SDL 2.0
SDL_audio.h File Reference
#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_AudioSpecSDL_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

◆ 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
userdataAn application-specific parameter saved in the SDL_AudioSpec structure
streamA pointer to the audio data buffer.
lenThe 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

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

Enumerator
SDL_AUDIO_STOPPED 
SDL_AUDIO_PLAYING 
SDL_AUDIO_PAUSED 

Definition at line 395 of file SDL_audio.h.

396{
SDL_AudioStatus
Definition: SDL_audio.h:396
@ SDL_AUDIO_STOPPED
Definition: SDL_audio.h:397
@ SDL_AUDIO_PLAYING
Definition: SDL_audio.h:398
@ SDL_AUDIO_PAUSED
Definition: SDL_audio.h:399

Function Documentation

◆ SDL_AudioInit()

int SDL_AudioInit ( const char *  driver_name)

Definition at line 940 of file SDL_audio.c.

941{
942 int i = 0;
943 int initialized = 0;
944 int tried_to_init = 0;
945
947 SDL_AudioQuit(); /* shutdown driver if already running. */
948 }
949
952
953 /* Select the proper audio driver */
954 if (driver_name == NULL) {
955 driver_name = SDL_getenv("SDL_AUDIODRIVER");
956 }
957
958 for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
959 /* make sure we should even try this driver before doing so... */
960 const AudioBootStrap *backend = bootstrap[i];
961 if ((driver_name && (SDL_strncasecmp(backend->name, driver_name, SDL_strlen(driver_name)) != 0)) ||
962 (!driver_name && backend->demand_only)) {
963 continue;
964 }
965
966 tried_to_init = 1;
968 current_audio.name = backend->name;
969 current_audio.desc = backend->desc;
970 initialized = backend->init(&current_audio.impl);
971 }
972
973 if (!initialized) {
974 /* specific drivers will set the error message if they fail... */
975 if (!tried_to_init) {
976 if (driver_name) {
977 SDL_SetError("Audio target '%s' not available", driver_name);
978 } else {
979 SDL_SetError("No available audio device");
980 }
981 }
982
984 return -1; /* No driver was available, so fail. */
985 }
986
988
990
991 /* Make sure we have a list of devices available at startup. */
993
994#ifdef HAVE_LIBSAMPLERATE_H
995 LoadLibSampleRate();
996#endif
997
998 return 0;
999}
#define SDL_INIT_AUDIO
Definition: SDL.h:78
void SDL_AudioQuit(void)
Definition: SDL_audio.c:1590
static void finish_audio_entry_points_init(void)
Definition: SDL_audio.c:331
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
static SDL_AudioDevice * open_devices[16]
Definition: SDL_audio.c:34
static const AudioBootStrap *const bootstrap[]
Definition: SDL_audio.c:37
#define SDL_SetError
#define SDL_WasInit
#define SDL_getenv
#define SDL_strlen
#define SDL_CreateMutex
#define SDL_strncasecmp
#define SDL_zero(x)
Definition: SDL_stdinc.h:416
return Display return Display Bool Bool int int int return Display XEvent Bool(*) XPointer return Display return Display Drawable _Xconst char unsigned int unsigned int return Display Pixmap Pixmap XColor XColor unsigned int unsigned int return Display _Xconst char char int char return Display Visual unsigned int int int char unsigned int unsigned int in i)
Definition: SDL_x11sym.h:50
#define NULL
Definition: begin_code.h:167
const char * name
Definition: SDL_sysaudio.h:178
const char * desc
Definition: SDL_sysaudio.h:179
int(* init)(SDL_AudioDriverImpl *impl)
Definition: SDL_sysaudio.h:180
const char * name
Definition: SDL_sysaudio.h:111
SDL_mutex * detectionLock
Definition: SDL_sysaudio.h:120
const char * desc
Definition: SDL_sysaudio.h:115
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:117
void(* DetectDevices)(void)
Definition: SDL_sysaudio.h:67

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, and SDL_zero.

◆ SDL_AudioQuit()

◆ 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 1646 of file SDL_audiocvt.c.

1647{
1648 return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
1649}
size_t SDL_CountDataQueue(SDL_DataQueue *queue)
GLuint GLuint stream

References 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 1652 of file SDL_audiocvt.c.

1653{
1654 if (!stream) {
1655 SDL_InvalidParamError("stream");
1656 } else {
1657 SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
1658 if (stream->reset_resampler_func) {
1659 stream->reset_resampler_func(stream);
1660 }
1661 stream->first_run = SDL_TRUE;
1662 stream->staging_buffer_filled = 0;
1663 }
1664}
void SDL_ClearDataQueue(SDL_DataQueue *queue, const size_t slack)
Definition: SDL_dataqueue.c:98
#define SDL_InvalidParamError(param)
Definition: SDL_error.h:54
@ SDL_TRUE
Definition: SDL_stdinc.h:164

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 1571 of file SDL_audiocvt.c.

