SDL/include/SDL3/SDL_audio.h

/*
  Simple DirectMedia Layer
  Copyright (C) 1997-2023 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.
*/

/**
 *  \file SDL_audio.h
 *
 *  \brief Audio functionality for the SDL library.
 */

#ifndef SDL_audio_h_
#define SDL_audio_h_

#include <SDL3/SDL_stdinc.h>
#include <SDL3/SDL_error.h>
#include <SDL3/SDL_endian.h>
#include <SDL3/SDL_mutex.h>
#include <SDL3/SDL_thread.h>
#include <SDL3/SDL_rwops.h>

#include <SDL3/SDL_begin_code.h>
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
#endif

/*
 *  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)
 */

/**
 *  \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_FLOAT         (1<<8)
#define SDL_AUDIO_MASK_BIG_ENDIAN    (1<<12)
#define SDL_AUDIO_MASK_SIGNED        (1<<15)
#define SDL_AUDIO_BITSIZE(x)         ((x) & SDL_AUDIO_MASK_BITSIZE)
#define SDL_AUDIO_BYTESIZE(x)        (SDL_AUDIO_BITSIZE(x) / 8)
#define SDL_AUDIO_ISFLOAT(x)         ((x) & SDL_AUDIO_MASK_FLOAT)
#define SDL_AUDIO_ISBIGENDIAN(x)     ((x) & SDL_AUDIO_MASK_BIG_ENDIAN)
#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
#define SDL_AUDIO_ISSIGNED(x)        ((x) & SDL_AUDIO_MASK_SIGNED)
#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))

/**
 *  \name Audio format flags
 *
 *  Defaults to LSB byte order.
 */
/* @{ */
#define SDL_AUDIO_U8        0x0008  /**< Unsigned 8-bit samples */
#define SDL_AUDIO_S8        0x8008  /**< Signed 8-bit samples */
#define SDL_AUDIO_S16LE     0x8010  /**< Signed 16-bit samples */
#define SDL_AUDIO_S16BE     0x9010  /**< As above, but big-endian byte order */
/* @} */

/**
 *  \name int32 support
 */
/* @{ */
#define SDL_AUDIO_S32LE     0x8020  /**< 32-bit integer samples */
#define SDL_AUDIO_S32BE     0x9020  /**< As above, but big-endian byte order */
/* @} */

/**
 *  \name float32 support
 */
/* @{ */
#define SDL_AUDIO_F32LE     0x8120  /**< 32-bit floating point samples */
#define SDL_AUDIO_F32BE     0x9120  /**< As above, but big-endian byte order */
/* @} */

/**
 *  \name Native audio byte ordering
 */
/* @{ */
#if SDL_BYTEORDER == SDL_LIL_ENDIAN
#define SDL_AUDIO_S16    SDL_AUDIO_S16LE
#define SDL_AUDIO_S32    SDL_AUDIO_S32LE
#define SDL_AUDIO_F32    SDL_AUDIO_F32LE
#else
#define SDL_AUDIO_S16    SDL_AUDIO_S16BE
#define SDL_AUDIO_S32    SDL_AUDIO_S32BE
#define SDL_AUDIO_F32    SDL_AUDIO_F32BE
#endif
/* @} */

/* @} *//* Audio flags */

/**
 * SDL Audio Device instance IDs.
 */
typedef Uint32 SDL_AudioDeviceID;

#define SDL_AUDIO_DEVICE_DEFAULT_OUTPUT ((SDL_AudioDeviceID) 0xFFFFFFFF)
#define SDL_AUDIO_DEVICE_DEFAULT_CAPTURE ((SDL_AudioDeviceID) 0xFFFFFFFE)

typedef struct SDL_AudioSpec
{
    SDL_AudioFormat format;     /**< Audio data format */
    int channels;               /**< Number of channels: 1 mono, 2 stereo, etc */
    int freq;                   /**< sample rate: sample frames per second */
} SDL_AudioSpec;

/* Calculate the size of each audio frame (in bytes) */
#define SDL_AUDIO_FRAMESIZE(x) (SDL_AUDIO_BYTESIZE((x).format) * (x).channels)

/* SDL_AudioStream is an audio conversion interface.
    - 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.
    - It can handle input/output format changes on the fly.
    - You push data as you have it, and pull it when you need it
    - It can also function as a basic audio data queue even if you
      just have sound that needs to pass from one place to another.
    - You can hook callbacks up to them when more data is added or
      requested, to manage data on-the-fly.
 */
struct SDL_AudioStream;  /* this is opaque to the outside world. */
typedef struct SDL_AudioStream SDL_AudioStream;


/* 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.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \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.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_GetNumAudioDrivers
 */
extern DECLSPEC const char *SDLCALL SDL_GetAudioDriver(int index);
/* @} */

/**
 * 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.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC const char *SDLCALL SDL_GetCurrentAudioDriver(void);

