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

#define SDL_AUDIOCVT_MAX_FILTERS   9
 Upper limit of filters in SDL_AudioCVT. More...
 
#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_ANY_CHANGE   (SDL_AUDIO_ALLOW_FREQUENCY_CHANGE|SDL_AUDIO_ALLOW_FORMAT_CHANGE|SDL_AUDIO_ALLOW_CHANNELS_CHANGE)
 

Typedefs

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

◆ AUDIO_F32

#define AUDIO_F32   AUDIO_F32LSB

Definition at line 114 of file SDL_audio.h.

Referenced by SDL_LoadWAV_RW().

◆ AUDIO_F32LSB

#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().

◆ AUDIO_F32MSB

#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().

◆ AUDIO_F32SYS

◆ AUDIO_S16

◆ AUDIO_S16LSB

#define AUDIO_S16LSB   0x8010

◆ AUDIO_S16MSB

#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().

◆ AUDIO_S16SYS

◆ AUDIO_S32

#define AUDIO_S32   AUDIO_S32LSB

◆ AUDIO_S32LSB

#define AUDIO_S32LSB   0x8020

32-bit integer samples

Definition at line 103 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat(), and SDL_SupportedAudioFormat().

◆ AUDIO_S32MSB

#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().

◆ AUDIO_S32SYS

#define AUDIO_S32SYS   AUDIO_S32LSB

Definition at line 124 of file SDL_audio.h.

Referenced by SDL_Convert_F32_to_S32_Scalar().

◆ AUDIO_S8

◆ AUDIO_U16

#define AUDIO_U16   AUDIO_U16LSB

◆ AUDIO_U16LSB

#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().

◆ AUDIO_U16MSB

#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().

◆ AUDIO_U16SYS

#define AUDIO_U16SYS   AUDIO_U16LSB

Definition at line 122 of file SDL_audio.h.

Referenced by SDL_Convert_F32_to_S32_Scalar(), and SDL_Convert_F32_to_U16_Scalar().

◆ AUDIO_U8

◆ 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.

Referenced by open_audio_device().

◆ SDL_AUDIO_ALLOW_FORMAT_CHANGE

#define SDL_AUDIO_ALLOW_FORMAT_CHANGE   0x00000002

Definition at line 141 of file SDL_audio.h.

Referenced by open_audio_device().

◆ SDL_AUDIO_ALLOW_FREQUENCY_CHANGE

#define SDL_AUDIO_ALLOW_FREQUENCY_CHANGE   0x00000001

Definition at line 140 of file SDL_audio.h.

Referenced by open_audio_device().

◆ SDL_AUDIO_BITSIZE

◆ SDL_AUDIO_ISBIGENDIAN

#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().

◆ SDL_AUDIO_ISFLOAT

#define SDL_AUDIO_ISFLOAT (   x)    (x & SDL_AUDIO_MASK_DATATYPE)

Definition at line 76 of file SDL_audio.h.

Referenced by SDL_BuildAudioTypeCVTFromFloat(), and SDL_BuildAudioTypeCVTToFloat().

◆ 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)

◆ 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 202 of file SDL_audio.h.

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

◆ SDL_AUDIOCVT_PACKED

#define SDL_AUDIOCVT_PACKED

Definition at line 222 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 450 of file SDL_audio.h.

Referenced by main().

◆ SDL_MIX_MAXVOLUME

#define SDL_MIX_MAXVOLUME   128

Definition at line 615 of file SDL_audio.h.

Referenced by SDL_MixAudioFormat().

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 162 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 329 of file SDL_audio.h.

◆ SDL_AudioFilter

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

Definition at line 192 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 394 of file SDL_audio.h.

Function Documentation

◆ SDL_AudioInit()

int SDL_AudioInit ( const char *  driver_name)

Definition at line 885 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.

886 {
887  int i = 0;
888  int initialized = 0;
889  int tried_to_init = 0;
890 
892  SDL_AudioQuit(); /* shutdown driver if already running. */
893  }
894 
897 
898  /* Select the proper audio driver */
899  if (driver_name == NULL) {
900  driver_name = SDL_getenv("SDL_AUDIODRIVER");
901  }
902 
903  for (i = 0; (!initialized) && (bootstrap[i]); ++i) {
904  /* make sure we should even try this driver before doing so... */
905  const AudioBootStrap *backend = bootstrap[i];
906  if ((driver_name && (SDL_strncasecmp(backend->name, driver_name, SDL_strlen(driver_name)) != 0)) ||
907  (!driver_name && backend->demand_only)) {
908  continue;
909  }
910 
911  tried_to_init = 1;
913  current_audio.name = backend->name;
914  current_audio.desc = backend->desc;
915  initialized = backend->init(&current_audio.impl);
916  }
917 
918  if (!initialized) {
919  /* specific drivers will set the error message if they fail... */
920  if (!tried_to_init) {
921  if (driver_name) {
922  SDL_SetError("Audio target '%s' not available", driver_name);
923  } else {
924  SDL_SetError("No available audio device");
925  }
926  }
927 
929  return -1; /* No driver was available, so fail. */
930  }
931 
933 
935 
936  /* Make sure we have a list of devices available at startup. */
938 
939 #ifdef HAVE_LIBSAMPLERATE_H
940  LoadLibSampleRate();
941 #endif
942 
943  return 0;
944 }
static SDL_AudioDevice * open_devices[16]
Definition: SDL_audio.c:34
const char * name
Definition: SDL_sysaudio.h:110
void(* DetectDevices)(void)
Definition: SDL_sysaudio.h:67
const char * name
Definition: SDL_sysaudio.h:177
#define SDL_CreateMutex
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
#define SDL_strncasecmp
SDL_mutex * detectionLock
Definition: SDL_sysaudio.h:119
const char * desc
Definition: SDL_sysaudio.h:114
void SDL_AudioQuit(void)
Definition: SDL_audio.c:1527
#define SDL_zero(x)
Definition: SDL_stdinc.h:416
#define SDL_getenv
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
static const AudioBootStrap *const bootstrap[]
Definition: SDL_audio.c:37
#define NULL
Definition: begin_code.h:164
#define SDL_SetError
#define SDL_INIT_AUDIO
Definition: SDL.h:77
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
#define SDL_strlen
int(* init)(SDL_AudioDriverImpl *impl)
Definition: SDL_sysaudio.h:179
const char * desc
Definition: SDL_sysaudio.h:178
#define SDL_WasInit
static void finish_audio_entry_points_init(void)
Definition: SDL_audio.c:334

◆ SDL_AudioQuit()

void SDL_AudioQuit ( void  )

Definition at line 1527 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.

Referenced by SDL_AudioInit().

1528 {
1530 
1531  if (!current_audio.name) { /* not initialized?! */
1532  return;
1533  }
1534 
1535  for (i = 0; i < SDL_arraysize(open_devices); i++) {
1537  }
1538 
1541 
1542  /* Free the driver data */
1544 
1546 
1549 
1550 #ifdef HAVE_LIBSAMPLERATE_H
1551  UnloadLibSampleRate();
1552 #endif
1553 
1555 }
static SDL_AudioDevice * open_devices[16]
Definition: SDL_audio.c:34
const char * name
Definition: SDL_sysaudio.h:110
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
SDL_mutex * detectionLock
Definition: SDL_sysaudio.h:119
void SDL_FreeResampleFilter(void)
Definition: SDL_audiocvt.c:464
void(* Deinitialize)(void)
Definition: SDL_sysaudio.h:83
#define SDL_zero(x)
Definition: SDL_stdinc.h:416
SDL_AudioDeviceItem * outputDevices
Definition: SDL_sysaudio.h:124
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
static void free_device_list(SDL_AudioDeviceItem **devices, int *devCount)
Definition: SDL_audio.c:415
#define SDL_DestroyMutex
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
Uint32 SDL_AudioDeviceID
Definition: SDL_audio.h:329
#define SDL_arraysize(array)
Definition: SDL_stdinc.h:93
SDL_AudioDeviceItem * inputDevices
Definition: SDL_sysaudio.h:125
static void close_audio_device(SDL_AudioDevice *device)
Definition: SDL_audio.c:1051

◆ 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 1636 of file SDL_audiocvt.c.