1572{
1573 if (!stream) {
1574 return SDL_InvalidParamError("stream");
1575 }
1576
1577 #if DEBUG_AUDIOSTREAM
1578 printf("AUDIOSTREAM: flushing! staging_buffer_filled=%d bytes\n", stream->staging_buffer_filled);
1579 #endif
1580
1581 /* shouldn't use a staging buffer if we're not resampling. */
1582 SDL_assert((stream->dst_rate != stream->src_rate) || (stream->staging_buffer_filled == 0));
1583
1584 if (stream->staging_buffer_filled > 0) {
1585 /* push the staging buffer + silence. We need to flush out not just
1586 the staging buffer, but the piece that the stream was saving off
1587 for right-side resampler padding. */
1588 const SDL_bool first_run = stream->first_run;
1589 const int filled = stream->staging_buffer_filled;
1590 int actual_input_frames = filled / stream->src_sample_frame_size;
1591 if (!first_run)
1592 actual_input_frames += stream->resampler_padding_samples / stream->pre_resample_channels;
1593
1594 if (actual_input_frames > 0) { /* don't bother if nothing to flush. */
1595 /* This is how many bytes we're expecting without silence appended. */
1596 int flush_remaining = ((int) SDL_ceil(actual_input_frames * stream->rate_incr)) * stream->dst_sample_frame_size;
1597
1599 printf("AUDIOSTREAM: flushing with padding to get max %d bytes!\n", flush_remaining);
1600 #endif
1601
1602 SDL_memset(stream->staging_buffer + filled, '\0', stream->staging_buffer_size - filled);
1603 if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
1604 return -1;
1605 }
1606
1607 /* we have flushed out (or initially filled) the pending right-side
1608 resampler padding, but we need to push more silence to guarantee
1609 the staging buffer is fully flushed out, too. */
1610 SDL_memset(stream->staging_buffer, '\0', filled);
1611 if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
1612 return -1;
1613 }
1614 }
1615 }
1616
1617 stream->staging_buffer_filled = 0;
1618 stream->first_run = SDL_TRUE;
1619
1620 return 0;
1621}
#define SDL_assert(condition)
Definition: SDL_assert.h:169
static int SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, int *maxputbytes)
#define DEBUG_AUDIOSTREAM
Definition: SDL_audiocvt.c:37
#define SDL_memset
#define SDL_ceil
SDL_bool
Definition: SDL_stdinc.h:162

References DEBUG_AUDIOSTREAM, 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
streamThe stream the audio is being requested from
bufA buffer to fill with audio data
lenThe 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 1625 of file SDL_audiocvt.c.

1626{
1627 #if DEBUG_AUDIOSTREAM
1628 printf("AUDIOSTREAM: want to get %d converted bytes\n", len);
1629 #endif
1630
1631 if (!stream) {
1632 return SDL_InvalidParamError("stream");
1633 } else if (!buf) {
1634 return SDL_InvalidParamError("buf");
1635 } else if (len <= 0) {
1636 return 0; /* nothing to do. */
1637 } else if ((len % stream->dst_sample_frame_size) != 0) {
1638 return SDL_SetError("Can't request partial sample frames");
1639 }
1640
1641 return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
1642}
size_t SDL_ReadFromDataQueue(SDL_DataQueue *queue, void *_buf, const size_t _len)
GLenum GLsizei len
GLenum GLuint GLenum GLsizei const GLchar * buf

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
streamThe stream the audio data is being added to
bufA pointer to the audio data to add
lenThe 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 1507 of file SDL_audiocvt.c.

1508{
1509 /* !!! FIXME: several converters can take advantage of SIMD, but only
1510 !!! FIXME: if the data is aligned to 16 bytes. EnsureStreamBufferSize()
1511 !!! FIXME: guarantees the buffer will align, but the
1512 !!! FIXME: converters will iterate over the data backwards if
1513 !!! FIXME: the output grows, and this means we won't align if buflen
1514 !!! FIXME: isn't a multiple of 16. In these cases, we should chop off
1515 !!! FIXME: a few samples at the end and convert them separately. */
1516
1517 #if DEBUG_AUDIOSTREAM
1518 printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
1519 #endif
1520
1521 if (!stream) {
1522 return SDL_InvalidParamError("stream");
1523 } else if (!buf) {
1524 return SDL_InvalidParamError("buf");
1525 } else if (len == 0) {
1526 return 0; /* nothing to do. */
1527 } else if ((len % stream->src_sample_frame_size) != 0) {
1528 return SDL_SetError("Can't add partial sample frames");
1529 }
1530
1531 if (!stream->cvt_before_resampling.needed &&
1532 (stream->dst_rate == stream->src_rate) &&
1533 !stream->cvt_after_resampling.needed) {
1534 #if DEBUG_AUDIOSTREAM
1535 printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", len);
1536 #endif
1537 return SDL_WriteToDataQueue(stream->queue, buf, len);
1538 }
1539
1540 while (len > 0) {
1541 int amount;
1542
1543 /* If we don't have a staging buffer or we're given enough data that
1544 we don't need to store it for later, skip the staging process.
1545 */
1546 if (!stream->staging_buffer_filled && len >= stream->staging_buffer_size) {
1548 }
1549
1550 /* If there's not enough data to fill the staging buffer, just save it */
1551 if ((stream->staging_buffer_filled + len) < stream->staging_buffer_size) {
1552 SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, len);
1553 stream->staging_buffer_filled += len;
1554 return 0;
1555 }
1556
1557 /* Fill the staging buffer, process it, and continue */
1558 amount = (stream->staging_buffer_size - stream->staging_buffer_filled);
1559 SDL_assert(amount > 0);
1560 SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, amount);
1561 stream->staging_buffer_filled = 0;
1562 if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, NULL) < 0) {
1563 return -1;
1564 }
1565 buf = (void *)((Uint8 *)buf + amount);
1566 len -= amount;
1567 }
1568 return 0;
1569}
int SDL_WriteToDataQueue(SDL_DataQueue *queue, const void *_data, const size_t _len)
#define SDL_memcpy
uint8_t Uint8
Definition: SDL_stdinc.h:179

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 878 of file SDL_audiocvt.c.

