#include "SDL_stdinc.h"
#include "SDL_error.h"
#include "SDL_endian.h"
#include "SDL_mutex.h"
#include "SDL_thread.h"
#include "SDL_rwops.h"
#include "begin_code.h"
#include "close_code.h"

Include dependency graph for SDL_audio.h:

This graph shows which files directly or indirectly include this file:

Go to the source code of this file.

Data Structures
struct	SDL_AudioSpec

struct	SDL_AudioCVT
	A structure to hold a set of audio conversion filters and buffers. More...

Macros
Audio flags
#define	SDL_AUDIO_MASK_BITSIZE (0xFF)

#define	SDL_AUDIO_MASK_DATATYPE (1<<8)

#define	SDL_AUDIO_MASK_ENDIAN (1<<12)

#define	SDL_AUDIO_MASK_SIGNED (1<<15)

#define	SDL_AUDIO_BITSIZE(x) (x & SDL_AUDIO_MASK_BITSIZE)

#define	SDL_AUDIO_ISFLOAT(x) (x & SDL_AUDIO_MASK_DATATYPE)

#define	SDL_AUDIO_ISBIGENDIAN(x) (x & SDL_AUDIO_MASK_ENDIAN)

#define	SDL_AUDIO_ISSIGNED(x) (x & SDL_AUDIO_MASK_SIGNED)

#define	SDL_AUDIO_ISINT(x) (!SDL_AUDIO_ISFLOAT(x))

#define	SDL_AUDIO_ISLITTLEENDIAN(x) (!SDL_AUDIO_ISBIGENDIAN(x))

#define	SDL_AUDIO_ISUNSIGNED(x) (!SDL_AUDIO_ISSIGNED(x))

Audio format flags
Defaults to LSB byte order.
#define	AUDIO_U8 0x0008

#define	AUDIO_S8 0x8008

#define	AUDIO_U16LSB 0x0010

#define	AUDIO_S16LSB 0x8010

#define	AUDIO_U16MSB 0x1010

#define	AUDIO_S16MSB 0x9010

#define	AUDIO_U16 AUDIO_U16LSB

#define	AUDIO_S16 AUDIO_S16LSB

int32 support
#define	AUDIO_S32LSB 0x8020

#define	AUDIO_S32MSB 0x9020

#define	AUDIO_S32 AUDIO_S32LSB

float32 support
#define	AUDIO_F32LSB 0x8120

#define	AUDIO_F32MSB 0x9120

#define	AUDIO_F32 AUDIO_F32LSB

Native audio byte ordering
#define	AUDIO_U16SYS AUDIO_U16LSB

#define	AUDIO_S16SYS AUDIO_S16LSB

#define	AUDIO_S32SYS AUDIO_S32LSB

#define	AUDIO_F32SYS AUDIO_F32LSB

Typedefs
typedef Uint16	SDL_AudioFormat
	Audio format flags. More...

Functions
Driver discovery functions
These functions return the list of built in audio drivers, in the order that they are normally initialized by default.
int	SDL_GetNumAudioDrivers (void)

const char *	SDL_GetAudioDriver (int index)

Audio lock functions
The lock manipulated by these functions protects the callback function. During a SDL_LockAudio()/SDL_UnlockAudio() pair, you can be guaranteed that the callback function is not running. Do not call these from the callback function or you will cause deadlock.
void	SDL_LockAudio (void)

void	SDL_LockAudioDevice (SDL_AudioDeviceID dev)

void	SDL_UnlockAudio (void)

void	SDL_UnlockAudioDevice (SDL_AudioDeviceID dev)

void	SDL_CloseAudio (void)

void	SDL_CloseAudioDevice (SDL_AudioDeviceID dev)

Allow change flags
Which audio format changes are allowed when opening a device.
#define	SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001

#define	SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002

#define	SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004

#define	SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008

#define	SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE\|SDL_AUDIO_ALLOW_FORMAT_CHANGE\|SDL_AUDIO_ALLOW_CHANNELS_CHANGE\|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)

#define	SDL_AUDIOCVT_MAX_FILTERS 9
	Upper limit of filters in SDL_AudioCVT. More...

#define	SDL_AUDIOCVT_PACKED

typedef void(*	SDL_AudioCallback) (void userdata, Uint8 stream, int len)

typedef void(*	SDL_AudioFilter) (struct SDL_AudioCVT *cvt, SDL_AudioFormat format)

Pause audio functions
These functions pause and unpause the audio callback processing. They should be called with a parameter of 0 after opening the audio device to start playing sound. This is so you can safely initialize data for your callback function after opening the audio device. Silence will be written to the audio device during the pause.
#define	SDL_LoadWAV(file, spec, audio_buf, audio_len) SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)

#define	SDL_MIX_MAXVOLUME 128

void	SDL_PauseAudio (int pause_on)

void	SDL_PauseAudioDevice (SDL_AudioDeviceID dev, int pause_on)

SDL_AudioSpec *	SDL_LoadWAV_RW (SDL_RWops src, int freesrc, SDL_AudioSpec spec, Uint8 *audio_buf, Uint32 audio_len)
	Load the audio data of a WAVE file into memory. More...

void	SDL_FreeWAV (Uint8 *audio_buf)

int	SDL_BuildAudioCVT (SDL_AudioCVT *cvt, SDL_AudioFormat src_format, Uint8 src_channels, int src_rate, SDL_AudioFormat dst_format, Uint8 dst_channels, int dst_rate)

int	SDL_ConvertAudio (SDL_AudioCVT *cvt)

SDL_AudioStream *	SDL_NewAudioStream (const SDL_AudioFormat src_format, const Uint8 src_channels, const int src_rate, const SDL_AudioFormat dst_format, const Uint8 dst_channels, const int dst_rate)

int	SDL_AudioStreamPut (SDL_AudioStream stream, const void buf, int len)

int	SDL_AudioStreamGet (SDL_AudioStream stream, void buf, int len)

int	SDL_AudioStreamAvailable (SDL_AudioStream *stream)

int	SDL_AudioStreamFlush (SDL_AudioStream *stream)

void	SDL_AudioStreamClear (SDL_AudioStream *stream)

void	SDL_FreeAudioStream (SDL_AudioStream *stream)

void	SDL_MixAudio (Uint8 dst, const Uint8 src, Uint32 len, int volume)

void	SDL_MixAudioFormat (Uint8 dst, const Uint8 src, SDL_AudioFormat format, Uint32 len, int volume)

int	SDL_QueueAudio (SDL_AudioDeviceID dev, const void *data, Uint32 len)

Uint32	SDL_DequeueAudio (SDL_AudioDeviceID dev, void *data, Uint32 len)

Uint32	SDL_GetQueuedAudioSize (SDL_AudioDeviceID dev)

void	SDL_ClearQueuedAudio (SDL_AudioDeviceID dev)

Audio state
Get the current audio state.
enum	SDL_AudioStatus { SDL_AUDIO_STOPPED = 0 , SDL_AUDIO_PLAYING , SDL_AUDIO_PAUSED }

SDL_AudioStatus	SDL_GetAudioStatus (void)

SDL_AudioStatus	SDL_GetAudioDeviceStatus (SDL_AudioDeviceID dev)

Initialization and cleanup
These functions are used internally, and should not be used unless you have a specific need to specify the audio driver you want to use. You should normally use SDL_Init() or SDL_InitSubSystem().
typedef Uint32	SDL_AudioDeviceID

int	SDL_AudioInit (const char *driver_name)

void	SDL_AudioQuit (void)

const char *	SDL_GetCurrentAudioDriver (void)

int	SDL_OpenAudio (SDL_AudioSpec desired, SDL_AudioSpec obtained)

int	SDL_GetNumAudioDevices (int iscapture)

const char *	SDL_GetAudioDeviceName (int index, int iscapture)

SDL_AudioDeviceID	SDL_OpenAudioDevice (const char device, int iscapture, const SDL_AudioSpec desired, SDL_AudioSpec *obtained, int allowed_changes)

Detailed Description

Access to the raw audio mixing buffer for the SDL library.

Definition in file SDL_audio.h.

Macro Definition Documentation

◆ AUDIO_F32

#define AUDIO_F32 AUDIO_F32LSB

Definition at line 114 of file SDL_audio.h.

◆ AUDIO_F32LSB

#define AUDIO_F32LSB 0x8120

32-bit floating point samples

Definition at line 112 of file SDL_audio.h.

◆ AUDIO_F32MSB

#define AUDIO_F32MSB 0x9120

As above, but big-endian byte order

Definition at line 113 of file SDL_audio.h.

◆ AUDIO_F32SYS

#define AUDIO_F32SYS AUDIO_F32LSB

Definition at line 125 of file SDL_audio.h.

◆ AUDIO_S16

#define AUDIO_S16 AUDIO_S16LSB

Definition at line 96 of file SDL_audio.h.

◆ AUDIO_S16LSB

#define AUDIO_S16LSB 0x8010

Signed 16-bit samples

Definition at line 92 of file SDL_audio.h.

◆ AUDIO_S16MSB

#define AUDIO_S16MSB 0x9010

As above, but big-endian byte order

Definition at line 94 of file SDL_audio.h.

◆ AUDIO_S16SYS

#define AUDIO_S16SYS AUDIO_S16LSB

Definition at line 123 of file SDL_audio.h.

◆ AUDIO_S32

#define AUDIO_S32 AUDIO_S32LSB

Definition at line 105 of file SDL_audio.h.

◆ AUDIO_S32LSB

#define AUDIO_S32LSB 0x8020

32-bit integer samples

Definition at line 103 of file SDL_audio.h.

◆ AUDIO_S32MSB

#define AUDIO_S32MSB 0x9020

As above, but big-endian byte order

Definition at line 104 of file SDL_audio.h.

◆ AUDIO_S32SYS

#define AUDIO_S32SYS AUDIO_S32LSB

Definition at line 124 of file SDL_audio.h.

◆ AUDIO_S8

#define AUDIO_S8 0x8008

Signed 8-bit samples

Definition at line 90 of file SDL_audio.h.

◆ AUDIO_U16

#define AUDIO_U16 AUDIO_U16LSB

Definition at line 95 of file SDL_audio.h.

◆ AUDIO_U16LSB

#define AUDIO_U16LSB 0x0010

Unsigned 16-bit samples

Definition at line 91 of file SDL_audio.h.

◆ AUDIO_U16MSB

#define AUDIO_U16MSB 0x1010

As above, but big-endian byte order

Definition at line 93 of file SDL_audio.h.

◆ AUDIO_U16SYS

#define AUDIO_U16SYS AUDIO_U16LSB

Definition at line 122 of file SDL_audio.h.

◆ AUDIO_U8

#define AUDIO_U8 0x0008

Unsigned 8-bit samples

Definition at line 89 of file SDL_audio.h.

◆ SDL_AUDIO_ALLOW_ANY_CHANGE

#define SDL_AUDIO_ALLOW_ANY_CHANGE (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE|SDL_AUDIO_ALLOW_SAMPLES_CHANGE)

Definition at line 144 of file SDL_audio.h.