References SDL_CountDataQueue().

1637 {
1638  return stream ? (int) SDL_CountDataQueue(stream->queue) : 0;
1639 }
GLuint GLuint stream
size_t SDL_CountDataQueue(SDL_DataQueue *queue)

◆ 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 1642 of file SDL_audiocvt.c.

References SDL_ClearDataQueue(), SDL_InvalidParamError, and SDL_TRUE.

1643 {
1644  if (!stream) {
1645  SDL_InvalidParamError("stream");
1646  } else {
1647  SDL_ClearDataQueue(stream->queue, stream->packetlen * 2);
1648  if (stream->reset_resampler_func) {
1649  stream->reset_resampler_func(stream);
1650  }
1651  stream->first_run = SDL_TRUE;
1652  stream->staging_buffer_filled = 0;
1653  }
1654 }
#define SDL_InvalidParamError(param)
Definition: SDL_error.h:54
GLuint GLuint stream
void SDL_ClearDataQueue(SDL_DataQueue *queue, const size_t slack)
Definition: SDL_dataqueue.c:98

◆ 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 1561 of file SDL_audiocvt.c.

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

1562 {
1563  if (!stream) {
1564  return SDL_InvalidParamError("stream");
1565  }
1566 
1567  #if DEBUG_AUDIOSTREAM
1568  printf("AUDIOSTREAM: flushing! staging_buffer_filled=%d bytes\n", stream->staging_buffer_filled);
1569  #endif
1570 
1571  /* shouldn't use a staging buffer if we're not resampling. */
1572  SDL_assert((stream->dst_rate != stream->src_rate) || (stream->staging_buffer_filled == 0));
1573 
1574  if (stream->staging_buffer_filled > 0) {
1575  /* push the staging buffer + silence. We need to flush out not just
1576  the staging buffer, but the piece that the stream was saving off
1577  for right-side resampler padding. */
1578  const SDL_bool first_run = stream->first_run;
1579  const int filled = stream->staging_buffer_filled;
1580  int actual_input_frames = filled / stream->src_sample_frame_size;
1581  if (!first_run)
1582  actual_input_frames += stream->resampler_padding_samples / stream->pre_resample_channels;
1583 
1584  if (actual_input_frames > 0) { /* don't bother if nothing to flush. */
1585  /* This is how many bytes we're expecting without silence appended. */
1586  int flush_remaining = ((int) SDL_ceil(actual_input_frames * stream->rate_incr)) * stream->dst_sample_frame_size;
1587 
1588  #if DEBUG_AUDIOSTREAM
1589  printf("AUDIOSTREAM: flushing with padding to get max %d bytes!\n", flush_remaining);
1590  #endif
1591 
1592  SDL_memset(stream->staging_buffer + filled, '\0', stream->staging_buffer_size - filled);
1593  if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
1594  return -1;
1595  }
1596 
1597  /* we have flushed out (or initially filled) the pending right-side
1598  resampler padding, but we need to push more silence to guarantee
1599  the staging buffer is fully flushed out, too. */
1600  SDL_memset(stream->staging_buffer, '\0', filled);
1601  if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, &flush_remaining) < 0) {
1602  return -1;
1603  }
1604  }
1605  }
1606 
1607  stream->staging_buffer_filled = 0;
1608  stream->first_run = SDL_TRUE;
1609 
1610  return 0;
1611 }
#define SDL_ceil
#define SDL_InvalidParamError(param)
Definition: SDL_error.h:54
GLuint GLuint stream
static int SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, int *maxputbytes)
#define SDL_assert(condition)
Definition: SDL_assert.h:169
SDL_bool
Definition: SDL_stdinc.h:139
#define SDL_memset
#define DEBUG_AUDIOSTREAM
Definition: SDL_audiocvt.c:37

◆ 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 1615 of file SDL_audiocvt.c.

References SDL_InvalidParamError, SDL_ReadFromDataQueue(), and SDL_SetError.

1616 {
1617  #if DEBUG_AUDIOSTREAM
1618  printf("AUDIOSTREAM: want to get %d converted bytes\n", len);
1619  #endif
1620 
1621  if (!stream) {
1622  return SDL_InvalidParamError("stream");
1623  } else if (!buf) {
1624  return SDL_InvalidParamError("buf");
1625  } else if (len <= 0) {
1626  return 0; /* nothing to do. */
1627  } else if ((len % stream->dst_sample_frame_size) != 0) {
1628  return SDL_SetError("Can't request partial sample frames");
1629  }
1630 
1631  return (int) SDL_ReadFromDataQueue(stream->queue, buf, len);
1632 }
#define SDL_InvalidParamError(param)
Definition: SDL_error.h:54
GLenum GLsizei len
GLuint GLuint stream
size_t SDL_ReadFromDataQueue(SDL_DataQueue *queue, void *_buf, const size_t _len)
GLenum GLuint GLenum GLsizei const GLchar * buf
#define 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 1497 of file SDL_audiocvt.c.

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

1498 {
1499  /* !!! FIXME: several converters can take advantage of SIMD, but only
1500  !!! FIXME: if the data is aligned to 16 bytes. EnsureStreamBufferSize()
1501  !!! FIXME: guarantees the buffer will align, but the
1502  !!! FIXME: converters will iterate over the data backwards if
1503  !!! FIXME: the output grows, and this means we won't align if buflen
1504  !!! FIXME: isn't a multiple of 16. In these cases, we should chop off
1505  !!! FIXME: a few samples at the end and convert them separately. */
1506 
1507  #if DEBUG_AUDIOSTREAM
1508  printf("AUDIOSTREAM: wants to put %d preconverted bytes\n", buflen);
1509  #endif
1510 
1511  if (!stream) {
1512  return SDL_InvalidParamError("stream");
1513  } else if (!buf) {
1514  return SDL_InvalidParamError("buf");
1515  } else if (len == 0) {
1516  return 0; /* nothing to do. */
1517  } else if ((len % stream->src_sample_frame_size) != 0) {
1518  return SDL_SetError("Can't add partial sample frames");
1519  }
1520 
1521  if (!stream->cvt_before_resampling.needed &&
1522  (stream->dst_rate == stream->src_rate) &&
1523  !stream->cvt_after_resampling.needed) {
1524  #if DEBUG_AUDIOSTREAM
1525  printf("AUDIOSTREAM: no conversion needed at all, queueing %d bytes.\n", len);
1526  #endif
1527  return SDL_WriteToDataQueue(stream->queue, buf, len);
1528  }
1529 
1530  while (len > 0) {
1531  int amount;
1532 
1533  /* If we don't have a staging buffer or we're given enough data that
1534  we don't need to store it for later, skip the staging process.
1535  */
1536  if (!stream->staging_buffer_filled && len >= stream->staging_buffer_size) {
1538  }
1539 
1540  /* If there's not enough data to fill the staging buffer, just save it */
1541  if ((stream->staging_buffer_filled + len) < stream->staging_buffer_size) {
1542  SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, len);
1543  stream->staging_buffer_filled += len;
1544  return 0;
1545  }
1546 
1547  /* Fill the staging buffer, process it, and continue */
1548  amount = (stream->staging_buffer_size - stream->staging_buffer_filled);
1549  SDL_assert(amount > 0);
1550  SDL_memcpy(stream->staging_buffer + stream->staging_buffer_filled, buf, amount);
1551  stream->staging_buffer_filled = 0;
1552  if (SDL_AudioStreamPutInternal(stream, stream->staging_buffer, stream->staging_buffer_size, NULL) < 0) {
1553  return -1;
1554  }
1555  buf = (void *)((Uint8 *)buf + amount);
1556  len -= amount;
1557  }
1558  return 0;
1559 }
int SDL_WriteToDataQueue(SDL_DataQueue *queue, const void *_data, const size_t _len)
#define SDL_InvalidParamError(param)
Definition: SDL_error.h:54
GLenum GLsizei len
#define SDL_memcpy
GLuint GLuint stream
uint8_t Uint8
Definition: SDL_stdinc.h:157
static int SDL_AudioStreamPutInternal(SDL_AudioStream *stream, const void *buf, int len, int *maxputbytes)
GLenum GLuint GLenum GLsizei const GLchar * buf
#define SDL_assert(condition)
Definition: SDL_assert.h:169
#define NULL
Definition: begin_code.h:164
#define SDL_SetError

◆ 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 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.