881{
882 /* Sanity check target pointer */
883 if (cvt == NULL) {
884 return SDL_InvalidParamError("cvt");
885 }
886
887 /* Make sure we zero out the audio conversion before error checking */
888 SDL_zerop(cvt);
889
890 if (!SDL_SupportedAudioFormat(src_fmt)) {
891 return SDL_SetError("Invalid source format");
892 } else if (!SDL_SupportedAudioFormat(dst_fmt)) {
893 return SDL_SetError("Invalid destination format");
894 } else if (!SDL_SupportedChannelCount(src_channels)) {
895 return SDL_SetError("Invalid source channels");
896 } else if (!SDL_SupportedChannelCount(dst_channels)) {
897 return SDL_SetError("Invalid destination channels");
898 } else if (src_rate <= 0) {
899 return SDL_SetError("Source rate is equal to or less than zero");
900 } else if (dst_rate <= 0) {
901 return SDL_SetError("Destination rate is equal to or less than zero");
902 } else if (src_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
903 return SDL_SetError("Source rate is too high");
904 } else if (dst_rate >= SDL_MAX_SINT32 / RESAMPLER_SAMPLES_PER_ZERO_CROSSING) {
905 return SDL_SetError("Destination rate is too high");
906 }
907
908#if DEBUG_CONVERT
909 printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
910 src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
911#endif
912
913 /* Start off with no conversion necessary */
914 cvt->src_format = src_fmt;
915 cvt->dst_format = dst_fmt;
916 cvt->needed = 0;
917 cvt->filter_index = 0;
918 SDL_zero(cvt->filters);
919 cvt->len_mult = 1;
920 cvt->len_ratio = 1.0;
921 cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
922
923 /* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
925
926 /* Type conversion goes like this now:
927 - byteswap to CPU native format first if necessary.
928 - convert to native Float32 if necessary.
929 - resample and change channel count if necessary.
930 - convert back to native format.
931 - byteswap back to foreign format if necessary.
932
933 The expectation is we can process data faster in float32
934 (possibly with SIMD), and making several passes over the same
935 buffer is likely to be CPU cache-friendly, avoiding the
936 biggest performance hit in modern times. Previously we had
937 (script-generated) custom converters for every data type and
938 it was a bloat on SDL compile times and final library size. */
939
940 /* see if we can skip float conversion entirely. */
941 if (src_rate == dst_rate && src_channels == dst_channels) {
942 if (src_fmt == dst_fmt) {
943 return 0;
944 }
945
946 /* just a byteswap needed? */
947 if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
949 return -1;
950 }
951 cvt->needed = 1;
952 return 1;
953 }
954 }
955
956 /* Convert data types, if necessary. Updates (cvt). */
957 if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
958 return -1; /* shouldn't happen, but just in case... */
959 }
960
961 /* Channel conversion */
962 if (src_channels < dst_channels) {
963 /* Upmixing */
964 /* Mono -> Stereo [-> ...] */
965 if ((src_channels == 1) && (dst_channels > 1)) {
967 return -1;
968 }
969 cvt->len_mult *= 2;
970 src_channels = 2;
971 cvt->len_ratio *= 2;
972 }
973 /* [Mono ->] Stereo -> 5.1 [-> 7.1] */
974 if ((src_channels == 2) && (dst_channels >= 6)) {
976 return -1;
977 }
978 src_channels = 6;
979 cvt->len_mult *= 3;
980 cvt->len_ratio *= 3;
981 }
982 /* Quad -> 5.1 [-> 7.1] */
983 if ((src_channels == 4) && (dst_channels >= 6)) {
985 return -1;
986 }
987 src_channels = 6;
988 cvt->len_mult = (cvt->len_mult * 3 + 1) / 2;
989 cvt->len_ratio *= 1.5;
990 }
991 /* [[Mono ->] Stereo ->] 5.1 -> 7.1 */
992 if ((src_channels == 6) && (dst_channels == 8)) {
994 return -1;
995 }
996 src_channels = 8;
997 cvt->len_mult = (cvt->len_mult * 4 + 2) / 3;
998 /* Should be numerically exact with every valid input to this
999 function */
1000 cvt->len_ratio = cvt->len_ratio * 4 / 3;
1001 }
1002 /* [Mono ->] Stereo -> Quad */
1003 if ((src_channels == 2) && (dst_channels == 4)) {
1005 return -1;
1006 }
1007 src_channels = 4;
1008 cvt->len_mult *= 2;
1009 cvt->len_ratio *= 2;
1010 }
1011 } else if (src_channels > dst_channels) {
1012 /* Downmixing */
1013 /* 7.1 -> 5.1 [-> Stereo [-> Mono]] */
1014 /* 7.1 -> 5.1 [-> Quad] */
1015 if ((src_channels == 8) && (dst_channels <= 6)) {
1017 return -1;
1018 }
1019 src_channels = 6;
1020 cvt->len_ratio *= 0.75;
1021 }
1022 /* [7.1 ->] 5.1 -> Stereo [-> Mono] */
1023 if ((src_channels == 6) && (dst_channels <= 2)) {
1025 return -1;
1026 }
1027 src_channels = 2;
1028 cvt->len_ratio /= 3;
1029 }
1030 /* 5.1 -> Quad */
1031 if ((src_channels == 6) && (dst_channels == 4)) {
1033 return -1;
1034 }
1035 src_channels = 4;
1036 cvt->len_ratio = cvt->len_ratio * 2 / 3;
1037 }
1038 /* Quad -> Stereo [-> Mono] */
1039 if ((src_channels == 4) && (dst_channels <= 2)) {
1041 return -1;
1042 }
1043 src_channels = 2;
1044 cvt->len_ratio /= 2;
1045 }
1046 /* [... ->] Stereo -> Mono */
1047 if ((src_channels == 2) && (dst_channels == 1)) {
1049
1050 #if HAVE_SSE3_INTRINSICS
1051 if (SDL_HasSSE3()) {
1052 filter = SDL_ConvertStereoToMono_SSE3;
1053 }
1054 #endif
1055
1056 if (!filter) {
1058 }
1059
1060 if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
1061 return -1;
1062 }
1063
1064 src_channels = 1;
1065 cvt->len_ratio /= 2;
1066 }
1067 }
1068
1069 if (src_channels != dst_channels) {
1070 /* All combinations of supported channel counts should have been
1071 handled by now, but let's be defensive */
1072 return SDL_SetError("Invalid channel combination");
1073 }
1074
1075 /* Do rate conversion, if necessary. Updates (cvt). */
1076 if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
1077 return -1; /* shouldn't happen, but just in case... */
1078 }
1079
1080 /* Move to final data type. */
1081 if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
1082 return -1; /* shouldn't happen, but just in case... */
1083 }
1084
1085 cvt->needed = (cvt->filter_index != 0);
1086 return (cvt->needed);
1087}
void(* SDL_AudioFilter)(struct SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audio.h:193
#define SDL_AUDIO_MASK_ENDIAN
Definition: SDL_audio.h:73
void SDL_ChooseAudioConverters(void)
static int SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
Definition: SDL_audiocvt.c:662
static void SDL_Convert51To71(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:335
#define RESAMPLER_SAMPLES_PER_ZERO_CROSSING
Definition: SDL_audiocvt.c:382
static void SDL_Convert71To51(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:151
static SDL_bool SDL_SupportedAudioFormat(const SDL_AudioFormat fmt)
Definition: SDL_audiocvt.c:831
static int SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
Definition: SDL_audiocvt.c:615
static void SDL_ConvertStereoToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:305
static SDL_bool SDL_SupportedChannelCount(const int channels)
Definition: SDL_audiocvt.c:854
static void SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:563
static void SDL_Convert51ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:181
static void SDL_ConvertStereoTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:234
static void SDL_Convert51ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:103
static int SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter)
Definition: SDL_audiocvt.c:601
static void SDL_ConvertStereoToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:81
static void SDL_ConvertQuadTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:268
static int SDL_BuildAudioResampleCVT(SDL_AudioCVT *cvt, const int dst_channels, const int src_rate, const int dst_rate)
Definition: SDL_audiocvt.c:782
static void SDL_ConvertMonoToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:210
static void SDL_ConvertQuadToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:128
#define SDL_HasSSE3
GLint GLint GLint GLint GLint GLint GLint GLbitfield GLenum filter
#define SDL_zerop(x)
Definition: SDL_stdinc.h:417
#define SDL_MAX_SINT32
A signed 32-bit integer type.
Definition: SDL_stdinc.h:195
int filter_index
Definition: SDL_audio.h:238
double len_ratio
Definition: SDL_audio.h:236
SDL_AudioFormat src_format
Definition: SDL_audio.h:229
SDL_AudioFormat dst_format
Definition: SDL_audio.h:230
double rate_incr
Definition: SDL_audio.h:231
SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS+1]
Definition: SDL_audio.h:237

