SDL  2.0
Data Structures | Macros | Typedefs | Functions
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

#define SDL_AUDIOCVT_MAX_FILTERS   9
 Upper limit of filters in SDL_AudioCVT.
#define SDL_AUDIOCVT_PACKED
#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
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
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)

Typedefs

typedef Uint16 SDL_AudioFormat
 Audio format flags.
typedef void(* SDL_AudioCallback )(void *userdata, Uint8 *stream, int len)
typedef void(* SDL_AudioFilter )(struct SDL_AudioCVT *cvt, SDL_AudioFormat format)
typedef Uint32 SDL_AudioDeviceID

Functions

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)
SDL_AudioSpec * SDL_LoadWAV_RW (SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len)
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)
void SDL_CloseAudio (void)
void SDL_CloseAudioDevice (SDL_AudioDeviceID dev)
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)
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().

int SDL_AudioInit (const char *driver_name)
void SDL_AudioQuit (void)
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.

void SDL_PauseAudio (int pause_on)
void SDL_PauseAudioDevice (SDL_AudioDeviceID dev, int pause_on)
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)

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)

Detailed Description

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

Definition in file SDL_audio.h.

Macro Definition Documentation

#define AUDIO_F32   AUDIO_F32LSB

Definition at line 114 of file SDL_audio.h.

Referenced by SDL_LoadWAV_RW().

#define AUDIO_F32LSB   0x8120

32-bit floating point samples

Definition at line 112 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), and SDL_SupportedAudioFormat().

#define AUDIO_F32MSB   0x9120

As above, but big-endian byte order

Definition at line 113 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), and SDL_SupportedAudioFormat().

#define AUDIO_F32SYS   AUDIO_F32LSB

Definition at line 125 of file SDL_audio.h.

Referenced by audio_initOpenCloseQuitAudio(), audio_pauseUnpauseAudio(), main(), SDL_Convert51To71(), SDL_Convert51ToQuad(), SDL_Convert51ToStereo(), SDL_Convert71To51(), SDL_Convert_S16_to_F32_Scalar(), SDL_Convert_S32_to_F32_Scalar(), SDL_Convert_S8_to_F32_Scalar(), SDL_Convert_U16_to_F32_Scalar(), SDL_Convert_U8_to_F32_Scalar(), SDL_ConvertMonoToStereo(), SDL_ConvertQuadTo51(), SDL_ConvertQuadToStereo(), SDL_ConvertStereoTo51(), SDL_ConvertStereoToMono(), SDL_ConvertStereoToQuad(), SDL_NewAudioStream(), and SDL_ResampleCVT().

#define AUDIO_S16   AUDIO_S16LSB

Definition at line 96 of file SDL_audio.h.

Referenced by prepare_audiospec(), SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), SDL_LoadWAV_RW(), SDLTest_CommonArg(), and SDLTest_CommonCreateState().

#define AUDIO_S16LSB   0x8010

Signed 16-bit samples

Definition at line 92 of file SDL_audio.h.

Referenced by audio_buildAudioCVT(), audio_buildAudioCVTNegative(), SDL_MixAudioFormat(), SDL_SupportedAudioFormat(), and SDLTest_CommonArg().

#define AUDIO_S16MSB   0x9010

As above, but big-endian byte order

Definition at line 94 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), SDL_SupportedAudioFormat(), and SDLTest_CommonArg().

#define AUDIO_S16SYS   AUDIO_S16LSB

Definition at line 123 of file SDL_audio.h.

Referenced by audio_initOpenCloseQuitAudio(), audio_lockUnlockOpenAudioDevice(), audio_openCloseAndGetAudioStatus(), audio_openCloseAudioDeviceConnected(), audio_pauseUnpauseAudio(), and SDL_Convert_F32_to_S16_Scalar().

#define AUDIO_S32   AUDIO_S32LSB

Definition at line 105 of file SDL_audio.h.

Referenced by SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), and SDL_LoadWAV_RW().

#define AUDIO_S32LSB   0x8020

32-bit integer samples

Definition at line 103 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), and SDL_SupportedAudioFormat().

#define AUDIO_S32MSB   0x9020

As above, but big-endian byte order

Definition at line 104 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), and SDL_SupportedAudioFormat().

#define AUDIO_S32SYS   AUDIO_S32LSB

Definition at line 124 of file SDL_audio.h.