Referenced by SDL_NewAudioStream().

875 {
876  /* Sanity check target pointer */
877  if (cvt == NULL) {
878  return SDL_InvalidParamError("cvt");
879  }
880 
881  /* Make sure we zero out the audio conversion before error checking */
882  SDL_zerop(cvt);
883 
884  if (!SDL_SupportedAudioFormat(src_fmt)) {
885  return SDL_SetError("Invalid source format");
886  } else if (!SDL_SupportedAudioFormat(dst_fmt)) {
887  return SDL_SetError("Invalid destination format");
888  } else if (!SDL_SupportedChannelCount(src_channels)) {
889  return SDL_SetError("Invalid source channels");
890  } else if (!SDL_SupportedChannelCount(dst_channels)) {
891  return SDL_SetError("Invalid destination channels");
892  } else if (src_rate == 0) {
893  return SDL_SetError("Source rate is zero");
894  } else if (dst_rate == 0) {
895  return SDL_SetError("Destination rate is zero");
896  }
897 
898 #if DEBUG_CONVERT
899  printf("Build format %04x->%04x, channels %u->%u, rate %d->%d\n",
900  src_fmt, dst_fmt, src_channels, dst_channels, src_rate, dst_rate);
901 #endif
902 
903  /* Start off with no conversion necessary */
904  cvt->src_format = src_fmt;
905  cvt->dst_format = dst_fmt;
906  cvt->needed = 0;
907  cvt->filter_index = 0;
908  SDL_zero(cvt->filters);
909  cvt->len_mult = 1;
910  cvt->len_ratio = 1.0;
911  cvt->rate_incr = ((double) dst_rate) / ((double) src_rate);
912 
913  /* Make sure we've chosen audio conversion functions (MMX, scalar, etc.) */
915 
916  /* Type conversion goes like this now:
917  - byteswap to CPU native format first if necessary.
918  - convert to native Float32 if necessary.
919  - resample and change channel count if necessary.
920  - convert back to native format.
921  - byteswap back to foreign format if necessary.
922 
923  The expectation is we can process data faster in float32
924  (possibly with SIMD), and making several passes over the same
925  buffer is likely to be CPU cache-friendly, avoiding the
926  biggest performance hit in modern times. Previously we had
927  (script-generated) custom converters for every data type and
928  it was a bloat on SDL compile times and final library size. */
929 
930  /* see if we can skip float conversion entirely. */
931  if (src_rate == dst_rate && src_channels == dst_channels) {
932  if (src_fmt == dst_fmt) {
933  return 0;
934  }
935 
936  /* just a byteswap needed? */
937  if ((src_fmt & ~SDL_AUDIO_MASK_ENDIAN) == (dst_fmt & ~SDL_AUDIO_MASK_ENDIAN)) {
939  return -1;
940  }
941  cvt->needed = 1;
942  return 1;
943  }
944  }
945 
946  /* Convert data types, if necessary. Updates (cvt). */
947  if (SDL_BuildAudioTypeCVTToFloat(cvt, src_fmt) < 0) {
948  return -1; /* shouldn't happen, but just in case... */
949  }
950 
951  /* Channel conversion */
952  if (src_channels < dst_channels) {
953  /* Upmixing */
954  /* Mono -> Stereo [-> ...] */
955  if ((src_channels == 1) && (dst_channels > 1)) {
957  return -1;
958  }
959  cvt->len_mult *= 2;
960  src_channels = 2;
961  cvt->len_ratio *= 2;
962  }
963  /* [Mono ->] Stereo -> 5.1 [-> 7.1] */
964  if ((src_channels == 2) && (dst_channels >= 6)) {
966  return -1;
967  }
968  src_channels = 6;
969  cvt->len_mult *= 3;
970  cvt->len_ratio *= 3;
971  }
972  /* Quad -> 5.1 [-> 7.1] */
973  if ((src_channels == 4) && (dst_channels >= 6)) {
975  return -1;
976  }
977  src_channels = 6;
978  cvt->len_mult = (cvt->len_mult * 3 + 1) / 2;
979  cvt->len_ratio *= 1.5;
980  }
981  /* [[Mono ->] Stereo ->] 5.1 -> 7.1 */
982  if ((src_channels == 6) && (dst_channels == 8)) {
984  return -1;
985  }
986  src_channels = 8;
987  cvt->len_mult = (cvt->len_mult * 4 + 2) / 3;
988  /* Should be numerically exact with every valid input to this
989  function */
990  cvt->len_ratio = cvt->len_ratio * 4 / 3;
991  }
992  /* [Mono ->] Stereo -> Quad */
993  if ((src_channels == 2) && (dst_channels == 4)) {
995  return -1;
996  }
997  src_channels = 4;
998  cvt->len_mult *= 2;
999  cvt->len_ratio *= 2;
1000  }
1001  } else if (src_channels > dst_channels) {
1002  /* Downmixing */
1003  /* 7.1 -> 5.1 [-> Stereo [-> Mono]] */
1004  /* 7.1 -> 5.1 [-> Quad] */
1005  if ((src_channels == 8) && (dst_channels <= 6)) {
1006  if (SDL_AddAudioCVTFilter(cvt, SDL_Convert71To51) < 0) {
1007  return -1;
1008  }
1009  src_channels = 6;
1010  cvt->len_ratio *= 0.75;
1011  }
1012  /* [7.1 ->] 5.1 -> Stereo [-> Mono] */
1013  if ((src_channels == 6) && (dst_channels <= 2)) {
1015  return -1;
1016  }
1017  src_channels = 2;
1018  cvt->len_ratio /= 3;
1019  }
1020  /* 5.1 -> Quad */
1021  if ((src_channels == 6) && (dst_channels == 4)) {
1023  return -1;
1024  }
1025  src_channels = 4;
1026  cvt->len_ratio = cvt->len_ratio * 2 / 3;
1027  }
1028  /* Quad -> Stereo [-> Mono] */
1029  if ((src_channels == 4) && (dst_channels <= 2)) {
1031  return -1;
1032  }
1033  src_channels = 2;
1034  cvt->len_ratio /= 2;
1035  }
1036  /* [... ->] Stereo -> Mono */
1037  if ((src_channels == 2) && (dst_channels == 1)) {
1039 
1040  #if HAVE_SSE3_INTRINSICS
1041  if (SDL_HasSSE3()) {
1042  filter = SDL_ConvertStereoToMono_SSE3;
1043  }
1044  #endif
1045 
1046  if (!filter) {
1047  filter = SDL_ConvertStereoToMono;
1048  }
1049 
1050  if (SDL_AddAudioCVTFilter(cvt, filter) < 0) {
1051  return -1;
1052  }
1053 
1054  src_channels = 1;
1055  cvt->len_ratio /= 2;
1056  }
1057  }
1058 
1059  if (src_channels != dst_channels) {
1060  /* All combinations of supported channel counts should have been
1061  handled by now, but let's be defensive */
1062  return SDL_SetError("Invalid channel combination");
1063  }
1064 
1065  /* Do rate conversion, if necessary. Updates (cvt). */
1066  if (SDL_BuildAudioResampleCVT(cvt, dst_channels, src_rate, dst_rate) < 0) {
1067  return -1; /* shouldn't happen, but just in case... */
1068  }
1069 
1070  /* Move to final data type. */
1071  if (SDL_BuildAudioTypeCVTFromFloat(cvt, dst_fmt) < 0) {
1072  return -1; /* shouldn't happen, but just in case... */
1073  }
1074 
1075  cvt->needed = (cvt->filter_index != 0);
1076  return (cvt->needed);
1077 }
static void SDL_Convert71To51(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:151
void SDL_ChooseAudioConverters(void)
int filter_index
Definition: SDL_audio.h:237
static void SDL_Convert51ToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:103
static SDL_bool SDL_SupportedAudioFormat(const SDL_AudioFormat fmt)
Definition: SDL_audiocvt.c:825
double len_ratio
Definition: SDL_audio.h:235
static void SDL_ConvertStereoToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:305
#define SDL_HasSSE3
static void SDL_Convert51ToQuad(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:181
#define SDL_AUDIO_MASK_ENDIAN
Definition: SDL_audio.h:73
#define SDL_InvalidParamError(param)
Definition: SDL_error.h:54
static int SDL_BuildAudioTypeCVTToFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat src_fmt)
Definition: SDL_audiocvt.c:615
#define SDL_zerop(x)
Definition: SDL_stdinc.h:417
static void SDL_Convert_Byteswap(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:563
static void SDL_Convert51To71(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:335
static int SDL_BuildAudioResampleCVT(SDL_AudioCVT *cvt, const int dst_channels, const int src_rate, const int dst_rate)
Definition: SDL_audiocvt.c:776
SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS+1]
Definition: SDL_audio.h:236
static void SDL_ConvertQuadToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:128
#define SDL_zero(x)
Definition: SDL_stdinc.h:416
SDL_AudioFormat src_format
Definition: SDL_audio.h:228
void(* SDL_AudioFilter)(struct SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audio.h:192
static void SDL_ConvertQuadTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:268
#define NULL
Definition: begin_code.h:164
static SDL_bool SDL_SupportedChannelCount(const int channels)
Definition: SDL_audiocvt.c:848
#define SDL_SetError
SDL_AudioFormat dst_format
Definition: SDL_audio.h:229
static void SDL_ConvertMonoToStereo(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:210
double rate_incr
Definition: SDL_audio.h:230
static int SDL_AddAudioCVTFilter(SDL_AudioCVT *cvt, const SDL_AudioFilter filter)
Definition: SDL_audiocvt.c:601
static int SDL_BuildAudioTypeCVTFromFloat(SDL_AudioCVT *cvt, const SDL_AudioFormat dst_fmt)
Definition: SDL_audiocvt.c:662
static void SDL_ConvertStereoTo51(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:234
static void SDL_ConvertStereoToMono(SDL_AudioCVT *cvt, SDL_AudioFormat format)
Definition: SDL_audiocvt.c:81
GLint GLint GLint GLint GLint GLint GLint GLbitfield GLenum filter

