foo_out_sdl: SDL3/SDL_audio.h comparison

comparison SDL3/SDL_audio.h @ 1:20d02a178406 default tip

*: check in everything else yay

author	Paper <paper@tflc.us>
date	Mon, 05 Jan 2026 02:15:46 -0500
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+:20d02a178406
+/*
+Simple DirectMedia Layer
+Copyright (C) 1997-2025 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.
+*/
+/**
+* # CategoryAudio
+*
+* Audio functionality for the SDL library.
+*
+* All audio in SDL3 revolves around SDL_AudioStream. Whether you want to play
+* or record audio, convert it, stream it, buffer it, or mix it, you're going
+* to be passing it through an audio stream.
+*
+* Audio streams are quite flexible; they can accept any amount of data at a
+* time, in any supported format, and output it as needed in any other format,
+* even if the data format changes on either side halfway through.
+*
+* An app opens an audio device and binds any number of audio streams to it,
+* feeding more data to the streams as available. When the device needs more
+* data, it will pull it from all bound streams and mix them together for
+* playback.
+*
+* Audio streams can also use an app-provided callback to supply data
+* on-demand, which maps pretty closely to the SDL2 audio model.
+*
+* SDL also provides a simple .WAV loader in SDL_LoadWAV (and SDL_LoadWAV_IO
+* if you aren't reading from a file) as a basic means to load sound data into
+* your program.
+*
+* ## Logical audio devices
+*
+* In SDL3, opening a physical device (like a SoundBlaster 16 Pro) gives you a
+* logical device ID that you can bind audio streams to. In almost all cases,
+* logical devices can be used anywhere in the API that a physical device is
+* normally used. However, since each device opening generates a new logical
+* device, different parts of the program (say, a VoIP library, or
+* text-to-speech framework, or maybe some other sort of mixer on top of SDL)
+* can have their own device opens that do not interfere with each other; each
+* logical device will mix its separate audio down to a single buffer, fed to
+* the physical device, behind the scenes. As many logical devices as you like
+* can come and go; SDL will only have to open the physical device at the OS
+* level once, and will manage all the logical devices on top of it
+* internally.
+*
+* One other benefit of logical devices: if you don't open a specific physical
+* device, instead opting for the default, SDL can automatically migrate those
+* logical devices to different hardware as circumstances change: a user
+* plugged in headphones? The system default changed? SDL can transparently
+* migrate the logical devices to the correct physical device seamlessly and
+* keep playing; the app doesn't even have to know it happened if it doesn't
+* want to.
+*
+* ## Simplified audio
+*
+* As a simplified model for when a single source of audio is all that's
+* needed, an app can use SDL_OpenAudioDeviceStream, which is a single
+* function to open an audio device, create an audio stream, bind that stream
+* to the newly-opened device, and (optionally) provide a callback for
+* obtaining audio data. When using this function, the primary interface is
+* the SDL_AudioStream and the device handle is mostly hidden away; destroying
+* a stream created through this function will also close the device, stream
+* bindings cannot be changed, etc. One other quirk of this is that the device
+* is started in a _paused_ state and must be explicitly resumed; this is
+* partially to offer a clean migration for SDL2 apps and partially because
+* the app might have to do more setup before playback begins; in the
+* non-simplified form, nothing will play until a stream is bound to a device,
+* so they start _unpaused_.
+*
+* ## Channel layouts
+*
+* Audio data passing through SDL is uncompressed PCM data, interleaved. One
+* can provide their own decompression through an MP3, etc, decoder, but SDL
+* does not provide this directly. Each interleaved channel of data is meant
+* to be in a specific order.
+*
+* Abbreviations:
+*
+* - FRONT = single mono speaker
+* - FL = front left speaker
+* - FR = front right speaker
+* - FC = front center speaker
+* - BL = back left speaker
+* - BR = back right speaker
+* - SR = surround right speaker
+* - SL = surround left speaker
+* - BC = back center speaker
+* - LFE = low-frequency speaker
+*
+* These are listed in the order they are laid out in memory, so "FL, FR"
+* means "the front left speaker is laid out in memory first, then the front
+* right, then it repeats for the next audio frame".
+*
+* - 1 channel (mono) layout: FRONT
+* - 2 channels (stereo) layout: FL, FR
+* - 3 channels (2.1) layout: FL, FR, LFE
+* - 4 channels (quad) layout: FL, FR, BL, BR
+* - 5 channels (4.1) layout: FL, FR, LFE, BL, BR
+* - 6 channels (5.1) layout: FL, FR, FC, LFE, BL, BR (last two can also be
+*   SL, SR)
+* - 7 channels (6.1) layout: FL, FR, FC, LFE, BC, SL, SR
+* - 8 channels (7.1) layout: FL, FR, FC, LFE, BL, BR, SL, SR
+*
+* This is the same order as DirectSound expects, but applied to all
+* platforms; SDL will swizzle the channels as necessary if a platform expects
+* something different.
+*
+* SDL_AudioStream can also be provided channel maps to change this ordering
+* to whatever is necessary, in other audio processing scenarios.
+*/
+#ifndef SDL_audio_h_
+#define SDL_audio_h_
+#include <SDL3/SDL_stdinc.h>
+#include <SDL3/SDL_endian.h>
+#include <SDL3/SDL_error.h>
+#include <SDL3/SDL_mutex.h>
+#include <SDL3/SDL_properties.h>
+#include <SDL3/SDL_iostream.h>
+#include <SDL3/SDL_begin_code.h>
+/* Set up for C function definitions, even when using C++ */
+#ifdef __cplusplus
+extern "C" {
+#endif
+/**
+* Mask of bits in an SDL_AudioFormat that contains the format bit size.
+*
+* Generally one should use SDL_AUDIO_BITSIZE instead of this macro directly.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_MASK_BITSIZE       (0xFFu)
+/**
+* Mask of bits in an SDL_AudioFormat that contain the floating point flag.
+*
+* Generally one should use SDL_AUDIO_ISFLOAT instead of this macro directly.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_MASK_FLOAT         (1u<<8)
+/**
+* Mask of bits in an SDL_AudioFormat that contain the bigendian flag.
+*
+* Generally one should use SDL_AUDIO_ISBIGENDIAN or SDL_AUDIO_ISLITTLEENDIAN
+* instead of this macro directly.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_MASK_BIG_ENDIAN    (1u<<12)
+/**
+* Mask of bits in an SDL_AudioFormat that contain the signed data flag.
+*
+* Generally one should use SDL_AUDIO_ISSIGNED instead of this macro directly.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_MASK_SIGNED        (1u<<15)
+/**
+* Define an SDL_AudioFormat value.