Referenced by SDL_Convert_F32_to_S32_Scalar().

#define AUDIO_S8   0x8008

Signed 8-bit samples

Definition at line 90 of file SDL_audio.h.

Referenced by audio_buildAudioCVT(), audio_buildAudioCVTNegative(), SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), SDL_Convert_F32_to_S8_Scalar(), SDL_MixAudioFormat(), SDL_SupportedAudioFormat(), and SDLTest_CommonArg().

#define AUDIO_U16   AUDIO_U16LSB

Definition at line 95 of file SDL_audio.h.

Referenced by SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), and SDLTest_CommonArg().

#define AUDIO_U16LSB   0x0010

Unsigned 16-bit samples

Definition at line 91 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), SDL_SupportedAudioFormat(), and SDLTest_CommonArg().

#define AUDIO_U16MSB   0x1010

As above, but big-endian byte order

Definition at line 93 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), SDL_SupportedAudioFormat(), and SDLTest_CommonArg().

#define AUDIO_U16SYS   AUDIO_U16LSB

Definition at line 122 of file SDL_audio.h.

Referenced by SDL_Convert_F32_to_U16_Scalar().

#define AUDIO_U8   0x0008

Unsigned 8-bit samples

Definition at line 89 of file SDL_audio.h.

Referenced by SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), SDL_CalculateAudioSpec(), SDL_Convert_F32_to_U8_Scalar(), SDL_LoadWAV_RW(), SDL_MixAudioFormat(), SDL_SupportedAudioFormat(), and SDLTest_CommonArg().

#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.

Referenced by audio_lockUnlockOpenAudioDevice(), audio_openCloseAndGetAudioStatus(), audio_openCloseAudioDeviceConnected(), main(), and SDL_OpenAudio().

#define SDL_AUDIO_ALLOW_CHANNELS_CHANGE   0x00000004

Definition at line 142 of file SDL_audio.h.

Referenced by open_audio_device().

#define SDL_AUDIO_ALLOW_FORMAT_CHANGE   0x00000002

Definition at line 141 of file SDL_audio.h.

Referenced by open_audio_device().

#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE   0x00000001

Definition at line 140 of file SDL_audio.h.

Referenced by open_audio_device().

#define SDL_AUDIO_ALLOW_SAMPLES_CHANGE   0x00000008

Definition at line 143 of file SDL_audio.h.

Referenced by open_audio_device().

#define SDL_AUDIO_BITSIZE (   x)    (x & SDL_AUDIO_MASK_BITSIZE)

Definition at line 75 of file SDL_audio.h.

Referenced by main(), SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), SDL_CalculateAudioSpec(), SDL_Convert_Byteswap(), SDL_LoadWAV_RW(), and SDL_NewAudioStream().

#define SDL_AUDIO_ISBIGENDIAN (   x)    (x & SDL_AUDIO_MASK_ENDIAN)

Definition at line 77 of file SDL_audio.h.

Referenced by SDL_BuildAudioTypeCVTFromFloat(), and SDL_BuildAudioTypeCVTToFloat().

#define SDL_AUDIO_ISFLOAT (   x)    (x & SDL_AUDIO_MASK_DATATYPE)

Definition at line 76 of file SDL_audio.h.

Referenced by main(), SDL_BuildAudioTypeCVTFromFloat(), and SDL_BuildAudioTypeCVTToFloat().

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

Definition at line 79 of file SDL_audio.h.

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

Definition at line 80 of file SDL_audio.h.

#define SDL_AUDIO_ISSIGNED (   x)    (x & SDL_AUDIO_MASK_SIGNED)

Definition at line 78 of file SDL_audio.h.

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

Definition at line 81 of file SDL_audio.h.

#define SDL_AUDIO_MASK_BITSIZE   (0xFF)

Definition at line 71 of file SDL_audio.h.

#define SDL_AUDIO_MASK_DATATYPE   (1<<8)

Definition at line 72 of file SDL_audio.h.

#define SDL_AUDIO_MASK_ENDIAN   (1<<12)

Definition at line 73 of file SDL_audio.h.

Referenced by SDL_BuildAudioCVT(), SDL_BuildAudioTypeCVTFromFloat(), SDL_BuildAudioTypeCVTToFloat(), and SDL_Convert_Byteswap().

