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//! Audio playback.
use crate::{c_char, c_int, c_void, rwops::*, stdinc::*};
/// Audio format flags.
///
/// Use the appropriate const functions to query the bits.
///
/// ```txt
/// ++-----------------------sample is signed if set
/// ||
/// || ++----------sample is big-endian if set
/// || ||
/// || || ++---sample is float if set
/// || || ||
/// || || || +---sample bit size---+
/// || || || | |
/// 15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00
/// ```
/// (Unspecified bits are always zero.)
#[derive(Debug, Clone, Copy, Default, PartialEq, Eq, PartialOrd, Ord, Hash)]
#[repr(transparent)]
pub struct SDL_AudioFormat(pub u16);
/// Pits per sample. eg: i16 = 16, f32 = 32.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_BITSIZE(af: SDL_AudioFormat) -> u16 {
af.0 & 0xFF
}
/// If the sample type is a floating type.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_ISFLOAT(af: SDL_AudioFormat) -> bool {
af.0 & (1 << 8) != 0
}
/// If the samples are big-endian.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_ISBIGENDIAN(af: SDL_AudioFormat) -> bool {
af.0 & (1 << 12) != 0
}
/// If the samples are signed values.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_ISSIGNED(af: SDL_AudioFormat) -> bool {
af.0 & (1 << 15) != 0
}
/// If the samples are an int type.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_ISINT(af: SDL_AudioFormat) -> bool {
!SDL_AUDIO_ISFLOAT(af)
}
/// If the samples are little-endian.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_ISLITTLEENDIAN(af: SDL_AudioFormat) -> bool {
!SDL_AUDIO_ISBIGENDIAN(af)
}
/// If the samples are unsigned values.
#[inline]
#[must_use]
pub const fn SDL_AUDIO_ISUNSIGNED(af: SDL_AudioFormat) -> bool {
!SDL_AUDIO_ISSIGNED(af)
}
/// Unsigned 8-bit samples
pub const AUDIO_U8: SDL_AudioFormat = SDL_AudioFormat(0x0008);
/// Signed 8-bit samples
pub const AUDIO_S8: SDL_AudioFormat = SDL_AudioFormat(0x8008);
/// Unsigned 16-bit samples
pub const AUDIO_U16LSB: SDL_AudioFormat = SDL_AudioFormat(0x0010);
/// Signed 16-bit samples
pub const AUDIO_S16LSB: SDL_AudioFormat = SDL_AudioFormat(0x8010);
/// As [`AUDIO_S16LSB`], but big-endian byte order
pub const AUDIO_U16MSB: SDL_AudioFormat = SDL_AudioFormat(0x1010);
/// As [`AUDIO_U16MSB`], but big-endian byte order
pub const AUDIO_S16MSB: SDL_AudioFormat = SDL_AudioFormat(0x9010);
/// Alias for [`AUDIO_U16LSB`]
pub const AUDIO_U16: SDL_AudioFormat = AUDIO_U16LSB;
/// Alias for [`AUDIO_S16LSB`]
pub const AUDIO_S16: SDL_AudioFormat = AUDIO_S16LSB;
/// 32-bit integer samples
pub const AUDIO_S32LSB: SDL_AudioFormat = SDL_AudioFormat(0x8020);
/// As [`AUDIO_S32MSB`], but big-endian byte order
pub const AUDIO_S32MSB: SDL_AudioFormat = SDL_AudioFormat(0x9020);
/// Alias for [`AUDIO_S32LSB`];
pub const AUDIO_S32: SDL_AudioFormat = AUDIO_S32LSB;
/// 32-bit floating point samples
pub const AUDIO_F32LSB: SDL_AudioFormat = SDL_AudioFormat(0x8120);
/// As [`AUDIO_F32LSB`], but big-endian byte order
pub const AUDIO_F32MSB: SDL_AudioFormat = SDL_AudioFormat(0x9120);
/// Alias for [`AUDIO_F32LSB`]
pub const AUDIO_F32: SDL_AudioFormat = AUDIO_F32LSB;
/// Native-endian u16 samples
pub const AUDIO_U16SYS: SDL_AudioFormat =
if cfg!(target_endian = "little") { AUDIO_U16LSB } else { AUDIO_U16MSB };
/// Native-endian i16 samples
pub const AUDIO_S16SYS: SDL_AudioFormat =
if cfg!(target_endian = "little") { AUDIO_S16LSB } else { AUDIO_S16MSB };
/// Native-endian i32 samples
pub const AUDIO_S32SYS: SDL_AudioFormat =
if cfg!(target_endian = "little") { AUDIO_S32LSB } else { AUDIO_S32MSB };
/// Native-endian f32 samples
pub const AUDIO_F32SYS: SDL_AudioFormat =
if cfg!(target_endian = "little") { AUDIO_F32LSB } else { AUDIO_F32MSB };
/// This function is called when the audio device needs more data.
