SDL3pp
A slim C++ wrapper for SDL3
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Audio Playback, Recording, and Mixing

Audio functionality for the SDL library. More...

Namespaces

namespace  SDL::prop
 Constants for Properties keys.
namespace  SDL::prop::AudioStream
 Properties for AudioStream.

Classes

class  SDL::AudioFormat
 Audio format. More...
struct  SDL::AudioDeviceBase
 Base class to AudioDevice. More...
struct  SDL::AudioDevice
 SDL Audio Device instance IDs. More...
struct  SDL::AudioStreamBase
 Base class to AudioStream. More...
struct  SDL::AudioStream
 The opaque handle that represents an audio stream. More...
class  SDL::AudioStreamLock
 Lock an audio stream for serialized access. More...

Typedefs

using SDL::AudioFormatRaw = SDL_AudioFormat
 Alias to raw representation for AudioFormat.
using SDL::AudioDeviceID = SDL_AudioDeviceID
 Alias to raw representation for AudioDevice.
using SDL::AudioDeviceRef = ResourceRefT<AudioDeviceBase>
 Reference for AudioDevice.
using SDL::AudioStreamRaw = SDL_AudioStream*
 Alias to raw representation for AudioStream.
using SDL::AudioStreamRef = ResourceRefT<AudioStreamBase>
 Reference for AudioStream.
using SDL::AudioSpec = SDL_AudioSpec
 Format specifier for audio data.
using SDL::AudioPostmixCallback
 A callback that fires when data is about to be fed to an audio device.
using SDL::AudioPostmixCB
 A callback that fires when data is about to be fed to an audio device.
using SDL::AudioStreamCallback
 A callback that fires when data passes through an AudioStream.
using SDL::AudioStreamCB
 A callback that fires when data passes through an AudioStream.
using SDL::AudioStreamDataCompleteCallback
 A callback that fires for completed PutAudioStreamDataNoCopy() data.
using SDL::AudioStreamDataCompleteCB
 A callback that fires for completed PutAudioStreamDataNoCopy() data.