#define SDL_AUDIO_MASK_SIGNED   (1<<15)

Definition at line 74 of file SDL_audio.h.

#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.

Referenced by SDL_AddAudioCVTFilter(), SDL_BuildAudioResampleCVT(), and SDL_ResampleCVT().

#define SDL_AUDIOCVT_PACKED

Definition at line 223 of file SDL_audio.h.

#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 451 of file SDL_audio.h.

Referenced by main().

#define SDL_MIX_MAXVOLUME   128

Definition at line 616 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat().

Typedef Documentation

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_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.

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

Definition at line 193 of file SDL_audio.h.

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

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

int SDL_AudioInit ( const char *  driver_name)

Definition at line 932 of file SDL_audio.c.

References 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, SDL_AudioQuit, SDL_CreateMutex, SDL_getenv, SDL_INIT_AUDIO, SDL_SetError, SDL_strlen, SDL_strncasecmp, SDL_WasInit, and SDL_zero.

{
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_zero(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;
}
void SDL_AudioQuit ( void  )

Definition at line 1582 of file SDL_audio.c.

References close_audio_device(), SDL_AudioDriverImpl::Deinitialize, SDL_AudioDriver::detectionLock, free_device_list(), i, SDL_AudioDriver::impl, SDL_AudioDriver::inputDeviceCount, SDL_AudioDriver::inputDevices, SDL_AudioDriver::name, SDL_AudioDriver::outputDeviceCount, SDL_AudioDriver::outputDevices, SDL_arraysize, SDL_DestroyMutex, SDL_FreeResampleFilter(), and SDL_zero.

{
SDL_AudioDeviceID i;
if (!current_audio.name) { /* not initialized?! */
return;
}
for (i = 0; i < SDL_arraysize(open_devices); i++) {
close_audio_device(open_devices[i]);
}
free_device_list(&current_audio.outputDevices, &current_audio.outputDeviceCount);
free_device_list(&current_audio.inputDevices, &current_audio.inputDeviceCount);
/* Free the driver data */
current_audio.impl.Deinitialize();
SDL_DestroyMutex(current_audio.detectionLock);
SDL_zero(current_audio);
SDL_zero(open_devices);
#ifdef HAVE_LIBSAMPLERATE_H
UnloadLibSampleRate();
#endif
SDL_FreeResampleFilter();
}
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 1636 of file SDL_audiocvt.c.

References SDL_CountDataQueue().

{
return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
}
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 1642 of file SDL_audiocvt.c.

References SDL_ClearDataQueue(), SDL_InvalidParamError, and SDL_TRUE.

{
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;
}
}
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 1561 of file SDL_audiocvt.c.

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

{
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;
}
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 1615 of file SDL_audiocvt.c.

References SDL_InvalidParamError, SDL_ReadFromDataQueue(), and SDL_SetError.

{
#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);
}
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 1497 of file SDL_audiocvt.c.

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

{
/* !!! 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;
}
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 872 of file SDL_audiocvt.c.

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, 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_SetError, SDL_SupportedAudioFormat(), SDL_SupportedChannelCount(), SDL_zero, SDL_zerop, and SDL_AudioCVT::src_format.

{
/* 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 zero");
} else if (dst_rate == 0) {
return SDL_SetError("Destination rate is zero");
}
#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_zero(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);
}
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 668 of file SDL_audio.c.

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

{
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);
}
void SDL_CloseAudio ( void  )

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

Definition at line 1576 of file SDL_audio.c.

References SDL_CloseAudioDevice.

{
SDL_CloseAudioDevice(1);
}
void SDL_CloseAudioDevice ( SDL_AudioDeviceID  dev)

Definition at line 1570 of file SDL_audio.c.

References close_audio_device(), and get_audio_device().

{
close_audio_device(get_audio_device(devid));
}
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.

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.

{
/* !!! 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;
}
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 625 of file SDL_audio.c.

References SDL_AudioDevice::buffer_queue, SDL_AudioSpec::callback, SDL_AudioDevice::callbackspec, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDevice::iscapture, SDL_AudioDriverImpl::LockDevice, SDL_BufferQueueFillCallback(), SDL_ReadFromDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

{
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;
}
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 1658 of file SDL_audiocvt.c.

References SDL_free, and SDL_FreeDataQueue().