◆ 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 623 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(), SDLCALL, and SDL_AudioDriverImpl::UnlockDevice.

624 {
626 
627  if (!device) {
628  return; /* nothing to do. */
629  }
630 
631  /* Blank out the device and release the mutex. Free it afterwards. */
632  current_audio.impl.LockDevice(device);
633 
634  /* Keep up to two packets in the pool to reduce future malloc pressure. */
636 
638 }
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:81
static SDL_AudioDeviceID device
Definition: loopwave.c:37
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
SDL_DataQueue * buffer_queue
Definition: SDL_sysaudio.h:165
void SDL_ClearDataQueue(SDL_DataQueue *queue, const size_t slack)
Definition: SDL_dataqueue.c:98
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
#define SDL_AUDIOBUFFERQUEUE_PACKETLEN
Definition: SDL_sysaudio.h:63
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ SDL_CloseAudio()

void SDL_CloseAudio ( void  )

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

Definition at line 1521 of file SDL_audio.c.

References SDL_CloseAudioDevice().

1522 {
1524 }
void SDL_CloseAudioDevice(SDL_AudioDeviceID devid)
Definition: SDL_audio.c:1515

◆ SDL_CloseAudioDevice()

void SDL_CloseAudioDevice ( SDL_AudioDeviceID  dev)

Definition at line 1515 of file SDL_audio.c.

References close_audio_device(), and get_audio_device().

Referenced by SDL_CloseAudio().

1516 {
1518 }
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197
static void close_audio_device(SDL_AudioDevice *device)
Definition: SDL_audio.c:1051

◆ 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.

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

Referenced by SDL_AudioStreamPutInternal().

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:231
int filter_index
Definition: SDL_audio.h:237
SDL_AudioFilter filters[SDL_AUDIOCVT_MAX_FILTERS+1]
Definition: SDL_audio.h:236
SDL_AudioFormat src_format
Definition: SDL_audio.h:228
#define NULL
Definition: begin_code.h:164
#define SDL_SetError

◆ 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 580 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.

581 {
583  Uint32 rc;
584 
585  if ( (len == 0) || /* nothing to do? */
586  (!device) || /* called with bogus device id */
587  (!device->iscapture) || /* playback devices can't dequeue */
588  (device->callbackspec.callback != SDL_BufferQueueFillCallback) ) { /* not set for queueing */
589  return 0; /* just report zero bytes dequeued. */
590  }
591 
592  current_audio.impl.LockDevice(device);
595  return rc;
596 }
GLint GLenum GLsizei GLsizei GLsizei GLint GLsizei const GLvoid * data
Definition: SDL_opengl.h:1974
uint32_t Uint32
Definition: SDL_stdinc.h:181
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:81
GLenum GLsizei len
static SDL_AudioDeviceID device
Definition: loopwave.c:37
SDL_bool iscapture
Definition: SDL_sysaudio.h:149
size_t SDL_ReadFromDataQueue(SDL_DataQueue *queue, void *_buf, const size_t _len)
SDL_AudioCallback callback
Definition: SDL_audio.h:186
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
SDL_DataQueue * buffer_queue
Definition: SDL_sysaudio.h:165
static void SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
Definition: SDL_audio.c:541
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
SDL_AudioSpec callbackspec
Definition: SDL_sysaudio.h:140
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ 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 1658 of file SDL_audiocvt.c.

References SDL_free, and SDL_FreeDataQueue().

Referenced by SDL_NewAudioStream().

1659 {
1660  if (stream) {
1661  if (stream->cleanup_resampler_func) {
1662  stream->cleanup_resampler_func(stream);
1663  }
1664  SDL_FreeDataQueue(stream->queue);
1665  SDL_free(stream->staging_buffer);
1666  SDL_free(stream->work_buffer_base);
1667  SDL_free(stream->resampler_padding);
1668  SDL_free(stream);
1669  }
1670 }
GLuint GLuint stream
#define SDL_free
void SDL_FreeDataQueue(SDL_DataQueue *queue)
Definition: SDL_dataqueue.c:88

◆ SDL_FreeWAV()

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.

673 {
674  SDL_free(audio_buf);
675 }
#define 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 1011 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.

1012 {
1013  const char *retval = NULL;
1014 
1015  if (!SDL_WasInit(SDL_INIT_AUDIO)) {
1016  SDL_SetError("Audio subsystem is not initialized");
1017  return NULL;
1018  }
1019 
1020  if ((iscapture) && (!current_audio.impl.HasCaptureSupport)) {
1021  SDL_SetError("No capture support");
1022  return NULL;
1023  }
1024 
1025  if (index >= 0) {
1026  SDL_AudioDeviceItem *item;
1027  int i;
1028 
1032  if (index < i) {
1033  for (i--; i > index; i--, item = item->next) {
1034  SDL_assert(item != NULL);
1035  }
1036  SDL_assert(item != NULL);
1037  retval = item->name;
1038  }
1040  }
1041 
1042  if (retval == NULL) {
1043  SDL_SetError("No such device");
1044  }
1045 
1046  return retval;
1047 }
#define SDL_LockMutex
struct SDL_AudioDeviceItem * next
Definition: SDL_sysaudio.h:101
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
SDL_mutex * detectionLock
Definition: SDL_sysaudio.h:119
SDL_bool retval
GLuint index
SDL_AudioDeviceItem * outputDevices
Definition: SDL_sysaudio.h:124
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 SDL_assert(condition)
Definition: SDL_assert.h:169
#define NULL
Definition: begin_code.h:164
#define SDL_SetError
#define SDL_INIT_AUDIO
Definition: SDL.h:77
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
#define SDL_UnlockMutex
#define SDL_WasInit
SDL_AudioDeviceItem * inputDevices
Definition: SDL_sysaudio.h:125

◆ SDL_GetAudioDeviceStatus()

SDL_AudioStatus SDL_GetAudioDeviceStatus ( SDL_AudioDeviceID  dev)

Definition at line 1443 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.