/**
 * Get a list of currently-connected audio output devices.
 *
 * This returns of list of available devices that play sound, perhaps to
 * speakers or headphones ("output" devices). If you want devices that record
 * audio, like a microphone ("capture" devices), use
 * SDL_GetAudioCaptureDevices() instead.
 *
 * This only returns a list of physical devices; it will not have any device
 * IDs returned by SDL_OpenAudioDevice().
 *
 * \param count a pointer filled in with the number of devices returned
 * \returns a 0 terminated array of device instance IDs which should be freed
 *          with SDL_free(), or NULL on error; call SDL_GetError() for more
 *          details.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_OpenAudioDevice
 * \sa SDL_GetAudioCaptureDevices
 */
extern DECLSPEC SDL_AudioDeviceID *SDLCALL SDL_GetAudioOutputDevices(int *count);

/**
 * Get a list of currently-connected audio capture devices.
 *
 * This returns of list of available devices that record audio, like a
 * microphone ("capture" devices). If you want devices that play sound,
 * perhaps to speakers or headphones ("output" devices), use
 * SDL_GetAudioOutputDevices() instead.
 *
 * This only returns a list of physical devices; it will not have any device
 * IDs returned by SDL_OpenAudioDevice().
 *
 * \param count a pointer filled in with the number of devices returned
 * \returns a 0 terminated array of device instance IDs which should be freed
 *          with SDL_free(), or NULL on error; call SDL_GetError() for more
 *          details.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_OpenAudioDevice
 * \sa SDL_GetAudioOutputDevices
 */
extern DECLSPEC SDL_AudioDeviceID *SDLCALL SDL_GetAudioCaptureDevices(int *count);

/**
 * Get the human-readable name of a specific audio device.
 *
 * The string returned by this function is UTF-8 encoded. The caller should
 * call SDL_free on the return value when done with it.
 *
 * \param devid the instance ID of the device to query.
 * \returns the name of the audio device, or NULL on error.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_GetNumAudioDevices
 * \sa SDL_GetDefaultAudioInfo
 */
extern DECLSPEC char *SDLCALL SDL_GetAudioDeviceName(SDL_AudioDeviceID devid);

/**
 * Get the current audio format of a specific audio device.
 *
 * For an opened device, this will report the format the device is currently
 * using. If the device isn't yet opened, this will report the device's
 * preferred format (or a reasonable default if this can't be determined).
 *
 * You may also specify SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
 * SDL_AUDIO_DEVICE_DEFAULT_CAPTURE here, which is useful for getting a
 * reasonable recommendation before opening the system-recommended default
 * device.
 *
 * \param devid the instance ID of the device to query.
 * \param spec On return, will be filled with device details.
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC int SDLCALL SDL_GetAudioDeviceFormat(SDL_AudioDeviceID devid, SDL_AudioSpec *spec);


/**
 * Open a specific audio device.
 *
 * You can open both output and capture devices through this function. Output
 * devices will take data from bound audio streams, mix it, and send it to the
 * hardware. Capture devices will feed any bound audio streams with a copy of
 * any incoming data.
 *
 * An opened audio device starts out with no audio streams bound. To start
 * audio playing, bind a stream and supply audio data to it. Unlike SDL2,
 * there is no audio callback; you only bind audio streams and make sure they
 * have data flowing into them (however, you can simulate SDL2's semantics
 * fairly closely by using SDL_OpenAudioDeviceStream instead of this
 * function).
 *
 * If you don't care about opening a specific device, pass a `devid` of either
 * `SDL_AUDIO_DEVICE_DEFAULT_OUTPUT` or `SDL_AUDIO_DEVICE_DEFAULT_CAPTURE`. In
 * this case, SDL will try to pick the most reasonable default, and may also
 * switch between physical devices seamlessly later, if the most reasonable
 * default changes during the lifetime of this opened device (user changed the
 * default in the OS's system preferences, the default got unplugged so the
 * system jumped to a new default, the user plugged in headphones on a mobile
 * device, etc). Unless you have a good reason to choose a specific device,
 * this is probably what you want.
 *
 * You may request a specific format for the audio device, but there is no
 * promise the device will honor that request for several reasons. As such,
 * it's only meant to be a hint as to what data your app will provide. Audio
 * streams will accept data in whatever format you specify and manage
 * conversion for you as appropriate. SDL_GetAudioDeviceFormat can tell you
 * the preferred format for the device before opening and the actual format
 * the device is using after opening.
 *
 * It's legal to open the same device ID more than once; each successful open
 * will generate a new logical SDL_AudioDeviceID that is managed separately
 * from others on the same physical device. This allows libraries to open a
 * device separately from the main app and bind its own streams without
 * conflicting.
 *
 * It is also legal to open a device ID returned by a previous call to this
 * function; doing so just creates another logical device on the same physical
 * device. This may be useful for making logical groupings of audio streams.
 *
 * This function returns the opened device ID on success. This is a new,
 * unique SDL_AudioDeviceID that represents a logical device.
 *
 * Some backends might offer arbitrary devices (for example, a networked audio
 * protocol that can connect to an arbitrary server). For these, as a change
 * from SDL2, you should open a default device ID and use an SDL hint to
 * specify the target if you care, or otherwise let the backend figure out a
 * reasonable default. Most backends don't offer anything like this, and often
 * this would be an end user setting an environment variable for their custom
 * need, and not something an application should specifically manage.
 *
 * When done with an audio device, possibly at the end of the app's life, one
 * should call SDL_CloseAudioDevice() on the returned device id.
 *
 * \param devid the device instance id to open, or
 *              SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
 *              SDL_AUDIO_DEVICE_DEFAULT_CAPTURE for the most reasonable
 *              default device.
 * \param spec the requested device configuration. Can be NULL to use
 *             reasonable defaults.
 * \returns The device ID on success, 0 on error; call SDL_GetError() for more
 *          information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CloseAudioDevice
 * \sa SDL_GetAudioDeviceFormat
 */
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec);