///
/// * `userdata` An application-specific parameter saved in the `SDL_AudioSpec`
/// structure
/// * `stream` A pointer to the audio data buffer.
/// * `len` The length of that buffer in bytes.
///
/// The buffer is uninitialized when the callback is called.
///
/// The callback *must* initialize the entire buffer.
///
/// Once the callback returns, the buffer will no longer be valid.
///
/// See [`SDL_AudioSpec`] for default sample ordering.
///
/// You can choose to avoid callbacks and use SDL_QueueAudio() instead, if you
/// like. Just open your audio device with a NULL callback.
pub type SDL_AudioCallback = Option<
unsafe extern "C" fn(userdata: *mut c_void, stream: *mut Uint8, len: c_int),
>;
/// The calculated values in this structure are calculated by SDL_OpenAudio().
///
/// For multi-channel audio, the default SDL channel mapping is:
/// ```txt
/// 2: FL FR (stereo)
/// 3: FL FR LFE (2.1 surround)
/// 4: FL FR BL BR (quad)
/// 5: FL FR FC BL BR (quad + center)
/// 6: FL FR FC LFE SL SR (5.1 surround - last two can also be BL BR)
/// 7: FL FR FC LFE BC SL SR (6.1 surround)
/// 8: FL FR FC LFE BL BR SL SR (7.1 surround)
/// ```
#[repr(C)]
pub struct SDL_AudioSpec {
/// DSP frequency -- samples per second
pub freq: c_int,
/// Audio data format
pub format: SDL_AudioFormat,
/// Number of channels: 1 mono, 2 stereo, etc
pub channels: Uint8,
/// Audio buffer silence value (calculated)
pub silence: Uint8,
/// Audio buffer size in sample FRAMES (total samples divided by channel
/// count)
pub samples: Uint16,
/// Necessary for some compile environments
pub padding: Uint16,
/// Audio buffer size in bytes (calculated)
pub size: Uint32,
/// Callback that feeds the audio device (NULL to use SDL_QueueAudio()).
pub callback: SDL_AudioCallback,
/// Userdata passed to callback (ignored for NULL callbacks).
pub userdata: *mut c_void,
}
/// Filter function used by [`SDL_AudioCVT`]
pub type SDL_AudioFilter =
Option<unsafe extern "C" fn(cvt: *mut SDL_AudioCVT, format: SDL_AudioFormat)>;
/// A structure to hold a set of audio conversion filters and buffers.
///
/// Note that various parts of the conversion pipeline can take advantage
/// of SIMD operations (like SSE2, for example). SDL_AudioCVT doesn't require
/// you to pass it aligned data, but can possibly run much faster if you
/// set both its (buf) field to a pointer that is aligned to 16 bytes, and its
/// (len) field to something that's a multiple of 16, if possible.