+*
+* SDL does not support custom audio formats, so this macro is not of much use
+* externally, but it can be illustrative as to what the various bits of an
+* SDL_AudioFormat mean.
+*
+* For example, SDL_AUDIO_S32LE looks like this:
+*
+* ```c
+* SDL_DEFINE_AUDIO_FORMAT(1, 0, 0, 32)
+* ```
+*
+* \param signed 1 for signed data, 0 for unsigned data.
+* \param bigendian 1 for bigendian data, 0 for littleendian data.
+* \param flt 1 for floating point data, 0 for integer data.
+* \param size number of bits per sample.
+* \returns a format value in the style of SDL_AudioFormat.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_DEFINE_AUDIO_FORMAT(signed, bigendian, flt, size) \
+(((Uint16)(signed) << 15) | ((Uint16)(bigendian) << 12) | ((Uint16)(flt) << 8) | ((size) & SDL_AUDIO_MASK_BITSIZE))
+/**
+* Audio format.
+*
+* \since This enum is available since SDL 3.2.0.
+*
+* \sa SDL_AUDIO_BITSIZE
+* \sa SDL_AUDIO_BYTESIZE
+* \sa SDL_AUDIO_ISINT
+* \sa SDL_AUDIO_ISFLOAT
+* \sa SDL_AUDIO_ISBIGENDIAN
+* \sa SDL_AUDIO_ISLITTLEENDIAN
+* \sa SDL_AUDIO_ISSIGNED
+* \sa SDL_AUDIO_ISUNSIGNED
+*/
+typedef enum SDL_AudioFormat
+{
+SDL_AUDIO_UNKNOWN   = 0x0000u,  /**< Unspecified audio format */
+SDL_AUDIO_U8        = 0x0008u,  /**< Unsigned 8-bit samples */
+/* SDL_DEFINE_AUDIO_FORMAT(0, 0, 0, 8), */
+SDL_AUDIO_S8        = 0x8008u,  /**< Signed 8-bit samples */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 0, 0, 8), */
+SDL_AUDIO_S16LE     = 0x8010u,  /**< Signed 16-bit samples */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 0, 0, 16), */
+SDL_AUDIO_S16BE     = 0x9010u,  /**< As above, but big-endian byte order */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 1, 0, 16), */
+SDL_AUDIO_S32LE     = 0x8020u,  /**< 32-bit integer samples */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 0, 0, 32), */
+SDL_AUDIO_S32BE     = 0x9020u,  /**< As above, but big-endian byte order */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 1, 0, 32), */
+SDL_AUDIO_F32LE     = 0x8120u,  /**< 32-bit floating point samples */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 0, 1, 32), */
+SDL_AUDIO_F32BE     = 0x9120u,  /**< As above, but big-endian byte order */
+/* SDL_DEFINE_AUDIO_FORMAT(1, 1, 1, 32), */
+/* These represent the current system's byteorder. */
+#if SDL_BYTEORDER == SDL_LIL_ENDIAN
+SDL_AUDIO_S16 = SDL_AUDIO_S16LE,
+SDL_AUDIO_S32 = SDL_AUDIO_S32LE,
+SDL_AUDIO_F32 = SDL_AUDIO_F32LE
+#else
+SDL_AUDIO_S16 = SDL_AUDIO_S16BE,
+SDL_AUDIO_S32 = SDL_AUDIO_S32BE,
+SDL_AUDIO_F32 = SDL_AUDIO_F32BE
+#endif
+} SDL_AudioFormat;
+/**
+* Retrieve the size, in bits, from an SDL_AudioFormat.
+*
+* For example, `SDL_AUDIO_BITSIZE(SDL_AUDIO_S16)` returns 16.
+*
+* \param x an SDL_AudioFormat value.
+* \returns data size in bits.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_BITSIZE(x)         ((x) & SDL_AUDIO_MASK_BITSIZE)
+/**
+* Retrieve the size, in bytes, from an SDL_AudioFormat.
+*
+* For example, `SDL_AUDIO_BYTESIZE(SDL_AUDIO_S16)` returns 2.
+*
+* \param x an SDL_AudioFormat value.
+* \returns data size in bytes.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_BYTESIZE(x)        (SDL_AUDIO_BITSIZE(x) / 8)
+/**
+* Determine if an SDL_AudioFormat represents floating point data.
+*
+* For example, `SDL_AUDIO_ISFLOAT(SDL_AUDIO_S16)` returns 0.
+*
+* \param x an SDL_AudioFormat value.
+* \returns non-zero if format is floating point, zero otherwise.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_ISFLOAT(x)         ((x) & SDL_AUDIO_MASK_FLOAT)
+/**
+* Determine if an SDL_AudioFormat represents bigendian data.
+*
+* For example, `SDL_AUDIO_ISBIGENDIAN(SDL_AUDIO_S16LE)` returns 0.
+*
+* \param x an SDL_AudioFormat value.
+* \returns non-zero if format is bigendian, zero otherwise.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_ISBIGENDIAN(x)     ((x) & SDL_AUDIO_MASK_BIG_ENDIAN)
+/**
+* Determine if an SDL_AudioFormat represents littleendian data.
+*
+* For example, `SDL_AUDIO_ISLITTLEENDIAN(SDL_AUDIO_S16BE)` returns 0.
+*
+* \param x an SDL_AudioFormat value.
+* \returns non-zero if format is littleendian, zero otherwise.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_ISLITTLEENDIAN(x)  (!SDL_AUDIO_ISBIGENDIAN(x))
+/**
+* Determine if an SDL_AudioFormat represents signed data.
+*
+* For example, `SDL_AUDIO_ISSIGNED(SDL_AUDIO_U8)` returns 0.
+*
+* \param x an SDL_AudioFormat value.
+* \returns non-zero if format is signed, zero otherwise.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_ISSIGNED(x)        ((x) & SDL_AUDIO_MASK_SIGNED)
+/**
+* Determine if an SDL_AudioFormat represents integer data.
+*
+* For example, `SDL_AUDIO_ISINT(SDL_AUDIO_F32)` returns 0.
+*
+* \param x an SDL_AudioFormat value.
+* \returns non-zero if format is integer, zero otherwise.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_ISINT(x)           (!SDL_AUDIO_ISFLOAT(x))
+/**
+* Determine if an SDL_AudioFormat represents unsigned data.
+*
+* For example, `SDL_AUDIO_ISUNSIGNED(SDL_AUDIO_S16)` returns 0.
+*
+* \param x an SDL_AudioFormat value.
+* \returns non-zero if format is unsigned, zero otherwise.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_ISUNSIGNED(x)      (!SDL_AUDIO_ISSIGNED(x))
+/**
+* SDL Audio Device instance IDs.