/**
 * Use this function to pause audio playback on a specified device.
 *
 * This function pauses audio processing for a given device. Any bound audio
 * streams will not progress, and no audio will be generated. Pausing one
 * device does not prevent other unpaused devices from running.
 *
 * Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
 * has to bind a stream before any audio will flow. Pausing a paused device is
 * a legal no-op.
 *
 * Pausing a device can be useful to halt all audio without unbinding all the
 * audio streams. This might be useful while a game is paused, or a level is
 * loading, etc.
 *
 * Physical devices can not be paused or unpaused, only logical devices
 * created through SDL_OpenAudioDevice() can be.
 *
 * \param dev a device opened by SDL_OpenAudioDevice()
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_ResumeAudioDevice
 * \sa SDL_IsAudioDevicePaused
 */
extern DECLSPEC int SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID dev);

/**
 * Use this function to unpause audio playback on a specified device.
 *
 * This function unpauses audio processing for a given device that has
 * previously been paused with SDL_PauseAudioDevice(). Once unpaused, any
 * bound audio streams will begin to progress again, and audio can be
 * generated.
 *
 * Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
 * has to bind a stream before any audio will flow. Unpausing an unpaused
 * device is a legal no-op.
 *
 * Physical devices can not be paused or unpaused, only logical devices
 * created through SDL_OpenAudioDevice() can be.
 *
 * \param dev a device opened by SDL_OpenAudioDevice()
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_ResumeAudioDevice
 * \sa SDL_IsAudioDevicePaused
 */
extern DECLSPEC int SDLCALL SDL_ResumeAudioDevice(SDL_AudioDeviceID dev);

/**
 * Use this function to query if an audio device is paused.
 *
 * Unlike in SDL2, audio devices start in an _unpaused_ state, since an app
 * has to bind a stream before any audio will flow.
 *
 * Physical devices can not be paused or unpaused, only logical devices
 * created through SDL_OpenAudioDevice() can be. Physical and invalid device
 * IDs will report themselves as unpaused here.
 *
 * \param dev a device opened by SDL_OpenAudioDevice()
 * \returns SDL_TRUE if device is valid and paused, SDL_FALSE otherwise.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_PauseAudioDevice
 * \sa SDL_ResumeAudioDevice
 * \sa SDL_IsAudioDevicePaused
 */
extern DECLSPEC SDL_bool SDLCALL SDL_IsAudioDevicePaused(SDL_AudioDeviceID dev);

/**
 * Close a previously-opened audio device.
 *
 * The application should close open audio devices once they are no longer
 * needed.
 *
 * 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 if terminating immediately afterwards.
 *
 * \param devid an audio device id previously returned by
 *              SDL_OpenAudioDevice()
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_OpenAudioDevice
 */
extern DECLSPEC void SDLCALL SDL_CloseAudioDevice(SDL_AudioDeviceID devid);

/**
 * Bind a list of audio streams to an audio device.
 *
 * Audio data will flow through any bound streams. For an output device, data
 * for all bound streams will be mixed together and fed to the device. For a
 * capture device, a copy of recorded data will be provided to each bound
 * stream.
 *
 * Audio streams can only be bound to an open device. This operation is
 * atomic--all streams bound in the same call will start processing at the
 * same time, so they can stay in sync. Also: either all streams will be bound
 * or none of them will be.
 *
 * It is an error to bind an already-bound stream; it must be explicitly
 * unbound first.
 *
 * Binding a stream to a device will set its output format for output devices,
 * and its input format for capture devices, so they match the device's
 * settings. The caller is welcome to change the other end of the stream's
 * format at any time.
 *
 * \param devid an audio device to bind a stream to.
 * \param streams an array of audio streams to unbind.
 * \param num_streams Number streams listed in the `streams` array.
 * \returns 0 on success, -1 on error; call SDL_GetError() for more
 *          information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_BindAudioStreams
 * \sa SDL_UnbindAudioStreams
 * \sa SDL_UnbindAudioStream
 * \sa SDL_GetAudioStreamDevice
 */
extern DECLSPEC int SDLCALL SDL_BindAudioStreams(SDL_AudioDeviceID devid, SDL_AudioStream **streams, int num_streams);