{
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);
}
}
void SDL_FreeWAV ( Uint8 *  audio_buf)

This function frees data previously allocated with SDL_LoadWAV_RW()

Definition at line 672 of file SDL_wave.c.

References SDL_free.

{
SDL_free(audio_buf);
}
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 1062 of file SDL_audio.c.

References 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.

{
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;
}
SDL_AudioStatus SDL_GetAudioDeviceStatus ( SDL_AudioDeviceID  dev)

Definition at line 1498 of file SDL_audio.c.

References device, SDL_AudioDevice::enabled, get_audio_device(), SDL_AudioDevice::paused, SDL_AtomicGet, SDL_AUDIO_PAUSED, SDL_AUDIO_PLAYING, and SDL_AUDIO_STOPPED.

{
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;
}
const char* SDL_GetAudioDriver ( int  index)

Definition at line 923 of file SDL_audio.c.

References AudioBootStrap::name, NULL, and SDL_GetNumAudioDrivers.

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

Definition at line 1514 of file SDL_audio.c.

References SDL_GetAudioDeviceStatus.

{
return SDL_GetAudioDeviceStatus(1);
}
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 997 of file SDL_audio.c.

References SDL_AudioDriver::name.

{
return current_audio.name;
}
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 1037 of file SDL_audio.c.

References SDL_AudioDriver::captureDevicesRemoved, clean_out_device_list(), 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.

{
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;
}
int SDL_GetNumAudioDrivers ( void  )

Definition at line 917 of file SDL_audio.c.

References SDL_arraysize.

{
return SDL_arraysize(bootstrap) - 1;
}
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 644 of file SDL_audio.c.

References SDL_AudioDevice::buffer_queue, SDL_AudioSpec::callback, SDL_AudioDevice::callbackspec, device, get_audio_device(), SDL_AudioDriverImpl::GetPendingBytes, SDL_AudioDriver::impl, SDL_AudioDriverImpl::LockDevice, retval, SDL_BufferQueueDrainCallback(), SDL_BufferQueueFillCallback(), SDL_CountDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

{
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) {
current_audio.impl.LockDevice(device);
retval = ((Uint32) SDL_CountDataQueue(device->buffer_queue)) + current_audio.impl.GetPendingBytes(device);
current_audio.impl.UnlockDevice(device);
} else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
current_audio.impl.LockDevice(device);
retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
current_audio.impl.UnlockDevice(device);
}
return retval;
}
SDL_AudioSpec* SDL_LoadWAV_RW ( SDL_RWops *  src,
int  freesrc,
SDL_AudioSpec *  spec,
Uint8 **  audio_buf,
Uint32 *  audio_len 
)

This function loads a WAVE from the data source, automatically freeing that source if freesrc is non-zero. For example, to load a WAVE file, you could do:

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

If this function succeeds, it returns the given SDL_AudioSpec, filled with the audio data format of the wave data, and sets *audio_buf to a malloc()'d buffer containing the audio data, and sets *audio_len to the length of that audio buffer, in bytes. You need to free the audio buffer with SDL_FreeWAV() when you are done with it.

This function returns NULL and sets the SDL error message if the wave file cannot be opened, uses an unknown data format, or is corrupt. Currently raw and MS-ADPCM WAVE files are supported.

Definition at line 448 of file SDL_wave.c.

References AUDIO_F32, AUDIO_S16, AUDIO_S32, AUDIO_U8, BEXT, WaveFMT::bitspersample, WaveFMT::channels, SDL_AudioSpec::channels, ConvertSint24ToSint32(), DATA, Chunk::data, done, WaveFMT::encoding, EXTENSIBLE_CODE, extensible_ieee_guid, extensible_pcm_guid, FACT, FMT, SDL_AudioSpec::format, SDL_AudioSpec::freq, WaveFMT::frequency, IEEE_FLOAT_CODE, IMA_ADPCM_CODE, IMA_ADPCM_decode(), InitIMA_ADPCM(), InitMS_ADPCM(), JUNK, Chunk::length, LIST, Chunk::magic, MP3_CODE, MS_ADPCM_CODE, MS_ADPCM_decode(), NULL, PCM_CODE, ReadChunk(), RIFF, RW_SEEK_CUR, SDL_AudioSpec::samples, SDL_AUDIO_BITSIZE, SDL_free, SDL_memcmp, SDL_ReadLE32, SDL_RWclose, SDL_RWseek, SDL_SetError, SDL_SwapLE16, SDL_SwapLE32, SDL_zero, SDL_zerop, WaveExtensibleFMT::size, WaveExtensibleFMT::subformat, and WAVE.

