/* SDL - Simple DirectMedia Layer Copyright (C) 1997-2012 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 MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 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 slouken@libsdl.org*//** * @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 __cplusplusextern "C" {#endif/** * 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 *//*@{*/#if SDL_BYTEORDER == SDL_LIL_ENDIAN#define AUDIO_U16SYS AUDIO_U16LSB#define AUDIO_S16SYS AUDIO_S16LSB#else#define AUDIO_U16SYS AUDIO_U16MSB#define AUDIO_S16SYS AUDIO_S16MSB#endif/*@}*//*@}*//** 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_AUDIO_STOPPED = 0, SDL_AUDIO_PLAYING, SDL_AUDIO_PAUSED} 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);#define SDL_MIX_MAXVOLUME 128/** * 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}#endif#include "close_code.h"#endif /* _SDL_audio_h */