/**
 * Bind a single audio stream to an audio device.
 *
 * This is a convenience function, equivalent to calling
 * `SDL_BindAudioStreams(devid, &stream, 1)`.
 *
 * \param devid an audio device to bind a stream to.
 * \param stream an audio stream to bind to a device.
 * \returns 0 on success, -1 on error; call SDL_GetError() for more
 *          information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_BindAudioStreams
 * \sa SDL_UnbindAudioStreams
 * \sa SDL_UnbindAudioStream
 * \sa SDL_GetAudioStreamDevice
 */
extern DECLSPEC int SDLCALL SDL_BindAudioStream(SDL_AudioDeviceID devid, SDL_AudioStream *stream);

/**
 * Unbind a list of audio streams from their audio devices.
 *
 * The streams being unbound do not all have to be on the same device. All
 * streams on the same device will be unbound atomically (data will stop
 * flowing through them all unbound streams on the same device at the same
 * time).
 *
 * Unbinding a stream that isn't bound to a device is a legal no-op.
 *
 * \param streams an array of audio streams to unbind.
 * \param num_streams Number streams listed in the `streams` array.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_BindAudioStreams
 * \sa SDL_BindAudioStream
 * \sa SDL_UnbindAudioStream
 * \sa SDL_GetAudioStreamDevice
 */
extern DECLSPEC void SDLCALL SDL_UnbindAudioStreams(SDL_AudioStream **streams, int num_streams);

/**
 * Unbind a single audio stream from its audio device.
 *
 * This is a convenience function, equivalent to calling
 * `SDL_UnbindAudioStreams(&stream, 1)`.
 *
 * \param stream an audio stream to unbind from a device.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_BindAudioStream
 * \sa SDL_BindAudioStreams
 * \sa SDL_UnbindAudioStreams
 * \sa SDL_GetAudioStreamDevice
 */
extern DECLSPEC void SDLCALL SDL_UnbindAudioStream(SDL_AudioStream *stream);

/**
 * Query an audio stream for its currently-bound device.
 *
 * This reports the audio device that an audio stream is currently bound to.
 *
 * If not bound, or invalid, this returns zero, which is not a valid device
 * ID.
 *
 * \param stream the audio stream to query.
 * \returns The bound audio device, or 0 if not bound or invalid.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_BindAudioStream
 * \sa SDL_BindAudioStreams
 * \sa SDL_UnbindAudioStream
 * \sa SDL_UnbindAudioStreams
 */
extern DECLSPEC SDL_AudioDeviceID SDLCALL SDL_GetAudioStreamDevice(SDL_AudioStream *stream);


/**
 * Create a new audio stream.
 *
 * \param src_spec The format details of the input audio
 * \param dst_spec The format details of the output audio
 * \returns 0 on success, or -1 on error.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_FlushAudioStream
 * \sa SDL_ClearAudioStream
 * \sa SDL_ChangeAudioStreamOutput
 * \sa SDL_DestroyAudioStream
 */
extern DECLSPEC SDL_AudioStream *SDLCALL SDL_CreateAudioStream(const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec);


/**
 * Query the current format of an audio stream.
 *
 * \param stream the SDL_AudioStream to query.
 * \param src_spec Where to store the input audio format; ignored if NULL.
 * \param dst_spec Where to store the output audio format; ignored if NULL.
 * \returns 0 on success, or -1 on error.
 *
 * \threadsafety It is safe to call this function from any thread, as it holds
 *               a stream-specific mutex while running.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC int SDLCALL SDL_GetAudioStreamFormat(SDL_AudioStream *stream,
                                                     SDL_AudioSpec *src_spec,
                                                     SDL_AudioSpec *dst_spec);

/**
 * Change the input and output formats of an audio stream.
 *
 * Future calls to and SDL_GetAudioStreamAvailable and SDL_GetAudioStreamData
 * will reflect the new format, and future calls to SDL_PutAudioStreamData
 * must provide data in the new input formats.
 *
 * \param stream The stream the format is being changed
 * \param src_spec The new format of the audio input; if NULL, it is not
 *                 changed.
 * \param dst_spec The new format of the audio output; if NULL, it is not
 *                 changed.
 * \returns 0 on success, or -1 on error.
 *
 * \threadsafety It is safe to call this function from any thread, as it holds
 *               a stream-specific mutex while running.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_GetAudioStreamFormat
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_SetAudioStreamFrequencyRatio
 */
extern DECLSPEC int SDLCALL SDL_SetAudioStreamFormat(SDL_AudioStream *stream,
                                                     const SDL_AudioSpec *src_spec,
                                                     const SDL_AudioSpec *dst_spec);