#[repr(C)]
pub struct SDL_AudioCVT {
/// Set to 1 if conversion possible
pub needed: c_int,
/// Source audio format
pub src_format: SDL_AudioFormat,
/// Target audio format
pub dst_format: SDL_AudioFormat,
/// Rate conversion increment
pub rate_incr: f64,
/// Buffer to hold entire audio data
pub buf: *mut Uint8,
/// Length of original audio buffer
pub len: c_int,
/// Length of converted audio buffer
pub len_cvt: c_int,
/// buffer must be len*len_mult big
pub len_mult: c_int,
/// Given len, final size is len*len_ratio
pub len_ratio: f64,
/// NULL-terminated list of filter functions
pub filters: [SDL_AudioFilter; 10],
/// Current audio conversion function
pub filter_index: c_int,
}
/// You can only use 9 slots in the `SDL_AudioCVT.filters` field. The 10th is
/// reserved as a null terminator.
pub const SDL_AUDIOCVT_MAX_FILTERS: u32 = 9;
/// Identifies an audio device.
///
/// A successful call to [`SDL_OpenAudio`] is always device id 1, and legacy SDL
/// audio APIs assume you want this device ID. [`SDL_OpenAudioDevice`] calls
/// always returns devices >= 2 on success. The legacy calls are good both for
/// backwards compatibility and when you don't care about multiple devices,
/// specific devices, or capture devices.
#[derive(Debug, Clone, Copy, Default, PartialEq, Eq, PartialOrd, Ord, Hash)]
#[repr(transparent)]
pub struct SDL_AudioDeviceID(pub u32);
/// The status of an audio device.
#[derive(Debug, Clone, Copy, Default, PartialEq, Eq, PartialOrd, Ord, Hash)]
#[repr(transparent)]
pub struct SDL_AudioStatus(pub u32);
#[allow(missing_docs)]
pub const SDL_AUDIO_STOPPED: SDL_AudioStatus = SDL_AudioStatus(0);
#[allow(missing_docs)]
pub const SDL_AUDIO_PLAYING: SDL_AudioStatus = SDL_AudioStatus(1);
#[allow(missing_docs)]
pub const SDL_AUDIO_PAUSED: SDL_AudioStatus = SDL_AudioStatus(2);
/// SDL_AudioStream is a new audio conversion interface.
///
/// The benefits vs SDL_AudioCVT:
/// * it can handle re-sampling data in chunks without generating artifacts,
/// when it doesn't have the complete buffer available.
/// * it can handle incoming data in any variable size.
/// * You push data as you have it, and pull it when you need it.
#[allow(unused)]
#[repr(transparent)]
pub struct SDL_AudioStream(c_void);
/// Maximum volume value, for use with [`SDL_MixAudio`].
pub const SDL_MIX_MAXVOLUME: c_int = 128;
extern "C" {
/// The number of audio drivers. See [`SDL_GetAudioDriver`]
pub fn SDL_GetNumAudioDrivers() -> c_int;
/// The name of an audio driver. See [`SDL_GetAudioDriver`]
pub fn SDL_GetAudioDriver(index: c_int) -> *const c_char;
// skipped because they're mostly internal use: SDL_AudioInit / SDL_AudioQuit
/// The name of the current audio driver (or null if no driver is
/// initialized).
pub fn SDL_GetCurrentAudioDriver() -> *const c_char;
/// This function opens the audio device with the desired parameters, and
/// returns 0 if successful, placing the actual hardware parameters in the
/// structure pointed to by `obtained`. If `obtained` is NULL, the audio
/// data passed to the callback function will be guaranteed to be in the
/// requested format, and will be automatically converted to the hardware
/// audio format if necessary. This function returns -1 if it failed
/// to open the audio device, or couldn't set up the audio thread.
///
/// When filling in the desired audio spec structure,
/// - `desired->freq` should be the desired audio frequency in samples-per-
/// second.
/// - `desired->format` should be the desired audio format.
/// - `desired->samples` is the desired size of the audio buffer, in samples.
/// This number should be a power of two, and may be adjusted by the audio
/// driver to a value more suitable for the hardware. Good values seem to
/// range between 512 and 8096 inclusive, depending on the application and
/// CPU speed. Smaller values yield faster response time, but can lead to
/// underflow if the application is doing heavy processing and cannot fill
/// the audio buffer in time. A stereo sample consists of both right and
/// left channels in LR ordering. Note that the number of samples is
/// directly related to time by the following formula: `ms =
/// (samples*1000)/freq`
/// - `desired->size` is the size in bytes of the audio buffer, and is
/// calculated by SDL_OpenAudio().