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_zero, 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
devThe device ID of which to clear the audio queue.
See also
SDL_QueueAudio
SDL_GetQueuedAudioSize

Definition at line 662 of file SDL_audio.c.

663{
665
666 if (!device) {
667 return; /* nothing to do. */
668 }
669
670 /* Blank out the device and release the mutex. Free it afterwards. */
672
673 /* Keep up to two packets in the pool to reduce future malloc pressure. */
675
677}
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:200
#define SDL_AUDIOBUFFERQUEUE_PACKETLEN
Definition: SDL_sysaudio.h:63
static SDL_AudioDeviceID device
Definition: loopwave.c:37
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:79

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 1578 of file SDL_audio.c.

1579{
1581}
static void close_audio_device(SDL_AudioDevice *device)
Definition: SDL_audio.c:1110

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 540 of file SDL_audiocvt.c.

541{
542 /* !!! FIXME: (cvt) should be const; stack-copy it here. */
543 /* !!! FIXME: (actually, we can't...len_cvt needs to be updated. Grr.) */
544
545 /* Make sure there's data to convert */
546 if (cvt->buf == NULL) {
547 return SDL_SetError("No buffer allocated for conversion");
548 }
549
550 /* Return okay if no conversion is necessary */
551 cvt->len_cvt = cvt->len;
552 if (cvt->filters[0] == NULL) {
553 return 0;
554 }
555
556 /* Set up the conversion and go! */
557 cvt->filter_index = 0;
558 cvt->filters[0] (cvt, cvt->src_format);
559 return 0;
560}
Uint8 * buf
Definition: SDL_audio.h:232

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
devThe device ID from which we will dequeue audio.
dataA pointer into where audio data should be copied.
lenThe 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 621 of file SDL_audio.c.

622{
624 Uint32 rc;
625
626 if ( (len == 0) || /* nothing to do? */
627 (!device) || /* called with bogus device id */
628 (!device->iscapture) || /* playback devices can't dequeue */
629 (device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
630 return 0; /* just report zero bytes dequeued. */
631 }
632
634 rc = (Uint32) SDL_ReadFromDataQueue(device->buffer_queue, data, len);
636 return rc;
637}
static void SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
Definition: SDL_audio.c:582
GLint GLenum GLsizei GLsizei GLsizei GLint GLsizei const GLvoid * data
Definition: SDL_opengl.h:1974
uint32_t Uint32
Definition: SDL_stdinc.h:203

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 1668 of file SDL_audiocvt.c.

1669{
1670 if (stream) {
1671 if (stream->cleanup_resampler_func) {
1672 stream->cleanup_resampler_func(stream);
1673 }
1674 SDL_FreeDataQueue(stream->queue);
1675 SDL_free(stream->staging_buffer);
1676 SDL_free(stream->work_buffer_base);
1677 SDL_free(stream->resampler_padding);
1679 }
1680}
void SDL_FreeDataQueue(SDL_DataQueue *queue)
Definition: SDL_dataqueue.c:88
#define SDL_free

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 2148 of file SDL_wave.c.

2149{
2150 SDL_free(audio_buf);
2151}

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 1070 of file SDL_audio.c.

1071{
1072 const char *retval = NULL;
1073
1075 SDL_SetError("Audio subsystem is not initialized");
1076 return NULL;
1077 }
1078
1079 if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
1080 SDL_SetError("No capture support");
1081 return NULL;
1082 }
1083
1084 if (index >= 0) {
1085 SDL_AudioDeviceItem *item;
1086 int i;
1087
1091 if (index < i) {
1092 for (i--; i > index; i--, item = item->next) {
1093 SDL_assert(item != NULL);
1094 }
1095 SDL_assert(item != NULL);
1096 retval = item->name;
1097 }
1099 }
1100
1101 if (retval == NULL) {
1102 SDL_SetError("No such device");
1103 }
1104
1105 return retval;
1106}
#define SDL_LockMutex
#define SDL_UnlockMutex
GLuint index
struct SDL_AudioDeviceItem * next
Definition: SDL_sysaudio.h:103
SDL_AudioDeviceItem * inputDevices
Definition: SDL_sysaudio.h:126
SDL_AudioDeviceItem * outputDevices
Definition: SDL_sysaudio.h:125
SDL_bool 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 1506 of file SDL_audio.c.