/**
 * Get the frequency ratio of an audio stream.
 *
 * \param stream the SDL_AudioStream to query.
 * \returns the frequency ratio of the stream, or 0.0 on error
 *
 * \threadsafety It is safe to call this function from any thread, as it holds
 *               a stream-specific mutex while running.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_SetAudioStreamFrequencyRatio
 */
extern DECLSPEC float SDLCALL SDL_GetAudioStreamFrequencyRatio(SDL_AudioStream *stream);

/**
 * Change the frequency ratio of an audio stream.
 *
 * The frequency ratio is used to adjust the rate at which input data is
 * consumed. Changing this effectively modifies the speed and pitch of the
 * audio. A value greater than 1.0 will play the audio faster, and at a higher
 * pitch. A value less than 1.0 will play the audio slower, and at a lower
 * pitch.
 *
 * This is applied during SDL_GetAudioStreamData, and can be continuously
 * changed to create various effects.
 *
 * \param stream The stream the frequency ratio is being changed
 * \param ratio The frequency ratio. 1.0 is normal speed. Must be between 0.01
 *              and 100.
 * \returns 0 on success, or -1 on error.
 *
 * \threadsafety It is safe to call this function from any thread, as it holds
 *               a stream-specific mutex while running.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_GetAudioStreamFrequencyRatio
 * \sa SDL_SetAudioStreamFormat
 */
extern DECLSPEC int SDLCALL SDL_SetAudioStreamFrequencyRatio(SDL_AudioStream *stream, float ratio);

/**
 * Add data to be converted/resampled to the stream.
 *
 * This data must match the format/channels/samplerate specified in the latest
 * call to SDL_SetAudioStreamFormat, or the format specified when creating the
 * stream if it hasn't been changed.
 *
 * Note that this call simply queues unconverted data for later. This is
 * different than SDL2, where data was converted during the Put call and the
 * Get call would just dequeue the previously-converted data.
 *
 * \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 a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread, but if the
 *               stream has a callback set, the caller might need to manage
 *               extra locking.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 * \sa SDL_GetAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_FlushAudioStream
 * \sa SDL_ClearAudioStream
 * \sa SDL_DestroyAudioStream
 */
extern DECLSPEC int SDLCALL SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len);

/**
 * Get converted/resampled data from the stream.
 *
 * The input/output data format/channels/samplerate is specified when creating
 * the stream, and can be changed after creation by calling
 * SDL_SetAudioStreamFormat.
 *
 * Note that any conversion and resampling necessary is done during this call,
 * and SDL_PutAudioStreamData simply queues unconverted data for later. This
 * is different than SDL2, where that work was done while inputting new data
 * to the stream and requesting the output just copied the converted data.
 *
 * \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
 *
 * \threadsafety It is safe to call this function from any thread, but if the
 *               stream has a callback set, the caller might need to manage
 *               extra locking.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_SetAudioStreamFormat
 * \sa SDL_FlushAudioStream
 * \sa SDL_ClearAudioStream
 * \sa SDL_DestroyAudioStream
 */
extern DECLSPEC int SDLCALL SDL_GetAudioStreamData(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.
 *
 * If the stream has so much data that it would overflow an int, the return
 * value is clamped to a maximum value, but no queued data is lost; if there
 * are gigabytes of data queued, the app might need to read some of it with
 * SDL_GetAudioStreamData before this function's return value is no longer
 * clamped.
 *
 * \param stream The audio stream to query
 * \returns the number of converted/resampled bytes available.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamData
 * \sa SDL_FlushAudioStream
 * \sa SDL_ClearAudioStream
 * \sa SDL_DestroyAudioStream
 */
extern DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(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.
 *
 * \param stream The audio stream to flush
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_ClearAudioStream
 * \sa SDL_DestroyAudioStream
 */
extern DECLSPEC int SDLCALL SDL_FlushAudioStream(SDL_AudioStream *stream);

/**
 * Clear any pending data in the stream without converting it
 *
 * \param stream The audio stream to clear
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_FlushAudioStream
 * \sa SDL_DestroyAudioStream
 */
extern DECLSPEC int SDLCALL SDL_ClearAudioStream(SDL_AudioStream *stream);