Referenced by SDL_GetAudioStatus().

1444 {
1447  if (device && SDL_AtomicGet(&device->enabled)) {
1448  if (SDL_AtomicGet(&device->paused)) {
1449  status = SDL_AUDIO_PAUSED;
1450  } else {
1451  status = SDL_AUDIO_PLAYING;
1452  }
1453  }
1454  return status;
1455 }
SDL_atomic_t enabled
Definition: SDL_sysaudio.h:147
SDL_atomic_t paused
Definition: SDL_sysaudio.h:148
SDL_AudioStatus
Definition: SDL_audio.h:394
static SDL_AudioDeviceID device
Definition: loopwave.c:37
#define SDL_AtomicGet
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ SDL_GetAudioDriver()

const char* SDL_GetAudioDriver ( int  index)

Definition at line 876 of file SDL_audio.c.

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

877 {
878  if (index >= 0 && index < SDL_GetNumAudioDrivers()) {
879  return bootstrap[index]->name;
880  }
881  return NULL;
882 }
int SDL_GetNumAudioDrivers(void)
Definition: SDL_audio.c:870
const char * name
Definition: SDL_sysaudio.h:177
GLuint index
static const AudioBootStrap *const bootstrap[]
Definition: SDL_audio.c:37
#define NULL
Definition: begin_code.h:164

◆ SDL_GetAudioStatus()

SDL_AudioStatus SDL_GetAudioStatus ( void  )

Definition at line 1459 of file SDL_audio.c.

References SDL_GetAudioDeviceStatus().

1460 {
1461  return SDL_GetAudioDeviceStatus(1);
1462 }
SDL_AudioStatus SDL_GetAudioDeviceStatus(SDL_AudioDeviceID devid)
Definition: SDL_audio.c:1443

◆ 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 950 of file SDL_audio.c.

References SDL_AudioDriver::name.

951 {
952  return current_audio.name;
953 }
const char * name
Definition: SDL_sysaudio.h:110
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33

◆ 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 985 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_FALSE, SDL_INIT_AUDIO, SDL_LockMutex, SDL_UnlockMutex, and SDL_WasInit.

986 {
987  int retval = 0;
988 
989  if (!SDL_WasInit(SDL_INIT_AUDIO)) {
990  return -1;
991  }
992 
994  if (iscapture && current_audio.captureDevicesRemoved) {
996  }
997 
998  if (!iscapture && current_audio.outputDevicesRemoved) {
1001  }
1002 
1005 
1006  return retval;
1007 }
#define SDL_LockMutex
SDL_bool captureDevicesRemoved
Definition: SDL_sysaudio.h:120
static void clean_out_device_list(SDL_AudioDeviceItem **devices, int *devCount, SDL_bool *removedFlag)
Definition: SDL_audio.c:957
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
SDL_mutex * detectionLock
Definition: SDL_sysaudio.h:119
SDL_bool retval
SDL_AudioDeviceItem * outputDevices
Definition: SDL_sysaudio.h:124
SDL_bool outputDevicesRemoved
Definition: SDL_sysaudio.h:121
#define SDL_INIT_AUDIO
Definition: SDL.h:77
#define SDL_UnlockMutex
#define SDL_WasInit
SDL_AudioDeviceItem * inputDevices
Definition: SDL_sysaudio.h:125

◆ SDL_GetNumAudioDrivers()

int SDL_GetNumAudioDrivers ( void  )

Definition at line 870 of file SDL_audio.c.

References SDL_arraysize.

Referenced by SDL_GetAudioDriver().

871 {
872  return SDL_arraysize(bootstrap) - 1;
873 }
static const AudioBootStrap *const bootstrap[]
Definition: SDL_audio.c:37
#define SDL_arraysize(array)
Definition: SDL_stdinc.h:93

◆ 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 599 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.

600 {
601  Uint32 retval = 0;
603 
604  if (!device) {
605  return 0;
606  }
607 
608  /* Nothing to do unless we're set up for queueing. */
610  current_audio.impl.LockDevice(device);
613  } else if (device->callbackspec.callback == SDL_BufferQueueFillCallback) {
614  current_audio.impl.LockDevice(device);
615  retval = (Uint32) SDL_CountDataQueue(device->buffer_queue);
617  }
618 
619  return retval;
620 }
uint32_t Uint32
Definition: SDL_stdinc.h:181
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:81
static SDL_AudioDeviceID device
Definition: loopwave.c:37
SDL_bool retval
SDL_AudioCallback callback
Definition: SDL_audio.h:186
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
static void SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
Definition: SDL_audio.c:520
SDL_DataQueue * buffer_queue
Definition: SDL_sysaudio.h:165
static void SDL_BufferQueueFillCallback(void *userdata, Uint8 *stream, int len)
Definition: SDL_audio.c:541
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
size_t SDL_CountDataQueue(SDL_DataQueue *queue)
int(* GetPendingBytes)(_THIS)
Definition: SDL_sysaudio.h:74
SDL_AudioSpec callbackspec
Definition: SDL_sysaudio.h:140
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ SDL_LoadWAV_RW()

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.