/// - `desired->silence` is the value used to set the buffer to silence, and
/// is calculated by SDL_OpenAudio().
/// - `desired->callback` should be set to a function that will be called when
/// the audio device is ready for more data. It is passed a pointer to the
/// audio buffer, and the length in bytes of the audio buffer. This function
/// usually runs in a separate thread, and so you should protect data
/// structures that it accesses by calling SDL_LockAudio() and
/// SDL_UnlockAudio() in your code. Alternately, you may pass a NULL pointer
/// here, and call SDL_QueueAudio() with some frequency, to queue more audio
/// samples to be played (or for capture devices, call SDL_DequeueAudio()
/// with some frequency, to obtain audio samples).
/// - `desired->userdata` is passed as the first parameter to your callback
/// function. If you passed a NULL callback, this value is ignored.
///
/// The audio device starts out playing silence when it's opened, and should
/// be enabled for playing by calling [`SDL_PauseAudio(0)`](SDL_PauseAudio)
/// when you are ready for your audio callback function to be called.
/// Since the audio driver may modify the requested size of the audio
/// buffer, you should allocate any local mixing buffers after you open the
/// audio device.
pub fn SDL_OpenAudio(
desired: *mut SDL_AudioSpec, obtained: *mut SDL_AudioSpec,
) -> c_int;
/// Get the number of available devices exposed by the current driver.
///
/// Only valid after a successfully initializing the audio subsystem.
/// Returns -1 if an explicit list of devices can't be determined; this is
/// not an error. For example, if SDL is set up to talk to a remote audio
/// server, it can't list every one available on the Internet, but it will
/// still allow a specific host to be specified to [`SDL_OpenAudioDevice`].
///
/// In many common cases, when this function returns a value <= 0, it can
/// still successfully open the default device (NULL for first argument of
/// [`SDL_OpenAudioDevice`]).
pub fn SDL_GetNumAudioDevices(iscapture: c_int) -> c_int;
/// Get the human-readable name of a specific audio device.
///
/// Must be a value between 0 and (number of audio devices-1).
/// Only valid after a successfully initializing the audio subsystem.
/// The values returned by this function reflect the latest call to
/// [`SDL_GetNumAudioDevices`]; Call it again to re-detect available
/// hardware.
///
/// The string returned by this function is UTF-8 encoded, read-only, and
/// managed internally. You are not to free it. If you need to keep the
/// string for any length of time, you should make your own copy of it, as it
/// will be invalid next time any of several other SDL functions is called.
pub fn SDL_GetAudioDeviceName(
index: c_int, iscapture: c_int,
) -> *const c_char;
/// Open a specific audio device.
///
/// Passing in a device name of NULL requests
/// the most reasonable default (and is equivalent to calling
/// [`SDL_OpenAudio`]).
///
/// The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(),
/// but some drivers allow arbitrary and driver-specific strings, such as a
/// hostname/IP address for a remote audio server, or a filename in the
/// disk audio driver.
///
/// **Returns:** 0 on error, a valid device ID that is >= 2 on success.
///
/// [`SDL_OpenAudio`], unlike this function, always acts on device ID 1.
pub fn SDL_OpenAudioDevice(
device: *const c_char, iscapture: c_int, desired: *const SDL_AudioSpec,
obtained: *mut SDL_AudioSpec, allowed_changes: c_int,
) -> SDL_AudioDeviceID;
/// Gets the current audio status (stopped / playing / paused).
pub fn SDL_GetAudioStatus() -> SDL_AudioStatus;
/// Gets the current audio status (stopped / playing / paused).
pub fn SDL_GetAudioDeviceStatus(dev: SDL_AudioDeviceID) -> SDL_AudioStatus;
/// Pause and unpause the audio callback processing.