Functions

constexpr AudioFormat SDL::DefineAudioFormat (bool sign, bool bigendian, bool flt, Uint16 size)
 Define an AudioFormat value.
constexpr Uint16 SDL::AudioBitSize (AudioFormatRaw x)
 Retrieve the size, in bits, from an AudioFormat.
constexpr Uint16 SDL::AudioByteSize (AudioFormatRaw x)
 Retrieve the size, in bytes, from an AudioFormat.
constexpr bool SDL::IsAudioFloat (AudioFormatRaw x)
 Determine if an AudioFormat represents floating point data.
constexpr bool SDL::IsAudioBigENDIAN (AudioFormatRaw x)
 Determine if an AudioFormat represents bigendian data.
constexpr bool SDL::IsAudioLittleEndian (AudioFormatRaw x)
 Determine if an AudioFormat represents littleendian data.
constexpr bool SDL::IsAudioSigned (AudioFormatRaw x)
 Determine if an AudioFormat represents signed data.
constexpr bool SDL::IsAudioInt (AudioFormatRaw x)
 Determine if an AudioFormat represents integer data.
constexpr bool SDL::IsAudioUnsigned (AudioFormatRaw x)
 Determine if an AudioFormat represents unsigned data.
constexpr int SDL::AudioFrameSize (const AudioSpec &x)
 Calculate the size of each audio frame (in bytes) from an AudioSpec.
int SDL::GetNumAudioDrivers ()
 Use this function to get the number of built-in audio drivers.
const char * SDL::GetAudioDriver (int index)
 Use this function to get the name of a built in audio driver.
const char * SDL::GetCurrentAudioDriver ()
 Get the name of the current audio driver.
OwnArray< AudioDeviceRefSDL::GetAudioPlaybackDevices ()
 Get a list of currently-connected audio playback devices.
OwnArray< AudioDeviceRefSDL::GetAudioRecordingDevices ()
 Get a list of currently-connected audio recording devices.
const char * SDL::GetAudioDeviceName (AudioDeviceRef devid)
 Get the human-readable name of a specific audio device.
AudioSpec SDL::GetAudioDeviceFormat (AudioDeviceRef devid, int *sample_frames=nullptr)
 Get the current audio format of a specific audio device.
OwnArray< int > SDL::GetAudioDeviceChannelMap (AudioDeviceRef devid)
 Get the current channel map of an audio device.
AudioDevice SDL::OpenAudioDevice (AudioDeviceRef devid, OptionalRef< const AudioSpec > spec)
 Open a specific audio device.
bool SDL::IsAudioDevicePhysical (AudioDeviceRef devid)
 Determine if an audio device is physical (instead of logical).
bool SDL::IsAudioDevicePlayback (AudioDeviceRef devid)
 Determine if an audio device is a playback device (instead of recording).
void SDL::PauseAudioDevice (AudioDeviceRef devid)
 Use this function to pause audio playback on a specified device.
void SDL::ResumeAudioDevice (AudioDeviceRef devid)
 Use this function to unpause audio playback on a specified device.
bool SDL::AudioDevicePaused (AudioDeviceRef devid)
 Use this function to query if an audio device is paused.
float SDL::GetAudioDeviceGain (AudioDeviceRef devid)
 Get the gain of an audio device.
void SDL::SetAudioDeviceGain (AudioDeviceRef devid, float gain)
 Change the gain of an audio device.
void SDL::CloseAudioDevice (AudioDeviceID devid)
 Close a previously-opened audio device.
void SDL::BindAudioStreams (AudioDeviceRef devid, std::span< AudioStreamRef > streams)
 Bind a list of audio streams to an audio device.
void SDL::BindAudioStream (AudioDeviceRef devid, AudioStreamRef stream)
 Bind a single audio stream to an audio device.
void SDL::UnbindAudioStreams (std::span< AudioStreamRef > streams)
 Unbind a list of audio streams from their audio devices.
void SDL::UnbindAudioStream (AudioStreamRef stream)
 Unbind a single audio stream from its audio device.
AudioDeviceRef SDL::GetAudioStreamDevice (AudioStreamRef stream)
 Query an audio stream for its currently-bound device.
AudioStream SDL::CreateAudioStream (OptionalRef< const AudioSpec > src_spec, OptionalRef< const AudioSpec > dst_spec)
 Create a new audio stream.
PropertiesRef SDL::GetAudioStreamProperties (AudioStreamRef stream)
 Get the properties associated with an audio stream.
void SDL::GetAudioStreamFormat (AudioStreamRef stream, AudioSpec *src_spec, AudioSpec *dst_spec)
 Query the current format of an audio stream.
void SDL::SetAudioStreamFormat (AudioStreamRef stream, OptionalRef< const AudioSpec > src_spec, OptionalRef< const AudioSpec > dst_spec)
 Change the input and output formats of an audio stream.
float SDL::GetAudioStreamFrequencyRatio (AudioStreamRef stream)
 Get the frequency ratio of an audio stream.
void SDL::SetAudioStreamFrequencyRatio (AudioStreamRef stream, float ratio)
 Change the frequency ratio of an audio stream.
float SDL::GetAudioStreamGain (AudioStreamRef stream)
 Get the gain of an audio stream.
void SDL::SetAudioStreamGain (AudioStreamRef stream, float gain)
 Change the gain of an audio stream.
OwnArray< int > SDL::GetAudioStreamInputChannelMap (AudioStreamRef stream)
 Get the current input channel map of an audio stream.
OwnArray< int > SDL::GetAudioStreamOutputChannelMap (AudioStreamRef stream)
 Get the current output channel map of an audio stream.
void SDL::SetAudioStreamInputChannelMap (AudioStreamRef stream, std::span< int > chmap)
 Set the current input channel map of an audio stream.
void SDL::SetAudioStreamOutputChannelMap (AudioStreamRef stream, std::span< int > chmap)
 Set the current output channel map of an audio stream.
void SDL::PutAudioStreamData (AudioStreamRef stream, SourceBytes buf)
 Add data to the stream.
void SDL::PutAudioStreamDataNoCopy (AudioStreamRef stream, SourceBytes buf, AudioStreamDataCompleteCallback callback, void *userdata)
 Add external data to an audio stream without copying it.
void SDL::PutAudioStreamDataNoCopy (AudioStreamRef stream, SourceBytes buf, AudioStreamDataCompleteCB callback)
 Add external data to an audio stream without copying it.
void SDL::PutAudioStreamPlanarData (AudioStreamRef stream, const void *const *channel_buffers, int num_channels, int num_samples)
 Add data to the stream with each channel in a separate array.
int SDL::GetAudioStreamData (AudioStreamRef stream, TargetBytes buf)
 Get converted/resampled data from the stream.
int SDL::GetAudioStreamAvailable (AudioStreamRef stream)
 Get the number of converted/resampled bytes available.
int SDL::GetAudioStreamQueued (AudioStreamRef stream)
 Get the number of bytes currently queued.
void SDL::FlushAudioStream (AudioStreamRef stream)
 Tell the stream that you're done sending data, and anything being buffered should be converted/resampled and made available immediately.
void SDL::ClearAudioStream (AudioStreamRef stream)
 Clear any pending data in the stream.
void SDL::PauseAudioStreamDevice (AudioStreamRef stream)
 Use this function to pause audio playback on the audio device associated with an audio stream.
void SDL::ResumeAudioStreamDevice (AudioStreamRef stream)
 Use this function to unpause audio playback on the audio device associated with an audio stream.
bool SDL::AudioStreamDevicePaused (AudioStreamRef stream)
 Use this function to query if an audio device associated with a stream is paused.
void SDL::LockAudioStream (AudioStreamRef stream)
 Lock an audio stream for serialized access.
void SDL::UnlockAudioStream (AudioStreamRef stream)
 Unlock an audio stream for serialized access.
void SDL::SetAudioStreamGetCallback (AudioStreamRef stream, AudioStreamCallback callback, void *userdata)
 Set a callback that runs when data is requested from an audio stream.
void SDL::SetAudioStreamGetCallback (AudioStreamRef stream, AudioStreamCB callback)
 Set a callback that runs when data is requested from an audio stream.
void SDL::SetAudioStreamPutCallback (AudioStreamRef stream, AudioStreamCallback callback, void *userdata)
 Set a callback that runs when data is added to an audio stream.
void SDL::SetAudioStreamPutCallback (AudioStreamRef stream, AudioStreamCB callback)
 Set a callback that runs when data is added to an audio stream.
void SDL::DestroyAudioStream (AudioStreamRaw stream)
 Free an audio stream.
AudioStream SDL::OpenAudioDeviceStream (AudioDeviceRef devid, OptionalRef< const AudioSpec > spec, AudioStreamCallback callback=nullptr, void *userdata=nullptr)
 Convenience function for straightforward audio init for the common case.
AudioStream SDL::OpenAudioDeviceStream (AudioDeviceRef devid, OptionalRef< const AudioSpec > spec, AudioStreamCB callback)
 Convenience function for straightforward audio init for the common case.
void SDL::SetAudioPostmixCallback (AudioDeviceRef devid, AudioPostmixCallback callback, void *userdata)
 Set a callback that fires when data is about to be fed to an audio device.
void SDL::SetAudioPostmixCallback (AudioDeviceRef devid, AudioPostmixCB callback)
 Set a callback that fires when data is about to be fed to an audio device.
OwnArray< Uint8SDL::LoadWAV_IO (IOStreamRef src, AudioSpec *spec, bool closeio=false)
 Load the audio data of a WAVE file into memory.
OwnArray< Uint8SDL::LoadWAV (StringParam path, AudioSpec *spec)
 Loads a WAV from a file path.
void SDL::MixAudio (Uint8 *dst, SourceBytes src, AudioFormat format, float volume)
 Mix audio data in a specified format.
void SDL::MixAudio (TargetBytes dst, SourceBytes src, AudioFormat format, float volume)
 Mix audio data in a specified format.
OwnArray< Uint8SDL::ConvertAudioSamples (const AudioSpec &src_spec, SourceBytes src_data, const AudioSpec &dst_spec)
 Convert some audio data of one format to another format.
const char * SDL::GetAudioFormatName (AudioFormatRaw format)
 Get the human readable name of an audio format.
int SDL::GetSilenceValueForFormat (AudioFormatRaw format)
 Get the appropriate memset value for silencing an audio format.
constexpr SDL::AudioFormat::AudioFormat (bool sign, bool bigendian, bool flt, Uint16 size)
 Define an AudioFormat value.
constexpr Uint16 SDL::AudioFormat::GetBitSize () const
 Retrieve the size, in bits, from an AudioFormat.
constexpr Uint16 SDL::AudioFormat::GetByteSize () const
 Retrieve the size, in bytes, from an AudioFormat.
constexpr bool SDL::AudioFormat::IsFloat () const
 Determine if an AudioFormat represents floating point data.
constexpr bool SDL::AudioFormat::IsBigEndian () const
 Determine if an AudioFormat represents bigendian data.
constexpr bool SDL::AudioFormat::IsLittleEndian () const
 Determine if an AudioFormat represents littleendian data.
constexpr bool SDL::AudioFormat::IsSigned () const
 Determine if an AudioFormat represents signed data.
constexpr bool SDL::AudioFormat::IsInt () const
 Determine if an AudioFormat represents integer data.
constexpr bool SDL::AudioFormat::IsUnsigned () const
 Determine if an AudioFormat represents unsigned data.
const char * SDL::AudioDeviceBase::GetName () const
 Get the human-readable name of a specific audio device.
AudioSpec SDL::AudioDeviceBase::GetFormat (int *sample_frames=nullptr) const
 Get the current audio format of a specific audio device.
OwnArray< int > SDL::AudioDeviceBase::GetChannelMap () const
 Get the current channel map of an audio device.
 SDL::AudioDevice::AudioDevice (AudioDeviceRef devid, OptionalRef< const AudioSpec > spec)
 Open a specific audio device.
bool SDL::AudioDeviceBase::IsPhysical () const
 Determine if an audio device is physical (instead of logical).
bool SDL::AudioDeviceBase::IsPlayback () const
 Determine if an audio device is a playback device (instead of recording).
void SDL::AudioDeviceBase::Pause ()
 Use this function to pause audio playback on a specified device.
void SDL::AudioDeviceBase::Resume ()
 Use this function to unpause audio playback on a specified device.
bool SDL::AudioDeviceBase::Paused () const
 Use this function to query if an audio device is paused.
float SDL::AudioDeviceBase::GetGain () const
 Get the gain of an audio device.
void SDL::AudioDeviceBase::SetGain (float gain)
 Change the gain of an audio device.
void SDL::AudioDeviceBase::Close ()
 Close a previously-opened audio device.
void SDL::AudioDeviceBase::BindAudioStreams (std::span< AudioStreamRef > streams)
 Bind a list of audio streams to an audio device.
void SDL::AudioDeviceBase::BindAudioStream (AudioStreamRef stream)
 Bind a single audio stream to an audio device.
void SDL::AudioStreamBase::Unbind ()
 Unbind a single audio stream from its audio device.
AudioDeviceRef SDL::AudioStreamBase::GetDevice () const
 Query an audio stream for its currently-bound device.
 SDL::AudioStream::AudioStream (OptionalRef< const AudioSpec > src_spec, OptionalRef< const AudioSpec > dst_spec)
 Create a new audio stream.
 SDL::AudioStream::AudioStream (AudioDeviceRef devid, OptionalRef< const AudioSpec > spec=std::nullopt, AudioStreamCallback callback=nullptr, void *userdata=nullptr)
 Convenience function for straightforward audio init for the common case.
 SDL::AudioStream::AudioStream (AudioDeviceRef devid, OptionalRef< const AudioSpec > spec, AudioStreamCB callback)
 Convenience function for straightforward audio init for the common case.
PropertiesRef SDL::AudioStreamBase::GetProperties () const
 Get the properties associated with an audio stream.
void SDL::AudioStreamBase::GetFormat (AudioSpec *src_spec, AudioSpec *dst_spec) const
 Query the current format of an audio stream.
void SDL::AudioStreamBase::SetFormat (OptionalRef< const AudioSpec > src_spec, OptionalRef< const AudioSpec > dst_spec)
 Change the input and output formats of an audio stream.
float SDL::AudioStreamBase::GetFrequencyRatio () const
 Get the frequency ratio of an audio stream.
void SDL::AudioStreamBase::SetFrequencyRatio (float ratio)
 Change the frequency ratio of an audio stream.
float SDL::AudioStreamBase::GetGain () const
 Get the gain of an audio stream.
void SDL::AudioStreamBase::SetGain (float gain)
 Change the gain of an audio stream.
OwnArray< int > SDL::AudioStreamBase::GetInputChannelMap () const
 Get the current input channel map of an audio stream.
OwnArray< int > SDL::AudioStreamBase::GetOutputChannelMap () const
 Get the current output channel map of an audio stream.
void SDL::AudioStreamBase::SetInputChannelMap (std::span< int > chmap)
 Set the current input channel map of an audio stream.
void SDL::AudioStreamBase::SetOutputChannelMap (std::span< int > chmap)
 Set the current output channel map of an audio stream.
void SDL::AudioStreamBase::PutData (SourceBytes buf)
 Add data to the stream.
void SDL::AudioStreamBase::PutDataNoCopy (SourceBytes buf, AudioStreamDataCompleteCallback callback, void *userdata)
 Add external data to an audio stream without copying it.
void SDL::AudioStreamBase::PutDataNoCopy (SourceBytes buf, AudioStreamDataCompleteCB callback)
 Add external data to an audio stream without copying it.
void SDL::AudioStreamBase::PutPlanarData (const void *const *channel_buffers, int num_channels, int num_samples)
 Add data to the stream with each channel in a separate array.
int SDL::AudioStreamBase::GetData (TargetBytes buf)
 Get converted/resampled data from the stream.
int SDL::AudioStreamBase::GetAvailable () const
 Get the number of converted/resampled bytes available.
int SDL::AudioStreamBase::GetQueued () const
 Get the number of bytes currently queued.
void SDL::AudioStreamBase::Flush ()
 Tell the stream that you're done sending data, and anything being buffered should be converted/resampled and made available immediately.
void SDL::AudioStreamBase::Clear ()
 Clear any pending data in the stream.
void SDL::AudioStreamBase::PauseDevice ()
 Use this function to pause audio playback on the audio device associated with an audio stream.
void SDL::AudioStreamBase::ResumeDevice ()
 Use this function to unpause audio playback on the audio device associated with an audio stream.
bool SDL::AudioStreamBase::DevicePaused () const
 Use this function to query if an audio device associated with a stream is paused.
AudioStreamLock SDL::AudioStreamBase::Lock ()
 Lock an audio stream for serialized access.
 SDL::AudioStreamLock::AudioStreamLock (AudioStreamRef resource)
 Lock an audio stream for serialized access.
void SDL::AudioStreamBase::Unlock (AudioStreamLock &&lock)
 Unlock an audio stream for serialized access.
void SDL::AudioStreamLock::reset ()
 Unlock an audio stream for serialized access.
void SDL::AudioStreamBase::SetGetCallback (AudioStreamCallback callback, void *userdata)
 Set a callback that runs when data is requested from an audio stream.
void SDL::AudioStreamBase::SetGetCallback (AudioStreamCB callback)
 Set a callback that runs when data is requested from an audio stream.
void SDL::AudioStreamBase::SetPutCallback (AudioStreamCallback callback, void *userdata)
 Set a callback that runs when data is added to an audio stream.
void SDL::AudioStreamBase::SetPutCallback (AudioStreamCB callback)
 Set a callback that runs when data is added to an audio stream.
void SDL::AudioStreamBase::Destroy ()
 Free an audio stream.
AudioStream SDL::AudioDeviceBase::OpenStream (OptionalRef< const AudioSpec > spec, AudioStreamCallback callback, void *userdata)
 Convenience function for straightforward audio init for the common case.
AudioStream SDL::AudioDeviceBase::OpenStream (OptionalRef< const AudioSpec > spec, AudioStreamCB callback)
 Convenience function for straightforward audio init for the common case.
void SDL::AudioDeviceBase::SetPostmixCallback (AudioPostmixCallback callback, void *userdata)
 Set a callback that fires when data is about to be fed to an audio device.
void SDL::AudioDeviceBase::SetPostmixCallback (AudioPostmixCB callback)
 Set a callback that fires when data is about to be fed to an audio device.
const char * SDL::AudioFormat::GetName () const
 Get the human readable name of an audio format.
int SDL::AudioFormat::GetSilenceValue () const
 Get the appropriate memset value for silencing an audio format.