1507{
1510 if (device && SDL_AtomicGet(&device->enabled)) {
1511 if (SDL_AtomicGet(&device->paused)) {
1512 status = SDL_AUDIO_PAUSED;
1513 } else {
1514 status = SDL_AUDIO_PLAYING;
1515 }
1516 }
1517 return status;
1518}
#define SDL_AtomicGet

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 931 of file SDL_audio.c.

932{
933 if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
934 return bootstrap[index]->name;
935 }
936 return NULL;
937}
int SDL_GetNumAudioDrivers(void)
Definition: SDL_audio.c:925

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 1005 of file SDL_audio.c.

1006{
1007 return current_audio.name;
1008}

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 1045 of file SDL_audio.c.

1046{
1047 int retval = 0;
1048
1050 return -1;
1051 }
1052
1054 if (iscapture && current_audio.captureDevicesRemoved) {
1056 }
1057
1058 if (!iscapture && current_audio.outputDevicesRemoved) {
1060 }
1061
1064
1065 return retval;
1066}
static void clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *removedFlag)
Definition: SDL_audio.c:1012
SDL_bool captureDevicesRemoved
Definition: SDL_sysaudio.h:121
SDL_bool outputDevicesRemoved
Definition: SDL_sysaudio.h:122

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 925 of file SDL_audio.c.

926{
927 return SDL_arraysize(bootstrap) - 1;
928}
#define SDL_arraysize(array)
Definition: SDL_stdinc.h:115

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
devThe 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 640 of file SDL_audio.c.

641{
642 Uint32 retval = 0;
644
645 if (!device) {
646 return 0;
647 }
648
649 /* Nothing to do unless we're set up for queueing. */
650 if (device->callbackspec.callback == SDL_BufferQueueDrainCallback ||
651 device->callbackspec.callback == SDL_BufferQueueFillCallback)
652 {
654 retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
656 }
657
658 return retval;
659}
static void SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
Definition: SDL_audio.c:561

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, ...);
#define SDL_RWFromFile
#define SDL_LoadWAV_RW
Parameters
srcThe data source with the WAVE data
freesrcA integer value that makes the function close the data source if non-zero
specA pointer filled with the audio format of the audio data
audio_bufA pointer filled with the audio data allocated by the function
audio_lenA pointer filled with the length of the audio data buffer in bytes
Returns
NULL on error, or non-NULL on success.

Definition at line 2095 of file SDL_wave.c.

2096{
2097 int result;
2098 WaveFile file;
2099
2100 SDL_zero(file);
2101
2102 /* Make sure we are passed a valid data source */
2103 if (src == NULL) {
2104 /* Error may come from RWops. */
2105 return NULL;
2106 } else if (spec == NULL) {
2107 SDL_InvalidParamError("spec");
2108 return NULL;
2109 } else if (audio_buf == NULL) {
2110 SDL_InvalidParamError("audio_buf");
2111 return NULL;
2112 } else if (audio_len == NULL) {
2113 SDL_InvalidParamError("audio_len");
2114 return NULL;
2115 }
2116
2117 *audio_buf = NULL;
2118 *audio_len = 0;
2119
2123
2124 result = WaveLoad(src, &file, spec, audio_buf, audio_len);
2125 if (result < 0) {
2126 SDL_free(*audio_buf);
2127 spec = NULL;
2128 audio_buf = NULL;
2129 audio_len = 0;
2130 }
2131
2132 /* Cleanup */
2133 if (freesrc) {
2135 } else {
2137 }
2138 WaveFreeChunkData(&file.chunk);
2139 SDL_free(file.decoderdata);
2140
2141 return spec;
2142}
#define SDL_RWseek
#define SDL_RWclose
GLenum src
GLuint64EXT * result
#define RW_SEEK_SET
Definition: SDL_rwops.h:174
static int WaveLoad(SDL_RWops *src, WaveFile *file, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
Definition: SDL_wave.c:1787
static WaveTruncationHint WaveGetTruncationHint()
Definition: SDL_wave.c:1471
static WaveRiffSizeHint WaveGetRiffSizeHint()
Definition: SDL_wave.c:1451
static void WaveFreeChunkData(WaveChunk *chunk)
Definition: SDL_wave.c:1511
static WaveFactChunkHint WaveGetFactChunkHint()
Definition: SDL_wave.c:1491
SDL_AudioSpec spec
Definition: loopwave.c:31
Sint64 position
Definition: SDL_wave.h:98
WaveChunk chunk
Definition: SDL_wave.h:132
WaveFactChunkHint facthint
Definition: SDL_wave.h:146
void * decoderdata
Definition: SDL_wave.h:142
WaveRiffSizeHint riffhint
Definition: SDL_wave.h:144
WaveTruncationHint trunchint
Definition: SDL_wave.h:145

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 1546 of file SDL_audio.c.

1547{
1548 /* Obtain a lock on the mixing buffers */
1550 if (device) {
1552 }
1553}

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 1689 of file SDL_audio.c.

1690{
1691 /* Mix the user-level audio format */
1693 if (device != NULL) {
1694 SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
1695 }
1696}
#define SDL_MixAudioFormat
GLenum GLenum dst

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.