+*
+* Zero is used to signify an invalid/null device.
+*
+* \since This datatype is available since SDL 3.2.0.
+*/
+typedef Uint32 SDL_AudioDeviceID;
+/**
+* A value used to request a default playback audio device.
+*
+* Several functions that require an SDL_AudioDeviceID will accept this value
+* to signify the app just wants the system to choose a default device instead
+* of the app providing a specific one.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK ((SDL_AudioDeviceID) 0xFFFFFFFFu)
+/**
+* A value used to request a default recording audio device.
+*
+* Several functions that require an SDL_AudioDeviceID will accept this value
+* to signify the app just wants the system to choose a default device instead
+* of the app providing a specific one.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_DEVICE_DEFAULT_RECORDING ((SDL_AudioDeviceID) 0xFFFFFFFEu)
+/**
+* Format specifier for audio data.
+*
+* \since This struct is available since SDL 3.2.0.
+*
+* \sa SDL_AudioFormat
+*/
+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) from an SDL_AudioSpec.
+*
+* This reports on the size of an audio sample frame: stereo Sint16 data (2
+* channels of 2 bytes each) would be 4 bytes per frame, for example.
+*
+* \param x an SDL_AudioSpec to query.
+* \returns the number of bytes used per sample frame.
+*
+* \threadsafety It is safe to call this macro from any thread.
+*
+* \since This macro is available since SDL 3.2.0.
+*/
+#define SDL_AUDIO_FRAMESIZE(x) (SDL_AUDIO_BYTESIZE((x).format) * (x).channels)
+/**
+* The opaque handle that represents an audio stream.
+*
+* 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.
+* - It can remap audio channels between inputs and outputs.
+* - 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.
+*
+* Audio streams are the core of the SDL3 audio interface. You create one or
+* more of them, bind them to an opened audio device, and feed data to them
+* (or for recording, consume data from them).
+*
+* \since This struct is available since SDL 3.2.0.
+*
+* \sa SDL_CreateAudioStream
+*/
+typedef struct SDL_AudioStream SDL_AudioStream;
+/* Function prototypes */
+/**
+* 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 3.2.0.
+*
+* \sa SDL_GetAudioDriver
+*/
+extern SDL_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 "wasapi". 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 3.2.0.
+*
+* \sa SDL_GetNumAudioDrivers
+*/
+extern SDL_DECLSPEC const char * SDLCALL SDL_GetAudioDriver(int index);
+/**
+* Get the name of the current audio driver.
+*
+* The names of drivers are all simple, low-ASCII identifiers, like "alsa",
+* "coreaudio" or "wasapi". These never have Unicode characters, and are not
+* meant to be proper names.
+*
+* \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 3.2.0.
+*/
+extern SDL_DECLSPEC const char * SDLCALL SDL_GetCurrentAudioDriver(void);
+/**
+* Get a list of currently-connected audio playback devices.
+*
+* This returns of list of available devices that play sound, perhaps to
+* speakers or headphones ("playback" devices). If you want devices that
+* record audio, like a microphone ("recording" devices), use
+* SDL_GetAudioRecordingDevices() instead.
+*
+* This only returns a list of physical devices; it will not have any device
+* IDs returned by SDL_OpenAudioDevice().
+*
+* If this function returns NULL, to signify an error, `*count` will be set to
+* zero.
+*
+* \param count a pointer filled in with the number of devices returned, may
+*              be NULL.
+* \returns a 0 terminated array of device instance IDs or NULL on error; call
+*          SDL_GetError() for more information. This should be freed with
+*          SDL_free() when it is no longer needed.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_OpenAudioDevice
+* \sa SDL_GetAudioRecordingDevices
+*/
+extern SDL_DECLSPEC SDL_AudioDeviceID * SDLCALL SDL_GetAudioPlaybackDevices(int *count);
+/**
+* Get a list of currently-connected audio recording devices.
+*
+* This returns of list of available devices that record audio, like a
+* microphone ("recording" devices). If you want devices that play sound,
+* perhaps to speakers or headphones ("playback" devices), use
+* SDL_GetAudioPlaybackDevices() instead.
+*
+* This only returns a list of physical devices; it will not have any device
+* IDs returned by SDL_OpenAudioDevice().
+*
+* If this function returns NULL, to signify an error, `*count` will be set to
+* zero.
+*
+* \param count a pointer filled in with the number of devices returned, may
+*              be NULL.
+* \returns a 0 terminated array of device instance IDs, or NULL on failure;
+*          call SDL_GetError() for more information. This should be freed
+*          with SDL_free() when it is no longer needed.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_OpenAudioDevice
+* \sa SDL_GetAudioPlaybackDevices
+*/
+extern SDL_DECLSPEC SDL_AudioDeviceID * SDLCALL SDL_GetAudioRecordingDevices(int *count);
+/**
+* Get the human-readable name of a specific audio device.
+*
+* **WARNING**: this function will work with SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK
+* and SDL_AUDIO_DEVICE_DEFAULT_RECORDING, returning the current default
+* physical devices' names. However, as the default device may change at any
+* time, it is likely better to show a generic name to the user, like "System
+* default audio device" or perhaps "default [currently %s]". Do not store
+* this name to disk to reidentify the device in a later run of the program,
+* as the default might change in general, and the string will be the name of
+* a specific device and not the abstract system default.
+*
+* \param devid the instance ID of the device to query.
+* \returns the name of the audio device, or NULL 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 3.2.0.
+*
+* \sa SDL_GetAudioPlaybackDevices
+* \sa SDL_GetAudioRecordingDevices
+*/
+extern SDL_DECLSPEC const 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_PLAYBACK or
+* SDL_AUDIO_DEVICE_DEFAULT_RECORDING here, which is useful for getting a
+* reasonable recommendation before opening the system-recommended default
+* device.
+*
+* You can also use this to request the current device buffer size. This is
+* specified in sample frames and represents the amount of data SDL will feed
+* to the physical hardware in each chunk. This can be converted to
+* milliseconds of audio with the following equation:
+*
+* `ms = (int) ((((Sint64) frames) * 1000) / spec.freq);`
+*
+* Buffer size is only important if you need low-level control over the audio
+* playback timing. Most apps do not need this.
+*
+* \param devid the instance ID of the device to query.
+* \param spec on return, will be filled with device details.
+* \param sample_frames pointer to store device buffer size, in sample frames.
+*                      Can be NULL.
+* \returns true on success or false 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 3.2.0.
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_GetAudioDeviceFormat(SDL_AudioDeviceID devid, SDL_AudioSpec *spec, int *sample_frames);
+/**
+* Get the current channel map of an audio device.