Variables

constexpr Uint32 SDL::AUDIO_MASK_BITSIZE = SDL_AUDIO_MASK_BITSIZE
 Mask of bits in an AudioFormat that contains the format bit size.
constexpr Uint32 SDL::AUDIO_MASK_FLOAT = SDL_AUDIO_MASK_FLOAT
 Mask of bits in an AudioFormat that contain the floating point flag.
constexpr Uint32 SDL::AUDIO_MASK_BIG_ENDIAN = SDL_AUDIO_MASK_BIG_ENDIAN
 Mask of bits in an AudioFormat that contain the bigendian flag.
constexpr Uint32 SDL::AUDIO_MASK_SIGNED = SDL_AUDIO_MASK_SIGNED
 Mask of bits in an AudioFormat that contain the signed data flag.
constexpr AudioFormat SDL::AUDIO_UNKNOWN
 Unspecified audio format.
constexpr AudioFormat SDL::AUDIO_U8 = SDL_AUDIO_U8
 Unsigned 8-bit samples.
constexpr AudioFormat SDL::AUDIO_S8 = SDL_AUDIO_S8
 Signed 8-bit samples.
constexpr AudioFormat SDL::AUDIO_S16LE = SDL_AUDIO_S16LE
 Signed 16-bit samples.
constexpr AudioFormat SDL::AUDIO_S16BE
 As above, but big-endian byte order.
constexpr AudioFormat SDL::AUDIO_S32LE = SDL_AUDIO_S32LE
 32-bit integer samples
constexpr AudioFormat SDL::AUDIO_S32BE
 As above, but big-endian byte order.
constexpr AudioFormat SDL::AUDIO_F32LE
 32-bit floating point samples
constexpr AudioFormat SDL::AUDIO_F32BE
 As above, but big-endian byte order.
constexpr AudioFormat SDL::AUDIO_S16 = SDL_AUDIO_S16
 AUDIO_S16.
constexpr AudioFormat SDL::AUDIO_S32 = SDL_AUDIO_S32
 AUDIO_S32.
constexpr AudioFormat SDL::AUDIO_F32 = SDL_AUDIO_F32
 AUDIO_F32.
constexpr AudioDeviceID SDL::AUDIO_DEVICE_DEFAULT_PLAYBACK
 A value used to request a default playback audio device.
constexpr AudioDeviceID SDL::AUDIO_DEVICE_DEFAULT_RECORDING
 A value used to request a default recording audio device.

Detailed Description

Audio functionality for the SDL library.

All audio in SDL3 revolves around 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 LoadWAV (and 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 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 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.

AudioStream can also be provided channel maps to change this ordering to whatever is necessary, in other audio processing scenarios.

Typedef Documentation

◆ AudioDeviceRef

Reference for AudioDevice.

This does not take ownership!

◆ AudioPostmixCallback

Initial value:
void(SDLCALL*)(void* userdata,
const AudioSpec* spec,
float* buffer,
int buflen)
SDL_AudioSpec AudioSpec
Format specifier for audio data.
Definition SDL3pp_audio.h:201

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 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.

Parameters
userdataa pointer provided by the app through SetAudioPostmixCallback, for its own use.
specthe current format of audio that is to be submitted to the audio device.
bufferthe buffer of audio samples to be submitted. The callback can inspect and/or modify this data.
buflenthe size of buffer in bytes.
Thread safety:
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.
See also
SetAudioPostmixCallback

◆ AudioPostmixCB

Initial value:
MakeFrontCallback<void(const AudioSpec* spec, float* buffer, int buflen)>
Definition SDL3pp_callbackWrapper.h:169

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 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.

Parameters
specthe current format of audio that is to be submitted to the audio device.
bufferthe buffer of audio samples to be submitted. The callback can inspect and/or modify this data.
Thread safety:
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.
See also
SetAudioPostmixCallback
AudioPostmixCallback

◆ AudioSpec

using SDL::AudioSpec = SDL_AudioSpec

Format specifier for audio data.

Since
This struct is available since SDL 3.2.0.
See also
AudioFormat

◆ AudioStreamCallback

Initial value:
void(SDLCALL*)(void* userdata,
int additional_amount,
int total_amount)
SDL_AudioStream * AudioStreamRaw
Alias to raw representation for AudioStream.
Definition SDL3pp_audio.h:145

A callback that fires when data passes through an AudioStream.

Apps can (optionally) register a callback with an audio stream that is called when data is added with PutAudioStreamData, or requested with 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.

Parameters
streamthe SDL audio stream associated with this callback.
additional_amountthe amount of data, in bytes, that is needed right now.
total_amountthe total amount of data requested, in bytes, that is requested or available.
userdataan opaque pointer provided by the app for their personal use.
Thread safety:
This callbacks may run from any thread, so if you need to protect shared data, you should use 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.
See also
SetAudioStreamGetCallback
SetAudioStreamPutCallback

◆ AudioStreamCB

Initial value:
void(AudioStreamRaw stream, int additional_amount, int total_amount)>

A callback that fires when data passes through an AudioStream.

Apps can (optionally) register a callback with an audio stream that is called when data is added with PutAudioStreamData, or requested with 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.

Parameters
streamthe SDL audio stream associated with this callback.
additional_amountthe amount of data, in bytes, that is needed right now.
total_amountthe total amount of data requested, in bytes, that is requested or available.
Thread safety:
This callbacks may run from any thread, so if you need to protect shared data, you should use 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.
See also
SetAudioStreamGetCallback
SetAudioStreamPutCallback
AudioStreamCallback

◆ AudioStreamDataCompleteCallback

Initial value:
void(SDLCALL*)(void* userdata,
const void* buf,
int buflen)

A callback that fires for completed PutAudioStreamDataNoCopy() data.

When using PutAudioStreamDataNoCopy() to provide data to an 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.

Parameters
userdataan opaque pointer provided by the app for their personal use.
bufthe pointer provided to PutAudioStreamDataNoCopy().
buflenthe size of buffer, in bytes, provided to PutAudioStreamDataNoCopy().
Thread safety:
This callbacks may run from any thread, so if you need to protect shared data, you should use 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.
See also
SetAudioStreamGetCallback
SetAudioStreamPutCallback

◆ AudioStreamDataCompleteCB

Initial value:
std::function<void(const void* buf, int buflen)>

A callback that fires for completed PutAudioStreamDataNoCopy() data.

When using PutAudioStreamDataNoCopy() to provide data to an 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.

Parameters
bufthe pointer provided to PutAudioStreamDataNoCopy().
buflenthe size of buffer, in bytes, provided to PutAudioStreamDataNoCopy().
Thread safety:
This callbacks may run from any thread, so if you need to protect shared data, you should use 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.
See also
SetAudioStreamGetCallback
SetAudioStreamPutCallback
AudioStreamDataCompleteCallback

◆ AudioStreamRef

Reference for AudioStream.

This does not take ownership!

Function Documentation

◆ AudioBitSize()

Uint16 SDL::AudioBitSize ( AudioFormatRaw x)
constexpr

Retrieve the size, in bits, from an AudioFormat.

For example, AudioBitSize(AUDIO_S16) returns 16.

Parameters
xan AudioFormat value.
Returns
data size in bits.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ AudioByteSize()

Uint16 SDL::AudioByteSize ( AudioFormatRaw x)
constexpr

Retrieve the size, in bytes, from an AudioFormat.

For example, AudioByteSize(AUDIO_S16) returns 2.

Parameters
xan AudioFormat value.
Returns
data size in bytes.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ AudioDevice()

SDL::AudioDevice::AudioDevice ( AudioDeviceRef devid,
OptionalRef< const AudioSpec > spec )
inline

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 OpenAudioDeviceStream instead of this function).

If you don't care about opening a specific device, pass a devid of either AUDIO_DEVICE_DEFAULT_PLAYBACK or 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. 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 AudioDevice 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 AudioDevice 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 CloseAudioDevice() on the returned device id.

Parameters
devidthe device instance id to open, or AUDIO_DEVICE_DEFAULT_PLAYBACK or AUDIO_DEVICE_DEFAULT_RECORDING for the most reasonable default device.
specthe requested device configuration. Can be nullptr to use reasonable defaults.
Postcondition
the device ID on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
CloseAudioDevice
GetAudioDeviceFormat

◆ AudioDevicePaused()

bool SDL::AudioDevicePaused ( AudioDeviceRef devid)
inline

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 OpenAudioDevice() can be. Physical and invalid device IDs will report themselves as unpaused here.

Parameters
devida device opened by OpenAudioDevice().
Returns
true if device is valid and paused, false otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PauseAudioDevice
ResumeAudioDevice

◆ AudioFormat()

SDL::AudioFormat::AudioFormat ( bool sign,
bool bigendian,
bool flt,
Uint16 size )
constexpr

Define an 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 AudioFormat mean.

For example, AUDIO_S32LE looks like this:

AudioFormat format(1, 0, 0, 32)
constexpr AudioFormat(AudioFormatRaw audioFormat={}) noexcept
Wraps AudioFormat.
Definition SDL3pp_audio.h:227
Parameters
sign1 for signed data, 0 for unsigned data.
bigendian1 for bigendian data, 0 for littleendian data.
flt1 for floating point data, 0 for integer data.
sizenumber of bits per sample.
Postcondition
a format value in the style of AudioFormat.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ AudioFrameSize()

int SDL::AudioFrameSize ( const AudioSpec & x)
constexpr

Calculate the size of each audio frame (in bytes) from an 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.

Parameters
xan AudioSpec to query.
Returns
the number of bytes used per sample frame.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ AudioStream() [1/3]

SDL::AudioStream::AudioStream ( AudioDeviceRef devid,
OptionalRef< const AudioSpec > spec,
AudioStreamCB callback )
inline

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 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 AudioStream.ResumeDevice();

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 nullptr, the system will choose the format, and the app can use AudioStream.GetFormat() to obtain this information later.

If you don't care about opening a specific audio device, you can (and probably should), use AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and AUDIO_DEVICE_DEFAULT_RECORDING for recording.

One can optionally provide a callback function; if nullptr, 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 AudioStream.Destroy will also close the audio device associated with this stream.

Parameters
devidan audio device to open, or AUDIO_DEVICE_DEFAULT_PLAYBACK or AUDIO_DEVICE_DEFAULT_RECORDING.
specthe audio stream's data format. Can be nullptr.
callbacka callback where the app will provide new data for playback, or receive new data for recording.
Postcondition
an audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamDevice
ResumeAudioStreamDevice

◆ AudioStream() [2/3]

SDL::AudioStream::AudioStream ( AudioDeviceRef devid,
OptionalRef< const AudioSpec > spec = std::nullopt,
AudioStreamCallback callback = nullptr,
void * userdata = nullptr )
inline

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 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 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 nullptr, the system will choose the format, and the app can use GetAudioStreamFormat() to obtain this information later.

If you don't care about opening a specific audio device, you can (and probably should), use AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and AUDIO_DEVICE_DEFAULT_RECORDING for recording.

One can optionally provide a callback function; if nullptr, 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 DestroyAudioStream will also close the audio device associated with this stream.