///
/// Should be called with a parameter of 0 after opening the audio
/// device to start playing sound. This is so you can safely initialize
/// data for your callback function after opening the audio device.
/// Silence will be written to the audio device during the pause.
pub fn SDL_PauseAudio(pause_on: c_int);
/// Pause and unpause the audio callback processing.
///
/// Should be called with a parameter of 0 after opening the audio
/// device to start playing sound. This is so you can safely initialize
/// data for your callback function after opening the audio device.
/// Silence will be written to the audio device during the pause.
pub fn SDL_PauseAudioDevice(dev: SDL_AudioDeviceID, pause_on: c_int);
/// Load the audio data of a WAVE file into memory
///
/// Loading a WAVE file requires `src`, `spec`, `audio_buf`, and `audio_len`
/// to be valid pointers. The entire data portion of the file is then loaded
/// into memory and decoded if necessary.
///
/// If `freesrc` is non-zero, the data source gets automatically closed and
/// freed before the function returns.
///
/// Supported 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 µ-law (8 bits). Other formats are currently unsupported and
/// cause an error.
///
/// If this function succeeds, the pointer returned by it is equal to `spec`
/// and the pointer to the audio data allocated by the function is written to
/// `audio_buf` and its length in bytes to `audio_len`. The [`SDL_AudioSpec`]
/// members `freq`, `channels`, and `format` are set to the values of the
/// audio data in the buffer. The `samples` member is set to a sane default
/// and all others are set to zero.
///
/// It's necessary to use [`SDL_FreeWAV`] to free the audio data returned in
/// `audio_buf` when it is no longer used.
///
/// Because of the under-specification of the Waveform 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:
/// ```txt
/// SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
/// ```
///
/// * `src` The data source with the WAVE data
/// * `freesrc` A integer value that makes the function close the data source
/// if non-zero
/// * `spec` A pointer filled with the audio format of the audio data
/// * `audio_buf` A pointer filled with the audio data allocated by the
/// function
/// * `audio_len` A pointer filled with the length of the audio data buffer in
/// bytes
///
/// **Return:** NULL on error, or non-NULL on success.
pub fn SDL_LoadWAV_RW(
src: *mut SDL_RWops, freesrc: c_int, spec: *mut SDL_AudioSpec,
audio_buf: *mut *mut Uint8, audio_len: *mut Uint32,
) -> *mut SDL_AudioSpec;
/// This function frees data previously allocated with [`SDL_LoadWAV_RW`]
pub fn SDL_FreeWAV(audio_buf: *mut Uint8);
/// Initializes an [`SDL_AudioCVT`].
///
/// This function takes a source format and rate and a destination format
/// and rate, and initializes the `cvt` structure with information needed
/// by [`SDL_ConvertAudio`] to convert a buffer of audio data from one format
/// to the other. An unsupported format causes an error and -1 will be
/// returned.
///
/// **Return:** 0 if no conversion is needed, 1 if the audio filter is set up,
/// or -1 on error.
pub fn SDL_BuildAudioCVT(
cvt: *mut SDL_AudioCVT, src_format: SDL_AudioFormat, src_channels: Uint8,
src_rate: c_int, dst_format: SDL_AudioFormat, dst_channels: Uint8,
dst_rate: c_int,
) -> c_int;
/// Performs an in-place audio conversion.
///
/// Once you have initialized the `cvt` structure using [`SDL_BuildAudioCVT`],
/// created an audio buffer `cvt->buf`, and filled it with `cvt->len` bytes of
/// audio data in the source format, this function will convert it in-place
/// to the desired format.
///
/// The data conversion may expand the size of the audio data, so the buffer
/// `cvt->buf` should be allocated after the `cvt` structure is initialized by
/// [`SDL_BuildAudioCVT`], and should be `cvt->len * cvt->len_mult` bytes
/// long.
