic files in the wild that cause issues with strict decoders. To + * provide compatibility with these files, this decoder is lenient in regards + * to the truncation of the file, the fact chunk, and the size of the RIFF + * chunk. The hints `SDL_HINT_WAVE_RIFF_CHUNK_SIZE`, + * `SDL_HINT_WAVE_TRUNCATION`, and `SDL_HINT_WAVE_FACT_CHUNK` can be used to + * tune the behavior of the loading process. + * + * Any file that is invalid (due to truncation, corruption, or wrong values in + * the headers), too big, or unsupported causes an error. Additionally, any + * critical I/O error from the data source will terminate the loading process + * with an error. The function returns NULL on error and in all cases (with + * the exception of `src` being NULL), an appropriate error message will be + * set. + * + * It is required that the data source supports seeking. + * + * Example: + * + * ```c + * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, &spec, &buf, &len); + * ``` + * + * Note that the SDL_LoadWAV macro does this same thing for you, but in a less + * messy way: + * + * ```c + * SDL_LoadWAV("sample.wav", &spec, &buf, &len); + * ``` + * + * \param src The data source for the WAVE data + * \param freesrc If non-zero, SDL will _always_ free the data source + * \param spec An SDL_AudioSpec that will be filled in with the wave file's + * format details + * \param audio_buf A pointer filled with the audio data, allocated by the + * function. + * \param audio_len A pointer filled with the length of the audio data buffer + * in bytes + * \returns This function, if successfully called, returns `spec`, which will + * be filled with the audio data format of the wave source data. + * `audio_buf` will be filled with a pointer to an allocated buffer + * containing the audio data, and `audio_len` is filled with the + * length of that audio buffer in bytes. + * + * This function returns NULL if the .WAV file cannot be opened, uses + * an unknown data format, or is corrupt; call SDL_GetError() for + * more information. + * + * When the application is done with the data returned in + * `audio_buf`, it should call SDL_FreeWAV() to dispose of it. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_FreeWAV + * \sa SDL_LoadWAV + */ +extern DECLSPEC SDL_AudioSpec *SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, + int freesrc, + SDL_AudioSpec * spec, + Uint8 ** audio_buf, + Uint32 * audio_len); + +/** + * Loads a WAV from a file. + * Compatibility convenience function. + */ +#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \ + SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len) + +/** + * Free data previously allocated with SDL_LoadWAV() or SDL_LoadWAV_RW(). + * + * After a WAVE file has been opened with SDL_LoadWAV() or SDL_LoadWAV_RW() + * its data can eventually be freed with SDL_FreeWAV(). It is safe to call + * this function with a NULL pointer. + * + * \param audio_buf a pointer to the buffer created by SDL_LoadWAV() or + * SDL_LoadWAV_RW() + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_LoadWAV + * \sa SDL_LoadWAV_RW + */ +extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 * audio_buf); + +/** + * Initialize an SDL_AudioCVT structure for conversion. + * + * Before an SDL_AudioCVT structure can be used to convert audio data it must + * be initialized with source and destination information. + * + * This function will zero out every field of the SDL_AudioCVT, so it must be + * called before the application fills in the final buffer information. + * + * Once this function has returned successfully, and reported that a + * conversion is necessary, the application fills in the rest of the fields in + * SDL_AudioCVT, now that it knows how large a buffer it needs to allocate, + * and then can call SDL_ConvertAudio() to complete the conversion. + * + * \param cvt an SDL_AudioCVT structure filled in with audio conversion + * information + * \param src_format the source format of the audio data; for more info see + * SDL_AudioFormat + * \param src_channels the number of channels in the source + * \param src_rate the frequency (sample-frames-per-second) of the source + * \param dst_format the destination format of the audio data; for more info + * see SDL_AudioFormat + * \param dst_channels the number of channels in the destination + * \param dst_rate the frequency (sample-frames-per-second) of the destination + * \returns 1 if the audio filter is prepared, 0 if no conversion is needed, + * or a negative error code on failure; call SDL_GetError() for more + * information. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_ConvertAudio + */ +extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT * cvt, + SDL_AudioFormat src_format, + Uint8 src_channels, + int src_rate, + SDL_AudioFormat dst_format, + Uint8 dst_channels, + int dst_rate); + +/** + * Convert audio data to a desired audio format. + * + * This function does the actual audio data conversion, after the application + * has called SDL_BuildAudioCVT() to prepare the conversion information and + * then filled in the buffer details. + * + * Once the application has initialized the `cvt` structure using + * SDL_BuildAudioCVT(), allocated an audio buffer and filled it with audio + * data in the source format, this function will convert the buffer, in-place, + * to the desired format. + * + * The data conversion may go through several passes; any given pass may + * possibly temporarily increase the size of the data. For example, SDL might + * expand 16-bit data to 32 bits before resampling to a lower frequency, + * shrinking the data size after having grown it briefly. Since the supplied + * buffer will be both the source and destination, converting as necessary + * in-place, the application must allocate a buffer that will fully contain + * the data during its largest conversion pass. After SDL_BuildAudioCVT() + * returns, the application should set the `cvt->len` field to the size, in + * bytes, of the source data, and allocate a buffer that is `cvt->len * + * cvt->len_mult` bytes long for the `buf` field. + * + * The source data should be copied into this buffer before the call to + * SDL_ConvertAudio(). Upon successful return, this buffer will contain the + * converted audio, and `cvt->len_cvt` will be the size of the converted data, + * in bytes. Any bytes in the buffer past `cvt->len_cvt` are undefined once + * this function returns. + * + * \param cvt an SDL_AudioCVT structure that was previously set up by + * SDL_BuildAudioCVT(). + * \returns 0 if the conversion was completed successfully or a negative error + * code on failure; call SDL_GetError() for more information. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_BuildAudioCVT + */ +extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT * cvt); + +/* SDL_AudioStream is a new audio conversion interface. + The benefits vs SDL_AudioCVT: + - it can handle resampling data in chunks without generating + artifacts, when it doesn't have the complete buffer available. + - it can handle incoming data in any variable size. + - You push data as you have it, and pull it when you need it + */ +/* this is opaque to the outside world. */ +struct _SDL_AudioStream; +typedef struct _SDL_AudioStream SDL_AudioStream; + +/** + * Create a new audio stream. + * + * \param src_format The format of the source audio + * \param src_channels The number of channels of the source audio + * \param src_rate The sampling rate of the source audio + * \param dst_format The format of the desired audio output + * \param dst_channels The number of channels of the desired audio output + * \param dst_rate The sampling rate of the desired audio output + * \returns 0 on success, or -1 on error. + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_AudioStreamPut + * \sa SDL_AudioStreamGet + * \sa SDL_AudioStreamAvailable + * \sa SDL_AudioStreamFlush + * \sa SDL_AudioStreamClear + * \sa SDL_FreeAudioStream + */ +extern DECLSPEC SDL_AudioStream * SDLCALL SDL_NewAudioStream(const SDL_AudioFormat src_format, + const Uint8 src_channels, + const int src_rate, + const SDL_AudioFormat dst_format, + const Uint8 dst_channels, + const int dst_rate); + +/** + * Add data to be converted/resampled to the stream. + * + * \param stream The stream the audio data is being added to + * \param buf A pointer to the audio data to add + * \param len The number of bytes to write to the stream + * \returns 0 on success, or -1 on error. + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_NewAudioStream + * \sa SDL_AudioStreamGet + * \sa SDL_AudioStreamAvailable + * \sa SDL_AudioStreamFlush + * \sa SDL_AudioStreamClear + * \sa SDL_FreeAudioStream + */ +extern DECLSPEC int SDLCALL SDL_AudioStreamPut(SDL_AudioStream *stream, const void *buf, int len); + +/** + * Get converted/resampled data from the stream + * + * \param stream The stream the audio is being requested from + * \param buf A buffer to fill with audio data + * \param len The maximum number of bytes to fill + * \returns the number of bytes read from the stream, or -1 on error + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_NewAudioStream + * \sa SDL_AudioStreamPut + * \sa SDL_AudioStreamAvailable + * \sa SDL_AudioStreamFlush + * \sa SDL_AudioStreamClear + * \sa SDL_FreeAudioStream + */ +extern DECLSPEC int SDLCALL SDL_AudioStreamGet(SDL_AudioStream *stream, void *buf, int len); + +/** + * Get the number of converted/resampled bytes available. + * + * The stream may be buffering data behind the scenes until it has enough to + * resample correctly, so this number might be lower than what you expect, or + * even be zero. Add more data or flush the stream if you need the data now. + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_NewAudioStream + * \sa SDL_AudioStreamPut + * \sa SDL_AudioStreamGet + * \sa SDL_AudioStreamFlush + * \sa SDL_AudioStreamClear + * \sa SDL_FreeAudioStream + */ +extern DECLSPEC int SDLCALL SDL_AudioStreamAvailable(SDL_AudioStream *stream); + +/** + * Tell the stream that you're done sending data, and anything being buffered + * should be converted/resampled and made available immediately. + * + * It is legal to add more data to a stream after flushing, but there will be + * audio gaps in the output. Generally this is intended to signal the end of + * input, so the complete output becomes available. + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_NewAudioStream + * \sa SDL_AudioStreamPut + * \sa SDL_AudioStreamGet + * \sa SDL_AudioStreamAvailable + * \sa SDL_AudioStreamClear + * \sa SDL_FreeAudioStream + */ +extern DECLSPEC int SDLCALL SDL_AudioStreamFlush(SDL_AudioStream *stream); + +/** + * Clear any pending data in the stream without converting it + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_NewAudioStream + * \sa SDL_AudioStreamPut + * \sa SDL_AudioStreamGet + * \sa SDL_AudioStreamAvailable + * \sa SDL_AudioStreamFlush + * \sa SDL_FreeAudioStream + */ +extern DECLSPEC void SDLCALL SDL_AudioStreamClear(SDL_AudioStream *stream); + +/** + * Free an audio stream + * + * \since This function is available since SDL 2.0.7. + * + * \sa SDL_NewAudioStream + * \sa SDL_AudioStreamPut + * \sa SDL_AudioStreamGet + * \sa SDL_AudioStreamAvailable + * \sa SDL_AudioStreamFlush + * \sa SDL_AudioStreamClear + */ +extern DECLSPEC void SDLCALL SDL_FreeAudioStream(SDL_AudioStream *stream); + +#define SDL_MIX_MAXVOLUME 128 + +/** + * This function is a legacy means of mixing audio. + * + * This function is equivalent to calling... + * + * ```c + * SDL_MixAudioFormat(dst, src, format, len, volume); + * ``` + * + * ...where `format` is the obtained format of the audio device from the + * legacy SDL_OpenAudio() function. + * + * \param dst the destination for the mixed audio + * \param src the source audio buffer to be mixed + * \param len the length of the audio buffer in bytes + * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME + * for full audio volume + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_MixAudioFormat + */ +extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 * dst, const Uint8 * src, + Uint32 len, int volume); + +/** + * Mix audio data in a specified format. + * + * This takes an audio buffer `src` of `len` bytes of `format` data and mixes + * it into `dst`, performing addition, volume adjustment, and overflow + * clipping. The buffer pointed to by `dst` must also be `len` bytes of + * `format` data. + * + * This is provided for convenience -- you can mix your own audio data. + * + * Do not use this function for mixing together more than two streams of + * sample data. The output from repeated application of this function may be + * distorted by clipping, because there is no accumulator with greater range + * than the input (not to mention this being an inefficient way of doing it). + * + * It is a common misconception that this function is required to write audio + * data to an output stream in an audio callback. While you can do that, + * SDL_MixAudioFormat() is really only needed when you're mixing a single + * audio stream with a volume adjustment. + * + * \param dst the destination for the mixed audio + * \param src the source audio buffer to be mixed + * \param format the SDL_AudioFormat structure representing the desired audio + * format + * \param len the length of the audio buffer in bytes + * \param volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME + * for full audio volume + * + * \since This function is available since SDL 2.0.0. + */ +extern DECLSPEC void SDLCALL SDL_MixAudioFormat(Uint8 * dst, + const Uint8 * src, + SDL_AudioFormat format, + Uint32 len, int volume); + +/** + * Queue more audio on non-callback devices. + * + * If you are looking to retrieve queued audio from a non-callback capture + * device, you want SDL_DequeueAudio() instead. SDL_QueueAudio() will return + * -1 to signify an error if you use it with capture devices. + * + * SDL offers two ways to feed audio to the device: you can either supply a + * callback that SDL triggers with some frequency to obtain more audio (pull + * method), or you can supply no callback, and then SDL will expect you to + * supply data at regular intervals (push method) with this function. + * + * There are no limits on the amount of data you can queue, short of + * exhaustion of address space. Queued data will drain to the device as + * necessary without further intervention from you. If the device needs audio + * but there is not enough queued, it will play silence to make up the + * difference. This means you will have skips in your audio playback if you + * aren't routinely queueing sufficient data. + * + * This function copies the supplied data, so you are safe to free it when the + * function returns. This function is thread-safe, but queueing to the same + * device from two threads at once does not promise which buffer will be + * queued first. + * + * You may not queue audio on a device that is using an application-supplied + * callback; doing so returns an error. You have to use the audio callback or + * queue audio with this function, but not both. + * + * You should not call SDL_LockAudio() on the device before queueing; SDL + * handles locking internally for this function. + * + * Note that SDL2 does not support planar audio. You will need to resample + * from planar audio formats into a non-planar one (see SDL_AudioFormat) + * before queuing audio. + * + * \param dev the device ID to which we will queue audio + * \param data the data to queue to the device for later playback + * \param len the number of bytes (not samples!) to which `data` points + * \returns 0 on success or a negative error code on failure; call + * SDL_GetError() for more information. + * + * \since This function is