Parameters
devidan audio device to open, or AUDIO_DEVICE_DEFAULT_PLAYBACK or AUDIO_DEVICE_DEFAULT_RECORDING.
specthe audio stream's data format. Can be nullptr.
callbacka callback where the app will provide new data for playback, or receive new data for recording. Can be nullptr, in which case the app will need to call PutAudioStreamData or GetAudioStreamData as necessary.
userdataapp-controlled pointer passed to callback. Can be nullptr. Ignored if callback is nullptr.
Postcondition
an audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamDevice
ResumeAudioStreamDevice

◆ AudioStream() [3/3]

SDL::AudioStream::AudioStream ( OptionalRef< const AudioSpec > src_spec,
OptionalRef< const AudioSpec > dst_spec )
inline

Create a new audio stream.

Parameters
src_specthe format details of the input audio.
dst_specthe format details of the output audio.
Postcondition
a new audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PutAudioStreamData
GetAudioStreamData
GetAudioStreamAvailable
FlushAudioStream
ClearAudioStream
SetAudioStreamFormat
DestroyAudioStream

◆ AudioStreamDevicePaused()

bool SDL::AudioStreamDevicePaused ( AudioStreamRef stream)
inline

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.

Parameters
streamthe audio stream associated with the audio device to query.
Returns
true if device is valid and paused, false otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PauseAudioStreamDevice
ResumeAudioStreamDevice

◆ AudioStreamLock()

SDL::AudioStreamLock::AudioStreamLock ( AudioStreamRef resource)
inline

Lock an audio stream for serialized access.

Each 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 LockMutex for an internal lock; it has all the same attributes (recursive locks are allowed, etc).

Parameters
resourcethe audio stream to lock.
Postcondition
true on success or false on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
UnlockAudioStream

◆ BindAudioStream() [1/2]

void SDL::AudioDeviceBase::BindAudioStream ( AudioStreamRef stream)
inline

Bind a single audio stream to an audio device.

This is a convenience function, equivalent to calling BindAudioStreams(devid, &stream, 1).

Parameters
streaman audio stream to bind to a device.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStreams
UnbindAudioStream
GetAudioStreamDevice

◆ BindAudioStream() [2/2]

void SDL::BindAudioStream ( AudioDeviceRef devid,
AudioStreamRef stream )
inline

Bind a single audio stream to an audio device.

This is a convenience function, equivalent to calling BindAudioStreams(devid, &stream, 1).

Parameters
devidan audio device to bind a stream to.
streaman audio stream to bind to a device.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStreams
UnbindAudioStream
GetAudioStreamDevice

◆ BindAudioStreams() [1/2]

void SDL::AudioDeviceBase::BindAudioStreams ( std::span< AudioStreamRef > streams)
inline

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 SetAudioStreamFormat(). If the other end of the stream's format has never been set (the audio stream was created with a nullptr audio spec), this function will set it to match the device end's format.

Parameters
streamsan array of audio streams to bind.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStreams
UnbindAudioStream
GetAudioStreamDevice

◆ BindAudioStreams() [2/2]

void SDL::BindAudioStreams ( AudioDeviceRef devid,
std::span< AudioStreamRef > streams )
inline

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 SetAudioStreamFormat(). If the other end of the stream's format has never been set (the audio stream was created with a nullptr audio spec), this function will set it to match the device end's format.

Parameters
devidan audio device to bind a stream to.
streamsan array of audio streams to bind.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStreams
UnbindAudioStream
GetAudioStreamDevice

◆ Clear()

void SDL::AudioStreamBase::Clear ( )
inline

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.

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamAvailable
GetAudioStreamData
GetAudioStreamQueued
PutAudioStreamData

◆ ClearAudioStream()

void SDL::ClearAudioStream ( AudioStreamRef stream)
inline

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.

Parameters
streamthe audio stream to clear.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamAvailable
GetAudioStreamData
GetAudioStreamQueued
PutAudioStreamData

◆ Close()

void SDL::AudioDeviceBase::Close ( )
inline

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.

Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
OpenAudioDevice

◆ CloseAudioDevice()

void SDL::CloseAudioDevice ( AudioDeviceID devid)
inline

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.

Parameters
devidan audio device id previously returned by OpenAudioDevice().
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
OpenAudioDevice

◆ ConvertAudioSamples()

OwnArray< Uint8 > SDL::ConvertAudioSamples ( const AudioSpec & src_spec,
SourceBytes src_data,
const AudioSpec & dst_spec )
inline

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 AudioStream, which is designed for this situation.

Internally, this function creates and destroys an AudioStream on each use, so it's also less efficient than using one directly, if you need to convert multiple times.

Parameters
src_specthe format details of the input audio.
src_datathe audio data to be converted.
dst_specthe format details of the output audio.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ CreateAudioStream()

AudioStream SDL::CreateAudioStream ( OptionalRef< const AudioSpec > src_spec,
OptionalRef< const AudioSpec > dst_spec )
inline

Create a new audio stream.

Parameters
src_specthe format details of the input audio.
dst_specthe format details of the output audio.
Returns
a new audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PutAudioStreamData
GetAudioStreamData
GetAudioStreamAvailable
FlushAudioStream
ClearAudioStream
SetAudioStreamFormat
DestroyAudioStream

◆ DefineAudioFormat()

AudioFormat SDL::DefineAudioFormat ( bool sign,
bool bigendian,
bool flt,
Uint16 size )
constexpr

Define an 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 AudioFormat mean.

For example, AUDIO_S32LE looks like this:

DefineAudioFormat(1, 0, 0, 32)
constexpr AudioFormat DefineAudioFormat(bool sign, bool bigendian, bool flt, Uint16 size)
Define an AudioFormat value.
Definition SDL3pp_audio.h:454
Parameters
sign1 for signed data, 0 for unsigned data.
bigendian1 for bigendian data, 0 for littleendian data.
flt1 for floating point data, 0 for integer data.
sizenumber of bits per sample.
Returns
a format value in the style of AudioFormat.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ Destroy()

void SDL::AudioStreamBase::Destroy ( )
inline

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 OpenAudioDeviceStream, the audio device that was opened alongside this stream's creation will be closed, too.

Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
CreateAudioStream

◆ DestroyAudioStream()

void SDL::DestroyAudioStream ( AudioStreamRaw stream)
inline

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 OpenAudioDeviceStream, the audio device that was opened alongside this stream's creation will be closed, too.

Parameters
streamthe audio stream to destroy.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
CreateAudioStream

◆ DevicePaused()

bool SDL::AudioStreamBase::DevicePaused ( ) const
inline

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.

Returns
true if device is valid and paused, false otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PauseAudioStreamDevice
ResumeAudioStreamDevice

◆ Flush()

void SDL::AudioStreamBase::Flush ( )
inline

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.

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PutAudioStreamData

◆ FlushAudioStream()

void SDL::FlushAudioStream ( AudioStreamRef stream)
inline

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.

Parameters
streamthe audio stream to flush.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PutAudioStreamData

◆ GetAudioDeviceChannelMap()

OwnArray< int > SDL::GetAudioDeviceChannelMap ( AudioDeviceRef devid)
inline

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 expectsorder that SDL expects.

Audio devices usually have no remapping applied. This is represented by returning nullptr, and does not signify an error.

Parameters
devidthe instance ID of the device to query.
Returns
an array of the current channel mapping, with as many elements as the current output spec's channels, or nullptr if default
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamInputChannelMap

◆ GetAudioDeviceFormat()

AudioSpec SDL::GetAudioDeviceFormat ( AudioDeviceRef devid,
int * sample_frames = nullptr )
inline

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 AUDIO_DEVICE_DEFAULT_PLAYBACK or 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.

Parameters
devidthe instance ID of the device to query.
sample_framespointer to store device buffer size, in sample frames. Can be nullptr.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetAudioDeviceGain()

float SDL::GetAudioDeviceGain ( AudioDeviceRef devid)
inline

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.

Parameters
devidthe audio device to query.
Returns
the gain of the device or -1.0f on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioDeviceGain

◆ GetAudioDeviceName()

const char * SDL::GetAudioDeviceName ( AudioDeviceRef devid)
inline

Get the human-readable name of a specific audio device.

WARNING: this function will work with AUDIO_DEVICE_DEFAULT_PLAYBACK and 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.

Parameters
devidthe instance ID of the device to query.
Returns
the name of the audio device on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioPlaybackDevices
GetAudioRecordingDevices

◆ GetAudioDriver()

const char * SDL::GetAudioDriver ( int index)
inline

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.

Parameters
indexthe index of the audio driver; the value ranges from 0 to GetNumAudioDrivers() - 1.
Returns
the name of the audio driver at the requested index, or nullptr if an invalid index was specified.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetNumAudioDrivers

◆ GetAudioFormatName()

const char * SDL::GetAudioFormatName ( AudioFormatRaw format)
inline

Get the human readable name of an audio format.

Parameters
formatthe audio format to query.
Returns
the human readable name of the specified audio format or "AUDIO_UNKNOWN" if the format isn't recognized.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetAudioPlaybackDevices()

OwnArray< AudioDeviceRef > SDL::GetAudioPlaybackDevices ( )
inline

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 GetAudioRecordingDevices() instead.

This only returns a list of physical devices; it will not have any device IDs returned by OpenAudioDevice().

If this function returns nullptr, to signify an error, *count will be set to zero.

Returns
a 0 terminated array of device instance IDs on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
OpenAudioDevice
GetAudioRecordingDevices

◆ GetAudioRecordingDevices()

OwnArray< AudioDeviceRef > SDL::GetAudioRecordingDevices ( )
inline

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 GetAudioPlaybackDevices() instead.

This only returns a list of physical devices; it will not have any device IDs returned by OpenAudioDevice().

If this function returns nullptr, to signify an error, *count will be set to zero.

Returns
a 0 terminated array of device instance IDs on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
OpenAudioDevice
GetAudioPlaybackDevices

◆ GetAudioStreamAvailable()

int SDL::GetAudioStreamAvailable ( AudioStreamRef stream)
inline

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 GetAudioStreamData before this function's return value is no longer clamped.

Parameters
streamthe audio stream to query.
Returns
the number of converted/resampled bytes available or -1 on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamData
PutAudioStreamData

◆ GetAudioStreamData()

int SDL::GetAudioStreamData ( AudioStreamRef stream,
TargetBytes buf )
inline

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 SetAudioStreamFormat.

Note that any conversion and resampling necessary is done during this call, and 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.

Parameters
streamthe stream the audio is being requested from.
bufa buffer to fill with audio data.
Returns
the number of bytes read from the stream or -1 on failure; call GetError() for more information.
Thread safety:
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.
See also
ClearAudioStream
GetAudioStreamAvailable
PutAudioStreamData

◆ GetAudioStreamDevice()

AudioDeviceRef SDL::GetAudioStreamDevice ( AudioStreamRef stream)
inline

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.