◆ SDL_AUDIO_ALLOW_CHANNELS_CHANGE

#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE 0x00000004

Definition at line 142 of file SDL_audio.h.

◆ SDL_AUDIO_ALLOW_FORMAT_CHANGE

#define SDL_AUDIO_ALLOW_FORMAT_CHANGE 0x00000002

Definition at line 141 of file SDL_audio.h.

◆ SDL_AUDIO_ALLOW_FREQUENCY_CHANGE

#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE 0x00000001

Definition at line 140 of file SDL_audio.h.

◆ SDL_AUDIO_ALLOW_SAMPLES_CHANGE

#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE 0x00000008

Definition at line 143 of file SDL_audio.h.

◆ SDL_AUDIO_BITSIZE

#define SDL_AUDIO_BITSIZE ( x ) (x & SDL_AUDIO_MASK_BITSIZE)

Definition at line 75 of file SDL_audio.h.

◆ SDL_AUDIO_ISBIGENDIAN

#define SDL_AUDIO_ISBIGENDIAN ( x ) (x & SDL_AUDIO_MASK_ENDIAN)

Definition at line 77 of file SDL_audio.h.

◆ SDL_AUDIO_ISFLOAT

#define SDL_AUDIO_ISFLOAT ( x ) (x & SDL_AUDIO_MASK_DATATYPE)

Definition at line 76 of file SDL_audio.h.

◆ SDL_AUDIO_ISINT

#define SDL_AUDIO_ISINT ( x ) (!SDL_AUDIO_ISFLOAT(x))

Definition at line 79 of file SDL_audio.h.

◆ SDL_AUDIO_ISLITTLEENDIAN

#define SDL_AUDIO_ISLITTLEENDIAN ( x ) (!SDL_AUDIO_ISBIGENDIAN(x))

Definition at line 80 of file SDL_audio.h.

◆ SDL_AUDIO_ISSIGNED

#define SDL_AUDIO_ISSIGNED ( x ) (x & SDL_AUDIO_MASK_SIGNED)

Definition at line 78 of file SDL_audio.h.

◆ SDL_AUDIO_ISUNSIGNED

#define SDL_AUDIO_ISUNSIGNED ( x ) (!SDL_AUDIO_ISSIGNED(x))

Definition at line 81 of file SDL_audio.h.

◆ SDL_AUDIO_MASK_BITSIZE

#define SDL_AUDIO_MASK_BITSIZE (0xFF)

Definition at line 71 of file SDL_audio.h.

◆ SDL_AUDIO_MASK_DATATYPE

#define SDL_AUDIO_MASK_DATATYPE (1<<8)

Definition at line 72 of file SDL_audio.h.

◆ SDL_AUDIO_MASK_ENDIAN

#define SDL_AUDIO_MASK_ENDIAN (1<<12)

Definition at line 73 of file SDL_audio.h.

◆ SDL_AUDIO_MASK_SIGNED

#define SDL_AUDIO_MASK_SIGNED (1<<15)

Definition at line 74 of file SDL_audio.h.

◆ SDL_AUDIOCVT_MAX_FILTERS

#define SDL_AUDIOCVT_MAX_FILTERS 9

Upper limit of filters in SDL_AudioCVT.

The maximum number of SDL_AudioFilter functions in SDL_AudioCVT is currently limited to 9. The SDL_AudioCVT.filters array has 10 pointers, one of which is the terminating NULL pointer.

Definition at line 203 of file SDL_audio.h.

◆ SDL_AUDIOCVT_PACKED

#define SDL_AUDIOCVT_PACKED

Definition at line 223 of file SDL_audio.h.

◆ SDL_LoadWAV

#define SDL_LoadWAV	(	file,
		spec,
		audio_buf,
		audio_len
	)	SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)

Loads a WAV from a file. Compatibility convenience function.

Definition at line 484 of file SDL_audio.h.

◆ SDL_MIX_MAXVOLUME

#define SDL_MIX_MAXVOLUME 128

Definition at line 649 of file SDL_audio.h.

Typedef Documentation

◆ SDL_AudioCallback

typedef void( * SDL_AudioCallback) (void *userdata, Uint8 *stream, int len)

This function is called when the audio device needs more data.

Parameters

userdata	An application-specific parameter saved in the SDL_AudioSpec structure
stream	A pointer to the audio data buffer.
len	The length of that buffer in bytes.

Once the callback returns, the buffer will no longer be valid. Stereo samples are stored in a LRLRLR ordering.

You can choose to avoid callbacks and use SDL_QueueAudio() instead, if you like. Just open your audio device with a NULL callback.

Definition at line 163 of file SDL_audio.h.

◆ SDL_AudioDeviceID

SDL_PRINTF_FORMAT_STRING const char int SDL_PRINTF_FORMAT_STRING const char int SDL_PRINTF_FORMAT_STRING const char int SDL_PRINTF_FORMAT_STRING const char const char SDL_SCANF_FORMAT_STRING const char return SDL_ThreadFunction const char void return Uint32 return Uint32 SDL_AssertionHandler void SDL_SpinLock SDL_atomic_t int int return SDL_atomic_t return void void void return void return int return SDL_AudioSpec SDL_AudioSpec return int int return return int SDL_RWops int SDL_AudioSpec Uint8 Uint32 return SDL_AudioCVT SDL_AudioFormat Uint8 int SDL_AudioFormat Uint8 int return Uint8 const Uint8 Uint32 int const char return return return return return return return return Uint32 return Uint32 SDL_Event return SDL_Event int return SDL_EventFilter void SDL_EventFilter void SDL_EventFilter void int return const char const char return SDL_JoystickGUID return int return int return SDL_GameController return int return const char return SDL_GameController SDL_GameControllerAxis return const char return SDL_GameController SDL_GameControllerButton return SDL_GameController SDL_RWops return SDL_TouchID SDL_RWops return int return int return return SDL_Joystick return SDL_Haptic SDL_Haptic return SDL_Haptic return SDL_Haptic SDL_HapticEffect return SDL_Haptic int Uint32 return SDL_Haptic int SDL_Haptic int return SDL_Haptic return SDL_Haptic return SDL_Haptic return SDL_Haptic return const char const char return const char SDL_HintCallback void int return SDL_Joystick return SDL_Joystick return const char return SDL_Joystick return SDL_Joystick return SDL_Joystick return int return SDL_Joystick int return SDL_Joystick int return return return SDL_Scancode return SDL_Scancode return SDL_Keycode return return const char return void int SDL_LogPriority SDL_LogOutputFunction void Uint32 const char const char SDL_Window return int int return SDL_Window int int return SDL_Surface int int return SDL_Cursor return int return SDL_mutex return SDL_mutex return Uint32 return SDL_sem return SDL_sem Uint32 return SDL_sem return SDL_cond SDL_cond return SDL_cond SDL_mutex Uint32 return Uint32 int Uint32 Uint32 Uint32 Uint32 return Uint32 return int return SDL_Palette const SDL_Color int int return const SDL_PixelFormat Uint8 Uint8 Uint8 return Uint32 const SDL_PixelFormat Uint8 Uint8 Uint8 float Uint16 int int return const SDL_Rect const SDL_Rect SDL_Rect return const SDL_Point int const SDL_Rect SDL_Rect return return int int Uint32 SDL_Window SDL_Renderer return SDL_Surface return SDL_Renderer SDL_RendererInfo return SDL_Renderer Uint32 int int int return SDL_Texture Uint32 int int int return SDL_Texture Uint8 Uint8 Uint8 return SDL_Texture Uint8 return SDL_Texture SDL_BlendMode return SDL_Texture const SDL_Rect const Uint8 int const Uint8 int const Uint8 int return SDL_Texture SDL_Renderer SDL_Texture return SDL_Renderer int int return SDL_Renderer const SDL_Rect return SDL_Renderer const SDL_Rect return SDL_Renderer float float return SDL_Renderer Uint8 Uint8 Uint8 Uint8 return SDL_Renderer SDL_BlendMode return SDL_Renderer return SDL_Renderer const SDL_Point int return SDL_Renderer const SDL_Point int return SDL_Renderer const SDL_Rect int return SDL_Renderer const SDL_Rect int return SDL_Renderer SDL_Texture const SDL_Rect const SDL_Rect const double const SDL_Point const SDL_RendererFlip return SDL_Renderer SDL_Renderer SDL_Texture return void int return return SDL_RWops return SDL_RWops return SDL_RWops return SDL_RWops return SDL_RWops Uint16 return SDL_RWops Uint32 return SDL_RWops Uint64 return const char unsigned int unsigned int unsigned int unsigned int Uint32 return SDL_Window SDL_Surface SDL_WindowShapeMode return size_t return void size_t return const char return void size_t size_t int(*) a int return int return int size_t return size_t return const wchar_t return const wchar_t size_t return const char size_t return const char size_t return char return char return const char int return int char int return long char int return Sint64 char int return const char return const char char int return const char char int return const char char return const char const char size_t return const char const char size_t return double return double return double return double return double return double return double int return float return const char const char return SDL_iconv_t const char size_t char size_t return Uint32 int int int Uint32 Uint32 Uint32 Uint32 return SDL_Surface SDL_Surface return SDL_RWops int return SDL_Surface int return SDL_Surface Uint32 return SDL_Surface Uint8 Uint8 Uint8 return SDL_Surface Uint8 return SDL_Surface SDL_BlendMode return SDL_Surface SDL_Rect SDL_Surface Uint32 Uint32 return SDL_Surface const SDL_Rect Uint32 return SDL_Surface const SDL_Rect SDL_Surface SDL_Rect return SDL_Surface const SDL_Rect SDL_Surface const SDL_Rect return SDL_Surface SDL_Rect SDL_Surface SDL_Rect return SDL_Thread return SDL_Thread return SDL_Thread int return SDL_TLSID const void void(*) return return Uint32 SDL_TimerID return int return SDL_TouchID int return return return const char return return int return int return int SDL_DisplayMode return int const SDL_DisplayMode SDL_DisplayMode return SDL_Window const SDL_DisplayMode return SDL_Window return const void return Uint32 return SDL_Window const char SDL_Window SDL_Surface SDL_Window const char return SDL_Window int int SDL_Window int int SDL_Window int int SDL_Window int int SDL_Window SDL_Window SDL_Window SDL_Window Uint32 return SDL_Window return SDL_Window SDL_bool SDL_Window float return SDL_Window const Uint16 const Uint16 const Uint16 return SDL_Window const char return SDL_GLattr int return SDL_Window return return SDL_Window int int return SDL_GLContext SDL_RWops int return return void return int int return double return SDL_bool return int int return SDL_AudioDeviceID const void Uint32 return SDL_AudioDeviceID int float float float return SDL_JoystickID return int SDL_Rect return SDL_Window float return SDL_Window return SDL_Renderer SDL_bool return SDL_AudioDeviceID

SDL Audio Device IDs.