///
/// **Return:** 0 on success, or -1 if `cvt->buf` is NULL.
pub fn SDL_ConvertAudio(cvt: *mut SDL_AudioCVT) -> c_int;
/// Create a new audio stream
///
/// * `src_format` The format of the source audio
/// * `src_channels` The number of channels of the source audio
/// * `src_rate` The sampling rate of the source audio
/// * `dst_format` The format of the desired audio output
/// * `dst_channels` The number of channels of the desired audio output
/// * `dst_rate` The sampling rate of the desired audio output
///
/// **Return:** 0 on success, or -1 on error.
pub fn SDL_NewAudioStream(
src_format: SDL_AudioFormat, src_channels: Uint8, src_rate: c_int,
dst_format: SDL_AudioFormat, dst_channels: Uint8, dst_rate: c_int,
) -> *mut SDL_AudioStream;
/// Add data to be converted/re-sampled to the stream
///
/// * `stream` The stream the audio data is being added to
/// * `buf` A pointer to the audio data to add
/// * `len` The number of bytes to write to the stream
///
/// **Return:** 0 on success, or -1 on error.
pub fn SDL_AudioStreamPut(
stream: *mut SDL_AudioStream, buf: *const c_void, len: c_int,
) -> c_int;
/// Get converted/re-sampled data from the stream
///
/// * `stream` The stream the audio is being requested from
/// * `buf` A buffer to fill with audio data
/// * `len` The maximum number of bytes to fill
///
/// **Return:** The number of bytes read from the stream, or -1 on error
pub fn SDL_AudioStreamGet(
stream: *mut SDL_AudioStream, buf: *mut c_void, len: c_int,
) -> c_int;
/// Get the number of converted/re-sampled 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.
pub fn SDL_AudioStreamAvailable(stream: *mut SDL_AudioStream) -> c_int;
/// Flush the stream.
///
/// This tell the stream that you're done sending data, and anything being
/// buffered should be converted/re-sampled and made available immediately.
///
/// It is legal to add more data to a stream after flushing, but there will
/// be audio gaps in the output. Generally this is intended to signal the
/// end of input, so the complete output becomes available.
pub fn SDL_AudioStreamFlush(stream: *mut SDL_AudioStream) -> c_int;
/// Clear any pending data in the stream without converting it
pub fn SDL_AudioStreamClear(stream: *mut SDL_AudioStream);
/// Free an audio stream
pub fn SDL_FreeAudioStream(stream: *mut SDL_AudioStream);
/// In-place mixes two audio sources.
///
/// This takes two audio buffers of the playing audio format and mixes
/// them, performing addition, volume adjustment, and overflow clipping.
/// The volume ranges from 0 - 128, and should be set to [`SDL_MIX_MAXVOLUME`]
/// for full audio volume. Note this does not change hardware volume.
///
/// This is provided for convenience, you can mix your own audio data.
pub fn SDL_MixAudio(
dst: *mut Uint8, src: *const Uint8, len: Uint32, volume: c_int,
);
/// Mix according to a format.
///
/// This works like [`SDL_MixAudio`], but you specify the audio format instead
/// of using the format of audio device 1. Thus it can be used when no audio
/// device is open at all.
pub fn SDL_MixAudioFormat(
dst: *mut Uint8, src: *const Uint8, format: SDL_AudioFormat, len: Uint32,
volume: c_int,
);
/// Queue more audio on non-callback devices.
///
/// (If you are looking to retrieve queued audio from a non-callback capture
/// device, you want [`SDL_DequeueAudio`] instead. This will return -1 to
/// signify an error if you use it with capture devices.)
///
/// SDL offers two ways to feed audio to the device: you can either supply a
/// callback that SDL triggers with some frequency to obtain more audio
/// (pull method), or you can supply no callback, and then SDL will expect
/// you to supply data at regular intervals (push method) with this function.
///
/// There are no limits on the amount of data you can queue, short of
/// exhaustion of address space. Queued data will drain to the device as
/// necessary without further intervention from you. If the device needs
/// audio but there is not enough queued, it will play silence to make up
/// the difference. This means you will have skips in your audio playback
/// if you aren't routinely queueing sufficient data.