Parameters
streamthe audio stream to query.
Returns
the bound audio device, or 0 if not bound or invalid.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStream
BindAudioStreams

◆ GetAudioStreamFormat()

void SDL::GetAudioStreamFormat ( AudioStreamRef stream,
AudioSpec * src_spec,
AudioSpec * dst_spec )
inline

Query the current format of an audio stream.

Parameters
streamthe AudioStream to query.
src_specwhere to store the input audio format; ignored if nullptr.
dst_specwhere to store the output audio format; ignored if nullptr.
Exceptions
Erroron failure.
Thread safety:
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.
See also
SetAudioStreamFormat

◆ GetAudioStreamFrequencyRatio()

float SDL::GetAudioStreamFrequencyRatio ( AudioStreamRef stream)
inline

Get the frequency ratio of an audio stream.

Parameters
streamthe AudioStream to query.
Returns
the frequency ratio of the stream or 0.0 on failure; call GetError() for more information.
Thread safety:
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.
See also
SetAudioStreamFrequencyRatio

◆ GetAudioStreamGain()

float SDL::GetAudioStreamGain ( AudioStreamRef stream)
inline

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).

Parameters
streamthe AudioStream to query.
Returns
the gain of the stream or -1.0f on failure; call GetError() for more information.
Thread safety:
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.
See also
SetAudioStreamGain

◆ GetAudioStreamInputChannelMap()

OwnArray< int > SDL::GetAudioStreamInputChannelMap ( AudioStreamRef stream)
inline

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 expectsorder that SDL expects.

Audio streams default to no remapping applied. This is represented by returning nullptr, and does not signify an error.

Parameters
streamthe AudioStream to query.
Returns
an array of the current channel mapping, with as many elements as the current output spec's channels, or nullptr if default.
Thread safety:
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.
See also
SetAudioStreamInputChannelMap

◆ GetAudioStreamOutputChannelMap()

OwnArray< int > SDL::GetAudioStreamOutputChannelMap ( AudioStreamRef stream)
inline

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 expectsorder that SDL expects.

Audio streams default to no remapping applied. This is represented by returning nullptr, and does not signify an error.

Parameters
streamthe AudioStream to query.
Returns
an array of the current channel mapping, with as many elements as the current output spec's channels, or nullptr if default.
Thread safety:
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.
See also
SetAudioStreamInputChannelMap

◆ GetAudioStreamProperties()

PropertiesRef SDL::GetAudioStreamProperties ( AudioStreamRef stream)
inline

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:

  • 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 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.
Parameters
streamthe AudioStream to query.
Returns
a valid property ID on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetAudioStreamQueued()

int SDL::GetAudioStreamQueued ( AudioStreamRef stream)
inline

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 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 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 GetAudioStreamData before this function's return value is no longer clamped.

Parameters
streamthe audio stream to query.
Returns
the number of bytes queued or -1 on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PutAudioStreamData
ClearAudioStream

◆ GetAvailable()

int SDL::AudioStreamBase::GetAvailable ( ) const
inline

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 GetAudioStreamData before this function's return value is no longer clamped.

Returns
the number of converted/resampled bytes available or -1 on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamData
PutAudioStreamData

◆ GetBitSize()

Uint16 SDL::AudioFormat::GetBitSize ( ) const
constexpr

Retrieve the size, in bits, from an AudioFormat.

For example, AudioBitSize(AUDIO_S16) returns 16.

Returns
data size in bits.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetByteSize()

Uint16 SDL::AudioFormat::GetByteSize ( ) const
constexpr

Retrieve the size, in bytes, from an AudioFormat.

For example, AudioByteSize(AUDIO_S16) returns 2.

Returns
data size in bytes.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetChannelMap()

OwnArray< int > SDL::AudioDeviceBase::GetChannelMap ( ) const
inline

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 expectsorder that SDL expects.

Audio devices usually have no remapping applied. This is represented by returning nullptr, and does not signify an error.

Returns
an array of the current channel mapping, with as many elements as the current output spec's channels, or nullptr if default
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamInputChannelMap

◆ GetCurrentAudioDriver()

const char * SDL::GetCurrentAudioDriver ( )
inline

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 nullptr if no driver has been initialized.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetData()

int SDL::AudioStreamBase::GetData ( TargetBytes buf)
inline

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 SetAudioStreamFormat.

Note that any conversion and resampling necessary is done during this call, and 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.

Parameters
bufa buffer to fill with audio data.
Returns
the number of bytes read from the stream or -1 on failure; call GetError() for more information.
Thread safety:
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.
See also
ClearAudioStream
GetAudioStreamAvailable
PutAudioStreamData

◆ GetDevice()

AudioDeviceRef SDL::AudioStreamBase::GetDevice ( ) const
inline

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.

Returns
the bound audio device, or 0 if not bound or invalid.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStream
BindAudioStreams

◆ GetFormat() [1/2]

AudioSpec SDL::AudioDeviceBase::GetFormat ( int * sample_frames = nullptr) const
inline

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 AUDIO_DEVICE_DEFAULT_PLAYBACK or 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.

Parameters
sample_framespointer to store device buffer size, in sample frames. Can be nullptr.
Returns
the device format details on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetFormat() [2/2]

void SDL::AudioStreamBase::GetFormat ( AudioSpec * src_spec,
AudioSpec * dst_spec ) const
inline

Query the current format of an audio stream.

Parameters
src_specwhere to store the input audio format; ignored if nullptr.
dst_specwhere to store the output audio format; ignored if nullptr.
Exceptions
Erroron failure.
Thread safety:
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.
See also
SetAudioStreamFormat

◆ GetFrequencyRatio()

float SDL::AudioStreamBase::GetFrequencyRatio ( ) const
inline

Get the frequency ratio of an audio stream.

Returns
the frequency ratio of the stream or 0.0 on failure; call GetError() for more information.
Thread safety:
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.
See also
SetAudioStreamFrequencyRatio

◆ GetGain() [1/2]

float SDL::AudioDeviceBase::GetGain ( ) const
inline

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.

Returns
the gain of the device or -1.0f on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioDeviceGain

◆ GetGain() [2/2]

float SDL::AudioStreamBase::GetGain ( ) const
inline

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).

Returns
the gain of the stream or -1.0f on failure; call GetError() for more information.
Thread safety:
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.
See also
SetAudioStreamGain

◆ GetInputChannelMap()

OwnArray< int > SDL::AudioStreamBase::GetInputChannelMap ( ) const
inline

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 expectsorder that SDL expects.

Audio streams default to no remapping applied. This is represented by returning nullptr, and does not signify an error.

Returns
an array of the current channel mapping, with as many elements as the current output spec's channels, or nullptr if default.
Thread safety:
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.
See also
SetAudioStreamInputChannelMap

◆ GetName() [1/2]

const char * SDL::AudioDeviceBase::GetName ( ) const
inline

Get the human-readable name of a specific audio device.

WARNING: this function will work with AUDIO_DEVICE_DEFAULT_PLAYBACK and 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.

Returns
the name of the audio device on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioPlaybackDevices
GetAudioRecordingDevices

◆ GetName() [2/2]

const char * SDL::AudioFormat::GetName ( ) const
inline

Get the human readable name of an audio format.

Returns
the human readable name of the specified audio format or "AUDIO_UNKNOWN" if the format isn't recognized.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetNumAudioDrivers()

int SDL::GetNumAudioDrivers ( )
inline

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.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioDriver

◆ GetOutputChannelMap()

OwnArray< int > SDL::AudioStreamBase::GetOutputChannelMap ( ) const
inline

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 expectsorder that SDL expects.

Audio streams default to no remapping applied. This is represented by returning nullptr, and does not signify an error.

Returns
an array of the current channel mapping, with as many elements as the current output spec's channels, or nullptr if default.
Thread safety:
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.
See also
SetAudioStreamInputChannelMap

◆ GetProperties()

PropertiesRef SDL::AudioStreamBase::GetProperties ( ) const
inline

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:

  • 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 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.
Returns
a valid property ID on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetQueued()

int SDL::AudioStreamBase::GetQueued ( ) const
inline

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 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 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 GetAudioStreamData before this function's return value is no longer clamped.

Returns
the number of bytes queued or -1 on failure; call GetError() for more information.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PutAudioStreamData
ClearAudioStream

◆ GetSilenceValue()

int SDL::AudioFormat::GetSilenceValue ( ) const
inline

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 memset) to set an audio buffer in a specific format to silence.

Returns
a byte value that can be passed to memset.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ GetSilenceValueForFormat()

int SDL::GetSilenceValueForFormat ( AudioFormatRaw format)
inline

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 memset) to set an audio buffer in a specific format to silence.

Parameters
formatthe audio data format to query.
Returns
a byte value that can be passed to memset.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioBigENDIAN()

bool SDL::IsAudioBigENDIAN ( AudioFormatRaw x)
constexpr

Determine if an AudioFormat represents bigendian data.

For example, IsAudioBigENDIAN(AUDIO_S16LE) returns 0.

Parameters
xan AudioFormat value.
Returns
non-zero if format is bigendian, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioDevicePhysical()

bool SDL::IsAudioDevicePhysical ( AudioDeviceRef devid)
inline

Determine if an audio device is physical (instead of logical).

An AudioDevice 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 OpenAudioDevice or 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.

Parameters
devidthe device ID to query.
Returns
true if devid is a physical device, false if it is logical.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioDevicePlayback()

bool SDL::IsAudioDevicePlayback ( AudioDeviceRef devid)
inline

Determine if an audio device is a playback device (instead of recording).

This function may return either true or false for invalid device IDs.

Parameters
devidthe device ID to query.
Returns
true if devid is a playback device, false if it is recording.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioFloat()

bool SDL::IsAudioFloat ( AudioFormatRaw x)
constexpr

Determine if an AudioFormat represents floating point data.

For example, IsAudioFloat(AUDIO_S16) returns 0.

Parameters
xan AudioFormat value.
Returns
non-zero if format is floating point, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioInt()

bool SDL::IsAudioInt ( AudioFormatRaw x)
constexpr

Determine if an AudioFormat represents integer data.

For example, IsAudioInt(AUDIO_F32) returns 0.

Parameters
xan AudioFormat value.
Returns
non-zero if format is integer, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioLittleEndian()

bool SDL::IsAudioLittleEndian ( AudioFormatRaw x)
constexpr

Determine if an AudioFormat represents littleendian data.

For example, IsAudioLittleEndian(AUDIO_S16BE) returns 0.

Parameters
xan AudioFormat value.
Returns
non-zero if format is littleendian, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioSigned()

bool SDL::IsAudioSigned ( AudioFormatRaw x)
constexpr

Determine if an AudioFormat represents signed data.

For example, IsAudioSigned(AUDIO_U8) returns 0.