92{
93 if (volume == 0) {
94 return;
95 }
96
97 switch (format) {
98
99 case AUDIO_U8:
100 {
101#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
102 SDL_MixAudio_m68k_U8((char *) dst, (char *) src,
103 (unsigned long) len, (long) volume,
104 (char *) mix8);
105#else
106 Uint8 src_sample;
107
108 while (len--) {
109 src_sample = *src;
110 ADJUST_VOLUME_U8(src_sample, volume);
111 *dst = mix8[*dst + src_sample];
112 ++dst;
113 ++src;
114 }
115#endif
116 }
117 break;
118
119 case AUDIO_S8:
120 {
121 Sint8 *dst8, *src8;
122 Sint8 src_sample;
123 int dst_sample;
124 const int max_audioval = ((1 << (8 - 1)) - 1);
125 const int min_audioval = -(1 << (8 - 1));
126
127 src8 = (Sint8 *) src;
128 dst8 = (Sint8 *) dst;
129 while (len--) {
130 src_sample = *src8;
131 ADJUST_VOLUME(src_sample, volume);
132 dst_sample = *dst8 + src_sample;
133 if (dst_sample > max_audioval) {
134 *dst8 = max_audioval;
135 } else if (dst_sample < min_audioval) {
136 *dst8 = min_audioval;
137 } else {
138 *dst8 = dst_sample;
139 }
140 ++dst8;
141 ++src8;
142 }
143 }
144 break;
145
146 case AUDIO_S16LSB:
147 {
148 Sint16 src1, src2;
149 int dst_sample;
150 const int max_audioval = ((1 << (16 - 1)) - 1);
151 const int min_audioval = -(1 << (16 - 1));
152
153 len /= 2;
154 while (len--) {
155 src1 = ((src[1]) << 8 | src[0]);
156 ADJUST_VOLUME(src1, volume);
157 src2 = ((dst[1]) << 8 | dst[0]);
158 src += 2;
159 dst_sample = src1 + src2;
160 if (dst_sample > max_audioval) {
161 dst_sample = max_audioval;
162 } else if (dst_sample < min_audioval) {
163 dst_sample = min_audioval;
164 }
165 dst[0] = dst_sample & 0xFF;
166 dst_sample >>= 8;
167 dst[1] = dst_sample & 0xFF;
168 dst += 2;
169 }
170 }
171 break;
172
173 case AUDIO_S16MSB:
174 {
175#if defined(__GNUC__) && defined(__M68000__) && !defined(__mcoldfire__) && defined(SDL_ASSEMBLY_ROUTINES)
176 SDL_MixAudio_m68k_S16MSB((short *) dst, (short *) src,
177 (unsigned long) len, (long) volume);
178#else
179 Sint16 src1, src2;
180 int dst_sample;
181 const int max_audioval = ((1 << (16 - 1)) - 1);
182 const int min_audioval = -(1 << (16 - 1));
183
184 len /= 2;
185 while (len--) {
186 src1 = ((src[0]) << 8 | src[1]);
187 ADJUST_VOLUME(src1, volume);
188 src2 = ((dst[0]) << 8 | dst[1]);
189 src += 2;
190 dst_sample = src1 + src2;
191 if (dst_sample > max_audioval) {
192 dst_sample = max_audioval;
193 } else if (dst_sample < min_audioval) {
194 dst_sample = min_audioval;
195 }
196 dst[1] = dst_sample & 0xFF;
197 dst_sample >>= 8;
198 dst[0] = dst_sample & 0xFF;
199 dst += 2;
200 }
201#endif
202 }
203 break;
204
205 case AUDIO_U16LSB:
206 {
207 Uint16 src1, src2;
208 int dst_sample;
209 const int max_audioval = 0xFFFF;
210
211 len /= 2;
212 while (len--) {
213 src1 = ((src[1]) << 8 | src[0]);
214 ADJUST_VOLUME(src1, volume);
215 src2 = ((dst[1]) << 8 | dst[0]);
216 src += 2;
217 dst_sample = src1 + src2;
218 if (dst_sample > max_audioval) {
219 dst_sample = max_audioval;
220 }
221 dst[0] = dst_sample & 0xFF;
222 dst_sample >>= 8;
223 dst[1] = dst_sample & 0xFF;
224 dst += 2;
225 }
226 }
227 break;
228
229 case AUDIO_U16MSB:
230 {
231 Uint16 src1, src2;
232 int dst_sample;
233 const int max_audioval = 0xFFFF;
234
235 len /= 2;
236 while (len--) {
237 src1 = ((src[0]) << 8 | src[1]);
238 ADJUST_VOLUME(src1, volume);
239 src2 = ((dst[0]) << 8 | dst[1]);
240 src += 2;
241 dst_sample = src1 + src2;
242 if (dst_sample > max_audioval) {
243 dst_sample = max_audioval;
244 }
245 dst[1] = dst_sample & 0xFF;
246 dst_sample >>= 8;
247 dst[0] = dst_sample & 0xFF;
248 dst += 2;
249 }
250 }
251 break;
252
253 case AUDIO_S32LSB:
254 {
255 const Uint32 *src32 = (Uint32 *) src;
256 Uint32 *dst32 = (Uint32 *) dst;
257 Sint64 src1, src2;
258 Sint64 dst_sample;
259 const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
260 const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
261
262 len /= 4;
263 while (len--) {
264 src1 = (Sint64) ((Sint32) SDL_SwapLE32(*src32));
265 src32++;
266 ADJUST_VOLUME(src1, volume);
267 src2 = (Sint64) ((Sint32) SDL_SwapLE32(*dst32));