A successful call to SDL_OpenAudio() is always device id 1, and legacy SDL audio APIs assume you want this device ID. SDL_OpenAudioDevice() calls always returns devices >= 2 on success. The legacy calls are good both for backwards compatibility and when you don't care about multiple, specific, or capture devices.

Definition at line 330 of file SDL_audio.h.

◆ SDL_AudioFilter

typedef void( * SDL_AudioFilter) (struct SDL_AudioCVT *cvt, SDL_AudioFormat format)

Definition at line 193 of file SDL_audio.h.

◆ SDL_AudioFormat

typedef Uint16 SDL_AudioFormat

Audio format flags.

These are what the 16 bits in SDL_AudioFormat currently mean... (Unspecified bits are always zero).

++-----------------------sample is signed if set
||
||       ++-----------sample is bigendian if set
||       ||
||       ||          ++---sample is float if set
||       ||          ||
||       ||          || +---sample bit size---+
||       ||          || |                     |
15 14 13 12 11 10 09 08 07 06 05 04 03 02 01 00

There are macros in SDL 2.0 and later to query these bits.

Definition at line 64 of file SDL_audio.h.

Enumeration Type Documentation

◆ SDL_AudioStatus

enum SDL_AudioStatus

Enumerator
SDL_AUDIO_STOPPED
SDL_AUDIO_PLAYING
SDL_AUDIO_PAUSED

Definition at line 395 of file SDL_audio.h.

 {
     SDL_AUDIO_STOPPED = 0,
     SDL_AUDIO_PLAYING,
     SDL_AUDIO_PAUSED
 } SDL_AudioStatus;

Function Documentation

◆ SDL_AudioInit()

int SDL_AudioInit ( const char * driver_name )

Definition at line 943 of file SDL_audio.c.

 {
     int i = 0;
     int initialized = 0;
     int tried_to_init = 0;
  
     if (SDL_WasInit(SDL_INIT_AUDIO)) {
         SDL_AudioQuit();        /* shutdown driver if already running. */
     }
  
     SDL_zero(current_audio);
     SDL_zeroa(open_devices);
  
     /* Select the proper audio driver */
     if (driver_name == NULL) {
         driver_name = SDL_getenv("SDL_AUDIODRIVER");
     }
  
     for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
         /* make sure we should even try this driver before doing so... */
         const AudioBootStrap *backend = bootstrap[i];
         if ((driver_name && (SDL_strncasecmp(backend->name, driver_name, SDL_strlen(driver_name)) != 0)) ||
             (!driver_name && backend->demand_only)) {
             continue;
         }
  
         tried_to_init = 1;
         SDL_zero(current_audio);
         current_audio.name = backend->name;
         current_audio.desc = backend->desc;
         initialized = backend->init(&current_audio.impl);
     }
  
     if (!initialized) {
         /* specific drivers will set the error message if they fail... */
         if (!tried_to_init) {
             if (driver_name) {
                 SDL_SetError("Audio target '%s' not available", driver_name);
             } else {
                 SDL_SetError("No available audio device");
             }
         }
  
         SDL_zero(current_audio);
         return -1;            /* No driver was available, so fail. */
     }
  
     current_audio.detectionLock = SDL_CreateMutex();
  
     finish_audio_entry_points_init();
  
     /* Make sure we have a list of devices available at startup. */
     current_audio.impl.DetectDevices();
  
 #ifdef HAVE_LIBSAMPLERATE_H
     LoadLibSampleRate();
 #endif
  
     return 0;
 }

References bootstrap, current_audio, AudioBootStrap::demand_only, SDL_AudioDriver::desc, AudioBootStrap::desc, SDL_AudioDriverImpl::DetectDevices, SDL_AudioDriver::detectionLock, finish_audio_entry_points_init(), i, SDL_AudioDriver::impl, AudioBootStrap::init, SDL_AudioDriver::name, AudioBootStrap::name, NULL, open_devices, SDL_AudioQuit(), SDL_CreateMutex, SDL_getenv, SDL_INIT_AUDIO, SDL_SetError, SDL_strlen, SDL_strncasecmp, SDL_WasInit, SDL_zero, and SDL_zeroa.

◆ SDL_AudioQuit()

void SDL_AudioQuit ( void )

◆ SDL_AudioStreamAvailable()

int SDL_AudioStreamAvailable ( SDL_AudioStream * stream )

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.

See also: SDL_NewAudioStream; SDL_AudioStreamPut; SDL_AudioStreamGet; SDL_AudioStreamFlush; SDL_AudioStreamClear; SDL_FreeAudioStream

Definition at line 1645 of file SDL_audiocvt.c.

 {
     return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
 }

References int(), and SDL_CountDataQueue().

◆ SDL_AudioStreamClear()

void SDL_AudioStreamClear ( SDL_AudioStream * stream )

Clear any pending data in the stream without converting it

See also: SDL_NewAudioStream; SDL_AudioStreamPut; SDL_AudioStreamGet; SDL_AudioStreamAvailable; SDL_AudioStreamFlush; SDL_FreeAudioStream

Definition at line 1651 of file SDL_audiocvt.c.

 {
     if (!stream) {
         SDL_InvalidParamError("stream");
     } else {
         SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
         if (stream->reset_resampler_func) {
             stream->reset_resampler_func(stream);
         }
         stream->first_run = SDL_TRUE;
         stream->staging_buffer_filled = 0;
     }
 }

References SDL_ClearDataQueue(), SDL_InvalidParamError, and SDL_TRUE.

◆ SDL_AudioStreamFlush()

int SDL_AudioStreamFlush ( SDL_AudioStream * stream )

Tell the stream that you're done sending data, and anything being buffered should be converted/resampled and made available immediately.

It is legal to add more data to a stream after flushing, but there will be audio gaps in the output. Generally this is intended to signal the end of input, so the complete output becomes available.

See also: SDL_NewAudioStream; SDL_AudioStreamPut; SDL_AudioStreamGet; SDL_AudioStreamAvailable; SDL_AudioStreamClear; SDL_FreeAudioStream

Definition at line 1570 of file SDL_audiocvt.c.

 {
     if (!stream) {
         return SDL_InvalidParamError("stream");
     }
  
     #if DEBUG_AUDIOSTREAM
     printf("AUDIOSTREAM: flushing! staging_buffer_filled=%d bytes\n", stream->staging_buffer_filled);
     #endif
  
     /* shouldn't use a staging buffer if we're not resampling. */
     SDL_assert((stream->dst_rate != stream->src_rate) || (stream->staging_buffer_filled == 0));
  
     if (stream->staging_buffer_filled > 0) {
         /* push the staging buffer + silence. We need to flush out not just
            the staging buffer, but the piece that the stream was saving off
            for right-side resampler padding. */
         const SDL_bool first_run = stream->first_run;
         const int filled = stream->staging_buffer_filled;
         int actual_input_frames = filled / stream->src_sample_frame_size;
         if (!first_run)
             actual_input_frames += stream->resampler_padding_samples / stream->pre_resample_channels;
  
         if (actual_input_frames > 0) {  /* don't bother if nothing to flush. */
             /* This is how many bytes we're expecting without silence appended. */
             int flush_remaining = ((int) SDL_ceil(actual_input_frames * stream->rate_incr)) * stream->dst_sample_frame_size;
  
             #if DEBUG_AUDIOSTREAM
             printf("AUDIOSTREAM: flushing with padding to get max %d bytes!\n", flush_remaining);
             #endif
  
             SDL_memset(stream->staging_buffer + filled, '\0', stream->staging_buffer_size - filled);
             if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
                 return -1;
             }
  
             /* we have flushed out (or initially filled) the pending right-side
                resampler padding, but we need to push more silence to guarantee
                the staging buffer is fully flushed out, too. */
             SDL_memset(stream->staging_buffer, '\0', filled);
             if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
                 return -1;
             }
         }
     }
  
     stream->staging_buffer_filled = 0;
     stream->first_run = SDL_TRUE;
  
     return 0;
 }

References DEBUG_AUDIOSTREAM, int(), SDL_assert, SDL_AudioStreamPutInternal(), SDL_ceil, SDL_InvalidParamError, SDL_memset, and SDL_TRUE.

◆ SDL_AudioStreamGet()

int SDL_AudioStreamGet	(	SDL_AudioStream *	stream,
		void *	buf,
		int	len
	)

Get converted/resampled data from the stream

Parameters

stream	The stream the audio is being requested from
buf	A buffer to fill with audio data
len	The maximum number of bytes to fill

Returns: The number of bytes read from the stream, or -1 on error

See also: SDL_NewAudioStream; SDL_AudioStreamPut; SDL_AudioStreamAvailable; SDL_AudioStreamFlush; SDL_AudioStreamClear; SDL_FreeAudioStream

Definition at line 1624 of file SDL_audiocvt.c.

 {
     #if DEBUG_AUDIOSTREAM
     printf("AUDIOSTREAM: want to get %d converted bytes\n", len);
     #endif
  
     if (!stream) {
         return SDL_InvalidParamError("stream");
     } else if (!buf) {
         return SDL_InvalidParamError("buf");
     } else if (len <= 0) {
         return 0;  /* nothing to do. */
     } else if ((len % stream->dst_sample_frame_size) != 0) {
         return SDL_SetError("Can't request partial sample frames");
     }
  
     return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
 }

References SDL_InvalidParamError, SDL_ReadFromDataQueue(), and SDL_SetError.

◆ SDL_AudioStreamPut()

int SDL_AudioStreamPut	(	SDL_AudioStream *	stream,
		const void *	buf,
		int	len
	)

Add data to be converted/resampled to the stream

Parameters

stream	The stream the audio data is being added to
buf	A pointer to the audio data to add
len	The number of bytes to write to the stream

Returns: 0 on success, or -1 on error.

See also: SDL_NewAudioStream; SDL_AudioStreamGet; SDL_AudioStreamAvailable; SDL_AudioStreamFlush; SDL_AudioStreamClear; SDL_FreeAudioStream

Definition at line 1506 of file SDL_audiocvt.c.

