diff options
Diffstat (limited to 'dependencies/SDL/SDL_audio.h')
-rw-r--r-- | dependencies/SDL/SDL_audio.h | 1500 |
1 files changed, 0 insertions, 1500 deletions
diff --git a/dependencies/SDL/SDL_audio.h b/dependencies/SDL/SDL_audio.h deleted file mode 100644 index c42de3e..0000000 --- a/dependencies/SDL/SDL_audio.h +++ /dev/null @@ -1,1500 +0,0 @@ -/* - 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: */ |