+*
+* Channel maps are optional; most things do not need them, instead passing
+* data in the [order that SDL expects](CategoryAudio#channel-layouts).
+*
+* Audio devices usually have no remapping applied. This is represented by
+* returning NULL, and does not signify an error.
+*
+* \param devid the instance ID of the device to query.
+* \param count On output, set to number of channels in the map. Can be NULL.
+* \returns an array of the current channel mapping, with as many elements as
+*          the current output spec's channels, or NULL if default. This
+*          should be freed with SDL_free() when it is no longer needed.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_SetAudioStreamInputChannelMap
+*/
+extern SDL_DECLSPEC int * SDLCALL SDL_GetAudioDeviceChannelMap(SDL_AudioDeviceID devid, int *count);
+/**
+* Open a specific audio device.
+*
+* You can open both playback and recording devices through this function.
+* Playback devices will take data from bound audio streams, mix it, and send
+* it to the hardware. Recording 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_PLAYBACK` or
+* `SDL_AUDIO_DEVICE_DEFAULT_RECORDING`. 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_PLAYBACK or
+*              SDL_AUDIO_DEVICE_DEFAULT_RECORDING 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 or 0 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 3.2.0.
+*
+* \sa SDL_CloseAudioDevice
+* \sa SDL_GetAudioDeviceFormat
+*/
+extern SDL_DECLSPEC SDL_AudioDeviceID SDLCALL SDL_OpenAudioDevice(SDL_AudioDeviceID devid, const SDL_AudioSpec *spec);
+/**
+* Determine if an audio device is physical (instead of logical).
+*
+* An SDL_AudioDeviceID that represents physical hardware is a physical
+* device; there is one for each piece of hardware that SDL can see. Logical
+* devices are created by calling SDL_OpenAudioDevice or
+* SDL_OpenAudioDeviceStream, and while each is associated with a physical
+* device, there can be any number of logical devices on one physical device.
+*
+* For the most part, logical and physical IDs are interchangeable--if you try
+* to open a logical device, SDL understands to assign that effort to the
+* underlying physical device, etc. However, it might be useful to know if an
+* arbitrary device ID is physical or logical. This function reports which.
+*
+* This function may return either true or false for invalid device IDs.
+*
+* \param devid the device ID to query.
+* \returns true if devid is a physical device, false if it is logical.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_IsAudioDevicePhysical(SDL_AudioDeviceID devid);
+/**
+* Determine if an audio device is a playback device (instead of recording).
+*
+* This function may return either true or false for invalid device IDs.
+*
+* \param devid the device ID to query.
+* \returns true if devid is a playback device, false if it is recording.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_IsAudioDevicePlayback(SDL_AudioDeviceID devid);
+/**
+* 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 devid a device opened by SDL_OpenAudioDevice().
+* \returns true on success or false 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 3.2.0.
+*
+* \sa SDL_ResumeAudioDevice
+* \sa SDL_AudioDevicePaused
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_PauseAudioDevice(SDL_AudioDeviceID devid);
+/**
+* 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 devid a device opened by SDL_OpenAudioDevice().
+* \returns true on success or false 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 3.2.0.
+*
+* \sa SDL_AudioDevicePaused
+* \sa SDL_PauseAudioDevice
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_ResumeAudioDevice(SDL_AudioDeviceID devid);
+/**
+* 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 devid a device opened by SDL_OpenAudioDevice().
+* \returns true if device is valid and paused, false otherwise.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_PauseAudioDevice
+* \sa SDL_ResumeAudioDevice
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_AudioDevicePaused(SDL_AudioDeviceID devid);
+/**
+* Get the gain of an audio device.
+*
+* The gain of a device is its volume; a larger gain means a louder output,
+* with a gain of zero being silence.
+*
+* Audio devices default to a gain of 1.0f (no change in output).
+*
+* Physical devices may not have their gain changed, only logical devices, and
+* this function will always return -1.0f when used on physical devices.
+*
+* \param devid the audio device to query.
+* \returns the gain of the device or -1.0f 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 3.2.0.
+*
+* \sa SDL_SetAudioDeviceGain
+*/
+extern SDL_DECLSPEC float SDLCALL SDL_GetAudioDeviceGain(SDL_AudioDeviceID devid);
+/**
+* Change the gain of an audio device.
+*
+* The gain of a device is its volume; a larger gain means a louder output,
+* with a gain of zero being silence.
+*
+* Audio devices default to a gain of 1.0f (no change in output).
+*
+* Physical devices may not have their gain changed, only logical devices, and
+* this function will always return false when used on physical devices. While
+* it might seem attractive to adjust several logical devices at once in this
+* way, it would allow an app or library to interfere with another portion of
+* the program's otherwise-isolated devices.
+*
+* This is applied, along with any per-audiostream gain, during playback to
+* the hardware, and can be continuously changed to create various effects. On
+* recording devices, this will adjust the gain before passing the data into
+* an audiostream; that recording audiostream can then adjust its gain further
+* when outputting the data elsewhere, if it likes, but that second gain is
+* not applied until the data leaves the audiostream again.
+*
+* \param devid the audio device on which to change gain.
+* \param gain the gain. 1.0f is no change, 0.0f is silence.
+* \returns true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_GetAudioDeviceGain
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioDeviceGain(SDL_AudioDeviceID devid, float gain);
+/**
+* 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 3.2.0.
+*
+* \sa SDL_OpenAudioDevice
+*/
+extern SDL_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 a playback device, data
+* for all bound streams will be mixed together and fed to the device. For a
+* recording 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 playback
+* devices, and its input format for recording 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 with SDL_SetAudioStreamFormat(). If the other
+* end of the stream's format has never been set (the audio stream was created
+* with a NULL audio spec), this function will set it to match the device
+* end's format.
+*
+* \param devid an audio device to bind a stream to.
+* \param streams an array of audio streams to bind.
+* \param num_streams number streams listed in the `streams` array.
+* \returns true on success or false 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 3.2.0.
+*
+* \sa SDL_BindAudioStreams
+* \sa SDL_UnbindAudioStream
+* \sa SDL_GetAudioStreamDevice
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_BindAudioStreams(SDL_AudioDeviceID devid, SDL_AudioStream * const *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 true on success or false 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 3.2.0.
+*
+* \sa SDL_BindAudioStreams
+* \sa SDL_UnbindAudioStream
+* \sa SDL_GetAudioStreamDevice
+*/
+extern SDL_DECLSPEC bool 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 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. Can be NULL or contain
+*                NULL.
+* \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 3.2.0.