available since SDL 2.0.4. + * + * \sa SDL_ClearQueuedAudio + * \sa SDL_GetQueuedAudioSize + */ +extern DECLSPEC int SDLCALL SDL_QueueAudio(SDL_AudioDeviceID dev, const void *data, Uint32 len); + +/** + * Dequeue more audio on non-callback devices. + * + * If you are looking to queue audio for output on a non-callback playback + * device, you want SDL_QueueAudio() instead. SDL_DequeueAudio() will always + * return 0 if you use it with playback devices. + * + * SDL offers two ways to retrieve audio from a capture device: you can either + * supply a callback that SDL triggers with some frequency as the device + * records more audio data, (push method), or you can supply no callback, and + * then SDL will expect you to retrieve data at regular intervals (pull + * method) with this function. + * + * There are no limits on the amount of data you can queue, short of + * exhaustion of address space. Data from the device will keep queuing as + * necessary without further intervention from you. This means you will + * eventually run out of memory if you aren't routinely dequeueing data. + * + * Capture devices will not queue data when paused; if you are expecting to + * not need captured audio for some length of time, use SDL_PauseAudioDevice() + * to stop the capture device from queueing more data. This can be useful + * during, say, level loading times. When unpaused, capture devices will start + * queueing data from that point, having flushed any capturable data available + * while paused. + * + * This function is thread-safe, but dequeueing from the same device from two + * threads at once does not promise which thread will dequeue data first. + * + * You may not dequeue audio from a device that is using an + * application-supplied callback; doing so returns an error. You have to use + * the audio callback, or dequeue audio with this function, but not both. + * + * You should not call SDL_LockAudio() on the device before dequeueing; SDL + * handles locking internally for this function. + * + * \param dev the device ID from which we will dequeue audio + * \param data a pointer into where audio data should be copied + * \param len the number of bytes (not samples!) to which (data) points + * \returns the number of bytes dequeued, which could be less than requested; + * call SDL_GetError() for more information. + * + * \since This function is available since SDL 2.0.5. + * + * \sa SDL_ClearQueuedAudio + * \sa SDL_GetQueuedAudioSize + */ +extern DECLSPEC Uint32 SDLCALL SDL_DequeueAudio(SDL_AudioDeviceID dev, void *data, Uint32 len); + +/** + * Get the number of bytes of still-queued audio. + * + * For playback devices: this is the number of bytes that have been queued for + * playback with SDL_QueueAudio(), but have not yet been sent to the hardware. + * + * Once we've sent it to the hardware, this function can not decide the exact + * byte boundary of what has been played. It's possible that we just gave the + * hardware several kilobytes right before you called this function, but it + * hasn't played any of it yet, or maybe half of it, etc. + * + * For capture devices, this is the number of bytes that have been captured by + * the device and are waiting for you to dequeue. This number may grow at any + * time, so this only informs of the lower-bound of available data. + * + * You may not queue or dequeue audio on a device that is using an + * application-supplied callback; calling this function on such a device + * always returns 0. You have to use the audio callback or queue audio, but + * not both. + * + * You should not call SDL_LockAudio() on the device before querying; SDL + * handles locking internally for this function. + * + * \param dev the device ID of which we will query queued audio size + * \returns the number of bytes (not samples!) of queued audio. + * + * \since This function is available since SDL 2.0.4. + * + * \sa SDL_ClearQueuedAudio + * \sa SDL_QueueAudio + * \sa SDL_DequeueAudio + */ +extern DECLSPEC Uint32 SDLCALL SDL_GetQueuedAudioSize(SDL_AudioDeviceID dev); + +/** + * Drop any queued audio data waiting to be sent to the hardware. + * + * Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For + * output devices, the hardware will start playing silence if more audio isn't + * queued. For capture devices, the hardware will start filling the empty + * queue with new data if the capture device isn't paused. + * + * This will not prevent playback of queued audio that's already been sent to + * the hardware, as we can not undo that, so expect there to be some fraction + * of a second of audio that might still be heard. This can be useful if you + * want to, say, drop any pending music or any unprocessed microphone input + * during a level change in your game. + * + * You may not queue or dequeue audio on a device that is using an + * application-supplied callback; calling this function on such a device + * always returns 0. You have to use the audio callback or queue audio, but + * not both. + * + * You should not call SDL_LockAudio() on the device before clearing the + * queue; SDL handles locking internally for this function. + * + * This function always succeeds and thus returns void. + * + * \param dev the device ID of which to clear the