450 {
451  int was_error;
452  Chunk chunk;
453  int lenread;
454  int IEEE_float_encoded, MS_ADPCM_encoded, IMA_ADPCM_encoded;
455  int samplesize;
456 
457  /* WAV magic header */
458  Uint32 RIFFchunk;
459  Uint32 wavelen = 0;
460  Uint32 WAVEmagic;
461  Uint32 headerDiff = 0;
462 
463  /* FMT chunk */
464  WaveFMT *format = NULL;
465  WaveExtensibleFMT *ext = NULL;
466 
467  SDL_zero(chunk);
468 
469  /* Make sure we are passed a valid data source */
470  was_error = 0;
471  if (src == NULL) {
472  was_error = 1;
473  goto done;
474  }
475 
476  /* Check the magic header */
477  RIFFchunk = SDL_ReadLE32(src);
478  wavelen = SDL_ReadLE32(src);
479  if (wavelen == WAVE) { /* The RIFFchunk has already been read */
480  WAVEmagic = wavelen;
481  wavelen = RIFFchunk;
482  RIFFchunk = RIFF;
483  } else {
484  WAVEmagic = SDL_ReadLE32(src);
485  }
486  if ((RIFFchunk != RIFF) || (WAVEmagic != WAVE)) {
487  SDL_SetError("Unrecognized file type (not WAVE)");
488  was_error = 1;
489  goto done;
490  }
491  headerDiff += sizeof(Uint32); /* for WAVE */
492 
493  /* Read the audio data format chunk */
494  chunk.data = NULL;
495  do {
496  SDL_free(chunk.data);
497  chunk.data = NULL;
498  lenread = ReadChunk(src, &chunk);
499  if (lenread < 0) {
500  was_error = 1;
501  goto done;
502  }
503  /* 2 Uint32's for chunk header+len, plus the lenread */
504  headerDiff += lenread + 2 * sizeof(Uint32);
505  } while ((chunk.magic == FACT) || (chunk.magic == LIST) || (chunk.magic == BEXT) || (chunk.magic == JUNK));
506 
507  /* Decode the audio data format */
508  format = (WaveFMT *) chunk.data;
509  if (chunk.magic != FMT) {
510  SDL_SetError("Complex WAVE files not supported");
511  was_error = 1;
512  goto done;
513  }
514  IEEE_float_encoded = MS_ADPCM_encoded = IMA_ADPCM_encoded = 0;
515  switch (SDL_SwapLE16(format->encoding)) {
516  case PCM_CODE:
517  /* We can understand this */
518  break;
519  case IEEE_FLOAT_CODE:
520  IEEE_float_encoded = 1;
521  /* We can understand this */
522  break;
523  case MS_ADPCM_CODE:
524  /* Try to understand this */
525  if (InitMS_ADPCM(format) < 0) {
526  was_error = 1;
527  goto done;
528  }
529  MS_ADPCM_encoded = 1;
530  break;
531  case IMA_ADPCM_CODE:
532  /* Try to understand this */
533  if (InitIMA_ADPCM(format) < 0) {
534  was_error = 1;
535  goto done;
536  }
537  IMA_ADPCM_encoded = 1;
538  break;
539  case EXTENSIBLE_CODE:
540  /* note that this ignores channel masks, smaller valid bit counts
541  inside a larger container, and most subtypes. This is just enough
542  to get things that didn't really _need_ WAVE_FORMAT_EXTENSIBLE
543  to be useful working when they use this format flag. */
544  ext = (WaveExtensibleFMT *) format;
545  if (SDL_SwapLE16(ext->size) < 22) {
546  SDL_SetError("bogus extended .wav header");
547  was_error = 1;
548  goto done;
549  }
550  if (SDL_memcmp(ext->subformat, extensible_pcm_guid, 16) == 0) {
551  break; /* cool. */
552  } else if (SDL_memcmp(ext->subformat, extensible_ieee_guid, 16) == 0) {
553  IEEE_float_encoded = 1;
554  break;
555  }
556  break;
557  case MP3_CODE:
558  SDL_SetError("MPEG Layer 3 data not supported");
559  was_error = 1;
560  goto done;
561  default:
562  SDL_SetError("Unknown WAVE data format: 0x%.4x",
563  SDL_SwapLE16(format->encoding));
564  was_error = 1;
565  goto done;
566  }
567  SDL_zerop(spec);
568  spec->freq = SDL_SwapLE32(format->frequency);
569 
570  if (IEEE_float_encoded) {
571  if ((SDL_SwapLE16(format->bitspersample)) != 32) {
572  was_error = 1;
573  } else {
574  spec->format = AUDIO_F32;
575  }
576  } else {
577  switch (SDL_SwapLE16(format->bitspersample)) {
578  case 4:
579  if (MS_ADPCM_encoded || IMA_ADPCM_encoded) {
580  spec->format = AUDIO_S16;
581  } else {
582  was_error = 1;
583  }
584  break;
585  case 8:
586  spec->format = AUDIO_U8;
587  break;
588  case 16:
589  spec->format = AUDIO_S16;
590  break;
591  case 24: /* convert this. */
592  spec->format = AUDIO_S32;
593  break;
594  case 32:
595  spec->format = AUDIO_S32;
596  break;
597  default:
598  was_error = 1;
599  break;
600  }
601  }
602 
603  if (was_error) {
604  SDL_SetError("Unknown %d-bit PCM data format",
605  SDL_SwapLE16(format->bitspersample));
606  goto done;
607  }
608  spec->channels = (Uint8) SDL_SwapLE16(format->channels);
609  spec->samples = 4096; /* Good default buffer size */
610 
611  /* Read the audio data chunk */
612  *audio_buf = NULL;
613  do {
614  SDL_free(*audio_buf);
615  *audio_buf = NULL;
616  lenread = ReadChunk(src, &chunk);
617  if (lenread < 0) {
618  was_error = 1;
619  goto done;
620  }
621  *audio_len = lenread;
622  *audio_buf = chunk.data;
623  if (chunk.magic != DATA)
624  headerDiff += lenread + 2 * sizeof(Uint32);
625  } while (chunk.magic != DATA);
626  headerDiff += 2 * sizeof(Uint32); /* for the data chunk and len */
627 
628  if (MS_ADPCM_encoded) {
629  if (MS_ADPCM_decode(audio_buf, audio_len) < 0) {
630  was_error = 1;
631  goto done;
632  }
633  }
634  if (IMA_ADPCM_encoded) {
635  if (IMA_ADPCM_decode(audio_buf, audio_len) < 0) {
636  was_error = 1;
637  goto done;
638  }
639  }
640 
641  if (SDL_SwapLE16(format->bitspersample) == 24) {
642  if (ConvertSint24ToSint32(audio_buf, audio_len) < 0) {
643  was_error = 1;
644  goto done;
645  }
646  }
647 
648  /* Don't return a buffer that isn't a multiple of samplesize */
649  samplesize = ((SDL_AUDIO_BITSIZE(spec->format)) / 8) * spec->channels;
650  *audio_len &= ~(samplesize - 1);
651 
652  done:
653  SDL_free(format);
654  if (src) {
655  if (freesrc) {
656  SDL_RWclose(src);
657  } else {
658  /* seek to the end of the file (given by the RIFF chunk) */
659  SDL_RWseek(src, wavelen - chunk.length - headerDiff, RW_SEEK_CUR);
660  }
661  }
662  if (was_error) {
663  spec = NULL;
664  }
665  return (spec);
666 }
static int InitIMA_ADPCM(WaveFMT *format)
Definition: SDL_wave.c:232
#define PCM_CODE
Definition: SDL_wave.h:36
#define RIFF
Definition: SDL_wave.h:28
#define LIST
Definition: SDL_wave.h:31
#define MP3_CODE
Definition: SDL_wave.h:40
Definition: SDL_wave.h:61
#define SDL_ReadLE32
Uint32 length
Definition: SDL_wave.h:64
Uint16 samples
Definition: SDL_audio.h:183
uint32_t Uint32
Definition: SDL_stdinc.h:181
Uint16 encoding
Definition: SDL_wave.h:52
#define SDL_zerop(x)
Definition: SDL_stdinc.h:417
static int MS_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
Definition: SDL_wave.c:119
Uint16 channels
Definition: SDL_wave.h:53
#define FMT
Definition: SDL_wave.h:34
#define SDL_RWseek(ctx, offset, whence)
Definition: SDL_rwops.h:185
static const Uint8 extensible_ieee_guid[16]
Definition: SDL_wave.c:445
GLint GLint GLsizei GLsizei GLsizei GLint GLenum format
Definition: SDL_opengl.h:1572
#define AUDIO_U8
Definition: SDL_audio.h:89
#define DATA
Definition: SDL_wave.h:35
#define FACT
Definition: SDL_wave.h:30
Uint8 channels
Definition: SDL_audio.h:181
static int IMA_ADPCM_decode(Uint8 **audio_buf, Uint32 *audio_len)
Definition: SDL_wave.c:338
#define EXTENSIBLE_CODE
Definition: SDL_wave.h:41
uint8_t Uint8
Definition: SDL_stdinc.h:157
#define SDL_free
#define SDL_AUDIO_BITSIZE(x)
Definition: SDL_audio.h:75
#define SDL_SwapLE32(X)
Definition: SDL_endian.h:233
static int ConvertSint24ToSint32(Uint8 **audio_buf, Uint32 *audio_len)
Definition: SDL_wave.c:408
int done
Definition: checkkeys.c:28
#define AUDIO_S32
Definition: SDL_audio.h:105
static const Uint8 extensible_pcm_guid[16]
Definition: SDL_wave.c:444
#define WAVE
Definition: SDL_wave.h:29
#define SDL_memcmp
#define JUNK
Definition: SDL_wave.h:33
Uint8 subformat[16]
Definition: SDL_wave.h:74
#define SDL_zero(x)
Definition: SDL_stdinc.h:416
Uint8 * data
Definition: SDL_wave.h:65
#define NULL
Definition: begin_code.h:164
#define IEEE_FLOAT_CODE
Definition: SDL_wave.h:38
#define SDL_RWclose(ctx)
Definition: SDL_rwops.h:189
#define IMA_ADPCM_CODE
Definition: SDL_wave.h:39
#define SDL_SetError
SDL_AudioFormat format
Definition: SDL_audio.h:180
static int InitMS_ADPCM(WaveFMT *format)
Definition: SDL_wave.c:49
Uint32 magic
Definition: SDL_wave.h:63
#define AUDIO_S16
Definition: SDL_audio.h:96
Uint32 frequency
Definition: SDL_wave.h:54
#define RW_SEEK_CUR
Definition: SDL_rwops.h:175
Uint16 bitspersample
Definition: SDL_wave.h:57
#define AUDIO_F32
Definition: SDL_audio.h:114
#define SDL_SwapLE16(X)
Definition: SDL_endian.h:232
#define BEXT
Definition: SDL_wave.h:32
static int ReadChunk(SDL_RWops *src, Chunk *chunk)
Definition: SDL_wave.c:678
#define MS_ADPCM_CODE
Definition: SDL_wave.h:37

◆ SDL_LockAudio()

void SDL_LockAudio ( void  )

Definition at line 1493 of file SDL_audio.c.