/**
 * Lock an audio stream for serialized access.
 *
 * Each SDL_AudioStream has an internal mutex it uses to protect its data
 * structures from threading conflicts. This function allows an app to lock
 * that mutex, which could be useful if registering callbacks on this stream.
 *
 * One does not need to lock a stream to use in it most cases, as the stream
 * manages this lock internally. However, this lock is held during callbacks,
 * which may run from arbitrary threads at any time, so if an app needs to
 * protect shared data during those callbacks, locking the stream guarantees
 * that the callback is not running while the lock is held.
 *
 * As this is just a wrapper over SDL_LockMutex for an internal lock, it has
 * all the same attributes (recursive locks are allowed, etc).
 *
 * \param stream The audio stream to lock.
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_UnlockAudioStream
 * \sa SDL_SetAudioStreamPutCallback
 * \sa SDL_SetAudioStreamGetCallback
 */
extern DECLSPEC int SDLCALL SDL_LockAudioStream(SDL_AudioStream *stream);


/**
 * Unlock an audio stream for serialized access.
 *
 * This unlocks an audio stream after a call to SDL_LockAudioStream.
 *
 * \param stream The audio stream to unlock.
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety You should only call this from the same thread that
 *               previously called SDL_LockAudioStream.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_LockAudioStream
 * \sa SDL_SetAudioStreamPutCallback
 * \sa SDL_SetAudioStreamGetCallback
 */
extern DECLSPEC int SDLCALL SDL_UnlockAudioStream(SDL_AudioStream *stream);

/**
 * A callback that fires when data passes through an SDL_AudioStream.
 *
 * Apps can (optionally) register a callback with an audio stream that
 * is called when data is added with SDL_PutAudioStreamData, or requested
 * with SDL_GetAudioStreamData. These callbacks may run from any
 * thread, so if you need to protect shared data, you should use
 * SDL_LockAudioStream to serialize access; this lock will be held by
 * before your callback is called, so your callback does not need to
 * manage the lock explicitly.
 *
 * \param stream The SDL audio stream associated with this callback.
 * \param approx_amount The _approximate_ amount of data, in bytes, that is requested or available.
 *                       This might be slightly overestimated due to buffering or
 *                       resampling, and may change from call to call.
 * \param userdata An opaque pointer provided by the app for their personal use.
 */
typedef void (SDLCALL *SDL_AudioStreamCallback)(void *userdata, SDL_AudioStream *stream, int approx_amount);

/**
 * Set a callback that runs when data is requested from an audio stream.
 *
 * This callback is called _before_ data is obtained from the stream, giving
 * the callback the chance to add more on-demand.
 *
 * The callback can (optionally) call SDL_PutAudioStreamData() to add more
 * audio to the stream during this call; if needed, the request that triggered
 * this callback will obtain the new data immediately.
 *
 * The callback's `approx_request` argument is roughly how many bytes of
 * _unconverted_ data (in the stream's input format) is needed by the caller,
 * although this may overestimate a little for safety. This takes into account
 * how much is already in the stream and only asks for any extra necessary to
 * resolve the request, which means the callback may be asked for zero bytes,
 * and a different amount on each call.
 *
 * The callback is not required to supply exact amounts; it is allowed to
 * supply too much or too little or none at all. The caller will get what's
 * available, up to the amount they requested, regardless of this callback's
 * outcome.
 *
 * Clearing or flushing an audio stream does not call this callback.
 *
 * This function obtains the stream's lock, which means any existing callback
 * (get or put) in progress will finish running before setting the new
 * callback.
 *
 * Setting a NULL function turns off the callback.
 *
 * \param stream the audio stream to set the new callback on.
 * \param callback the new callback function to call when data is added to the
 *                 stream.
 * \param userdata an opaque pointer provided to the callback for its own
 *                 personal use.
 * \returns 0 on success, -1 on error. This only fails if `stream` is NULL.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_SetAudioStreamPutCallback
 */
extern DECLSPEC int SDLCALL SDL_SetAudioStreamGetCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata);

/**
 * Set a callback that runs when data is added to an audio stream.
 *
 * This callback is called _after_ the data is added to the stream, giving the
 * callback the chance to obtain it immediately.
 *
 * The callback can (optionally) call SDL_GetAudioStreamData() to obtain audio
 * from the stream during this call.
 *
 * The callback's `approx_request` argument is how many bytes of _converted_
 * data (in the stream's output format) was provided by the caller, although
 * this may underestimate a little for safety. This value might be less than
 * what is currently available in the stream, if data was already there, and
 * might be less than the caller provided if the stream needs to keep a buffer
 * to aid in resampling. Which means the callback may be provided with zero
 * bytes, and a different amount on each call.
 *
 * The callback may call SDL_GetAudioStreamAvailable to see the total amount
 * currently available to read from the stream, instead of the total provided
 * by the current call.
 *
 * The callback is not required to obtain all data. It is allowed to read less
 * or none at all. Anything not read now simply remains in the stream for
 * later access.
 *
 * Clearing or flushing an audio stream does not call this callback.
 *
 * This function obtains the stream's lock, which means any existing callback
 * (get or put) in progress will finish running before setting the new
 * callback.
 *
 * Setting a NULL function turns off the callback.
 *
 * \param stream the audio stream to set the new callback on.
 * \param callback the new callback function to call when data is added to the
 *                 stream.
 * \param userdata an opaque pointer provided to the callback for its own
 *                 personal use.
 * \returns 0 on success, -1 on error. This only fails if `stream` is NULL.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_SetAudioStreamGetCallback
 */
extern DECLSPEC int SDLCALL SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata);


