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Diffstat (limited to 'src/SDL12/include/SDL_audio.h')
| -rw-r--r-- | src/SDL12/include/SDL_audio.h | 284 |
1 files changed, 0 insertions, 284 deletions
diff --git a/src/SDL12/include/SDL_audio.h b/src/SDL12/include/SDL_audio.h deleted file mode 100644 index e879c989..00000000 --- a/src/SDL12/include/SDL_audio.h +++ /dev/null @@ -1,284 +0,0 @@ -/* - 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 */ |