+*
+* \sa SDL_BindAudioStreams
+*/
+extern SDL_DECLSPEC void SDLCALL SDL_UnbindAudioStreams(SDL_AudioStream * const *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. Can be NULL.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_BindAudioStream
+*/
+extern SDL_DECLSPEC void SDLCALL SDL_UnbindAudioStream(SDL_AudioStream *stream);
+/**
+* Query an audio stream for its currently-bound device.
+*
+* This reports the logical 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 3.2.0.
+*
+* \sa SDL_BindAudioStream
+* \sa SDL_BindAudioStreams
+*/
+extern SDL_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 a new audio stream on success or NULL 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 3.2.0.
+*
+* \sa SDL_PutAudioStreamData
+* \sa SDL_GetAudioStreamData
+* \sa SDL_GetAudioStreamAvailable
+* \sa SDL_FlushAudioStream
+* \sa SDL_ClearAudioStream
+* \sa SDL_SetAudioStreamFormat
+* \sa SDL_DestroyAudioStream
+*/
+extern SDL_DECLSPEC SDL_AudioStream * SDLCALL SDL_CreateAudioStream(const SDL_AudioSpec *src_spec, const SDL_AudioSpec *dst_spec);
+/**
+* Get the properties associated with an audio stream.
+*
+* The application can hang any data it wants here, but the following
+* properties are understood by SDL:
+*
+* - `SDL_PROP_AUDIOSTREAM_AUTO_CLEANUP_BOOLEAN`: if true (the default), the
+*   stream be automatically cleaned up when the audio subsystem quits. If set
+*   to false, the streams will persist beyond that. This property is ignored
+*   for streams created through SDL_OpenAudioDeviceStream(), and will always
+*   be cleaned up. Streams that are not cleaned up will still be unbound from
+*   devices when the audio subsystem quits. This property was added in SDL
+*   3.4.0.
+*
+* \param stream the SDL_AudioStream to query.
+* \returns a valid property ID on success or 0 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 3.2.0.
+*/
+extern SDL_DECLSPEC SDL_PropertiesID SDLCALL SDL_GetAudioStreamProperties(SDL_AudioStream *stream);
+#define SDL_PROP_AUDIOSTREAM_AUTO_CLEANUP_BOOLEAN "SDL.audiostream.auto_cleanup"
+/**
+* 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 true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_SetAudioStreamFormat
+*/
+extern SDL_DECLSPEC bool 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.
+*
+* Data that was previously queued in the stream will still be operated on in
+* the format that was current when it was added, which is to say you can put
+* the end of a sound file in one format to a stream, change formats for the
+* next sound file, and start putting that new data while the previous sound
+* file is still queued, and everything will still play back correctly.
+*
+* If a stream is bound to a device, then the format of the side of the stream
+* bound to a device cannot be changed (src_spec for recording devices,
+* dst_spec for playback devices). Attempts to make a change to this side will
+* be ignored, but this will not report an error. The other side's format can
+* be changed.
+*
+* \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 true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_GetAudioStreamFormat
+* \sa SDL_SetAudioStreamFrequencyRatio
+*/
+extern SDL_DECLSPEC bool 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 failure; call
+*          SDL_GetError() for more information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_SetAudioStreamFrequencyRatio
+*/
+extern SDL_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 1.0f will play the audio faster, and at a
+* higher pitch. A value less than 1.0f will play the audio slower, and at a
+* lower pitch. 1.0f means play at normal speed.
+*
+* This is applied during SDL_GetAudioStreamData, and can be continuously
+* changed to create various effects.
+*
+* \param stream the stream on which the frequency ratio is being changed.
+* \param ratio the frequency ratio. 1.0 is normal speed. Must be between 0.01
+*              and 100.
+* \returns true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_GetAudioStreamFrequencyRatio
+* \sa SDL_SetAudioStreamFormat
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamFrequencyRatio(SDL_AudioStream *stream, float ratio);
+/**
+* Get the gain of an audio stream.
+*
+* The gain of a stream is its volume; a larger gain means a louder output,
+* with a gain of zero being silence.
+*
+* Audio streams default to a gain of 1.0f (no change in output).
+*
+* \param stream the SDL_AudioStream to query.
+* \returns the gain of the stream or -1.0f on failure; call SDL_GetError()
+*          for more information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_SetAudioStreamGain
+*/
+extern SDL_DECLSPEC float SDLCALL SDL_GetAudioStreamGain(SDL_AudioStream *stream);
+/**
+* Change the gain of an audio stream.
+*
+* The gain of a stream is its volume; a larger gain means a louder output,
+* with a gain of zero being silence.
+*
+* Audio streams default to a gain of 1.0f (no change in output).
+*
+* This is applied during SDL_GetAudioStreamData, and can be continuously
+* changed to create various effects.
+*
+* \param stream the stream on which the gain is being changed.
+* \param gain the gain. 1.0f is no change, 0.0f is silence.
+* \returns true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \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 3.2.0.
+*
+* \sa SDL_GetAudioStreamGain
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamGain(SDL_AudioStream *stream, float gain);
+/**
+* Get the current input channel map of an audio stream.
+*
+* Channel maps are optional; most things do not need them, instead passing
+* data in the [order that SDL expects](CategoryAudio#channel-layouts).
+*
+* Audio streams default to no remapping applied. This is represented by
+* returning NULL, and does not signify an error.
+*
+* \param stream the SDL_AudioStream to query.
+* \param count On output, set to number of channels in the map. Can be NULL.
+* \returns an array of the current channel mapping, with as many elements as
+*          the current output spec's channels, or NULL if default. This
+*          should be freed with SDL_free() when it is no longer needed.
+*
+* \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 3.2.0.
+*
+* \sa SDL_SetAudioStreamInputChannelMap
+*/
+extern SDL_DECLSPEC int * SDLCALL SDL_GetAudioStreamInputChannelMap(SDL_AudioStream *stream, int *count);
+/**
+* Get the current output channel map of an audio stream.
+*
+* Channel maps are optional; most things do not need them, instead passing
+* data in the [order that SDL expects](CategoryAudio#channel-layouts).
+*
+* Audio streams default to no remapping applied. This is represented by
+* returning NULL, and does not signify an error.
+*
+* \param stream the SDL_AudioStream to query.
+* \param count On output, set to number of channels in the map. Can be NULL.
+* \returns an array of the current channel mapping, with as many elements as
+*          the current output spec's channels, or NULL if default. This
+*          should be freed with SDL_free() when it is no longer needed.
+*
+* \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 3.2.0.
+*
+* \sa SDL_SetAudioStreamInputChannelMap
+*/
+extern SDL_DECLSPEC int * SDLCALL SDL_GetAudioStreamOutputChannelMap(SDL_AudioStream *stream, int *count);
+/**
+* Set the current input channel map of an audio stream.