 {
     /* !!! FIXME: several converters can take advantage of SIMD, but only
        !!! FIXME:  if the data is aligned to 16 bytes. EnsureStreamBufferSize()
        !!! FIXME:  guarantees the buffer will align, but the
        !!! FIXME:  converters will iterate over the data backwards if
        !!! FIXME:  the output grows, and this means we won't align if buflen
        !!! FIXME:  isn't a multiple of 16. In these cases, we should chop off
        !!! FIXME:  a few samples at the end and convert them separately. */
  
     #if DEBUG_AUDIOSTREAM
     printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
     #endif
  
     if (!stream) {
         return SDL_InvalidParamError("stream");
     } else if (!buf) {
         return SDL_InvalidParamError("buf");
     } else if (len == 0) {
         return 0;  /* nothing to do. */
     } else if ((len % stream->src_sample_frame_size) != 0) {
         return SDL_SetError("Can't add partial sample frames");
     }
  
     if (!stream->cvt_before_resampling.needed &&
         (stream->dst_rate == stream->src_rate) &&
         !stream->cvt_after_resampling.needed) {
         #if DEBUG_AUDIOSTREAM
         printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", len);
         #endif
         return SDL_WriteToDataQueue(stream->queue, buf, len);
     }
  
     while (len > 0) {
         int amount;
  
         /* If we don't have a staging buffer or we're given enough data that
            we don't need to store it for later, skip the staging process.
          */
         if (!stream->staging_buffer_filled && len >= stream->staging_buffer_size) {
             return SDL_AudioStreamPutInternal(stream, buf, len, NULL);
         }
  
         /* If there's not enough data to fill the staging buffer, just save it */
         if ((stream->staging_buffer_filled + len) < stream->staging_buffer_size) {
             SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, len);
             stream->staging_buffer_filled += len;
             return 0;
         }
  
         /* Fill the staging buffer, process it, and continue */
         amount = (stream->staging_buffer_size - stream->staging_buffer_filled);
         SDL_assert(amount > 0);
         SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, amount);
         stream->staging_buffer_filled = 0;
         if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, NULL) < 0) {
             return -1;
         }
         buf = (void *)((Uint8 *)buf + amount);
         len -= amount;
     }
     return 0;
 }

References NULL, SDL_assert, SDL_AudioStreamPutInternal(), SDL_InvalidParamError, SDL_memcpy, SDL_SetError, and SDL_WriteToDataQueue().

◆ SDL_BuildAudioCVT()

int SDL_BuildAudioCVT	(	SDL_AudioCVT *	cvt,
		SDL_AudioFormat	src_format,
		Uint8	src_channels,
		int	src_rate,
		SDL_AudioFormat	dst_format,
		Uint8	dst_channels,
		int	dst_rate
	)

This function takes a source format and rate and a destination format and rate, and initializes the cvt structure with information needed by SDL_ConvertAudio() to convert a buffer of audio data from one format to the other. An unsupported format causes an error and -1 will be returned.

Returns: 0 if no conversion is needed, 1 if the audio filter is set up, or -1 on error.

Definition at line 877 of file SDL_audiocvt.c.

 {
     /* Sanity check target pointer */
     if (cvt == NULL) {
         return SDL_InvalidParamError("cvt");
     }
  
     /* Make sure we zero out the audio conversion before error checking */
     SDL_zerop(cvt);
  
     if (!SDL_SupportedAudioFormat(src_fmt)) {
         return SDL_SetError("Invalid source format");
     } else if (!SDL_SupportedAudioFormat(dst_fmt)) {
         return SDL_SetError("Invalid destination format");
     } else if (!SDL_SupportedChannelCount(src_channels)) {
         return SDL_SetError("Invalid source channels");
     } else if (!SDL_SupportedChannelCount(dst_channels)) {
         return SDL_SetError("Invalid destination channels");
     } else if (src_rate <= 0) {
         return SDL_SetError("Source rate is equal to or less than zero");
     } else if (dst_rate <= 0) {
         return SDL_SetError("Destination rate is equal to or less than zero");
     } else if (src_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
         return SDL_SetError("Source rate is too high");
     } else if (dst_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
         return SDL_SetError("Destination rate is too high");
     }
  
 #if DEBUG_CONVERT
     printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
            src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
 #endif
  
     /* Start off with no conversion necessary */
     cvt->src_format = src_fmt;
     cvt->dst_format = dst_fmt;
     cvt->needed = 0;
     cvt->filter_index = 0;
     SDL_zeroa(cvt->filters);
     cvt->len_mult = 1;
     cvt->len_ratio = 1.0;
     cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
  
     /* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
     SDL_ChooseAudioConverters();
  
     /* Type conversion goes like this now:
         - byteswap to CPU native format first if necessary.
         - convert to native Float32 if necessary.
         - resample and change channel count if necessary.
         - convert back to native format.
         - byteswap back to foreign format if necessary.
  
        The expectation is we can process data faster in float32
        (possibly with SIMD), and making several passes over the same
        buffer is likely to be CPU cache-friendly, avoiding the
        biggest performance hit in modern times. Previously we had
        (script-generated) custom converters for every data type and
        it was a bloat on SDL compile times and final library size. */
  
     /* see if we can skip float conversion entirely. */
     if (src_rate == dst_rate && src_channels == dst_channels) {
         if (src_fmt == dst_fmt) {
             return 0;
         }
  
         /* just a byteswap needed? */
         if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_Convert_Byteswap) < 0) {
                 return -1;
             }
             cvt->needed = 1;
             return 1;
         }
     }
  
     /* Convert data types, if necessary. Updates (cvt). */
     if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
         return -1;              /* shouldn't happen, but just in case... */
     }
  
     /* Channel conversion */
     if (src_channels < dst_channels) {
         /* Upmixing */
         /* Mono -> Stereo [-> ...] */
         if ((src_channels == 1) && (dst_channels > 1)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertMonoToStereo) < 0) {
                 return -1;
             }
             cvt->len_mult *= 2;
             src_channels = 2;
             cvt->len_ratio *= 2;
         }
         /* [Mono ->] Stereo -> 5.1 [-> 7.1] */
         if ((src_channels == 2) && (dst_channels >= 6)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoTo51) < 0) {
                 return -1;
             }
             src_channels = 6;
             cvt->len_mult *= 3;
             cvt->len_ratio *= 3;
         }
         /* Quad -> 5.1 [-> 7.1] */
         if ((src_channels == 4) && (dst_channels >= 6)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadTo51) < 0) {
                 return -1;
             }
             src_channels = 6;
             cvt->len_mult = (cvt->len_mult * 3 + 1) / 2;
             cvt->len_ratio *= 1.5;
         }
         /* [[Mono ->] Stereo ->] 5.1 -> 7.1 */
         if ((src_channels == 6) && (dst_channels == 8)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51To71) < 0) {
                 return -1;
             }
             src_channels = 8;
             cvt->len_mult = (cvt->len_mult * 4 + 2) / 3;
             /* Should be numerically exact with every valid input to this
                function */
             cvt->len_ratio = cvt->len_ratio * 4 / 3;
         }
         /* [Mono ->] Stereo -> Quad */
         if ((src_channels == 2) && (dst_channels == 4)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertStereoToQuad) < 0) {
                 return -1;
             }
             src_channels = 4;
             cvt->len_mult *= 2;
             cvt->len_ratio *= 2;
         }
     } else if (src_channels > dst_channels) {
         /* Downmixing */
         /* 7.1 -> 5.1 [-> Stereo [-> Mono]] */
         /* 7.1 -> 5.1 [-> Quad] */
         if ((src_channels == 8) && (dst_channels <= 6)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_Convert71To51) < 0) {
                 return -1;
             }
             src_channels = 6;
             cvt->len_ratio *= 0.75;
         }
         /* [7.1 ->] 5.1 -> Stereo [-> Mono] */
         if ((src_channels == 6) && (dst_channels <= 2)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToStereo) < 0) {
                 return -1;
             }
             src_channels = 2;
             cvt->len_ratio /= 3;
         }
         /* 5.1 -> Quad */
         if ((src_channels == 6) && (dst_channels == 4)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_Convert51ToQuad) < 0) {
                 return -1;
             }
             src_channels = 4;
             cvt->len_ratio = cvt->len_ratio * 2 / 3;
         }
         /* Quad -> Stereo [-> Mono] */
         if ((src_channels == 4) && (dst_channels <= 2)) {
             if (SDL_AddAudioCVTFilter(cvt, SDL_ConvertQuadToStereo) < 0) {
                 return -1;
             }
             src_channels = 2;
             cvt->len_ratio /= 2;
         }
         /* [... ->] Stereo -> Mono */
         if ((src_channels == 2) && (dst_channels == 1)) {
             SDL_AudioFilter filter = NULL;
  
             #if HAVE_SSE3_INTRINSICS
             if (SDL_HasSSE3()) {
                 filter = SDL_ConvertStereoToMono_SSE3;
             }
             #endif
  
             if (!filter) {
                 filter = SDL_ConvertStereoToMono;
             }
  
             if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
                 return -1;
             }
  
             src_channels = 1;
             cvt->len_ratio /= 2;
         }
     }
  
     if (src_channels != dst_channels) {
         /* All combinations of supported channel counts should have been
            handled by now, but let's be defensive */
       return SDL_SetError("Invalid channel combination");
     }
     
     /* Do rate conversion, if necessary. Updates (cvt). */
     if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
         return -1;              /* shouldn't happen, but just in case... */
     }
  
     /* Move to final data type. */
     if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
         return -1;              /* shouldn't happen, but just in case... */
     }
  
     cvt->needed = (cvt->filter_index != 0);
     return (cvt->needed);
 }

References SDL_AudioCVT::dst_format, SDL_AudioCVT::filter_index, SDL_AudioCVT::filters, SDL_AudioCVT::len_mult, SDL_AudioCVT::len_ratio, SDL_AudioCVT::needed, NULL, SDL_AudioCVT::rate_incr, RESAMPLER_SAMPLES_PER_ZERO_CROSSING, SDL_AddAudioCVTFilter(), SDL_AUDIO_MASK_ENDIAN, SDL_BuildAudioResampleCVT(), SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), SDL_ChooseAudioConverters(), SDL_Convert51To71(), SDL_Convert51ToQuad(), SDL_Convert51ToStereo(), SDL_Convert71To51(), SDL_Convert_Byteswap(), SDL_ConvertMonoToStereo(), SDL_ConvertQuadTo51(), SDL_ConvertQuadToStereo(), SDL_ConvertStereoTo51(), SDL_ConvertStereoToMono(), SDL_ConvertStereoToQuad(), SDL_HasSSE3, SDL_InvalidParamError, SDL_MAX_SINT32, SDL_SetError, SDL_SupportedAudioFormat(), SDL_SupportedChannelCount(), SDL_zeroa, SDL_zerop, and SDL_AudioCVT::src_format.

Referenced by SDL_NewAudioStream().

◆ SDL_ClearQueuedAudio()

void SDL_ClearQueuedAudio ( SDL_AudioDeviceID dev )

Drop any queued audio data. For playback devices, this is any queued data still waiting to be submitted to the hardware. For capture devices, this is any data that was queued by the device that hasn't yet been dequeued by the application.

Immediately after this call, SDL_GetQueuedAudioSize() will return 0. For playback devices, the hardware will start playing silence if more audio isn't queued. Unpaused capture devices will start filling the queue again as soon as they have more data available (which, depending on the state of the hardware and the thread, could be before this function call returns!).

This will not prevent playback of queued audio that's already been sent to the hardware, as we can not undo that, so expect there to be some fraction of a second of audio that might still be heard. This can be useful if you want to, say, drop any pending music during a level change in your game.

You may not queue audio on a device that is using an application-supplied callback; calling this function on such a device is always a no-op. You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use the audio callback, but not both.

You should not call SDL_LockAudio() on the device before clearing the queue; SDL handles locking internally for this function.

This function always succeeds and thus returns void.

Parameters

dev	The device ID of which to clear the audio queue.

See also: SDL_QueueAudio; SDL_GetQueuedAudioSize

Definition at line 665 of file SDL_audio.c.