Parameters
xan AudioFormat value.
Returns
non-zero if format is signed, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsAudioUnsigned()

bool SDL::IsAudioUnsigned ( AudioFormatRaw x)
constexpr

Determine if an AudioFormat represents unsigned data.

For example, IsAudioUnsigned(AUDIO_S16) returns 0.

Parameters
xan AudioFormat value.
Returns
non-zero if format is unsigned, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsBigEndian()

bool SDL::AudioFormat::IsBigEndian ( ) const
constexpr

Determine if an AudioFormat represents bigendian data.

For example, IsAudioBigENDIAN(AUDIO_S16LE) returns 0.

Returns
non-zero if format is bigendian, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsFloat()

bool SDL::AudioFormat::IsFloat ( ) const
constexpr

Determine if an AudioFormat represents floating point data.

For example, IsAudioFloat(AUDIO_S16) returns 0.

Returns
non-zero if format is floating point, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsInt()

bool SDL::AudioFormat::IsInt ( ) const
constexpr

Determine if an AudioFormat represents integer data.

For example, IsAudioInt(AUDIO_F32) returns 0.

Returns
non-zero if format is integer, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsLittleEndian()

bool SDL::AudioFormat::IsLittleEndian ( ) const
constexpr

Determine if an AudioFormat represents littleendian data.

For example, IsAudioLittleEndian(AUDIO_S16BE) returns 0.

Returns
non-zero if format is littleendian, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsPhysical()

bool SDL::AudioDeviceBase::IsPhysical ( ) const
inline

Determine if an audio device is physical (instead of logical).

An AudioDevice 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 OpenAudioDevice or 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.

Returns
true if devid is a physical device, false if it is logical.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsPlayback()

bool SDL::AudioDeviceBase::IsPlayback ( ) const
inline

Determine if an audio device is a playback device (instead of recording).

This function may return either true or false for invalid device IDs.

Returns
true if devid is a playback device, false if it is recording.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsSigned()

bool SDL::AudioFormat::IsSigned ( ) const
constexpr

Determine if an AudioFormat represents signed data.

For example, IsAudioSigned(AUDIO_U8) returns 0.

Returns
non-zero if format is signed, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ IsUnsigned()

bool SDL::AudioFormat::IsUnsigned ( ) const
constexpr

Determine if an AudioFormat represents unsigned data.

For example, IsAudioUnsigned(AUDIO_S16) returns 0.

Returns
non-zero if format is unsigned, zero otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ LoadWAV()

OwnArray< Uint8 > SDL::LoadWAV ( StringParam path,
AudioSpec * spec )
inline

Loads a WAV from a file path.

This is a convenience function that is effectively the same as:

LoadWAV_IO(IOFromFile(path, "rb"), true, spec);
OwnArray< Uint8 > LoadWAV_IO(IOStreamRef src, AudioSpec *spec, bool closeio=false)
Load the audio data of a WAVE file into memory.
Definition SDL3pp_audio.h:5288
IOStream IOFromFile(StringParam file, StringParam mode)
Use this function to create a new IOStream structure for reading from and/or writing to a named file.
Definition SDL3pp_iostream.h:1683
Parameters
paththe file path of the WAV file to open.
speca pointer to an AudioSpec that will be set to the WAVE data's format details on successful return.
Returns
an allocated buffer containing the audio data on success, or nullptr on failure.

This function throws if the .WAV file cannot be opened, uses an unknown data format, or is corrupt,

Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
LoadWAV_IO

◆ LoadWAV_IO()

OwnArray< Uint8 > SDL::LoadWAV_IO ( IOStreamRef src,
AudioSpec * spec,
bool closeio = false )
inline

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 AudioSpec members freq, channels, and format are set to the values of the audio data in the buffer.

It's necessary to use 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 nullptr on error and in all cases (with the exception of src being nullptr), an appropriate error message will be set.

It is required that the data source supports seeking.

Example:

LoadWAV_IO(IOStream.FromFile("sample.wav", "rb"), spec);
static IOStream FromFile(StringParam file, StringParam mode)
Use this function to create a new IOStream structure for reading from and/or writing to a named file.
Definition SDL3pp_iostream.h:1688
The read/write operation structure.
Definition SDL3pp_iostream.h:1332

Note that the LoadWAV function does this same thing for you, but in a less messy way:

LoadWAV("sample.wav", &spec);
OwnArray< Uint8 > LoadWAV(StringParam path, AudioSpec *spec)
Loads a WAV from a file path.
Definition SDL3pp_audio.h:5322
Parameters
srcthe data source for the WAVE data.
speca pointer to an AudioSpec that will be set to the WAVE data's format details on successful return.
closeioif true, calls CloseIO() on src before returning, even in the case of an error.
Returns
an allocated buffer containing the audio data on success, or nullptr on failure.

This function throws if the .WAV file cannot be opened, uses an unknown data format, or is corrupt; call GetError() for more information.

Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
LoadWAV

◆ Lock()

AudioStreamLock SDL::AudioStreamBase::Lock ( )
inline

Lock an audio stream for serialized access.

Each 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 LockMutex for an internal lock; it has all the same attributes (recursive locks are allowed, etc).

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
UnlockAudioStream

◆ LockAudioStream()

void SDL::LockAudioStream ( AudioStreamRef stream)
inline

Lock an audio stream for serialized access.

Each 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 LockMutex for an internal lock; it has all the same attributes (recursive locks are allowed, etc).

Parameters
streamthe audio stream to lock.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
UnlockAudioStream

◆ MixAudio() [1/2]

void SDL::MixAudio ( TargetBytes dst,
SourceBytes src,
AudioFormat format,
float volume )
inline

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, MixAudio() is really only needed when you're mixing a single audio stream with a volume adjustment.

Parameters
dstthe destination for the mixed audio.
srcthe source audio buffer to be mixed.
formatthe AudioFormat structure representing the desired audio format.
volumeranges from 0.0 - 1.0, and should be set to 1.0 for full audio volume.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ MixAudio() [2/2]

void SDL::MixAudio ( Uint8 * dst,
SourceBytes src,
AudioFormat format,
float volume )
inline

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, MixAudio() is really only needed when you're mixing a single audio stream with a volume adjustment.

Parameters
dstthe destination for the mixed audio.
srcthe source audio buffer to be mixed.
formatthe AudioFormat structure representing the desired audio format.
volumeranges from 0.0 - 1.0, and should be set to 1.0 for full audio volume.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ OpenAudioDevice()

AudioDevice SDL::OpenAudioDevice ( AudioDeviceRef devid,
OptionalRef< const AudioSpec > spec )
inline

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 OpenAudioDeviceStream instead of this function).

If you don't care about opening a specific device, pass a devid of either AUDIO_DEVICE_DEFAULT_PLAYBACK or 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. 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 AudioDevice 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 AudioDevice 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 CloseAudioDevice() on the returned device id.

Parameters
devidthe device instance id to open, or AUDIO_DEVICE_DEFAULT_PLAYBACK or AUDIO_DEVICE_DEFAULT_RECORDING for the most reasonable default device.
specthe requested device configuration. Can be nullptr to use reasonable defaults.
Returns
the device ID on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
CloseAudioDevice
GetAudioDeviceFormat

◆ OpenAudioDeviceStream() [1/2]

AudioStream SDL::OpenAudioDeviceStream ( AudioDeviceRef devid,
OptionalRef< const AudioSpec > spec,
AudioStreamCallback callback = nullptr,
void * userdata = nullptr )
inline

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 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 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 nullptr, the system will choose the format, and the app can use GetAudioStreamFormat() to obtain this information later.

If you don't care about opening a specific audio device, you can (and probably should), use AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and AUDIO_DEVICE_DEFAULT_RECORDING for recording.

One can optionally provide a callback function; if nullptr, 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 DestroyAudioStream will also close the audio device associated with this stream.

Parameters
devidan audio device to open, or AUDIO_DEVICE_DEFAULT_PLAYBACK or AUDIO_DEVICE_DEFAULT_RECORDING.
specthe audio stream's data format. Can be nullptr.
callbacka callback where the app will provide new data for playback, or receive new data for recording. Can be nullptr, in which case the app will need to call PutAudioStreamData or GetAudioStreamData as necessary.
userdataapp-controlled pointer passed to callback. Can be nullptr. Ignored if callback is nullptr.
Returns
an audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamDevice
ResumeAudioStreamDevice

◆ OpenAudioDeviceStream() [2/2]

AudioStream SDL::OpenAudioDeviceStream ( AudioDeviceRef devid,
OptionalRef< const AudioSpec > spec,
AudioStreamCB callback )
inline

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 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 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 nullptr, the system will choose the format, and the app can use GetAudioStreamFormat() to obtain this information later.

If you don't care about opening a specific audio device, you can (and probably should), use AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and AUDIO_DEVICE_DEFAULT_RECORDING for recording.

One can optionally provide a callback function; if nullptr, 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 DestroyAudioStream will also close the audio device associated with this stream.

Parameters
devidan audio device to open, or AUDIO_DEVICE_DEFAULT_PLAYBACK or AUDIO_DEVICE_DEFAULT_RECORDING.
specthe audio stream's data format. Can be nullptr.
callbacka callback where the app will provide new data for playback, or receive new data for recording.
Returns
an audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamDevice
ResumeAudioStreamDevice

◆ OpenStream() [1/2]

AudioStream SDL::AudioDeviceBase::OpenStream ( OptionalRef< const AudioSpec > spec,
AudioStreamCallback callback,
void * userdata )
inline

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 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 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 nullptr, the system will choose the format, and the app can use GetAudioStreamFormat() to obtain this information later.

If you don't care about opening a specific audio device, you can (and probably should), use AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and AUDIO_DEVICE_DEFAULT_RECORDING for recording.

One can optionally provide a callback function; if nullptr, 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 DestroyAudioStream will also close the audio device associated with this stream.

Parameters
specthe audio stream's data format. Can be nullptr.
callbacka callback where the app will provide new data for playback, or receive new data for recording. Can be nullptr, in which case the app will need to call PutAudioStreamData or GetAudioStreamData as necessary.
userdataapp-controlled pointer passed to callback. Can be nullptr. Ignored if callback is nullptr.
Returns
an audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamDevice
ResumeAudioStreamDevice

◆ OpenStream() [2/2]

AudioStream SDL::AudioDeviceBase::OpenStream ( OptionalRef< const AudioSpec > spec,
AudioStreamCB callback )
inline

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 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 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 nullptr, the system will choose the format, and the app can use GetAudioStreamFormat() to obtain this information later.

If you don't care about opening a specific audio device, you can (and probably should), use AUDIO_DEVICE_DEFAULT_PLAYBACK for playback and AUDIO_DEVICE_DEFAULT_RECORDING for recording.