///
/// This function copies the supplied data, so you are safe to free it when
/// the function returns. This function is thread-safe, but queueing to the
/// same device from two threads at once does not promise which buffer will
/// be queued first.
///
/// You may not queue audio on a device that is using an application-supplied
/// callback; doing so returns an error. You have to use the audio callback
/// or queue audio with this function, but not both.
///
/// You should **not** call [`SDL_LockAudio`] on the device before queueing;
/// SDL handles locking internally for this function.
///
/// * `dev` The device ID to which we will queue audio.
/// * `data` The data to queue to the device for later playback.
/// * `len` The number of bytes (not samples!) to which (data) points.
///
/// **Return:** 0 on success, or -1 on error.
pub fn SDL_QueueAudio(
dev: SDL_AudioDeviceID, data: *const c_void, len: Uint32,
) -> c_int;
/// Dequeue more audio on non-callback devices.
///
/// (If you are looking to queue audio for output on a non-callback playback
/// device, you want [``SDL_QueueAudio`] instead. This will always return 0
/// if you use it with playback devices.)
///
/// SDL offers two ways to retrieve audio from a capture device: you can
/// either supply a callback that SDL triggers with some frequency as the
/// device records more audio data, (push method), or you can supply no
/// callback, and then SDL will expect you to retrieve data at regular
/// intervals (pull method) with this function.
///
/// There are no limits on the amount of data you can queue, short of
/// exhaustion of address space. Data from the device will keep queuing as
/// necessary without further intervention from you. This means you will
/// eventually run out of memory if you aren't routinely dequeueing data.
///
/// Capture devices will not queue data when paused; if you are expecting
/// to not need captured audio for some length of time, use
/// SDL_PauseAudioDevice() to stop the capture device from queueing more
/// data. This can be useful during, say, level loading times. When
/// un-paused, capture devices will start queueing data from that point,
/// having flushed any capturable data available while paused.
///
/// This function is thread-safe, but dequeueing from the same device from
/// two threads at once does not promise which thread will dequeued data
/// first.
///
/// You may not dequeue audio from a device that is using an
/// application-supplied callback; doing so returns an error. You have to use
/// the audio callback, or dequeue audio with this function, but not both.
///
/// You should not call SDL_LockAudio() on the device before queueing; SDL
/// handles locking internally for this function.
///
/// * `dev` The device ID from which we will dequeue audio.
/// * `data` A pointer into where audio data should be copied.
/// * `len` The number of bytes (not samples!) to which (data) points.
///
/// **Return:** number of bytes dequeued, which could be less than requested.
pub fn SDL_DequeueAudio(
dev: SDL_AudioDeviceID, data: *mut c_void, len: Uint32,
) -> Uint32;
/// Get the number of bytes of still-queued audio.
///
/// * For playback device:
/// * This is the number of bytes that have been queued for playback with
/// SDL_QueueAudio(), but have not yet been sent to the hardware. This
/// number may shrink at any time, so this only informs of pending data.
/// * Once we've sent it to the hardware, this function can not decide the
/// exact byte boundary of what has been played. It's possible that we just
/// gave the hardware several kilobytes right before you called this
/// function, but it hasn't played any of it yet, or maybe half of it, etc.
///
/// * For capture devices:
/// * This is the number of bytes that have been captured by the device and
/// are waiting for you to dequeue. This number may grow at any time, so
/// this only informs of the lower-bound of available data.
///
/// You may not queue audio on a device that is using an application-supplied
/// callback; calling this function on such a device always returns 0. You
/// have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use the
/// audio callback, but not both.
///
/// You should not call SDL_LockAudio() on the device before querying; SDL
/// handles locking internally for this function.
///
/// * `dev` The device ID of which we will query queued audio size.
///
/// **Return:** Number of bytes (not samples!) of queued audio.
pub fn SDL_GetQueuedAudioSize(dev: SDL_AudioDeviceID) -> Uint32;
/// Drop any queued audio data.