 {
     SDL_AudioDevice *device = get_audio_device(devid);
  
     if (!device) {
         return;  /* nothing to do. */
     }
  
     /* Blank out the device and release the mutex. Free it afterwards. */
     current_audio.impl.LockDevice(device);
  
     /* Keep up to two packets in the pool to reduce future malloc pressure. */
     SDL_ClearDataQueue(device->buffer_queue, SDL_AUDIOBUFFERQUEUE_PACKETLEN * 2);
  
     current_audio.impl.UnlockDevice(device);
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDriverImpl::LockDevice, SDL_AUDIOBUFFERQUEUE_PACKETLEN, SDL_ClearDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

◆ SDL_CloseAudio()

void SDL_CloseAudio ( void )

This function shuts down audio processing and closes the audio device.

Definition at line 127 of file SDL_dynapi_procs.h.

References SDL_CloseAudioDevice().

◆ SDL_CloseAudioDevice()

void SDL_CloseAudioDevice ( SDL_AudioDeviceID dev )

Definition at line 1581 of file SDL_audio.c.

 {
     close_audio_device(get_audio_device(devid));
 }

References close_audio_device(), and get_audio_device().

Referenced by SDL_CloseAudio().

◆ SDL_ConvertAudio()

int SDL_ConvertAudio ( SDL_AudioCVT * cvt )

Once you have initialized the cvt structure using SDL_BuildAudioCVT(), created an audio buffer cvt->buf, and filled it with cvt->len bytes of audio data in the source format, this function will convert it in-place to the desired format.

The data conversion may expand the size of the audio data, so the buffer cvt->buf should be allocated after the cvt structure is initialized by SDL_BuildAudioCVT(), and should be cvt->len*cvt->len_mult bytes long.

Returns: 0 on success or -1 if cvt->buf is NULL.

Definition at line 539 of file SDL_audiocvt.c.

 {
     /* !!! FIXME: (cvt) should be const; stack-copy it here. */
     /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
  
     /* Make sure there's data to convert */
     if (cvt->buf == NULL) {
         return SDL_SetError("No buffer allocated for conversion");
     }
  
     /* Return okay if no conversion is necessary */
     cvt->len_cvt = cvt->len;
     if (cvt->filters[0] == NULL) {
         return 0;
     }
  
     /* Set up the conversion and go! */
     cvt->filter_index = 0;
     cvt->filters[0] (cvt, cvt->src_format);
     return 0;
 }

References SDL_AudioCVT::buf, SDL_AudioCVT::filter_index, SDL_AudioCVT::filters, SDL_AudioCVT::len, SDL_AudioCVT::len_cvt, NULL, SDL_SetError, and SDL_AudioCVT::src_format.

Referenced by SDL_AudioStreamPutInternal().

◆ SDL_DequeueAudio()

Uint32 SDL_DequeueAudio	(	SDL_AudioDeviceID	dev,
		void *	data,
		Uint32	len
	)

Dequeue more audio on non-callback devices.

(If you are looking to queue audio for output on a non-callback playback device, you want SDL_QueueAudio() instead. This will always return 0 if you use it with playback devices.)

SDL offers two ways to retrieve audio from a capture device: you can either supply a callback that SDL triggers with some frequency as the device records more audio data, (push method), or you can supply no callback, and then SDL will expect you to retrieve data at regular intervals (pull method) with this function.

There are no limits on the amount of data you can queue, short of exhaustion of address space. Data from the device will keep queuing as necessary without further intervention from you. This means you will eventually run out of memory if you aren't routinely dequeueing data.

Capture devices will not queue data when paused; if you are expecting to not need captured audio for some length of time, use SDL_PauseAudioDevice() to stop the capture device from queueing more data. This can be useful during, say, level loading times. When unpaused, capture devices will start queueing data from that point, having flushed any capturable data available while paused.

This function is thread-safe, but dequeueing from the same device from two threads at once does not promise which thread will dequeued data first.

You may not dequeue audio from a device that is using an application-supplied callback; doing so returns an error. You have to use the audio callback, or dequeue audio with this function, but not both.

You should not call SDL_LockAudio() on the device before queueing; SDL handles locking internally for this function.

Parameters

dev	The device ID from which we will dequeue audio.
data	A pointer into where audio data should be copied.
len	The number of bytes (not samples!) to which (data) points.

Returns: number of bytes dequeued, which could be less than requested.

See also: SDL_GetQueuedAudioSize; SDL_ClearQueuedAudio

Definition at line 624 of file SDL_audio.c.

 {
     SDL_AudioDevice *device = get_audio_device(devid);
     Uint32 rc;
  
     if ( (len == 0) ||  /* nothing to do? */
          (!device) ||  /* called with bogus device id */
          (!device->iscapture) ||  /* playback devices can't dequeue */
          (device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
         return 0;  /* just report zero bytes dequeued. */
     }
  
     current_audio.impl.LockDevice(device);
     rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
     current_audio.impl.UnlockDevice(device);
     return rc;
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDriverImpl::LockDevice, SDL_BufferQueueFillCallback(), SDL_ReadFromDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

◆ SDL_FreeAudioStream()

void SDL_FreeAudioStream ( SDL_AudioStream * stream )

Free an audio stream

See also: SDL_NewAudioStream; SDL_AudioStreamPut; SDL_AudioStreamGet; SDL_AudioStreamAvailable; SDL_AudioStreamFlush; SDL_AudioStreamClear

Definition at line 1667 of file SDL_audiocvt.c.

 {
     if (stream) {
         if (stream->cleanup_resampler_func) {
             stream->cleanup_resampler_func(stream);
         }
         SDL_FreeDataQueue(stream->queue);
         SDL_free(stream->staging_buffer);
         SDL_free(stream->work_buffer_base);
         SDL_free(stream->resampler_padding);
         SDL_free(stream);
     }
 }

References SDL_free, and SDL_FreeDataQueue().

Referenced by SDL_NewAudioStream().

◆ SDL_FreeWAV()

void SDL_FreeWAV ( Uint8 * audio_buf )

This function frees data previously allocated with SDL_LoadWAV_RW()

Definition at line 2150 of file SDL_wave.c.

 {
     SDL_free(audio_buf);
 }

References SDL_free.

◆ SDL_GetAudioDeviceName()

const char* SDL_GetAudioDeviceName	(	int	index,
		int	iscapture
	)

Get the human-readable name of a specific audio device. Must be a value between 0 and (number of audio devices-1). Only valid after a successfully initializing the audio subsystem. The values returned by this function reflect the latest call to SDL_GetNumAudioDevices(); recall that function to redetect available hardware.

The string returned by this function is UTF-8 encoded, read-only, and managed internally. You are not to free it. If you need to keep the string for any length of time, you should make your own copy of it, as it will be invalid next time any of several other SDL functions is called.

Definition at line 1073 of file SDL_audio.c.

 {
     const char *retval = NULL;
  
     if (!SDL_WasInit(SDL_INIT_AUDIO)) {
         SDL_SetError("Audio subsystem is not initialized");
         return NULL;
     }
  
     if (iscapture && !current_audio.impl.HasCaptureSupport) {
         SDL_SetError("No capture support");
         return NULL;
     }
  
     if (index >= 0) {
         SDL_AudioDeviceItem *item;
         int i;
  
         SDL_LockMutex(current_audio.detectionLock);
         item = iscapture ? current_audio.inputDevices : current_audio.outputDevices;
         i = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
         if (index < i) {
             for (i--; i > index; i--, item = item->next) {
                 SDL_assert(item != NULL);
             }
             SDL_assert(item != NULL);
             retval = item->name;
         }
         SDL_UnlockMutex(current_audio.detectionLock);
     }
  
     if (retval == NULL) {
         SDL_SetError("No such device");
     }
  
     return retval;
 }

References current_audio, SDL_AudioDriver::detectionLock, SDL_AudioDriverImpl::HasCaptureSupport, i, SDL_AudioDriver::impl, SDL_AudioDriver::inputDeviceCount, SDL_AudioDriver::inputDevices, SDL_AudioDeviceItem::next, NULL, SDL_AudioDriver::outputDeviceCount, SDL_AudioDriver::outputDevices, retval, SDL_assert, SDL_INIT_AUDIO, SDL_LockMutex, SDL_SetError, SDL_UnlockMutex, and SDL_WasInit.

◆ SDL_GetAudioDeviceStatus()

SDL_AudioStatus SDL_GetAudioDeviceStatus ( SDL_AudioDeviceID dev )

Definition at line 1509 of file SDL_audio.c.

 {
     SDL_AudioDevice *device = get_audio_device(devid);
     SDL_AudioStatus status = SDL_AUDIO_STOPPED;
     if (device && SDL_AtomicGet(&device->enabled)) {
         if (SDL_AtomicGet(&device->paused)) {
             status = SDL_AUDIO_PAUSED;
         } else {
             status = SDL_AUDIO_PLAYING;
         }
     }
     return status;
 }

References device, get_audio_device(), SDL_AtomicGet, SDL_AUDIO_PAUSED, SDL_AUDIO_PLAYING, and SDL_AUDIO_STOPPED.

Referenced by SDL_GetAudioStatus().

◆ SDL_GetAudioDriver()

const char* SDL_GetAudioDriver ( int index )

Definition at line 934 of file SDL_audio.c.

 {
     if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
         return bootstrap[index]->name;
     }
     return NULL;
 }

References bootstrap, AudioBootStrap::name, NULL, and SDL_GetNumAudioDrivers().

◆ SDL_GetAudioStatus()

SDL_AudioStatus SDL_GetAudioStatus ( void )

Definition at line 113 of file SDL_dynapi_procs.h.

References SDL_GetAudioDeviceStatus().

◆ SDL_GetCurrentAudioDriver()

const char* SDL_GetCurrentAudioDriver ( void )

This function returns the name of the current audio driver, or NULL if no driver has been initialized.

Definition at line 1008 of file SDL_audio.c.

 {
     return current_audio.name;
 }

References current_audio, and SDL_AudioDriver::name.

◆ SDL_GetNumAudioDevices()

int SDL_GetNumAudioDevices ( int iscapture )

Get the number of available devices exposed by the current driver. Only valid after a successfully initializing the audio subsystem. Returns -1 if an explicit list of devices can't be determined; this is not an error. For example, if SDL is set up to talk to a remote audio server, it can't list every one available on the Internet, but it will still allow a specific host to be specified to SDL_OpenAudioDevice().

In many common cases, when this function returns a value <= 0, it can still successfully open the default device (NULL for first argument of SDL_OpenAudioDevice()).

Definition at line 1048 of file SDL_audio.c.

