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SDL - Simple DirectMedia Layer
Copyright (C) 1997-2009 Sam Lantinga
This library is free software; you can redistribute it and/or
modify it under the terms of the GNU Lesser General Public
License as published by the Free Software Foundation; either
version 2.1 of the License, or (at your option) any later version.
This library is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
Lesser General Public License for more details.
You should have received a copy of the GNU Lesser General Public
License along with this library; if not, write to the Free Software
Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
Sam Lantinga
* @file SDL_audio.h
* Access to the raw audio mixing buffer for the SDL library
#ifndef _SDL_audio_h
#define _SDL_audio_h
#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"
/* Set up for C function definitions, even when using C++ */
#ifdef __cplusplus
extern "C" {
* 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.
* - 'desired->userdata' is passed as the first parameter to your callback
* function.
* @note The calculated values in this structure are calculated by SDL_OpenAudio()
typedef struct SDL_AudioSpec {
int freq; /**< DSP frequency -- samples per second */
Uint16 format; /**< Audio data format */
Uint8 channels; /**< Number of channels: 1 mono, 2 stereo */
Uint8 silence; /**< Audio buffer silence value (calculated) */
Uint16 samples; /**< Audio buffer size in samples (power of 2) */
Uint16 padding; /**< Necessary for some compile environments */
Uint32 size; /**< Audio buffer size in bytes (calculated) */
* This function is called when the audio device needs more data.
* @param[out] stream A pointer to the audio data buffer
* @param[in] len The length of the audio buffer in bytes.
* Once the callback returns, the buffer will no longer be valid.
* Stereo samples are stored in a LRLRLR ordering.
void (SDLCALL *callback)(void *userdata, Uint8 *stream, int len);
void *userdata;
} SDL_AudioSpec;
* @name Audio format flags
* defaults to LSB byte order
#define AUDIO_U8 0x0008 /**< Unsigned 8-bit samples */
#define AUDIO_S8 0x8008 /**< Signed 8-bit samples */
#define AUDIO_U16LSB 0x0010 /**< Unsigned 16-bit samples */
#define AUDIO_S16LSB 0x8010 /**< Signed 16-bit samples */
#define AUDIO_U16MSB 0x1010 /**< As above, but big-endian byte order */
#define AUDIO_S16MSB 0x9010 /**< As above, but big-endian byte order */
#define AUDIO_U16 AUDIO_U16LSB
#define AUDIO_S16 AUDIO_S16LSB
* @name Native audio byte ordering
/** A structure to hold a set of audio conversion filters and buffers */
typedef struct SDL_AudioCVT {
int needed; /**< Set to 1 if conversion possible */
Uint16 src_format; /**< Source audio format */
Uint16 dst_format; /**< Target audio format */
double rate_incr; /**< Rate conversion increment */
Uint8 *buf; /**< Buffer to hold entire audio data */
int len; /**< Length of original audio buffer */
int len_cvt; /**< Length of converted audio buffer */
int len_mult; /**< buffer must be len*len_mult big */
double len_ratio; /**< Given len, final size is len*len_ratio */
void (SDLCALL *filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
int filter_index; /**< Current audio conversion function */
} SDL_AudioCVT;
/* Function prototypes */
* @name Audio Init and Quit
* 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().
extern DECLSPEC int SDLCALL SDL_AudioInit(const char *driver_name);
extern DECLSPEC void SDLCALL SDL_AudioQuit(void);
* This function fills the given character buffer with the name of the
* current audio driver, and returns a pointer to it if the audio driver has
* been initialized. It returns NULL if no driver has been initialized.
extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen);
* 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.
* 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.
* @sa SDL_AudioSpec
extern DECLSPEC int SDLCALL SDL_OpenAudio(SDL_AudioSpec *desired, SDL_AudioSpec *obtained);
typedef enum {
} SDL_audiostatus;
/** Get the current audio state */
extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void);
* This function pauses and unpauses the audio callback processing.
* It 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.
extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on);
* 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:
* @code SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); @endcode
* 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.
extern DECLSPEC SDL_AudioSpec * SDLCALL SDL_LoadWAV_RW(SDL_RWops *src, int freesrc, SDL_AudioSpec *spec, Uint8 **audio_buf, Uint32 *audio_len);
/** Compatibility convenience function -- loads a WAV from a file */
#define SDL_LoadWAV(file, spec, audio_buf, audio_len) \
SDL_LoadWAV_RW(SDL_RWFromFile(file, "rb"),1, spec,audio_buf,audio_len)
* This function frees data previously allocated with SDL_LoadWAV_RW()
extern DECLSPEC void SDLCALL SDL_FreeWAV(Uint8 *audio_buf);
* 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.
* @return This function returns 0, or -1 if there was an error.
extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt,
Uint16 src_format, Uint8 src_channels, int src_rate,
Uint16 dst_format, Uint8 dst_channels, int dst_rate);
* 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.
extern DECLSPEC int SDLCALL SDL_ConvertAudio(SDL_AudioCVT *cvt);
* 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.
extern DECLSPEC void SDLCALL SDL_MixAudio(Uint8 *dst, const Uint8 *src, Uint32 len, int volume);
* @name Audio Locks
* The lock manipulated by these functions protects the callback function.
* During a LockAudio/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.
extern DECLSPEC void SDLCALL SDL_LockAudio(void);
extern DECLSPEC void SDLCALL SDL_UnlockAudio(void);
* This function shuts down audio processing and closes the audio device.
extern DECLSPEC void SDLCALL SDL_CloseAudio(void);
/* Ends C function definitions when using C++ */
#ifdef __cplusplus
#include "close_code.h"
#endif /* _SDL_audio_h */