///
/// For playback devices, this is any queued data
/// still waiting to be submitted to the hardware. For capture devices, this
/// is any data that was queued by the device that hasn't yet been dequeued by
/// the application.
///
/// Immediately after this call, [`SDL_GetQueuedAudioSize`] will return 0. For
/// playback devices, the hardware will start playing silence if more audio
/// isn't queued. Un-paused capture devices will start filling the queue again
/// as soon as they have more data available (which, depending on the state
/// of the hardware and the thread, could be before this function call
/// returns!).
///
/// This will not prevent playback of queued audio that's already been sent
/// to the hardware, as we can not undo that, so expect there to be some
/// fraction of a second of audio that might still be heard. This can be
/// useful if you want to, say, drop any pending music during a level change
/// in your game.
///
/// You may not queue audio on a device that is using an application-supplied
/// callback; calling this function on such a device is always a no-op.
/// You have to queue audio with [`SDL_QueueAudio`]/[`SDL_DequeueAudio`], or
/// use the audio callback, but not both.
///
/// You should not call SDL_LockAudio() on the device before clearing the
/// queue; SDL handles locking internally for this function.
///
/// This function always succeeds and thus returns void.
///
/// * `dev` The device ID of which to clear the audio queue.
pub fn SDL_ClearQueuedAudio(dev: SDL_AudioDeviceID);
/// Locks the default audio device.
///
/// The lock manipulated by this function protects the audio callback
/// function. During an [`SDL_LockAudio`]/[`SDL_UnlockAudio`] pair, you can be
/// guaranteed that the callback function is not running.
///
/// Do not call these from the callback function or you will cause deadlock.
pub fn SDL_LockAudio();
/// Locks the given audio device.
///
/// See [`SDL_LockAudio`]
pub fn SDL_LockAudioDevice(dev: SDL_AudioDeviceID);
/// Unlocks the default audio device.
///
/// See [`SDL_LockAudio`]
pub fn SDL_UnlockAudio();
/// Unlocks the given audio device.
///
/// See [`SDL_LockAudio`]
pub fn SDL_UnlockAudioDevice(dev: SDL_AudioDeviceID);
/// This function shuts down audio processing and closes the default audio
/// device.
pub fn SDL_CloseAudio();
/// This function shuts down audio processing and closes the given audio
/// device.
pub fn SDL_CloseAudioDevice(dev: SDL_AudioDeviceID);
/// Get the preferred audio format of a specific audio device.
///
/// This function is only valid after a successfully initializing the audio
/// subsystem. The values returned by this function reflect the latest call to
/// [SDL_GetNumAudioDevices]; re-call that function to re-detect available
/// hardware.
///
/// `spec` will be filled with the sample rate, sample format, and channel
/// count. All other values in the structure are filled with 0. When the
/// supported struct members are 0, SDL was unable to get the property from
/// the backend.
///
/// * `index` the index of the audio device; valid values range from 0 to
/// [SDL_GetNumAudioDevices] - 1
/// * `iscapture` non-zero to query the list of recording devices, zero to
/// query the list of output devices.
/// * `spec` The [SDL_AudioSpec] to be initialized by this function.
/// * **Returns:** 0 on success, nonzero on error
///
/// See Also: [SDL_GetNumAudioDevices]
pub fn SDL_GetAudioDeviceSpec(
index: c_int, iscapture: c_int, spec: *mut SDL_AudioSpec,
) -> c_int;
}
/// Loads a WAV file.
///
/// Works like [`SDL_LoadWAV_RW`], but automatically creates and then frees the
/// intermediate [`SDL_RWops`] for you.
#[inline]
pub unsafe fn SDL_LoadWAV(
file: *const c_char, spec: *mut SDL_AudioSpec, audio_buf: *mut *mut Uint8,
audio_len: *mut Uint32,
) -> *mut SDL_AudioSpec {
SDL_LoadWAV_RW(
SDL_RWFromFile(file, b"rb\0".as_ptr().cast()),
1,
spec,
audio_buf,
audio_len,
)
}