 {
     int retval = 0;
  
     if (!SDL_WasInit(SDL_INIT_AUDIO)) {
         return -1;
     }
  
     SDL_LockMutex(current_audio.detectionLock);
     if (iscapture && current_audio.captureDevicesRemoved) {
         clean_out_device_list(&current_audio.inputDevices, &current_audio.inputDeviceCount, &current_audio.captureDevicesRemoved);
     }
  
     if (!iscapture && current_audio.outputDevicesRemoved) {
         clean_out_device_list(&current_audio.outputDevices, &current_audio.outputDeviceCount, &current_audio.outputDevicesRemoved);
     }
  
     retval = iscapture ? current_audio.inputDeviceCount : current_audio.outputDeviceCount;
     SDL_UnlockMutex(current_audio.detectionLock);
  
     return retval;
 }

References SDL_AudioDriver::captureDevicesRemoved, clean_out_device_list(), current_audio, SDL_AudioDriver::detectionLock, SDL_AudioDriver::inputDeviceCount, SDL_AudioDriver::inputDevices, SDL_AudioDriver::outputDeviceCount, SDL_AudioDriver::outputDevices, SDL_AudioDriver::outputDevicesRemoved, retval, SDL_INIT_AUDIO, SDL_LockMutex, SDL_UnlockMutex, and SDL_WasInit.

◆ SDL_GetNumAudioDrivers()

int SDL_GetNumAudioDrivers ( void )

Definition at line 928 of file SDL_audio.c.

 {
     return SDL_arraysize(bootstrap) - 1;
 }

References bootstrap, and SDL_arraysize.

Referenced by SDL_GetAudioDriver().

◆ SDL_GetQueuedAudioSize()

Uint32 SDL_GetQueuedAudioSize ( SDL_AudioDeviceID dev )

Get the number of bytes of still-queued audio.

For playback device:

This is the number of bytes that have been queued for playback with SDL_QueueAudio(), but have not yet been sent to the hardware. This number may shrink at any time, so this only informs of pending data.

Once we've sent it to the hardware, this function can not decide the exact byte boundary of what has been played. It's possible that we just gave the hardware several kilobytes right before you called this function, but it hasn't played any of it yet, or maybe half of it, etc.

For capture devices:

This is the number of bytes that have been captured by the device and are waiting for you to dequeue. This number may grow at any time, so this only informs of the lower-bound of available data.

You may not queue audio on a device that is using an application-supplied callback; calling this function on such a device always returns 0. You have to queue audio with SDL_QueueAudio()/SDL_DequeueAudio(), or use the audio callback, but not both.

You should not call SDL_LockAudio() on the device before querying; SDL handles locking internally for this function.

Parameters

dev	The device ID of which we will query queued audio size.

Returns: Number of bytes (not samples!) of queued audio.

See also: SDL_QueueAudio; SDL_ClearQueuedAudio

Definition at line 643 of file SDL_audio.c.

 {
     Uint32 retval = 0;
     SDL_AudioDevice *device = get_audio_device(devid);
  
     if (!device) {
         return 0;
     }
  
     /* Nothing to do unless we're set up for queueing. */
     if (device->callbackspec.callback == SDL_BufferQueueDrainCallback ||
         device->callbackspec.callback == SDL_BufferQueueFillCallback)
     {
         current_audio.impl.LockDevice(device);
         retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
         current_audio.impl.UnlockDevice(device);
     }
  
     return retval;
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDriverImpl::LockDevice, retval, SDL_BufferQueueDrainCallback(), SDL_BufferQueueFillCallback(), SDL_CountDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

◆ SDL_LoadWAV_RW()

SDL_AudioSpec* SDL_LoadWAV_RW	(	SDL_RWops *	src,
		int	freesrc,
		SDL_AudioSpec *	spec,
		Uint8 **	audio_buf,
		Uint32 *	audio_len
	)

Load the audio data of a WAVE file into memory.

Loading a WAVE file requires src, spec, audio_buf and audio_len to be valid pointers. The entire data portion of the file is then loaded into memory and decoded if necessary.

If freesrc is non-zero, the data source gets automatically closed and freed before the function returns.

Supported are RIFF WAVE files with the formats PCM (8, 16, 24, and 32 bits), IEEE Float (32 bits), Microsoft ADPCM and IMA ADPCM (4 bits), and A-law and µ-law (8 bits). Other formats are currently unsupported and cause an error.

If this function succeeds, the pointer returned by it is equal to spec and the pointer to the audio data allocated by the function is written to audio_buf and its length in bytes to audio_len. The SDL_AudioSpec members freq, channels, and format are set to the values of the audio data in the buffer. The samples member is set to a sane default and all others are set to zero.

It's necessary to use SDL_FreeWAV() to free the audio data returned in audio_buf when it is no longer used.

Because of the underspecification of the Waveform format, there are many problematic files in the wild that cause issues with strict decoders. To provide compatibility with these files, this decoder is lenient in regards to the truncation of the file, the fact chunk, and the size of the RIFF chunk. The hints SDL_HINT_WAVE_RIFF_CHUNK_SIZE, SDL_HINT_WAVE_TRUNCATION, and SDL_HINT_WAVE_FACT_CHUNK can be used to tune the behavior of the loading process.

Any file that is invalid (due to truncation, corruption, or wrong values in the headers), too big, or unsupported causes an error. Additionally, any critical I/O error from the data source will terminate the loading process with an error. The function returns NULL on error and in all cases (with the exception of src being NULL), an appropriate error message will be set.

It is required that the data source supports seeking.

Example:

SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);

SDL_RWFromFile

#define SDL_RWFromFile

Definition: SDL_dynapi_overrides.h:351

SDL_LoadWAV_RW

#define SDL_LoadWAV_RW

Definition: SDL_dynapi_overrides.h:86

Parameters

src	The data source with the WAVE data
freesrc	A integer value that makes the function close the data source if non-zero
spec	A pointer filled with the audio format of the audio data
audio_buf	A pointer filled with the audio data allocated by the function
audio_len	A pointer filled with the length of the audio data buffer in bytes

Returns: NULL on error, or non-NULL on success.

Definition at line 2097 of file SDL_wave.c.

 {
     int result;
     WaveFile file;
  
     SDL_zero(file);
  
     /* Make sure we are passed a valid data source */
     if (src == NULL) {
         /* Error may come from RWops. */
         return NULL;
     } else if (spec == NULL) {
         SDL_InvalidParamError("spec");
         return NULL;
     } else if (audio_buf == NULL) {
         SDL_InvalidParamError("audio_buf");
         return NULL;
     } else if (audio_len == NULL) {
         SDL_InvalidParamError("audio_len");
         return NULL;
     }
  
     *audio_buf = NULL;
     *audio_len = 0;
  
     file.riffhint = WaveGetRiffSizeHint();
     file.trunchint = WaveGetTruncationHint();
     file.facthint = WaveGetFactChunkHint();
  
     result = WaveLoad(src, &file, spec, audio_buf, audio_len);
     if (result < 0) {
         SDL_free(*audio_buf);
         spec = NULL;
         audio_buf = NULL;
         audio_len = 0;
     }
  
     /* Cleanup */
     if (freesrc) {
         SDL_RWclose(src);
     } else {
         SDL_RWseek(src, file.chunk.position, RW_SEEK_SET);
     }
     WaveFreeChunkData(&file.chunk);
     SDL_free(file.decoderdata);
  
     return spec;
 }

References WaveFile::chunk, WaveFile::decoderdata, WaveFile::facthint, NULL, WaveChunk::position, WaveFile::riffhint, RW_SEEK_SET, SDL_free, SDL_InvalidParamError, SDL_RWclose, SDL_RWseek, SDL_zero, spec, WaveFile::trunchint, WaveFreeChunkData(), WaveGetFactChunkHint(), WaveGetRiffSizeHint(), WaveGetTruncationHint(), and WaveLoad().

◆ SDL_LockAudio()

void SDL_LockAudio ( void )

Definition at line 123 of file SDL_dynapi_procs.h.

References SDL_LockAudioDevice().

◆ SDL_LockAudioDevice()

void SDL_LockAudioDevice ( SDL_AudioDeviceID dev )

Definition at line 1549 of file SDL_audio.c.

 {
     /* Obtain a lock on the mixing buffers */
     SDL_AudioDevice *device = get_audio_device(devid);
     if (device) {
         current_audio.impl.LockDevice(device);
     }
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, and SDL_AudioDriverImpl::LockDevice.

Referenced by SDL_LockAudio().

◆ SDL_MixAudio()

void SDL_MixAudio	(	Uint8 *	dst,
		const Uint8 *	src,
		Uint32	len,
		int	volume
	)

This takes two audio buffers of the playing audio format and mixes them, performing addition, volume adjustment, and overflow clipping. The volume ranges from 0 - 128, and should be set to SDL_MIX_MAXVOLUME for full audio volume. Note this does not change hardware volume. This is provided for convenience – you can mix your own audio data.

Definition at line 1703 of file SDL_audio.c.

 {
     /* Mix the user-level audio format */
     SDL_AudioDevice *device = get_audio_device(1);
     if (device != NULL) {
         SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
     }
 }

References device, get_audio_device(), NULL, and SDL_MixAudioFormat.

◆ SDL_MixAudioFormat()

void SDL_MixAudioFormat	(	Uint8 *	dst,
		const Uint8 *	src,
		SDL_AudioFormat	format,
		Uint32	len,
		int	volume
	)

This works like SDL_MixAudio(), but you specify the audio format instead of using the format of audio device 1. Thus it can be used when no audio device is open at all.

Definition at line 90 of file SDL_mixer.c.

 {
     if (volume == 0) {
         return;
     }
  
     switch (format) {
  
     case AUDIO_U8:
         {
 #if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
             SDL_MixAudio_m68k_U8((char *) dst, (char *) src,
                                  (unsigned long) len, (long) volume,
                                  (char *) mix8);
 #else
             Uint8 src_sample;
  
             while (len--) {
                 src_sample = *src;
                 ADJUST_VOLUME_U8(src_sample, volume);
                 *dst = mix8[*dst + src_sample];
                 ++dst;
                 ++src;
             }
 #endif
         }
         break;
  
     case AUDIO_S8:
         {
             Sint8 *dst8, *src8;
             Sint8 src_sample;
             int dst_sample;
             const int max_audioval = ((1 << (8 - 1)) - 1);
             const int min_audioval = -(1 << (8 - 1));
  
             src8 = (Sint8 *) src;
             dst8 = (Sint8 *) dst;
             while (len--) {
                 src_sample = *src8;
                 ADJUST_VOLUME(src_sample, volume);
                 dst_sample = *dst8 + src_sample;
                 if (dst_sample > max_audioval) {
                     *dst8 = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     *dst8 = min_audioval;
                 } else {
                     *dst8 = dst_sample;
                 }
                 ++dst8;
                 ++src8;
             }
         }
         break;
  
     case AUDIO_S16LSB:
         {
             Sint16 src1, src2;
             int dst_sample;
             const int max_audioval = ((1 << (16 - 1)) - 1);
             const int min_audioval = -(1 << (16 - 1));
  
             len /= 2;
             while (len--) {
                 src1 = ((src[1]) << 8 | src[0]);
                 ADJUST_VOLUME(src1, volume);
                 src2 = ((dst[1]) << 8 | dst[0]);
                 src += 2;
                 dst_sample = src1 + src2;
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     dst_sample = min_audioval;
                 }
                 dst[0] = dst_sample & 0xFF;
                 dst_sample >>= 8;
                 dst[1] = dst_sample & 0xFF;
                 dst += 2;
             }
         }
         break;
  