One can optionally provide a callback function; if nullptr, 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 DestroyAudioStream will also close the audio device associated with this stream.

Parameters
specthe audio stream's data format. Can be nullptr.
callbacka callback where the app will provide new data for playback, or receive new data for recording.
Returns
an audio stream on success.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
GetAudioStreamDevice
ResumeAudioStreamDevice

◆ Pause()

void SDL::AudioDeviceBase::Pause ( )
inline

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 OpenAudioDevice() can be.

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
ResumeAudioDevice
AudioDevicePaused

◆ PauseAudioDevice()

void SDL::PauseAudioDevice ( AudioDeviceRef devid)
inline

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 OpenAudioDevice() can be.

Parameters
devida device opened by OpenAudioDevice().
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
ResumeAudioDevice
AudioDevicePaused

◆ PauseAudioStreamDevice()

void SDL::PauseAudioStreamDevice ( AudioStreamRef stream)
inline

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.

Parameters
streamthe audio stream associated with the audio device to pause.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
ResumeAudioStreamDevice

◆ Paused()

bool SDL::AudioDeviceBase::Paused ( ) const
inline

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 OpenAudioDevice() can be. Physical and invalid device IDs will report themselves as unpaused here.

Returns
true if device is valid and paused, false otherwise.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PauseAudioDevice
ResumeAudioDevice

◆ PauseDevice()

void SDL::AudioStreamBase::PauseDevice ( )
inline

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.

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
ResumeAudioStreamDevice

◆ PutAudioStreamData()

void SDL::PutAudioStreamData ( AudioStreamRef stream,
SourceBytes buf )
inline

Add data to the stream.

This data must match the format/channels/samplerate specified in the latest call to 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.

Parameters
streamthe stream the audio data is being added to.
bufa pointer to the audio data to add.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutAudioStreamDataNoCopy() [1/2]

void SDL::PutAudioStreamDataNoCopy ( AudioStreamRef stream,
SourceBytes buf,
AudioStreamDataCompleteCallback callback,
void * userdata )
inline

Add external data to an audio stream without copying it.

Unlike 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 ClearAudioStream() or DestroyAudioStream().

The data must match the format/channels/samplerate specified in the latest call to 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 PutAudioStreamData(), which allocates and buffers in blocks.

Parameters
streamthe stream the audio data is being added to.
bufa pointer to the audio data to add.
callbackthe callback function to call when the data is no longer needed by the stream. May be nullptr.
userdataan opaque pointer provided to the callback for its own personal use.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutAudioStreamDataNoCopy() [2/2]

void SDL::PutAudioStreamDataNoCopy ( AudioStreamRef stream,
SourceBytes buf,
AudioStreamDataCompleteCB callback )
inline

Add external data to an audio stream without copying it.

Unlike 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 ClearAudioStream() or DestroyAudioStream().

The data must match the format/channels/samplerate specified in the latest call to 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 PutAudioStreamData(), which allocates and buffers in blocks.

Parameters
streamthe stream the audio data is being added to.
bufa pointer to the audio data to add.
callbackthe callback function to call when the data is no longer needed by the stream. May be nullptr.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutAudioStreamPlanarData()

void SDL::PutAudioStreamPlanarData ( AudioStreamRef stream,
const void *const * channel_buffers,
int num_channels,
int num_samples )
inline

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 SetAudioStreamFormat, or the format specified when creating the stream if it hasn't been changed.

The data will be interleaved and queued. Note that 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 nullptr; 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 nullptr (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 PutAudioStreamData, which wants the size of a single array in bytes.

Parameters
streamthe stream the audio data is being added to.
channel_buffersa pointer to an array of arrays, one array per channel.
num_channelsthe number of arrays in channel_buffers or -1.
num_samplesthe number of samples per array to write to the stream.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutData()

void SDL::AudioStreamBase::PutData ( SourceBytes buf)
inline

Add data to the stream.

This data must match the format/channels/samplerate specified in the latest call to 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.

Parameters
bufa pointer to the audio data to add.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutDataNoCopy() [1/2]

void SDL::AudioStreamBase::PutDataNoCopy ( SourceBytes buf,
AudioStreamDataCompleteCallback callback,
void * userdata )
inline

Add external data to an audio stream without copying it.

Unlike 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 ClearAudioStream() or DestroyAudioStream().

The data must match the format/channels/samplerate specified in the latest call to 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 PutAudioStreamData(), which allocates and buffers in blocks.

Parameters
bufa pointer to the audio data to add.
callbackthe callback function to call when the data is no longer needed by the stream. May be nullptr.
userdataan opaque pointer provided to the callback for its own personal use.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutDataNoCopy() [2/2]

void SDL::AudioStreamBase::PutDataNoCopy ( SourceBytes buf,
AudioStreamDataCompleteCB callback )
inline

Add external data to an audio stream without copying it.

Unlike 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 ClearAudioStream() or DestroyAudioStream().

The data must match the format/channels/samplerate specified in the latest call to 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 PutAudioStreamData(), which allocates and buffers in blocks.

Parameters
bufa pointer to the audio data to add.
callbackthe callback function to call when the data is no longer needed by the stream. May be nullptr.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ PutPlanarData()

void SDL::AudioStreamBase::PutPlanarData ( const void *const * channel_buffers,
int num_channels,
int num_samples )
inline

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 SetAudioStreamFormat, or the format specified when creating the stream if it hasn't been changed.

The data will be interleaved and queued. Note that 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 nullptr; 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 nullptr (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 PutAudioStreamData, which wants the size of a single array in bytes.

Parameters
channel_buffersa pointer to an array of arrays, one array per channel.
num_channelsthe number of arrays in channel_buffers or -1.
num_samplesthe number of samples per array to write to the stream.
Exceptions
Erroron failure.
Thread safety:
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.
See also
ClearAudioStream
FlushAudioStream
GetAudioStreamData
GetAudioStreamQueued

◆ reset()

void SDL::AudioStreamLock::reset ( )
inline

Unlock an audio stream for serialized access.

This unlocks an audio stream after a call to LockAudioStream.

Exceptions
Erroron failure.
Thread safety:
You should only call this from the same thread that previously called LockAudioStream.
Since
This function is available since SDL 3.2.0.
See also
LockAudioStream

◆ Resume()

void SDL::AudioDeviceBase::Resume ( )
inline

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 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 OpenAudioDevice() can be.

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
AudioDevicePaused
PauseAudioDevice

◆ ResumeAudioDevice()

void SDL::ResumeAudioDevice ( AudioDeviceRef devid)
inline

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 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 OpenAudioDevice() can be.

Parameters
devida device opened by OpenAudioDevice().
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
AudioDevicePaused
PauseAudioDevice

◆ ResumeAudioStreamDevice()

void SDL::ResumeAudioStreamDevice ( AudioStreamRef stream)
inline

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.

OpenAudioDeviceStream opens audio devices in a paused state, so this function call is required for audio playback to begin on such devices.

Parameters
streamthe audio stream associated with the audio device to resume.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PauseAudioStreamDevice

◆ ResumeDevice()

void SDL::AudioStreamBase::ResumeDevice ( )
inline

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.

OpenAudioDeviceStream opens audio devices in a paused state, so this function call is required for audio playback to begin on such devices.

Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
PauseAudioStreamDevice

◆ SetAudioDeviceGain()

void SDL::SetAudioDeviceGain ( AudioDeviceRef devid,
float gain )
inline

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.

Parameters
devidthe audio device on which to change gain.
gainthe gain. 1.0f is no change, 0.0f is silence.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioDeviceGain

◆ SetAudioPostmixCallback() [1/2]

void SDL::SetAudioPostmixCallback ( AudioDeviceRef devid,
AudioPostmixCallback callback,
void * userdata )
inline

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 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 nullptr callback function disables any previously-set callback.

Parameters
devidthe ID of an opened audio device.
callbacka callback function to be called. Can be nullptr.
userdataapp-controlled pointer passed to callback. Can be nullptr.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ SetAudioPostmixCallback() [2/2]

void SDL::SetAudioPostmixCallback ( AudioDeviceRef devid,
AudioPostmixCB callback )
inline

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 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 nullptr callback function disables any previously-set callback.

Parameters
devidthe ID of an opened audio device.
callbacka callback function to be called. Can be nullptr.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ SetAudioStreamFormat()

void SDL::SetAudioStreamFormat ( AudioStreamRef stream,
OptionalRef< const AudioSpec > src_spec,
OptionalRef< const AudioSpec > dst_spec )
inline

Change the input and output formats of an audio stream.

Future calls to and GetAudioStreamAvailable and GetAudioStreamData will reflect the new format, and future calls to 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.

Parameters
streamthe stream the format is being changed.
src_specthe new format of the audio input; if nullptr, it is not changed.
dst_specthe new format of the audio output; if nullptr, it is not changed.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioStreamFormat
SetAudioStreamFrequencyRatio

◆ SetAudioStreamFrequencyRatio()

void SDL::SetAudioStreamFrequencyRatio ( AudioStreamRef stream,
float ratio )
inline

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 GetAudioStreamData, and can be continuously changed to create various effects.

Parameters
streamthe stream on which the frequency ratio is being changed.
ratiothe frequency ratio. 1.0 is normal speed. Must be between 0.01 and 100.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioStreamFrequencyRatio
SetAudioStreamFormat

◆ SetAudioStreamGain()

void SDL::SetAudioStreamGain ( AudioStreamRef stream,
float gain )
inline

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 GetAudioStreamData, and can be continuously changed to create various effects.

Parameters
streamthe stream on which the gain is being changed.
gainthe gain. 1.0f is no change, 0.0f is silence.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioStreamGain

◆ SetAudioStreamGetCallback() [1/2]

void SDL::SetAudioStreamGetCallback ( AudioStreamRef stream,
AudioStreamCallback callback,
void * userdata )
inline

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 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 nullptr function turns off the callback.

Parameters
streamthe audio stream to set the new callback on.
callbackthe new callback function to call when data is requested from the stream.
userdataan opaque pointer provided to the callback for its own personal use.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamPutCallback

◆ SetAudioStreamGetCallback() [2/2]

void SDL::SetAudioStreamGetCallback ( AudioStreamRef stream,
AudioStreamCB callback )
inline

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 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 nullptr function turns off the callback.

Parameters
streamthe audio stream to set the new callback on.
callbackthe new callback function to call when data is requested from the stream.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamPutCallback

◆ SetAudioStreamInputChannelMap()

void SDL::SetAudioStreamInputChannelMap ( AudioStreamRef stream,
std::span< int > chmap )
inline

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 expectsorder that SDL expects.

The input channel map reorders data that is added to a stream via PutAudioStreamData. Future calls to 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 nullptr 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.