References SDL_LockAudioDevice().

1494 {
1496 }
void SDL_LockAudioDevice(SDL_AudioDeviceID devid)
Definition: SDL_audio.c:1483

◆ SDL_LockAudioDevice()

void SDL_LockAudioDevice ( SDL_AudioDeviceID  dev)

Definition at line 1483 of file SDL_audio.c.

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

Referenced by SDL_LockAudio().

1484 {
1485  /* Obtain a lock on the mixing buffers */
1487  if (device) {
1488  current_audio.impl.LockDevice(device);
1489  }
1490 }
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
static SDL_AudioDeviceID device
Definition: loopwave.c:37
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ 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 1626 of file SDL_audio.c.

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

1627 {
1628  /* Mix the user-level audio format */
1630  if (device != NULL) {
1631  SDL_MixAudioFormat(dst, src, device->callbackspec.format, len, volume);
1632  }
1633 }
GLenum GLenum dst
#define SDL_MixAudioFormat
GLenum src
GLenum GLsizei len
static SDL_AudioDeviceID device
Definition: loopwave.c:37
#define NULL
Definition: begin_code.h:164
SDL_AudioFormat format
Definition: SDL_audio.h:180
SDL_AudioSpec callbackspec
Definition: SDL_sysaudio.h:140
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ 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.

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.

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 SDL_MIX_MAXVOLUME
Definition: SDL_audio.h:615
#define ADJUST_VOLUME(s, v)
Definition: SDL_mixer.c:85
#define SDL_SwapFloatBE(X)
Definition: SDL_endian.h:239
#define AUDIO_S32MSB
Definition: SDL_audio.h:104
GLenum GLenum dst
#define AUDIO_U16LSB
Definition: SDL_audio.h:91
#define SDL_SwapFloatLE(X)
Definition: SDL_endian.h:235
uint32_t Uint32
Definition: SDL_stdinc.h:181
GLenum src
GLenum GLsizei len
#define SDL_SwapBE32(X)
Definition: SDL_endian.h:237
#define AUDIO_F32MSB
Definition: SDL_audio.h:113
GLint GLint GLsizei GLsizei GLsizei GLint GLenum format
Definition: SDL_opengl.h:1572
static const Uint8 mix8[]
Definition: SDL_mixer.c:34
#define AUDIO_U8
Definition: SDL_audio.h:89
int8_t Sint8
Definition: SDL_stdinc.h:151
uint8_t Uint8
Definition: SDL_stdinc.h:157
#define ADJUST_VOLUME_U8(s, v)
Definition: SDL_mixer.c:86
#define SDL_SwapLE32(X)
Definition: SDL_endian.h:233
#define AUDIO_F32LSB
Definition: SDL_audio.h:112
#define AUDIO_S32LSB
Definition: SDL_audio.h:103
int32_t Sint32
Definition: SDL_stdinc.h:175
#define SDL_SetError
#define AUDIO_S16MSB
Definition: SDL_audio.h:94
#define AUDIO_S16LSB
Definition: SDL_audio.h:92
uint16_t Uint16
Definition: SDL_stdinc.h:169
#define F(x, y, z)
Definition: SDL_test_md5.c:73
int64_t Sint64
Definition: SDL_stdinc.h:188
#define AUDIO_S8
Definition: SDL_audio.h:90
int16_t Sint16
Definition: SDL_stdinc.h:163
#define AUDIO_U16MSB
Definition: SDL_audio.h:93

◆ 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 1259 of file SDL_audiocvt.c.

References AUDIO_F32SYS, SDL_AudioStream::dst_channels, SDL_AudioStream::dst_format, SDL_AudioStream::dst_rate, NULL, SDL_AudioStream::packetlen, SDL_AudioStream::pre_resample_channels, 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(), SDL_TRUE, SDL_AudioStream::src_channels, SDL_AudioStream::src_format, and SDL_AudioStream::src_rate.

1265 {
1266  const int packetlen = 4096; /* !!! FIXME: good enough for now. */
1267  Uint8 pre_resample_channels;
1268  SDL_AudioStream *retval;
1269 
1270  retval = (SDL_AudioStream *) SDL_calloc(1, sizeof (SDL_AudioStream));
1271  if (!retval) {
1272  return NULL;
1273  }
1274 
1275  /* If increasing channels, do it after resampling, since we'd just
1276  do more work to resample duplicate channels. If we're decreasing, do
1277  it first so we resample the interpolated data instead of interpolating
1278  the resampled data (!!! FIXME: decide if that works in practice, though!). */
1279  pre_resample_channels = SDL_min(src_channels, dst_channels);
1280 
1281  retval->first_run = SDL_TRUE;
1282  retval->src_sample_frame_size = (SDL_AUDIO_BITSIZE(src_format) / 8) * src_channels;
1283  retval->src_format = src_format;
1284  retval->src_channels = src_channels;
1285  retval->src_rate = src_rate;
1286  retval->dst_sample_frame_size = (SDL_AUDIO_BITSIZE(dst_format) / 8) * dst_channels;
1287  retval->dst_format = dst_format;
1288  retval->dst_channels = dst_channels;
1289  retval->dst_rate = dst_rate;
1290  retval->pre_resample_channels = pre_resample_channels;
1291  retval->packetlen = packetlen;
1292  retval->rate_incr = ((double) dst_rate) / ((double) src_rate);
1293  retval->resampler_padding_samples = ResamplerPadding(retval->src_rate, retval->dst_rate) * pre_resample_channels;
1294  retval->resampler_padding = (float *) SDL_calloc(retval->resampler_padding_samples, sizeof (float));
1295 
1296  if (retval->resampler_padding == NULL) {
1297  SDL_FreeAudioStream(retval);
1298  SDL_OutOfMemory();
1299  return NULL;
1300  }
1301 
1302  retval->staging_buffer_size = ((retval->resampler_padding_samples / retval->pre_resample_channels) * retval->src_sample_frame_size);
1303  if (retval->staging_buffer_size > 0) {
1304  retval->staging_buffer = (Uint8 *) SDL_malloc(retval->staging_buffer_size);
1305  if (retval->staging_buffer == NULL) {
1306  SDL_FreeAudioStream(retval);
1307  SDL_OutOfMemory();
1308  return NULL;
1309  }
1310  }
1311 
1312  /* Not resampling? It's an easy conversion (and maybe not even that!) */
1313  if (src_rate == dst_rate) {
1314  retval->cvt_before_resampling.needed = SDL_FALSE;
1315  if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, src_format, src_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
1316  SDL_FreeAudioStream(retval);
1317  return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
1318  }
1319  } else {
1320  /* Don't resample at first. Just get us to Float32 format. */
1321  /* !!! FIXME: convert to int32 on devices without hardware float. */
1322  if (SDL_BuildAudioCVT(&retval->cvt_before_resampling, src_format, src_channels, src_rate, AUDIO_F32SYS, pre_resample_channels, src_rate) < 0) {
1323  SDL_FreeAudioStream(retval);
1324  return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
1325  }
1326 
1327 #ifdef HAVE_LIBSAMPLERATE_H
1328  SetupLibSampleRateResampling(retval);
1329 #endif
1330 
1331  if (!retval->resampler_func) {
1332  retval->resampler_state = SDL_calloc(retval->resampler_padding_samples, sizeof (float));
1333  if (!retval->resampler_state) {
1334  SDL_FreeAudioStream(retval);
1335  SDL_OutOfMemory();
1336  return NULL;
1337  }
1338 
1339  if (SDL_PrepareResampleFilter() < 0) {
1340  SDL_free(retval->resampler_state);
1341  retval->resampler_state = NULL;
1342  SDL_FreeAudioStream(retval);
1343  return NULL;
1344  }
1345 
1346  retval->resampler_func = SDL_ResampleAudioStream;
1347  retval->reset_resampler_func = SDL_ResetAudioStreamResampler;
1348  retval->cleanup_resampler_func = SDL_CleanupAudioStreamResampler;
1349  }
1350 
1351  /* Convert us to the final format after resampling. */
1352  if (SDL_BuildAudioCVT(&retval->cvt_after_resampling, AUDIO_F32SYS, pre_resample_channels, dst_rate, dst_format, dst_channels, dst_rate) < 0) {
1353  SDL_FreeAudioStream(retval);
1354  return NULL; /* SDL_BuildAudioCVT should have called SDL_SetError. */
1355  }
1356  }
1357 
1358  retval->queue = SDL_NewDataQueue(packetlen, packetlen * 2);
1359  if (!retval->queue) {
1360  SDL_FreeAudioStream(retval);
1361  return NULL; /* SDL_NewDataQueue should have called SDL_SetError. */
1362  }
1363 
1364  return retval;
1365 }
static int ResamplerPadding(const int inrate, const int outrate)
Definition: SDL_audiocvt.c:473
#define SDL_min(x, y)
Definition: SDL_stdinc.h:406
SDL_DataQueue * SDL_NewDataQueue(const size_t _packetlen, const size_t initialslack)
Definition: SDL_dataqueue.c:58
void SDL_FreeAudioStream(SDL_AudioStream *stream)
SDL_bool retval
static void SDL_ResetAudioStreamResampler(SDL_AudioStream *stream)
int SDL_PrepareResampleFilter(void)
Definition: SDL_audiocvt.c:435
#define AUDIO_F32SYS
Definition: SDL_audio.h:125
uint8_t Uint8
Definition: SDL_stdinc.h:157
#define SDL_free
#define SDL_AUDIO_BITSIZE(x)
Definition: SDL_audio.h:75
static int SDL_ResampleAudioStream(SDL_AudioStream *stream, const void *_inbuf, const int inbuflen, void *_outbuf, const int outbuflen)
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:872
#define NULL
Definition: begin_code.h:164
#define SDL_OutOfMemory()
Definition: SDL_error.h:52
#define SDL_calloc
static void SDL_CleanupAudioStreamResampler(SDL_AudioStream *stream)
#define SDL_malloc