     case AUDIO_S16MSB:
         {
 #if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
             SDL_MixAudio_m68k_S16MSB((short *) dst, (short *) src,
                                      (unsigned long) len, (long) volume);
 #else
             Sint16 src1, src2;
             int dst_sample;
             const int max_audioval = ((1 << (16 - 1)) - 1);
             const int min_audioval = -(1 << (16 - 1));
  
             len /= 2;
             while (len--) {
                 src1 = ((src[0]) << 8 | src[1]);
                 ADJUST_VOLUME(src1, volume);
                 src2 = ((dst[0]) << 8 | dst[1]);
                 src += 2;
                 dst_sample = src1 + src2;
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     dst_sample = min_audioval;
                 }
                 dst[1] = dst_sample & 0xFF;
                 dst_sample >>= 8;
                 dst[0] = dst_sample & 0xFF;
                 dst += 2;
             }
 #endif
         }
         break;
  
     case AUDIO_U16LSB:
         {
             Uint16 src1, src2;
             int dst_sample;
             const int max_audioval = 0xFFFF;
  
             len /= 2;
             while (len--) {
                 src1 = ((src[1]) << 8 | src[0]);
                 ADJUST_VOLUME(src1, volume);
                 src2 = ((dst[1]) << 8 | dst[0]);
                 src += 2;
                 dst_sample = src1 + src2;
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 }
                 dst[0] = dst_sample & 0xFF;
                 dst_sample >>= 8;
                 dst[1] = dst_sample & 0xFF;
                 dst += 2;
             }
         }
         break;
  
     case AUDIO_U16MSB:
         {
             Uint16 src1, src2;
             int dst_sample;
             const int max_audioval = 0xFFFF;
  
             len /= 2;
             while (len--) {
                 src1 = ((src[0]) << 8 | src[1]);
                 ADJUST_VOLUME(src1, volume);
                 src2 = ((dst[0]) << 8 | dst[1]);
                 src += 2;
                 dst_sample = src1 + src2;
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 }
                 dst[1] = dst_sample & 0xFF;
                 dst_sample >>= 8;
                 dst[0] = dst_sample & 0xFF;
                 dst += 2;
             }
         }
         break;
  
     case AUDIO_S32LSB:
         {
             const Uint32 *src32 = (Uint32 *) src;
             Uint32 *dst32 = (Uint32 *) dst;
             Sint64 src1, src2;
             Sint64 dst_sample;
             const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
             const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
  
             len /= 4;
             while (len--) {
                 src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
                 src32++;
                 ADJUST_VOLUME(src1, volume);
                 src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
                 dst_sample = src1 + src2;
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     dst_sample = min_audioval;
                 }
                 *(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
             }
         }
         break;
  
     case AUDIO_S32MSB:
         {
             const Uint32 *src32 = (Uint32 *) src;
             Uint32 *dst32 = (Uint32 *) dst;
             Sint64 src1, src2;
             Sint64 dst_sample;
             const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
             const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
  
             len /= 4;
             while (len--) {
                 src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
                 src32++;
                 ADJUST_VOLUME(src1, volume);
                 src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
                 dst_sample = src1 + src2;
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     dst_sample = min_audioval;
                 }
                 *(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
             }
         }
         break;
  
     case AUDIO_F32LSB:
         {
             const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
             const float fvolume = (float) volume;
             const float *src32 = (float *) src;
             float *dst32 = (float *) dst;
             float src1, src2;
             double dst_sample;
             /* !!! FIXME: are these right? */
             const double max_audioval = 3.402823466e+38F;
             const double min_audioval = -3.402823466e+38F;
  
             len /= 4;
             while (len--) {
                 src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
                 src2 = SDL_SwapFloatLE(*dst32);
                 src32++;
  
                 dst_sample = ((double) src1) + ((double) src2);
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     dst_sample = min_audioval;
                 }
                 *(dst32++) = SDL_SwapFloatLE((float) dst_sample);
             }
         }
         break;
  
     case AUDIO_F32MSB:
         {
             const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
             const float fvolume = (float) volume;
             const float *src32 = (float *) src;
             float *dst32 = (float *) dst;
             float src1, src2;
             double dst_sample;
             /* !!! FIXME: are these right? */
             const double max_audioval = 3.402823466e+38F;
             const double min_audioval = -3.402823466e+38F;
  
             len /= 4;
             while (len--) {
                 src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
                 src2 = SDL_SwapFloatBE(*dst32);
                 src32++;
  
                 dst_sample = ((double) src1) + ((double) src2);
                 if (dst_sample > max_audioval) {
                     dst_sample = max_audioval;
                 } else if (dst_sample < min_audioval) {
                     dst_sample = min_audioval;
                 }
                 *(dst32++) = SDL_SwapFloatBE((float) dst_sample);
             }
         }
         break;
  
     default:                   /* If this happens... FIXME! */
         SDL_SetError("SDL_MixAudioFormat(): unknown audio format");
         return;
     }
 }

References ADJUST_VOLUME, ADJUST_VOLUME_U8, AUDIO_F32LSB, AUDIO_F32MSB, AUDIO_S16LSB, AUDIO_S16MSB, AUDIO_S32LSB, AUDIO_S32MSB, AUDIO_S8, AUDIO_U16LSB, AUDIO_U16MSB, AUDIO_U8, mix8, SDL_MIX_MAXVOLUME, SDL_SetError, SDL_SwapBE32, SDL_SwapFloatBE, SDL_SwapFloatLE, and SDL_SwapLE32.

◆ SDL_NewAudioStream()

SDL_AudioStream* SDL_NewAudioStream	(	const SDL_AudioFormat	src_format,
		const Uint8	src_channels,
		const int	src_rate,
		const SDL_AudioFormat	dst_format,
		const Uint8	dst_channels,
		const int	dst_rate
	)

Create a new audio stream

Parameters

src_format	The format of the source audio
src_channels	The number of channels of the source audio
src_rate	The sampling rate of the source audio
dst_format	The format of the desired audio output
dst_channels	The number of channels of the desired audio output
dst_rate	The sampling rate of the desired audio output

Returns: 0 on success, or -1 on error.

See also: SDL_AudioStreamPut; SDL_AudioStreamGet; SDL_AudioStreamAvailable; SDL_AudioStreamFlush; SDL_AudioStreamClear; SDL_FreeAudioStream

Definition at line 1268 of file SDL_audiocvt.c.

 {
     const int packetlen = 4096;  /* !!! FIXME: good enough for now. */
     Uint8 pre_resample_channels;
     SDL_AudioStream *retval;
  
     retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
     if (!retval) {
         return NULL;
     }
  
     /* If increasing channels, do it after resampling, since we'd just
        do more work to resample duplicate channels. If we're decreasing, do
        it first so we resample the interpolated data instead of interpolating
        the resampled data (!!! FIXME: decide if that works in practice, though!). */
     pre_resample_channels = SDL_min(src_channels, dst_channels);
  
     retval->first_run = SDL_TRUE;
     retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
     retval->src_format = src_format;
     retval->src_channels = src_channels;
     retval->src_rate = src_rate;
     retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
     retval->dst_format = dst_format;
     retval->dst_channels = dst_channels;
     retval->dst_rate = dst_rate;
     retval->pre_resample_channels = pre_resample_channels;
     retval->packetlen = packetlen;
     retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
     retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels;
     retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples ? retval->resampler_padding_samples : 1, sizeof (float));
  
     if (retval->resampler_padding == NULL) {
         SDL_FreeAudioStream(retval);
         SDL_OutOfMemory();
         return NULL;
     }
  
     retval->staging_buffer_size = ((retval->resampler_padding_samples / retval->pre_resample_channels) * retval->src_sample_frame_size);
     if (retval->staging_buffer_size > 0) {
         retval->staging_buffer = (Uint8 *) SDL_malloc(retval->staging_buffer_size);
         if (retval->staging_buffer == NULL) {
             SDL_FreeAudioStream(retval);
             SDL_OutOfMemory();
             return NULL;
         }
     }
  
     /* Not resampling? It's an easy conversion (and maybe not even that!) */
     if (src_rate == dst_rate) {
         retval->cvt_before_resampling.needed = SDL_FALSE;
         if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
             SDL_FreeAudioStream(retval);
             return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
         }
     } else {
         /* Don't resample at first. Just get us to Float32 format. */
         /* !!! FIXME: convert to int32 on devices without hardware float. */
         if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
             SDL_FreeAudioStream(retval);
             return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
         }
  
 #ifdef HAVE_LIBSAMPLERATE_H
         SetupLibSampleRateResampling(retval);
 #endif
  
         if (!retval->resampler_func) {
             retval->resampler_state = SDL_calloc(retval->resampler_padding_samples, sizeof (float));
             if (!retval->resampler_state) {
                 SDL_FreeAudioStream(retval);
                 SDL_OutOfMemory();
                 return NULL;
             }
  
             if (SDL_PrepareResampleFilter() < 0) {
                 SDL_free(retval->resampler_state);
                 retval->resampler_state = NULL;
                 SDL_FreeAudioStream(retval);
                 return NULL;
             }
  
             retval->resampler_func = SDL_ResampleAudioStream;
             retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
             retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
         }
  
         /* Convert us to the final format after resampling. */
         if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
             SDL_FreeAudioStream(retval);
             return NULL;  /* SDL_BuildAudioCVT should have called SDL_SetError. */
         }
     }
  
     retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
     if (!retval->queue) {
         SDL_FreeAudioStream(retval);
         return NULL;  /* SDL_NewDataQueue should have called SDL_SetError. */
     }
  
     return retval;
 }

References AUDIO_F32SYS, NULL, ResamplerPadding(), retval, SDL_AUDIO_BITSIZE, SDL_BuildAudioCVT(), SDL_calloc, SDL_CleanupAudioStreamResampler(), SDL_FALSE, SDL_free, SDL_FreeAudioStream(), SDL_malloc, SDL_min, SDL_NewDataQueue(), SDL_OutOfMemory, SDL_PrepareResampleFilter(), SDL_ResampleAudioStream(), SDL_ResetAudioStreamResampler(), and SDL_TRUE.

◆ SDL_OpenAudio()

int SDL_OpenAudio	(	SDL_AudioSpec *	desired,
		SDL_AudioSpec *	obtained
	)

This function opens the audio device with the desired parameters, and returns 0 if successful, placing the actual hardware parameters in the structure pointed to by obtained. If obtained is NULL, the audio data passed to the callback function will be guaranteed to be in the requested format, and will be automatically converted to the hardware audio format if necessary. This function returns -1 if it failed to open the audio device, or couldn't set up the audio thread.