/**
 * Free an audio stream
 *
 * \param stream The audio stream to free
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 * \sa SDL_PutAudioStreamData
 * \sa SDL_GetAudioStreamData
 * \sa SDL_GetAudioStreamAvailable
 * \sa SDL_FlushAudioStream
 * \sa SDL_ClearAudioStream
 */
extern DECLSPEC void SDLCALL SDL_DestroyAudioStream(SDL_AudioStream *stream);


/**
 * Convenience function for straightforward audio init for the common case.
 *
 * If all your app intends to do is provide a single source of PCM audio, this
 * function allows you to do all your audio setup in a single call.
 *
 * This is intended to be a clean means to migrate apps from SDL2.
 *
 * This function will open an audio device, create a stream and bind it.
 * Unlike other methods of setup, the audio device will be closed when this
 * stream is destroyed, so the app can treat the returned SDL_AudioStream as
 * the only object needed to manage audio playback.
 *
 * Also unlike other functions, the audio device begins paused. This is to map
 * more closely to SDL2-style behavior, and since there is no extra step here
 * to bind a stream to begin audio flowing.
 *
 * This function works with both playback and capture devices.
 *
 * The `spec` parameter represents the app's side of the audio stream. That
 * is, for recording audio, this will be the output format, and for playing
 * audio, this will be the input format.
 *
 * If you don't care about opening a specific audio device, you can (and
 * probably _should_), use SDL_AUDIO_DEVICE_DEFAULT_OUTPUT for playback and
 * SDL_AUDIO_DEVICE_DEFAULT_CAPTURE for recording.
 *
 * One can optionally provide a callback function; if NULL, the app is
 * expected to queue audio data for playback (or unqueue audio data if
 * capturing). Otherwise, the callback will begin to fire once the device is
 * unpaused.
 *
 * \param devid an audio device to open, or SDL_AUDIO_DEVICE_DEFAULT_OUTPUT or
 *              SDL_AUDIO_DEVICE_DEFAULT_CAPTURE.
 * \param spec the audio stream's data format. Required.
 * \param callback A callback where the app will provide new data for
 *                 playback, or receive new data for capture. Can be NULL, in
 *                 which case the app will need to call SDL_PutAudioStreamData
 *                 or SDL_GetAudioStreamData as necessary.
 * \param userdata App-controlled pointer passed to callback. Can be NULL.
 *                 Ignored if callback is NULL.
 * \returns an audio stream on success, ready to use. NULL on error; call
 *          SDL_GetError() for more information. When done with this stream,
 *          call SDL_DestroyAudioStream to free resources and close the
 *          device.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC SDL_AudioStream *SDLCALL SDL_OpenAudioDeviceStream(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec, SDL_AudioStreamCallback callback, void *userdata);


/**
 * A callback that fires when data is about to be fed to an audio device.
 *
 * This is useful for accessing the final mix, perhaps for writing a
 * visualizer or applying a final effect to the audio data before playback.
 *
 * \sa SDL_SetAudioDevicePostmixCallback
 */
typedef void (SDLCALL *SDL_AudioPostmixCallback)(void *userdata, const SDL_AudioSpec *spec, float *buffer, int buflen);

/**
 * Set a callback that fires when data is about to be fed to an audio device.
 *
 * This is useful for accessing the final mix, perhaps for writing a
 * visualizer or applying a final effect to the audio data before playback.
 *
 * The buffer is the final mix of all bound audio streams on an opened
 * device; this callback will fire regularly for any device that is both
 * opened and unpaused. If there is no new data to mix, either because no
 * streams are bound to the device or all the streams are empty, this
 * callback will still fire with the entire buffer set to silence.
 *
 * This callback is allowed to make changes to the data; the contents of
 * the buffer after this call is what is ultimately passed along to the
 * hardware.
 *
 * The callback is always provided the data in float format (values from
 * -1.0f to 1.0f), but the number of channels or sample rate may be
 * different than the format the app requested when opening the device; SDL
 * might have had to manage a conversion behind the scenes, or the playback
 * might have jumped to new physical hardware when a system default changed,
 * etc. These details may change between calls. Accordingly, the size of the
 * buffer might change between calls as well.
 *
 * This callback can run at any time, and from any thread; if you need to
 * serialize access to your app's data, you should provide and use a mutex or
 * other synchronization device.
 *
 * All of this to say: there are specific needs this callback can fulfill,
 * but it is not the simplest interface. Apps should generally provide audio
 * in their preferred format through an SDL_AudioStream and let SDL handle
 * the difference.
 *
 * This function is extremely time-sensitive; the callback should do the
 * least amount of work possible and return as quickly as it can. The longer
 * the callback runs, the higher the risk of audio dropouts or other problems.
 *
 * This function will block until the audio device is in between iterations,
 * so any existing callback that might be running will finish before this
 * function sets the new callback and returns.
 *
 * Setting a NULL callback function disables any previously-set callback.
 *
 * \param devid The ID of an opened audio device.
 * \param callback A callback function to be called. Can be NULL.
 * \param userdata App-controlled pointer passed to callback. Can be NULL.
 * \returns zero on success, -1 on error; call SDL_GetError() for more
 *          information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC int SDLCALL SDL_SetAudioPostmixCallback(SDL_AudioDeviceID devid, SDL_AudioPostmixCallback callback, void *userdata);