268 dst_sample = src1 + src2;
269 if (dst_sample > max_audioval) {
270 dst_sample = max_audioval;
271 } else if (dst_sample < min_audioval) {
272 dst_sample = min_audioval;
273 }
274 *(dst32++) = SDL_SwapLE32((Uint32) ((Sint32) dst_sample));
275 }
276 }
277 break;
278
279 case AUDIO_S32MSB:
280 {
281 const Uint32 *src32 = (Uint32 *) src;
282 Uint32 *dst32 = (Uint32 *) dst;
283 Sint64 src1, src2;
284 Sint64 dst_sample;
285 const Sint64 max_audioval = ((((Sint64) 1) << (32 - 1)) - 1);
286 const Sint64 min_audioval = -(((Sint64) 1) << (32 - 1));
287
288 len /= 4;
289 while (len--) {
290 src1 = (Sint64) ((Sint32) SDL_SwapBE32(*src32));
291 src32++;
292 ADJUST_VOLUME(src1, volume);
293 src2 = (Sint64) ((Sint32) SDL_SwapBE32(*dst32));
294 dst_sample = src1 + src2;
295 if (dst_sample > max_audioval) {
296 dst_sample = max_audioval;
297 } else if (dst_sample < min_audioval) {
298 dst_sample = min_audioval;
299 }
300 *(dst32++) = SDL_SwapBE32((Uint32) ((Sint32) dst_sample));
301 }
302 }
303 break;
304
305 case AUDIO_F32LSB:
306 {
307 const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
308 const float fvolume = (float) volume;
309 const float *src32 = (float *) src;
310 float *dst32 = (float *) dst;
311 float src1, src2;
312 double dst_sample;
313 /* !!! FIXME: are these right? */
314 const double max_audioval = 3.402823466e+38F;
315 const double min_audioval = -3.402823466e+38F;
316
317 len /= 4;
318 while (len--) {
319 src1 = ((SDL_SwapFloatLE(*src32) * fvolume) * fmaxvolume);
320 src2 = SDL_SwapFloatLE(*dst32);
321 src32++;
322
323 dst_sample = ((double) src1) + ((double) src2);
324 if (dst_sample > max_audioval) {
325 dst_sample = max_audioval;
326 } else if (dst_sample < min_audioval) {
327 dst_sample = min_audioval;
328 }
329 *(dst32++) = SDL_SwapFloatLE((float) dst_sample);
330 }
331 }
332 break;
333
334 case AUDIO_F32MSB:
335 {
336 const float fmaxvolume = 1.0f / ((float) SDL_MIX_MAXVOLUME);
337 const float fvolume = (float) volume;
338 const float *src32 = (float *) src;
339 float *dst32 = (float *) dst;
340 float src1, src2;
341 double dst_sample;
342 /* !!! FIXME: are these right? */
343 const double max_audioval = 3.402823466e+38F;
344 const double min_audioval = -3.402823466e+38F;
345
346 len /= 4;
347 while (len--) {
348 src1 = ((SDL_SwapFloatBE(*src32) * fvolume) * fmaxvolume);
349 src2 = SDL_SwapFloatBE(*dst32);
350 src32++;
351
352 dst_sample = ((double) src1) + ((double) src2);
353 if (dst_sample > max_audioval) {
354 dst_sample = max_audioval;
355 } else if (dst_sample < min_audioval) {
356 dst_sample = min_audioval;
357 }
358 *(dst32++) = SDL_SwapFloatBE((float) dst_sample);
359 }
360 }
361 break;
362
363 default: /* If this happens... FIXME! */
364 SDL_SetError("SDL_MixAudioFormat(): unknown audio format");
365 return;
366 }
367}
#define AUDIO_U16LSB
Definition: SDL_audio.h:91
#define AUDIO_S16LSB
Definition: SDL_audio.h:92
#define AUDIO_F32MSB
Definition: SDL_audio.h:113
#define AUDIO_S32MSB
Definition: SDL_audio.h:104
#define AUDIO_S16MSB
Definition: SDL_audio.h:94
#define AUDIO_U16MSB
Definition: SDL_audio.h:93
#define AUDIO_U8
Definition: SDL_audio.h:89
#define AUDIO_S8
Definition: SDL_audio.h:90
#define AUDIO_S32LSB
Definition: SDL_audio.h:103
#define SDL_MIX_MAXVOLUME
Definition: SDL_audio.h:649
#define AUDIO_F32LSB
Definition: SDL_audio.h:112
#define SDL_SwapFloatBE(X)
Definition: SDL_endian.h:239
#define SDL_SwapLE32(X)
Definition: SDL_endian.h:233
#define SDL_SwapBE32(X)
Definition: SDL_endian.h:237
#define SDL_SwapFloatLE(X)
Definition: SDL_endian.h:235
#define ADJUST_VOLUME_U8(s, v)
Definition: SDL_mixer.c:86
#define ADJUST_VOLUME(s, v)
Definition: SDL_mixer.c:85
static const Uint8 mix8[]
Definition: SDL_mixer.c:34
GLint GLint GLsizei GLsizei GLsizei GLint GLenum format
Definition: SDL_opengl.h:1572
int8_t Sint8
Definition: SDL_stdinc.h:173
int32_t Sint32
Definition: SDL_stdinc.h:197
int16_t Sint16
Definition: SDL_stdinc.h:185
uint16_t Uint16
Definition: SDL_stdinc.h:191
int64_t Sint64
Definition: SDL_stdinc.h:210