+*
+* Channel maps are optional; most things do not need them, instead passing
+* data in the [order that SDL expects](CategoryAudio#channel-layouts).
+*
+* The input channel map reorders data that is added to a stream via
+* SDL_PutAudioStreamData. Future calls to SDL_PutAudioStreamData must provide
+* data in the new channel order.
+*
+* Each item in the array represents an input channel, and its value is the
+* channel that it should be remapped to. To reverse a stereo signal's left
+* and right values, you'd have an array of `{ 1, 0 }`. It is legal to remap
+* multiple channels to the same thing, so `{ 1, 1 }` would duplicate the
+* right channel to both channels of a stereo signal. An element in the
+* channel map set to -1 instead of a valid channel will mute that channel,
+* setting it to a silence value.
+*
+* You cannot change the number of channels through a channel map, just
+* reorder/mute them.
+*
+* Data that was previously queued in the stream will still be operated on in
+* the order that was current when it was added, which is to say you can put
+* the end of a sound file in one order to a stream, change orders for the
+* next sound file, and start putting that new data while the previous sound
+* file is still queued, and everything will still play back correctly.
+*
+* Audio streams default to no remapping applied. Passing a NULL channel map
+* is legal, and turns off remapping.
+*
+* SDL will copy the channel map; the caller does not have to save this array
+* after this call.
+*
+* If `count` is not equal to the current number of channels in the audio
+* stream's format, this will fail. This is a safety measure to make sure a
+* race condition hasn't changed the format while this call is setting the
+* channel map.
+*
+* Unlike attempting to change the stream's format, the input channel map on a
+* stream bound to a recording device is permitted to change at any time; any
+* data added to the stream from the device after this call will have the new
+* mapping, but previously-added data will still have the prior mapping.
+*
+* \param stream the SDL_AudioStream to change.
+* \param chmap the new channel map, NULL to reset to default.
+* \param count The number of channels in the map.
+* \returns true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \threadsafety It is safe to call this function from any thread, as it holds
+*               a stream-specific mutex while running. Don't change the
+*               stream's format to have a different number of channels from a
+*               different thread at the same time, though!
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_SetAudioStreamInputChannelMap
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamInputChannelMap(SDL_AudioStream *stream, const int *chmap, int count);
+/**
+* Set the current output channel map of an audio stream.
+*
+* Channel maps are optional; most things do not need them, instead passing
+* data in the [order that SDL expects](CategoryAudio#channel-layouts).
+*
+* The output channel map reorders data that is leaving a stream via
+* SDL_GetAudioStreamData.
+*
+* Each item in the array represents an input channel, and its value is the
+* channel that it should be remapped to. To reverse a stereo signal's left
+* and right values, you'd have an array of `{ 1, 0 }`. It is legal to remap
+* multiple channels to the same thing, so `{ 1, 1 }` would duplicate the
+* right channel to both channels of a stereo signal. An element in the
+* channel map set to -1 instead of a valid channel will mute that channel,
+* setting it to a silence value.
+*
+* You cannot change the number of channels through a channel map, just
+* reorder/mute them.
+*
+* The output channel map can be changed at any time, as output remapping is
+* applied during SDL_GetAudioStreamData.
+*
+* Audio streams default to no remapping applied. Passing a NULL channel map
+* is legal, and turns off remapping.
+*
+* SDL will copy the channel map; the caller does not have to save this array
+* after this call.
+*
+* If `count` is not equal to the current number of channels in the audio
+* stream's format, this will fail. This is a safety measure to make sure a
+* race condition hasn't changed the format while this call is setting the
+* channel map.
+*
+* Unlike attempting to change the stream's format, the output channel map on
+* a stream bound to a recording device is permitted to change at any time;
+* any data added to the stream after this call will have the new mapping, but
+* previously-added data will still have the prior mapping. When the channel
+* map doesn't match the hardware's channel layout, SDL will convert the data
+* before feeding it to the device for playback.
+*
+* \param stream the SDL_AudioStream to change.
+* \param chmap the new channel map, NULL to reset to default.
+* \param count The number of channels in the map.
+* \returns true on success or false on failure; call SDL_GetError() for more
+*          information.
+*
+* \threadsafety It is safe to call this function from any thread, as it holds
+*               a stream-specific mutex while running. Don't change the
+*               stream's format to have a different number of channels from a
+*               a different thread at the same time, though!
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_SetAudioStreamInputChannelMap
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamOutputChannelMap(SDL_AudioStream *stream, const int *chmap, int count);
+/**
+* Add data 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 copies the 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 true on success or false 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 3.2.0.
+*
+* \sa SDL_ClearAudioStream
+* \sa SDL_FlushAudioStream
+* \sa SDL_GetAudioStreamData
+* \sa SDL_GetAudioStreamQueued
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_PutAudioStreamData(SDL_AudioStream *stream, const void *buf, int len);
+/**
+* A callback that fires for completed SDL_PutAudioStreamDataNoCopy() data.
+*
+* When using SDL_PutAudioStreamDataNoCopy() to provide data to an
+* SDL_AudioStream, it's not safe to dispose of the data until the stream has
+* completely consumed it. Often times it's difficult to know exactly when
+* this has happened.
+*
+* This callback fires once when the stream no longer needs the buffer,
+* allowing the app to easily free or reuse it.
+*
+* \param userdata an opaque pointer provided by the app for their personal
+*                 use.
+* \param buf the pointer provided to SDL_PutAudioStreamDataNoCopy().
+* \param buflen the size of buffer, in bytes, provided to
+*               SDL_PutAudioStreamDataNoCopy().
+*
+* \threadsafety This 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 before your callback
+*               is called, so your callback does not need to manage the lock
+*               explicitly.
+*
+* \since This datatype is available since SDL 3.4.0.
+*
+* \sa SDL_SetAudioStreamGetCallback
+* \sa SDL_SetAudioStreamPutCallback
+*/
+typedef void (SDLCALL *SDL_AudioStreamDataCompleteCallback)(void *userdata, const void *buf, int buflen);
+/**
+* Add external data to an audio stream without copying it.
+*
+* Unlike SDL_PutAudioStreamData(), this function does not make a copy of the
+* provided data, instead storing the provided pointer. This means that the
+* put operation does not need to allocate and copy the data, but the original
+* data must remain available until the stream is done with it, either by
+* being read from the stream in its entirety, or a call to
+* SDL_ClearAudioStream() or SDL_DestroyAudioStream().
+*
+* The 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.
+*
+* An optional callback may be provided, which is called when the stream no
+* longer needs the data. Once this callback fires, the stream will not access
+* the data again. This callback will fire for any reason the data is no
+* longer needed, including clearing or destroying the stream.