{
int was_error;
Chunk chunk;
int lenread;
int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
int samplesize;
/* WAV magic header */
Uint32 RIFFchunk;
Uint32 wavelen = 0;
Uint32 WAVEmagic;
Uint32 headerDiff = 0;
/* FMT chunk */
WaveFMT *format = NULL;
WaveExtensibleFMT *ext = NULL;
SDL_zero(chunk);
/* Make sure we are passed a valid data source */
was_error = 0;
if (src == NULL) {
was_error = 1;
goto done;
}
/* Check the magic header */
RIFFchunk = SDL_ReadLE32(src);
wavelen = SDL_ReadLE32(src);
if (wavelen == WAVE) { /* The RIFFchunk has already been read */
WAVEmagic = wavelen;
wavelen = RIFFchunk;
RIFFchunk = RIFF;
} else {
WAVEmagic = SDL_ReadLE32(src);
}
if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) {
SDL_SetError("Unrecognized file type (not WAVE)");
was_error = 1;
goto done;
}
headerDiff += sizeof(Uint32); /* for WAVE */
/* Read the audio data format chunk */
chunk.data = NULL;
do {
SDL_free(chunk.data);
chunk.data = NULL;
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
/* 2 Uint32's for chunk header+len, plus the lenread */
headerDiff += lenread + 2 * sizeof(Uint32);
} while ((chunk.magic == FACT) || (chunk.magic == LIST) || (chunk.magic == BEXT) || (chunk.magic == JUNK));
/* Decode the audio data format */
format = (WaveFMT *) chunk.data;
if (chunk.magic != FMT) {
SDL_SetError("Complex WAVE files not supported");
was_error = 1;
goto done;
}
IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
switch (SDL_SwapLE16(format->encoding)) {
case PCM_CODE:
/* We can understand this */
break;
case IEEE_FLOAT_CODE:
IEEE_float_encoded = 1;
/* We can understand this */
break;
case MS_ADPCM_CODE:
/* Try to understand this */
if (InitMS_ADPCM(format) < 0) {
was_error = 1;
goto done;
}
MS_ADPCM_encoded = 1;
break;
case IMA_ADPCM_CODE:
/* Try to understand this */
if (InitIMA_ADPCM(format) < 0) {
was_error = 1;
goto done;
}
IMA_ADPCM_encoded = 1;
break;
case EXTENSIBLE_CODE:
/* note that this ignores channel masks, smaller valid bit counts
inside a larger container, and most subtypes. This is just enough
to get things that didn't really _need_ WAVE_FORMAT_EXTENSIBLE
to be useful working when they use this format flag. */
ext = (WaveExtensibleFMT *) format;
if (SDL_SwapLE16(ext->size) < 22) {
SDL_SetError("bogus extended .wav header");
was_error = 1;
goto done;
}
if (SDL_memcmp(ext->subformat, extensible_pcm_guid, 16) == 0) {
break; /* cool. */
} else if (SDL_memcmp(ext->subformat, extensible_ieee_guid, 16) == 0) {
IEEE_float_encoded = 1;
break;
}
break;
case MP3_CODE:
SDL_SetError("MPEG Layer 3 data not supported");
was_error = 1;
goto done;
default:
SDL_SetError("Unknown WAVE data format: 0x%.4x",
SDL_SwapLE16(format->encoding));
was_error = 1;
goto done;
}
SDL_zerop(spec);
spec->freq = SDL_SwapLE32(format->frequency);
if (IEEE_float_encoded) {
if ((SDL_SwapLE16(format->bitspersample)) != 32) {
was_error = 1;
} else {
spec->format = AUDIO_F32;
}
} else {
switch (SDL_SwapLE16(format->bitspersample)) {
case 4:
if (MS_ADPCM_encoded || IMA_ADPCM_encoded) {
spec->format = AUDIO_S16;
} else {
was_error = 1;
}
break;
case 8:
spec->format = AUDIO_U8;
break;
case 16:
spec->format = AUDIO_S16;
break;
case 24: /* convert this. */
spec->format = AUDIO_S32;
break;
case 32:
spec->format = AUDIO_S32;
break;
default:
was_error = 1;
break;
}
}
if (was_error) {
SDL_SetError("Unknown %d-bit PCM data format",
SDL_SwapLE16(format->bitspersample));
goto done;
}
spec->channels = (Uint8) SDL_SwapLE16(format->channels);
spec->samples = 4096; /* Good default buffer size */
/* Read the audio data chunk */
*audio_buf = NULL;
do {
SDL_free(*audio_buf);
*audio_buf = NULL;
lenread = ReadChunk(src, &chunk);
if (lenread < 0) {
was_error = 1;
goto done;
}
*audio_len = lenread;
*audio_buf = chunk.data;
if (chunk.magic != DATA)
headerDiff += lenread + 2 * sizeof(Uint32);
} while (chunk.magic != DATA);
headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */
if (MS_ADPCM_encoded) {
if (MS_ADPCM_decode(audio_buf, audio_len) < 0) {
was_error = 1;
goto done;
}
}
if (IMA_ADPCM_encoded) {
if (IMA_ADPCM_decode(audio_buf, audio_len) < 0) {
was_error = 1;
goto done;
}
}
if (SDL_SwapLE16(format->bitspersample) == 24) {
if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) {
was_error = 1;
goto done;
}
}
/* Don't return a buffer that isn't a multiple of samplesize */
samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
*audio_len &= ~(samplesize - 1);
done:
SDL_free(format);
if (src) {
if (freesrc) {
SDL_RWclose(src);
} else {
/* seek to the end of the file (given by the RIFF chunk) */
SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
}
}
if (was_error) {
spec = NULL;
}
return (spec);
}
void SDL_LockAudio ( void  )