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_formatThe format of the source audio
src_channelsThe number of channels of the source audio
src_rateThe sampling rate of the source audio
dst_formatThe format of the desired audio output
dst_channelsThe number of channels of the desired audio output
dst_rateThe 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 1269 of file SDL_audiocvt.c.

1275{
1276 const int packetlen = 4096; /* !!! FIXME: good enough for now. */
1277 Uint8 pre_resample_channels;
1278 SDL_AudioStream *retval;
1279
1280 retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
1281 if (!retval) {
1282 return NULL;
1283 }
1284
1285 /* If increasing channels, do it after resampling, since we'd just
1286 do more work to resample duplicate channels. If we're decreasing, do
1287 it first so we resample the interpolated data instead of interpolating
1288 the resampled data (!!! FIXME: decide if that works in practice, though!). */
1289 pre_resample_channels = SDL_min(src_channels, dst_channels);
1290
1291 retval->first_run = SDL_TRUE;
1292 retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
1293 retval->src_format = src_format;
1294 retval->src_channels = src_channels;
1295 retval->src_rate = src_rate;
1296 retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
1297 retval->dst_format = dst_format;
1298 retval->dst_channels = dst_channels;
1299 retval->dst_rate = dst_rate;
1300 retval->pre_resample_channels = pre_resample_channels;
1301 retval->packetlen = packetlen;
1302 retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
1303 retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels;
1304 retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples ? retval->resampler_padding_samples : 1, sizeof (float));
1305
1306 if (retval->resampler_padding == NULL) {
1309 return NULL;
1310 }
1311
1312 retval->staging_buffer_size = ((retval->resampler_padding_samples / retval->pre_resample_channels) * retval->src_sample_frame_size);
1313 if (retval->staging_buffer_size > 0) {
1314 retval->staging_buffer = (Uint8 *) SDL_malloc(retval->staging_buffer_size);
1315 if (retval->staging_buffer == NULL) {
1318 return NULL;
1319 }
1320 }
1321
1322 /* Not resampling? It's an easy conversion (and maybe not even that!) */
1323 if (src_rate == dst_rate) {
1324 retval->cvt_before_resampling.needed = SDL_FALSE;
1325 if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
1327 return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
1328 }
1329 } else {
1330 /* Don't resample at first. Just get us to Float32 format. */
1331 /* !!! FIXME: convert to int32 on devices without hardware float. */
1332 if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
1334 return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
1335 }
1336
1337#ifdef HAVE_LIBSAMPLERATE_H
1338 SetupLibSampleRateResampling(retval);
1339#endif
1340
1341 if (!retval->resampler_func) {
1342 retval->resampler_state = SDL_calloc(retval->resampler_padding_samples, sizeof (float));
1343 if (!retval->resampler_state) {
1346 return NULL;
1347 }
1348
1349 if (SDL_PrepareResampleFilter() < 0) {
1350 SDL_free(retval->resampler_state);
1351 retval->resampler_state = NULL;
1353 return NULL;
1354 }
1355
1356 retval->resampler_func = SDL_ResampleAudioStream;
1357 retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
1358 retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
1359 }
1360
1361 /* Convert us to the final format after resampling. */
1362 if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
1364 return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
1365 }
1366 }
1367
1368 retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
1369 if (!retval->queue) {
1371 return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */
1372 }
1373
1374 return retval;
1375}
#define SDL_AUDIO_BITSIZE(x)
Definition: SDL_audio.h:75
#define AUDIO_F32SYS
Definition: SDL_audio.h:125
int SDL_BuildAudioCVT(SDL_AudioCVT *cvt, SDL_AudioFormat src_fmt, Uint8 src_channels, int src_rate, SDL_AudioFormat dst_fmt, Uint8 dst_channels, int dst_rate)
Definition: SDL_audiocvt.c:878
int SDL_PrepareResampleFilter(void)
Definition: SDL_audiocvt.c:435
static int SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
static void SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
static void SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
static int ResamplerPadding(const int inrate, const int outrate)
Definition: SDL_audiocvt.c:473
void SDL_FreeAudioStream(SDL_AudioStream *stream)
SDL_DataQueue * SDL_NewDataQueue(const size_t _packetlen, const size_t initialslack)
Definition: SDL_dataqueue.c:58
#define SDL_malloc
#define SDL_calloc
#define SDL_OutOfMemory()
Definition: SDL_error.h:52
@ SDL_FALSE
Definition: SDL_stdinc.h:163
#define SDL_min(x, y)
Definition: SDL_stdinc.h:406

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
    GLsizei samples
  • 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 1461 of file SDL_audio.c.

1462{
1463 SDL_AudioDeviceID id = 0;
1464
1465 /* Start up the audio driver, if necessary. This is legacy behaviour! */
1468 return -1;
1469 }
1470 }
1471
1472 /* SDL_OpenAudio() is legacy and can only act on Device ID #1. */
1473 if (open_devices[0] != NULL) {
1474 SDL_SetError("Audio device is already opened");
1475 return -1;
1476 }
1477
1478 if (obtained) {
1479 id = open_audio_device(NULL, 0, desired, obtained,
1481 } else {
1482 SDL_AudioSpec _obtained;
1483 SDL_zero(_obtained);
1484 id = open_audio_device(NULL, 0, desired, &_obtained, 0, 1);
1485 /* On successful open, copy calculated values into 'desired'. */
1486 if (id > 0) {
1487 desired->size = _obtained.size;
1488 desired->silence = _obtained.silence;
1489 }
1490 }
1491
1492 SDL_assert((id == 0) || (id == 1));
1493 return (id == 0) ? -1 : 0;
1494}
static SDL_AudioDeviceID open_audio_device(const char *devname, int iscapture, const SDL_AudioSpec *desired, SDL_AudioSpec *obtained, int allowed_changes, int min_id)
Definition: SDL_audio.c:1216
#define SDL_AUDIO_ALLOW_ANY_CHANGE
Definition: SDL_audio.h:144
Uint32 SDL_AudioDeviceID
Definition: SDL_audio.h:330
#define SDL_InitSubSystem
Uint32 size
Definition: SDL_audio.h:186
Uint8 silence
Definition: SDL_audio.h:183

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 1497 of file SDL_audio.c.

1500{
1501 return open_audio_device(device, iscapture, desired, obtained,
1502 allowed_changes, 2);
1503}

References device, and open_audio_device().

◆ SDL_PauseAudio()

void SDL_PauseAudio ( int  pause_on)

Definition at line 1539 of file SDL_audio.c.

1540{
1541 SDL_PauseAudioDevice(1, pause_on);
1542}
void SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
Definition: SDL_audio.c:1528

References SDL_PauseAudioDevice().

◆ SDL_PauseAudioDevice()

void SDL_PauseAudioDevice ( SDL_AudioDeviceID  dev,
int  pause_on 
)

◆ 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
devThe device ID to which we will queue audio.
dataThe data to queue to the device for later playback.
lenThe 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 598 of file SDL_audio.c.

599{
601 int rc = 0;
602
603 if (!device) {
604 return -1; /* get_audio_device() will have set the error state */
605 } else if (device->iscapture) {
606 return SDL_SetError("This is a capture device, queueing not allowed");
607 } else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
608 return SDL_SetError("Audio device has a callback, queueing not allowed");
609 }
610
611 if (len > 0) {
613 rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
615 }
616
617 return rc;
618}

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 1562 of file SDL_audio.c.

1563{
1564 /* Obtain a lock on the mixing buffers */
1566 if (device) {
1568 }
1569}

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

Referenced by SDL_UnlockAudio().