+*
+* Note that there is still an allocation to store tracking information, so
+* this function is more efficient for larger blocks of data. If you're
+* planning to put a few samples at a time, it will be more efficient to use
+* SDL_PutAudioStreamData(), which allocates and buffers in blocks.
+*
+* \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 add to the stream.
+* \param callback the callback function to call when the data is no longer
+*                 needed by the stream. May be NULL.
+* \param userdata an opaque pointer provided to the callback for its own
+*                 personal use.
+* \returns true on success or false 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 3.4.0.
+*
+* \sa SDL_ClearAudioStream
+* \sa SDL_FlushAudioStream
+* \sa SDL_GetAudioStreamData
+* \sa SDL_GetAudioStreamQueued
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_PutAudioStreamDataNoCopy(SDL_AudioStream *stream, const void *buf, int len, SDL_AudioStreamDataCompleteCallback callback, void *userdata);
+/**
+* Add data to the stream with each channel in a separate array.
+*
+* 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.
+*
+* The data will be interleaved and queued. Note that SDL_AudioStream only
+* operates on interleaved data, so this is simply a convenience function for
+* easily queueing data from sources that provide separate arrays. There is no
+* equivalent function to retrieve planar data.
+*
+* The arrays in `channel_buffers` are ordered as they are to be interleaved;
+* the first array will be the first sample in the interleaved data. Any
+* individual array may be NULL; in this case, silence will be interleaved for
+* that channel.
+*
+* `num_channels` specifies how many arrays are in `channel_buffers`. This can
+* be used as a safety to prevent overflow, in case the stream format has
+* changed elsewhere. If more channels are specified than the current input
+* spec, they are ignored. If less channels are specified, the missing arrays
+* are treated as if they are NULL (silence is written to those channels). If
+* the count is -1, SDL will assume the array count matches the current input
+* spec.
+*
+* Note that `num_samples` is the number of _samples per array_. This can also
+* be thought of as the number of _sample frames_ to be queued. A value of 1
+* with stereo arrays will queue two samples to the stream. This is different
+* than SDL_PutAudioStreamData, which wants the size of a single array in
+* bytes.
+*
+* \param stream the stream the audio data is being added to.
+* \param channel_buffers a pointer to an array of arrays, one array per
+*                        channel.
+* \param num_channels the number of arrays in `channel_buffers` or -1.
+* \param num_samples the number of _samples_ per array to write to the
+*                    stream.
+* \returns true on success or false 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 3.4.0.
+*
+* \sa SDL_ClearAudioStream
+* \sa SDL_FlushAudioStream
+* \sa SDL_GetAudioStreamData
+* \sa SDL_GetAudioStreamQueued
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_PutAudioStreamPlanarData(SDL_AudioStream *stream, const void * const *channel_buffers, int num_channels, int num_samples);
+/**
+* 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 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 3.2.0.
+*
+* \sa SDL_ClearAudioStream
+* \sa SDL_GetAudioStreamAvailable
+* \sa SDL_PutAudioStreamData
+*/
+extern SDL_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 or -1 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 3.2.0.
+*
+* \sa SDL_GetAudioStreamData
+* \sa SDL_PutAudioStreamData
+*/
+extern SDL_DECLSPEC int SDLCALL SDL_GetAudioStreamAvailable(SDL_AudioStream *stream);
+/**
+* Get the number of bytes currently queued.
+*
+* This is the number of bytes put into a stream as input, not the number that
+* can be retrieved as output. Because of several details, it's not possible
+* to calculate one number directly from the other. If you need to know how
+* much usable data can be retrieved right now, you should use
+* SDL_GetAudioStreamAvailable() and not this function.
+*
+* Note that audio streams can change their input format at any time, even if
+* there is still data queued in a different format, so the returned byte
+* count will not necessarily match the number of _sample frames_ available.
+* Users of this API should be aware of format changes they make when feeding
+* a stream and plan accordingly.
+*
+* Queued data is not converted until it is consumed by
+* SDL_GetAudioStreamData, so this value should be representative of the exact
+* data that was put into the stream.
+*
+* 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 bytes queued or -1 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 3.2.0.
+*
+* \sa SDL_PutAudioStreamData
+* \sa SDL_ClearAudioStream
+*/
+extern SDL_DECLSPEC int SDLCALL SDL_GetAudioStreamQueued(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 may 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 true on success or false 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 3.2.0.
+*
+* \sa SDL_PutAudioStreamData
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_FlushAudioStream(SDL_AudioStream *stream);
+/**
+* Clear any pending data in the stream.
+*
+* This drops any queued data, so there will be nothing to read from the
+* stream until more is added.
+*
+* \param stream the audio stream to clear.
+* \returns true on success or false 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 3.2.0.
+*
+* \sa SDL_GetAudioStreamAvailable
+* \sa SDL_GetAudioStreamData
+* \sa SDL_GetAudioStreamQueued
+* \sa SDL_PutAudioStreamData
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_ClearAudioStream(SDL_AudioStream *stream);
+/**
+* Use this function to pause audio playback on the audio device associated
+* with an audio stream.
+*
+* 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.
+*
+* 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.
+*
+* \param stream the audio stream associated with the audio device to pause.
+* \returns true on success or false 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 3.2.0.
+*
+* \sa SDL_ResumeAudioStreamDevice
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_PauseAudioStreamDevice(SDL_AudioStream *stream);
+/**
+* Use this function to unpause audio playback on the audio device associated
+* with an audio stream.
+*
+* This function unpauses audio processing for a given device that has
+* previously been paused. Once unpaused, any bound audio streams will begin
+* to progress again, and audio can be generated.
+*
+* SDL_OpenAudioDeviceStream opens audio devices in a paused state, so this
+* function call is required for audio playback to begin on such devices.
+*
+* \param stream the audio stream associated with the audio device to resume.
+* \returns true on success or false 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 3.2.0.
+*
+* \sa SDL_PauseAudioStreamDevice
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_ResumeAudioStreamDevice(SDL_AudioStream *stream);
+/**
+* Use this function to query if an audio device associated with a stream 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.
+*
+* \param stream the audio stream associated with the audio device to query.
+* \returns true if device is valid and paused, false otherwise.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_PauseAudioStreamDevice
+* \sa SDL_ResumeAudioStreamDevice
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_AudioStreamDevicePaused(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 true on success or false 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 3.2.0.
+*
+* \sa SDL_UnlockAudioStream
+*/
+extern SDL_DECLSPEC bool 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 true on success or false 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 3.2.0.
+*
+* \sa SDL_LockAudioStream
+*/
+extern SDL_DECLSPEC bool 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.