Parameters
streamthe AudioStream to change.
chmapthe new channel map, nullptr to reset to default.
Exceptions
Erroron failure.
Thread safety:
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.
See also
SetAudioStreamOutputChannelMap

◆ SetAudioStreamOutputChannelMap()

void SDL::SetAudioStreamOutputChannelMap ( AudioStreamRef stream,
std::span< int > chmap )
inline

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 expectsorder that SDL expects.

The output channel map reorders data that is leaving a stream via 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 GetAudioStreamData.

Audio streams default to no remapping applied. Passing a nullptr 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.

Parameters
streamthe AudioStream to change.
chmapthe new channel map, nullptr to reset to default.
Exceptions
Erroron failure.
Thread safety:
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.
See also
SetAudioStreamInputChannelMap

◆ SetAudioStreamPutCallback() [1/2]

void SDL::SetAudioStreamPutCallback ( AudioStreamRef stream,
AudioStreamCallback callback,
void * userdata )
inline

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 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 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 nullptr function turns off the callback.

Parameters
streamthe audio stream to set the new callback on.
callbackthe new callback function to call when data is added to the stream.
userdataan opaque pointer provided to the callback for its own personal use.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamGetCallback

◆ SetAudioStreamPutCallback() [2/2]

void SDL::SetAudioStreamPutCallback ( AudioStreamRef stream,
AudioStreamCB callback )
inline

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 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 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 nullptr function turns off the callback.

Parameters
streamthe audio stream to set the new callback on.
callbackthe new callback function to call when data is added to the stream.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamGetCallback

◆ SetFormat()

void SDL::AudioStreamBase::SetFormat ( OptionalRef< const AudioSpec > src_spec,
OptionalRef< const AudioSpec > dst_spec )
inline

Change the input and output formats of an audio stream.

Future calls to and GetAudioStreamAvailable and GetAudioStreamData will reflect the new format, and future calls to 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.

Parameters
src_specthe new format of the audio input; if nullptr, it is not changed.
dst_specthe new format of the audio output; if nullptr, it is not changed.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioStreamFormat
SetAudioStreamFrequencyRatio

◆ SetFrequencyRatio()

void SDL::AudioStreamBase::SetFrequencyRatio ( float ratio)
inline

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 GetAudioStreamData, and can be continuously changed to create various effects.

Parameters
ratiothe frequency ratio. 1.0 is normal speed. Must be between 0.01 and 100.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioStreamFrequencyRatio
SetAudioStreamFormat

◆ SetGain() [1/2]

void SDL::AudioDeviceBase::SetGain ( float gain)
inline

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.

Parameters
gainthe gain. 1.0f is no change, 0.0f is silence.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioDeviceGain

◆ SetGain() [2/2]

void SDL::AudioStreamBase::SetGain ( float gain)
inline

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 GetAudioStreamData, and can be continuously changed to create various effects.

Parameters
gainthe gain. 1.0f is no change, 0.0f is silence.
Exceptions
Erroron failure.
Thread safety:
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.
See also
GetAudioStreamGain

◆ SetGetCallback() [1/2]

void SDL::AudioStreamBase::SetGetCallback ( AudioStreamCallback callback,
void * userdata )
inline

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 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 nullptr function turns off the callback.

Parameters
callbackthe new callback function to call when data is requested from the stream.
userdataan opaque pointer provided to the callback for its own personal use.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamPutCallback

◆ SetGetCallback() [2/2]

void SDL::AudioStreamBase::SetGetCallback ( AudioStreamCB callback)
inline

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 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 nullptr function turns off the callback.

Parameters
callbackthe new callback function to call when data is requested from the stream.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamPutCallback

◆ SetInputChannelMap()

void SDL::AudioStreamBase::SetInputChannelMap ( std::span< int > chmap)
inline

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 expectsorder that SDL expects.

The input channel map reorders data that is added to a stream via PutAudioStreamData. Future calls to 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 nullptr 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.

Parameters
chmapthe new channel map, nullptr to reset to default.
Exceptions
Erroron failure.
Thread safety:
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.
See also
SetAudioStreamOutputChannelMap

◆ SetOutputChannelMap()

void SDL::AudioStreamBase::SetOutputChannelMap ( std::span< int > chmap)
inline

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 expectsorder that SDL expects.

The output channel map reorders data that is leaving a stream via 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 GetAudioStreamData.

Audio streams default to no remapping applied. Passing a nullptr 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.

Parameters
chmapthe new channel map, nullptr to reset to default.
Exceptions
Erroron failure.
Thread safety:
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.
See also
SetAudioStreamInputChannelMap

◆ SetPostmixCallback() [1/2]

void SDL::AudioDeviceBase::SetPostmixCallback ( AudioPostmixCallback callback,
void * userdata )
inline

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 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 nullptr callback function disables any previously-set callback.

Parameters
callbacka callback function to be called. Can be nullptr.
userdataapp-controlled pointer passed to callback. Can be nullptr.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ SetPostmixCallback() [2/2]

void SDL::AudioDeviceBase::SetPostmixCallback ( AudioPostmixCB callback)
inline

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 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 nullptr callback function disables any previously-set callback.

Parameters
callbacka callback function to be called. Can be nullptr.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.

◆ SetPutCallback() [1/2]

void SDL::AudioStreamBase::SetPutCallback ( AudioStreamCallback callback,
void * userdata )
inline

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 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 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 nullptr function turns off the callback.

Parameters
callbackthe new callback function to call when data is added to the stream.
userdataan opaque pointer provided to the callback for its own personal use.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamGetCallback

◆ SetPutCallback() [2/2]

void SDL::AudioStreamBase::SetPutCallback ( AudioStreamCB callback)
inline

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 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 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 nullptr function turns off the callback.

Parameters
callbackthe new callback function to call when data is added to the stream.
Exceptions
Erroron failure.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
SetAudioStreamGetCallback

◆ Unbind()

void SDL::AudioStreamBase::Unbind ( )
inline

Unbind a single audio stream from its audio device.

This is a convenience function, equivalent to calling UnbindAudioStreams(&stream, 1).

Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStream

◆ UnbindAudioStream()

void SDL::UnbindAudioStream ( AudioStreamRef stream)
inline

Unbind a single audio stream from its audio device.

This is a convenience function, equivalent to calling UnbindAudioStreams(&stream, 1).

Parameters
streaman audio stream to unbind from a device. Can be nullptr.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStream

◆ UnbindAudioStreams()

void SDL::UnbindAudioStreams ( std::span< AudioStreamRef > streams)
inline

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.

Parameters
streamsan array of audio streams to unbind. Can be nullptr or contain nullptr.
Thread safety:
It is safe to call this function from any thread.
Since
This function is available since SDL 3.2.0.
See also
BindAudioStreams

◆ Unlock()

void SDL::AudioStreamBase::Unlock ( AudioStreamLock && lock)
inline

Unlock an audio stream for serialized access.

This unlocks an audio stream after a call to LockAudioStream.

Exceptions
Erroron failure.
Thread safety:
You should only call this from the same thread that previously called LockAudioStream.
Since
This function is available since SDL 3.2.0.
See also
LockAudioStream

◆ UnlockAudioStream()

void SDL::UnlockAudioStream ( AudioStreamRef stream)
inline

Unlock an audio stream for serialized access.

This unlocks an audio stream after a call to LockAudioStream.

Parameters
streamthe audio stream to unlock.
Exceptions
Erroron failure.
Thread safety:
You should only call this from the same thread that previously called LockAudioStream.
Since
This function is available since SDL 3.2.0.
See also
LockAudioStream

Variable Documentation

◆ AUDIO_DEVICE_DEFAULT_PLAYBACK

AudioDeviceID SDL::AUDIO_DEVICE_DEFAULT_PLAYBACK
constexpr
Initial value:
=
SDL_AUDIO_DEVICE_DEFAULT_PLAYBACK

A value used to request a default playback audio device.

Several functions that require an AudioDevice 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 constant is available since SDL 3.2.0.

◆ AUDIO_DEVICE_DEFAULT_RECORDING

AudioDeviceID SDL::AUDIO_DEVICE_DEFAULT_RECORDING
constexpr
Initial value:
=
SDL_AUDIO_DEVICE_DEFAULT_RECORDING

A value used to request a default recording audio device.

Several functions that require an AudioDevice 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 constant is available since SDL 3.2.0.

◆ AUDIO_F32BE

AudioFormat SDL::AUDIO_F32BE
constexpr
Initial value:
=
SDL_AUDIO_F32BE

As above, but big-endian byte order.

◆ AUDIO_F32LE

AudioFormat SDL::AUDIO_F32LE
constexpr
Initial value:
=
SDL_AUDIO_F32LE

32-bit floating point samples

◆ AUDIO_MASK_BIG_ENDIAN

Uint32 SDL::AUDIO_MASK_BIG_ENDIAN = SDL_AUDIO_MASK_BIG_ENDIAN
constexpr

Mask of bits in an AudioFormat that contain the bigendian flag.

Generally one should use IsAudioBigENDIAN or IsAudioLittleEndian instead of this constant directly.

Since
This constant is available since SDL 3.2.0.

◆ AUDIO_MASK_BITSIZE

Uint32 SDL::AUDIO_MASK_BITSIZE = SDL_AUDIO_MASK_BITSIZE
constexpr

Mask of bits in an AudioFormat that contains the format bit size.

Generally one should use AudioBitSize instead of this constant directly.

Since
This constant is available since SDL 3.2.0.

◆ AUDIO_MASK_FLOAT

Uint32 SDL::AUDIO_MASK_FLOAT = SDL_AUDIO_MASK_FLOAT
constexpr

Mask of bits in an AudioFormat that contain the floating point flag.

Generally one should use IsAudioFloat instead of this constant directly.

Since
This constant is available since SDL 3.2.0.

◆ AUDIO_MASK_SIGNED

Uint32 SDL::AUDIO_MASK_SIGNED = SDL_AUDIO_MASK_SIGNED
constexpr

Mask of bits in an AudioFormat that contain the signed data flag.

Generally one should use IsAudioSigned instead of this constant directly.

Since
This constant is available since SDL 3.2.0.

◆ AUDIO_S16BE

AudioFormat SDL::AUDIO_S16BE
constexpr
Initial value:
=
SDL_AUDIO_S16BE

As above, but big-endian byte order.

◆ AUDIO_S32BE

AudioFormat SDL::AUDIO_S32BE
constexpr
Initial value:
=
SDL_AUDIO_S32BE

As above, but big-endian byte order.

◆ AUDIO_UNKNOWN

AudioFormat SDL::AUDIO_UNKNOWN
constexpr
Initial value:
=
SDL_AUDIO_UNKNOWN

Unspecified audio format.