audio queue + * + * \since This function is available since SDL 2.0.4. + * + * \sa SDL_GetQueuedAudioSize + * \sa SDL_QueueAudio + * \sa SDL_DequeueAudio + */ +extern DECLSPEC void SDLCALL SDL_ClearQueuedAudio(SDL_AudioDeviceID dev); + + +/** + * \name Audio lock functions + * + * The lock manipulated by these functions protects the callback function. + * During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that + * the callback function is not running. Do not call these from the callback + * function or you will cause deadlock. + */ +/* @{ */ + +/** + * This function is a legacy means of locking the audio device. + * + * New programs might want to use SDL_LockAudioDevice() instead. This function + * is equivalent to calling... + * + * ```c + * SDL_LockAudioDevice(1); + * ``` + * + * ...and is only useful if you used the legacy SDL_OpenAudio() function. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_LockAudioDevice + * \sa SDL_UnlockAudio + * \sa SDL_UnlockAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_LockAudio(void); + +/** + * Use this function to lock out the audio callback function for a specified + * device. + * + * The lock manipulated by these functions protects the audio callback + * function specified in SDL_OpenAudioDevice(). During a + * SDL_LockAudioDevice()/SDL_UnlockAudioDevice() pair, you can be guaranteed + * that the callback function for that device is not running, even if the + * device is not paused. While a device is locked, any other unpaused, + * unlocked devices may still run their callbacks. + * + * Calling this function from inside your audio callback is unnecessary. SDL + * obtains this lock before calling your function, and releases it when the + * function returns. + * + * You should not hold the lock longer than absolutely necessary. If you hold + * it too long, you'll experience dropouts in your audio playback. Ideally, + * your application locks the device, sets a few variables and unlocks again. + * Do not do heavy work while holding the lock for a device. + * + * It is safe to lock the audio device multiple times, as long as you unlock + * it an equivalent number of times. The callback will not run until the + * device has been unlocked completely in this way. If your application fails + * to unlock the device appropriately, your callback will never run, you might + * hear repeating bursts of audio, and SDL_CloseAudioDevice() will probably + * deadlock. + * + * Internally, the audio device lock is a mutex; if you lock from two threads + * at once, not only will you block the audio callback, you'll block the other + * thread. + * + * \param dev the ID of the device to be locked + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_UnlockAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_LockAudioDevice(SDL_AudioDeviceID dev); + +/** + * This function is a legacy means of unlocking the audio device. + * + * New programs might want to use SDL_UnlockAudioDevice() instead. This + * function is equivalent to calling... + * + * ```c + * SDL_UnlockAudioDevice(1); + * ``` + * + * ...and is only useful if you used the legacy SDL_OpenAudio() function. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_LockAudio + * \sa SDL_UnlockAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_UnlockAudio(void); + +/** + * Use this function to unlock the audio callback function for a specified + * device. + * + * This function should be paired with a previous SDL_LockAudioDevice() call. + * + * \param dev the ID of the device to be unlocked + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_LockAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_UnlockAudioDevice(SDL_AudioDeviceID dev); +/* @} *//* Audio lock functions */ + +/** + * This function is a legacy means of closing the audio device. + * + * This function is equivalent to calling... + * + * ```c + * SDL_CloseAudioDevice(1); + * ``` + * + * ...and is only useful if you used the legacy SDL_OpenAudio() function. + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_OpenAudio + */ +extern DECLSPEC void SDLCALL SDL_CloseAudio(void); + +/** + * Use this function to shut down audio processing and close the audio device. + * + * The application should close open audio devices once they are no longer + * needed. Calling this function will wait until the device's audio callback + * is not running, release the audio hardware and then clean up internal + * state. No further audio will play from this device once this function + * returns. + * + * This function may block briefly while pending audio data is played by the + * hardware, so that applications don't drop the last buffer of data they + * supplied. + * + * The device ID is invalid as soon as the device is closed, and is eligible + * for reuse in a new SDL_OpenAudioDevice() call immediately. + * + * \param dev an audio device previously opened with SDL_OpenAudioDevice() + * + * \since This function is available since SDL 2.0.0. + * + * \sa SDL_OpenAudioDevice + */ +extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID dev); + +/* Ends C function definitions when using C++ */ +#ifdef __cplusplus +} +#endif +#include "close_code.h" + +#endif /* SDL_audio_h_ */ + +/* vi: set ts=4 sw=4 expandtab: */ |