+*
+* Two values are offered here: one is the amount of additional data needed to
+* satisfy the immediate request (which might be zero if the stream already
+* has enough data queued) and the other is the total amount being requested.
+* In a Get call triggering a Put callback, these values can be different. In
+* a Put call triggering a Get callback, these values are always the same.
+*
+* Byte counts might be slightly overestimated due to buffering or resampling,
+* and may change from call to call.
+*
+* This callback is not required to do anything. Generally this is useful for
+* adding/reading data on demand, and the app will often put/get data as
+* appropriate, but the system goes on with the data currently available to it
+* if this callback does nothing.
+*
+* \param stream the SDL audio stream associated with this callback.
+* \param additional_amount the amount of data, in bytes, that is needed right
+*                          now.
+* \param total_amount the total amount of data requested, in bytes, that is
+*                     requested or available.
+* \param userdata an opaque pointer provided by the app for their personal
+*                 use.
+*
+* \threadsafety This 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 before your callback
+*               is called, so your callback does not need to manage the lock
+*               explicitly.
+*
+* \since This datatype is available since SDL 3.2.0.
+*
+* \sa SDL_SetAudioStreamGetCallback
+* \sa SDL_SetAudioStreamPutCallback
+*/
+typedef void (SDLCALL *SDL_AudioStreamCallback)(void *userdata, SDL_AudioStream *stream, int additional_amount, int total_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 `additional_amount` 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 requested
+*                 from the stream.
+* \param userdata an opaque pointer provided to the callback for its own
+*                 personal use.
+* \returns true on success or false on failure; call SDL_GetError() for more
+*          information. 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 3.2.0.
+*
+* \sa SDL_SetAudioStreamPutCallback
+*/
+extern SDL_DECLSPEC bool 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 `additional_amount` 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 true on success or false on failure; call SDL_GetError() for more
+*          information. 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 3.2.0.
+*
+* \sa SDL_SetAudioStreamGetCallback
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_SetAudioStreamPutCallback(SDL_AudioStream *stream, SDL_AudioStreamCallback callback, void *userdata);
+/**
+* Free an audio stream.
+*
+* This will release all allocated data, including any audio that is still
+* queued. You do not need to manually clear the stream first.
+*
+* If this stream was bound to an audio device, it is unbound during this
+* call. If this stream was created with SDL_OpenAudioDeviceStream, the audio
+* device that was opened alongside this stream's creation will be closed,
+* too.
+*
+* \param stream the audio stream to destroy.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*
+* \sa SDL_CreateAudioStream
+*/
+extern SDL_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 also 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, since there is no extra step here to
+* bind a stream to begin audio flowing. The audio device should be resumed
+* with SDL_ResumeAudioStreamDevice().
+*
+* This function works with both playback and recording 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 spec is NULL, the system will
+* choose the format, and the app can use SDL_GetAudioStreamFormat() to obtain
+* this information later.
+*
+* If you don't care about opening a specific audio device, you can (and
+* probably _should_), use SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and
+* SDL_AUDIO_DEVICE_DEFAULT_RECORDING 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.
+*
+* Destroying the returned stream with SDL_DestroyAudioStream will also close
+* the audio device associated with this stream.
+*
+* \param devid an audio device to open, or SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK
+*              or SDL_AUDIO_DEVICE_DEFAULT_RECORDING.
+* \param spec the audio stream's data format. Can be NULL.
+* \param callback a callback where the app will provide new data for
+*                 playback, or receive new data for recording. 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, or NULL on failure; 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 3.2.0.
+*
+* \sa SDL_GetAudioStreamDevice
+* \sa SDL_ResumeAudioStreamDevice
+*/
+extern SDL_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.
+*
+* This callback should run as quickly as possible and not block for any
+* significant time, as this callback delays submission of data to the audio
+* device, which can cause audio playback problems.
+*
+* The postmix callback _must_ be able to handle any audio data format
+* specified in `spec`, which can change between callbacks if the audio device
+* changed. However, this only covers frequency and channel count; data is
+* always provided here in SDL_AUDIO_F32 format.
+*
+* The postmix callback runs _after_ logical device gain and audiostream gain
+* have been applied, which is to say you can make the output data louder at
+* this point than the gain settings would suggest.
+*
+* \param userdata a pointer provided by the app through
+*                 SDL_SetAudioPostmixCallback, for its own use.
+* \param spec the current format of audio that is to be submitted to the
+*             audio device.
+* \param buffer the buffer of audio samples to be submitted. The callback can
+*               inspect and/or modify this data.
+* \param buflen the size of `buffer` in bytes.
+*
+* \threadsafety This will run from a background thread owned by SDL. The
+*               application is responsible for locking resources the callback
+*               touches that need to be protected.
+*
+* \since This datatype is available since SDL 3.2.0.
+*
+* \sa SDL_SetAudioPostmixCallback
+*/
+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 true on success or false 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 3.2.0.
+*/
+extern SDL_DECLSPEC bool 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.
+*
+* 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_IO(SDL_IOFromFile("sample.wav", "rb"), true, &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 closeio if true, calls SDL_CloseIO() 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 true on success. `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 false 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 3.2.0.
+*
+* \sa SDL_free
+* \sa SDL_LoadWAV
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_LoadWAV_IO(SDL_IOStream *src, bool closeio, 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_IO(SDL_IOFromFile(path, "rb"), true, spec, audio_buf, audio_len);
+* ```
+*
+* \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 true on success. `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 false 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 3.2.0.
+*
+* \sa SDL_free
+* \sa SDL_LoadWAV_IO
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_LoadWAV(const char *path, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
+/**
+* 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_MixAudio() 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 0.0 - 1.0, and should be set to 1.0 for full
+*               audio volume.
+* \returns true on success or false 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 3.2.0.
+*/
+extern SDL_DECLSPEC bool SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, SDL_AudioFormat format, Uint32 len, float 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 true on success or false 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 3.2.0.
+*/
+extern SDL_DECLSPEC bool 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 human readable name of an audio format.
+*
+* \param format the audio format to query.
+* \returns the human readable name of the specified audio format or
+*          "SDL_AUDIO_UNKNOWN" if the format isn't recognized.
+*
+* \threadsafety It is safe to call this function from any thread.
+*
+* \since This function is available since SDL 3.2.0.
+*/
+extern SDL_DECLSPEC const char * SDLCALL SDL_GetAudioFormatName(SDL_AudioFormat format);
+/**
+* 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 3.2.0.
+*/
+extern SDL_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_ */

Mercurial > foo_out_sdl

comparison SDL3/SDL_audio.h @ 1:20d02a178406 default tip