/**
 * 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.
 *
 * 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 return value is zero 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_free() 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 function 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 SDL_TRUE, calls SDL_RWclose() on `src` before returning,
 *                even in the case of an error
 * \param spec A pointer to an SDL_AudioSpec that will be set to the WAVE
 *             data's format details on successful return
 * \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 0. `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 -1 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_free() to dispose of it.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_free
 * \sa SDL_LoadWAV
 */
extern DECLSPEC int SDLCALL SDL_LoadWAV_RW(SDL_RWops * src, SDL_bool freesrc,
                                           SDL_AudioSpec * spec, Uint8 ** audio_buf,
                                           Uint32 * audio_len);

/**
 * Loads a WAV from a file path.
 *
 * This is a convenience function that is effectively the same as:
 *
 * ```c
 * SDL_LoadWAV_RW(SDL_RWFromFile(path, "rb"), 1, spec, audio_buf, audio_len);
 * ```
 *
 * Note that in SDL2, this was a preprocessor macro and not a real function.
 *
 * \param path The file path of the WAV file to open.
 * \param spec A pointer to an SDL_AudioSpec that will be set to the WAVE
 *             data's format details on successful return.
 * \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 0. `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 -1 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_free() to dispose of it.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_free
 * \sa SDL_LoadWAV_RW
 */
extern DECLSPEC int SDLCALL SDL_LoadWAV(const char *path, SDL_AudioSpec * spec,
                                        Uint8 ** audio_buf, Uint32 * audio_len);



#define SDL_MIX_MAXVOLUME 128

/**
 * 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
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC int SDLCALL SDL_MixAudioFormat(Uint8 * dst,
                                                const Uint8 * src,
                                                SDL_AudioFormat format,
                                                Uint32 len, int volume);

/**
 * Convert some audio data of one format to another format.
 *
 * Please note that this function is for convenience, but should not be used
 * to resample audio in blocks, as it will introduce audio artifacts on the
 * boundaries. You should only use this function if you are converting audio
 * data in its entirety in one call. If you want to convert audio in smaller
 * chunks, use an SDL_AudioStream, which is designed for this situation.
 *
 * Internally, this function creates and destroys an SDL_AudioStream on each
 * use, so it's also less efficient than using one directly, if you need to
 * convert multiple times.
 *
 * \param src_spec The format details of the input audio
 * \param src_data The audio data to be converted
 * \param src_len The len of src_data
 * \param dst_spec The format details of the output audio
 * \param dst_data Will be filled with a pointer to converted audio data,
 *                 which should be freed with SDL_free(). On error, it will be
 *                 NULL.
 * \param dst_len Will be filled with the len of dst_data
 * \returns 0 on success or a negative error code on failure; call
 *          SDL_GetError() for more information.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 *
 * \sa SDL_CreateAudioStream
 */
extern DECLSPEC int SDLCALL SDL_ConvertAudioSamples(const SDL_AudioSpec *src_spec,
                                                    const Uint8 *src_data,
                                                    int src_len,
                                                    const SDL_AudioSpec *dst_spec,
                                                    Uint8 **dst_data,
                                                    int *dst_len);


/**
 * Get the appropriate memset value for silencing an audio format.
 *
 * The value returned by this function can be used as the second argument to
 * memset (or SDL_memset) to set an audio buffer in a specific format to
 * silence.
 *
 * \param format the audio data format to query.
 * \returns A byte value that can be passed to memset.
 *
 * \threadsafety It is safe to call this function from any thread.
 *
 * \since This function is available since SDL 3.0.0.
 */
extern DECLSPEC int SDLCALL SDL_GetSilenceValueForFormat(SDL_AudioFormat format);


/* Ends C function definitions when using C++ */
#ifdef __cplusplus
}
#endif
#include <SDL3/SDL_close_code.h>

#endif /* SDL_audio_h_ */