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Diffstat (limited to 'distrib/sdl-1.2.15/include/SDL_audio.h')
| -rw-r--r-- | distrib/sdl-1.2.15/include/SDL_audio.h | 284 |
1 files changed, 284 insertions, 0 deletions
diff --git a/distrib/sdl-1.2.15/include/SDL_audio.h b/distrib/sdl-1.2.15/include/SDL_audio.h new file mode 100644 index 0000000..e879c98 --- /dev/null +++ b/distrib/sdl-1.2.15/include/SDL_audio.h @@ -0,0 +1,284 @@ +/* + 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 __cplusplus +extern "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 */ |