When filling in the desired audio spec structure,

desired->freq should be the desired audio frequency in samples-per- second.
desired->format should be the desired audio format.
desired->samples is the desired size of the audio buffer, in samples. This number should be a power of two, and may be adjusted by the audio driver to a value more suitable for the hardware. Good values seem to range between 512 and 8096 inclusive, depending on the application and CPU speed. Smaller values yield faster response time, but can lead to underflow if the application is doing heavy processing and cannot fill the audio buffer in time. A stereo sample consists of both right and left channels in LR ordering. Note that the number of samples is directly related to time by the following formula:
ms = (samples*1000)/freq

samples
GLsizei samples
Definition: SDL_opengl_glext.h:1188
desired->size is the size in bytes of the audio buffer, and is calculated by SDL_OpenAudio().
desired->silence is the value used to set the buffer to silence, and is calculated by SDL_OpenAudio().
desired->callback should be set to a function that will be called when the audio device is ready for more data. It is passed a pointer to the audio buffer, and the length in bytes of the audio buffer. This function usually runs in a separate thread, and so you should protect data structures that it accesses by calling SDL_LockAudio() and SDL_UnlockAudio() in your code. Alternately, you may pass a NULL pointer here, and call SDL_QueueAudio() with some frequency, to queue more audio samples to be played (or for capture devices, call SDL_DequeueAudio() with some frequency, to obtain audio samples).
desired->userdata is passed as the first parameter to your callback function. If you passed a NULL callback, this value is ignored.

The audio device starts out playing silence when it's opened, and should be enabled for playing by calling SDL_PauseAudio(0) when you are ready for your audio callback function to be called. Since the audio driver may modify the requested size of the audio buffer, you should allocate any local mixing buffers after you open the audio device.

Definition at line 1464 of file SDL_audio.c.

 {
     SDL_AudioDeviceID id = 0;
  
     /* Start up the audio driver, if necessary. This is legacy behaviour! */
     if (!SDL_WasInit(SDL_INIT_AUDIO)) {
         if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
             return -1;
         }
     }
  
     /* SDL_OpenAudio() is legacy and can only act on Device ID #1. */
     if (open_devices[0] != NULL) {
         SDL_SetError("Audio device is already opened");
         return -1;
     }
  
     if (obtained) {
         id = open_audio_device(NULL, 0, desired, obtained,
                                SDL_AUDIO_ALLOW_ANY_CHANGE, 1);
     } else {
         SDL_AudioSpec _obtained;
         SDL_zero(_obtained);
         id = open_audio_device(NULL, 0, desired, &_obtained, 0, 1);
         /* On successful open, copy calculated values into 'desired'. */
         if (id > 0) {
             desired->size = _obtained.size;
             desired->silence = _obtained.silence;
         }
     }
  
     SDL_assert((id == 0) || (id == 1));
     return (id == 0) ? -1 : 0;
 }

References NULL, open_audio_device(), open_devices, SDL_assert, SDL_AUDIO_ALLOW_ANY_CHANGE, SDL_INIT_AUDIO, SDL_InitSubSystem, SDL_SetError, SDL_WasInit, SDL_zero, SDL_AudioSpec::silence, and SDL_AudioSpec::size.

◆ SDL_OpenAudioDevice()

SDL_AudioDeviceID SDL_OpenAudioDevice	(	const char *	device,
		int	iscapture,
		const SDL_AudioSpec *	desired,
		SDL_AudioSpec *	obtained,
		int	allowed_changes
	)

Open a specific audio device. Passing in a device name of NULL requests the most reasonable default (and is equivalent to calling SDL_OpenAudio()).

The device name is a UTF-8 string reported by SDL_GetAudioDeviceName(), but some drivers allow arbitrary and driver-specific strings, such as a hostname/IP address for a remote audio server, or a filename in the diskaudio driver.

Returns: 0 on error, a valid device ID that is >= 2 on success.

SDL_OpenAudio(), unlike this function, always acts on device ID 1.

Definition at line 1500 of file SDL_audio.c.

 {
     return open_audio_device(device, iscapture, desired, obtained,
                              allowed_changes, 2);
 }

References device, and open_audio_device().

◆ SDL_PauseAudio()

void SDL_PauseAudio ( int pause_on )

Definition at line 1542 of file SDL_audio.c.

 {
     SDL_PauseAudioDevice(1, pause_on);
 }

References SDL_PauseAudioDevice().

◆ SDL_PauseAudioDevice()

void SDL_PauseAudioDevice	(	SDL_AudioDeviceID	dev,
		int	pause_on
	)

Definition at line 1531 of file SDL_audio.c.

 {
     SDL_AudioDevice *device = get_audio_device(devid);
     if (device) {
         current_audio.impl.LockDevice(device);
         SDL_AtomicSet(&device->paused, pause_on ? 1 : 0);
         current_audio.impl.UnlockDevice(device);
     }
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDriverImpl::LockDevice, SDL_AtomicSet, and SDL_AudioDriverImpl::UnlockDevice.

Referenced by SDL_PauseAudio().

◆ SDL_QueueAudio()

int SDL_QueueAudio	(	SDL_AudioDeviceID	dev,
		const void *	data,
		Uint32	len
	)

Queue more audio on non-callback devices.

(If you are looking to retrieve queued audio from a non-callback capture device, you want SDL_DequeueAudio() instead. This will return -1 to signify an error if you use it with capture devices.)

SDL offers two ways to feed audio to the device: you can either supply a callback that SDL triggers with some frequency to obtain more audio (pull method), or you can supply no callback, and then SDL will expect you to supply data at regular intervals (push method) with this function.

There are no limits on the amount of data you can queue, short of exhaustion of address space. Queued data will drain to the device as necessary without further intervention from you. If the device needs audio but there is not enough queued, it will play silence to make up the difference. This means you will have skips in your audio playback if you aren't routinely queueing sufficient data.

This function copies the supplied data, so you are safe to free it when the function returns. This function is thread-safe, but queueing to the same device from two threads at once does not promise which buffer will be queued first.

You may not queue audio on a device that is using an application-supplied callback; doing so returns an error. You have to use the audio callback or queue audio with this function, but not both.

You should not call SDL_LockAudio() on the device before queueing; SDL handles locking internally for this function.

Parameters

dev	The device ID to which we will queue audio.
data	The data to queue to the device for later playback.
len	The number of bytes (not samples!) to which (data) points.

Returns: 0 on success, or -1 on error.

See also: SDL_GetQueuedAudioSize; SDL_ClearQueuedAudio

Definition at line 601 of file SDL_audio.c.

 {
     SDL_AudioDevice *device = get_audio_device(devid);
     int rc = 0;
  
     if (!device) {
         return -1;  /* get_audio_device() will have set the error state */
     } else if (device->iscapture) {
         return SDL_SetError("This is a capture device, queueing not allowed");
     } else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
         return SDL_SetError("Audio device has a callback, queueing not allowed");
     }
  
     if (len > 0) {
         current_audio.impl.LockDevice(device);
         rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
         current_audio.impl.UnlockDevice(device);
     }
  
     return rc;
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDriverImpl::LockDevice, SDL_BufferQueueDrainCallback(), SDL_SetError, SDL_WriteToDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

◆ SDL_UnlockAudio()

void SDL_UnlockAudio ( void )

Definition at line 125 of file SDL_dynapi_procs.h.

References SDL_UnlockAudioDevice().

◆ SDL_UnlockAudioDevice()

void SDL_UnlockAudioDevice ( SDL_AudioDeviceID dev )

Definition at line 1565 of file SDL_audio.c.

 {
     /* Obtain a lock on the mixing buffers */
     SDL_AudioDevice *device = get_audio_device(devid);
     if (device) {
         current_audio.impl.UnlockDevice(device);
     }
 }

References current_audio, device, get_audio_device(), SDL_AudioDriver::impl, and SDL_AudioDriverImpl::UnlockDevice.

Referenced by SDL_UnlockAudio().

Data Structures

Macros

Typedefs

Functions

Allow change flags

Pause audio functions

Audio state

Initialization and cleanup

Detailed Description

Macro Definition Documentation

◆ AUDIO_F32

◆ AUDIO_F32LSB

◆ AUDIO_F32MSB

◆ AUDIO_F32SYS

◆ AUDIO_S16

◆ AUDIO_S16LSB

◆ AUDIO_S16MSB

◆ AUDIO_S16SYS

◆ AUDIO_S32

◆ AUDIO_S32LSB

◆ AUDIO_S32MSB

◆ AUDIO_S32SYS

◆ AUDIO_S8

◆ AUDIO_U16

◆ AUDIO_U16LSB

◆ AUDIO_U16MSB

◆ AUDIO_U16SYS

◆ AUDIO_U8

◆ SDL_AUDIO_ALLOW_ANY_CHANGE

◆ SDL_AUDIO_ALLOW_CHANNELS_CHANGE

◆ SDL_AUDIO_ALLOW_FORMAT_CHANGE

◆ SDL_AUDIO_ALLOW_FREQUENCY_CHANGE

◆ SDL_AUDIO_ALLOW_SAMPLES_CHANGE

◆ SDL_AUDIO_BITSIZE

◆ SDL_AUDIO_ISBIGENDIAN

◆ SDL_AUDIO_ISFLOAT

◆ SDL_AUDIO_ISINT

◆ SDL_AUDIO_ISLITTLEENDIAN

◆ SDL_AUDIO_ISSIGNED

◆ SDL_AUDIO_ISUNSIGNED

◆ SDL_AUDIO_MASK_BITSIZE

◆ SDL_AUDIO_MASK_DATATYPE

◆ SDL_AUDIO_MASK_ENDIAN

◆ SDL_AUDIO_MASK_SIGNED

◆ SDL_AUDIOCVT_MAX_FILTERS

◆ SDL_AUDIOCVT_PACKED

◆ SDL_LoadWAV

◆ SDL_MIX_MAXVOLUME

Typedef Documentation

◆ SDL_AudioCallback

◆ SDL_AudioDeviceID

◆ SDL_AudioFilter

◆ SDL_AudioFormat

Enumeration Type Documentation

◆ SDL_AudioStatus

Function Documentation

◆ SDL_AudioInit()

◆ SDL_AudioQuit()

◆ SDL_AudioStreamAvailable()

◆ SDL_AudioStreamClear()

◆ SDL_AudioStreamFlush()

◆ SDL_AudioStreamGet()

◆ SDL_AudioStreamPut()

◆ SDL_BuildAudioCVT()

◆ SDL_ClearQueuedAudio()

◆ SDL_CloseAudio()

◆ SDL_CloseAudioDevice()

◆ SDL_ConvertAudio()

◆ SDL_DequeueAudio()

◆ SDL_FreeAudioStream()

◆ SDL_FreeWAV()

◆ SDL_GetAudioDeviceName()

◆ SDL_GetAudioDeviceStatus()

◆ SDL_GetAudioDriver()

◆ SDL_GetAudioStatus()

◆ SDL_GetCurrentAudioDriver()

◆ SDL_GetNumAudioDevices()

◆ SDL_GetNumAudioDrivers()

◆ SDL_GetQueuedAudioSize()

◆ SDL_LoadWAV_RW()