Definition at line 1548 of file SDL_audio.c.

References SDL_LockAudioDevice.

{
SDL_LockAudioDevice(1);
}
void SDL_LockAudioDevice ( SDL_AudioDeviceID  dev)

Definition at line 1538 of file SDL_audio.c.

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

{
/* Obtain a lock on the mixing buffers */
SDL_AudioDevice *device = get_audio_device(devid);
if (device) {
current_audio.impl.LockDevice(device);
}
}
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 1681 of file SDL_audio.c.

References SDL_AudioDevice::callbackspec, device, SDL_AudioSpec::format, get_audio_device(), NULL, and SDL_MixAudioFormat.

{
/* 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);
}
}
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.

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, F, mix8, SDL_MIX_MAXVOLUME, SDL_SetError, SDL_SwapBE32, SDL_SwapFloatBE, SDL_SwapFloatLE, and SDL_SwapLE32.

{
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;
}
}
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 1259 of file SDL_audiocvt.c.

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.

{
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;
}
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
  • 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 1453 of file SDL_audio.c.

References NULL, open_audio_device(), 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_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;
}
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 1489 of file SDL_audio.c.

References open_audio_device().

{
return open_audio_device(device, iscapture, desired, obtained,
allowed_changes, 2);
}
void SDL_PauseAudio ( int  pause_on)

Definition at line 1531 of file SDL_audio.c.

References SDL_PauseAudioDevice.

{
SDL_PauseAudioDevice(1, pause_on);
}
void SDL_PauseAudioDevice ( SDL_AudioDeviceID  dev,
int  pause_on 
)

Definition at line 1520 of file SDL_audio.c.

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

{
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);
}
}
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 602 of file SDL_audio.c.

References SDL_AudioDevice::buffer_queue, SDL_AudioSpec::callback, SDL_AudioDevice::callbackspec, device, get_audio_device(), SDL_AudioDriver::impl, SDL_AudioDevice::iscapture, SDL_AudioDriverImpl::LockDevice, SDL_BufferQueueDrainCallback(), SDL_SetError, SDL_WriteToDataQueue(), and SDL_AudioDriverImpl::UnlockDevice.

{
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;
}
void SDL_UnlockAudio ( void  )

Definition at line 1564 of file SDL_audio.c.

References SDL_UnlockAudioDevice.

{
SDL_UnlockAudioDevice(1);
}
void SDL_UnlockAudioDevice ( SDL_AudioDeviceID  dev)

Definition at line 1554 of file SDL_audio.c.

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

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