◆ 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
  • 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 1398 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.

1399 {
1400  SDL_AudioDeviceID id = 0;
1401 
1402  /* Start up the audio driver, if necessary. This is legacy behaviour! */
1403  if (!SDL_WasInit(SDL_INIT_AUDIO)) {
1404  if (SDL_InitSubSystem(SDL_INIT_AUDIO) < 0) {
1405  return -1;
1406  }
1407  }
1408 
1409  /* SDL_OpenAudio() is legacy and can only act on Device ID #1. */
1410  if (open_devices[0] != NULL) {
1411  SDL_SetError("Audio device is already opened");
1412  return -1;
1413  }
1414 
1415  if (obtained) {
1416  id = open_audio_device(NULL, 0, desired, obtained,
1418  } else {
1419  SDL_AudioSpec _obtained;
1420  SDL_zero(_obtained);
1421  id = open_audio_device(NULL, 0, desired, &_obtained, 0, 1);
1422  /* On successful open, copy calculated values into 'desired'. */
1423  if (id > 0) {
1424  desired->size = _obtained.size;
1425  desired->silence = _obtained.silence;
1426  }
1427  }
1428 
1429  SDL_assert((id == 0) || (id == 1));
1430  return (id == 0) ? -1 : 0;
1431 }
static SDL_AudioDevice * open_devices[16]
Definition: SDL_audio.c:34
Uint8 silence
Definition: SDL_audio.h:182
#define SDL_InitSubSystem
#define SDL_AUDIO_ALLOW_ANY_CHANGE
Definition: SDL_audio.h:143
#define SDL_zero(x)
Definition: SDL_stdinc.h:416
Uint32 size
Definition: SDL_audio.h:185
#define SDL_assert(condition)
Definition: SDL_assert.h:169
#define NULL
Definition: begin_code.h:164
#define SDL_SetError
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:1150
#define SDL_INIT_AUDIO
Definition: SDL.h:77
Uint32 SDL_AudioDeviceID
Definition: SDL_audio.h:329
#define SDL_WasInit

◆ 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 1434 of file SDL_audio.c.

References open_audio_device().

1437 {
1438  return open_audio_device(device, iscapture, desired, obtained,
1439  allowed_changes, 2);
1440 }
static SDL_AudioDeviceID device
Definition: loopwave.c:37
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:1150

◆ SDL_PauseAudio()

void SDL_PauseAudio ( int  pause_on)

Definition at line 1476 of file SDL_audio.c.

References SDL_PauseAudioDevice().

1477 {
1478  SDL_PauseAudioDevice(1, pause_on);
1479 }
void SDL_PauseAudioDevice(SDL_AudioDeviceID devid, int pause_on)
Definition: SDL_audio.c:1465

◆ SDL_PauseAudioDevice()

void SDL_PauseAudioDevice ( SDL_AudioDeviceID  dev,
int  pause_on 
)

Definition at line 1465 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.

Referenced by SDL_PauseAudio().

1466 {
1468  if (device) {
1469  current_audio.impl.LockDevice(device);
1470  SDL_AtomicSet(&device->paused, pause_on ? 1 : 0);
1472  }
1473 }
SDL_atomic_t paused
Definition: SDL_sysaudio.h:148
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:81
static SDL_AudioDeviceID device
Definition: loopwave.c:37
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
#define SDL_AtomicSet
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ 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 557 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.

558 {
560  int rc = 0;
561 
562  if (!device) {
563  return -1; /* get_audio_device() will have set the error state */
564  } else if (device->iscapture) {
565  return SDL_SetError("This is a capture device, queueing not allowed");
566  } else if (device->callbackspec.callback != SDL_BufferQueueDrainCallback) {
567  return SDL_SetError("Audio device has a callback, queueing not allowed");
568  }
569 
570  if (len > 0) {
571  current_audio.impl.LockDevice(device);
572  rc = SDL_WriteToDataQueue(device->buffer_queue, data, len);
574  }
575 
576  return rc;
577 }
int SDL_WriteToDataQueue(SDL_DataQueue *queue, const void *_data, const size_t _len)
GLint GLenum GLsizei GLsizei GLsizei GLint GLsizei const GLvoid * data
Definition: SDL_opengl.h:1974
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:81
GLenum GLsizei len
static SDL_AudioDeviceID device
Definition: loopwave.c:37
SDL_bool iscapture
Definition: SDL_sysaudio.h:149
SDL_AudioCallback callback
Definition: SDL_audio.h:186
void(* LockDevice)(_THIS)
Definition: SDL_sysaudio.h:80
static void SDL_BufferQueueDrainCallback(void *userdata, Uint8 *stream, int len)
Definition: SDL_audio.c:520
SDL_DataQueue * buffer_queue
Definition: SDL_sysaudio.h:165
#define SDL_SetError
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
SDL_AudioSpec callbackspec
Definition: SDL_sysaudio.h:140
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197

◆ SDL_UnlockAudio()

void SDL_UnlockAudio ( void  )

Definition at line 1509 of file SDL_audio.c.

References SDL_UnlockAudioDevice().

1510 {
1512 }
void SDL_UnlockAudioDevice(SDL_AudioDeviceID devid)
Definition: SDL_audio.c:1499

◆ SDL_UnlockAudioDevice()

void SDL_UnlockAudioDevice ( SDL_AudioDeviceID  dev)

Definition at line 1499 of file SDL_audio.c.

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

Referenced by SDL_UnlockAudio().

1500 {
1501  /* Obtain a lock on the mixing buffers */
1503  if (device) {
1505  }
1506 }
static SDL_AudioDriver current_audio
Definition: SDL_audio.c:33
void(* UnlockDevice)(_THIS)
Definition: SDL_sysaudio.h:81
static SDL_AudioDeviceID device
Definition: loopwave.c:37
SDL_AudioDriverImpl impl
Definition: SDL_sysaudio.h:116
static SDL_AudioDevice * get_audio_device(SDL_AudioDeviceID id)
Definition: SDL_audio.c:197