diff options
Diffstat (limited to 'code/SDL12/include/SDL_audio.h')
| -rw-r--r-- | code/SDL12/include/SDL_audio.h | 189 |
1 files changed, 110 insertions, 79 deletions
diff --git a/code/SDL12/include/SDL_audio.h b/code/SDL12/include/SDL_audio.h index 68ec475..3a8e7fa 100644 --- a/code/SDL12/include/SDL_audio.h +++ b/code/SDL12/include/SDL_audio.h @@ -1,6 +1,6 @@ /* SDL - Simple DirectMedia Layer - Copyright (C) 1997-2006 Sam Lantinga + 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 @@ -20,7 +20,10 @@ slouken@libsdl.org */ -/* Access to the raw audio mixing buffer for the SDL library */ +/** + * @file SDL_audio.h + * Access to the raw audio mixing buffer for the SDL library + */ #ifndef _SDL_audio_h #define _SDL_audio_h @@ -38,36 +41,75 @@ extern "C" { #endif -/* The calculated values in this structure are calculated by SDL_OpenAudio() */ +/** + * 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. - 'stream' is a pointer to the audio data buffer - 'len' is the 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. - */ + 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; -/* 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 */ +/** + * @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 -/* Native audio byte ordering */ +/** + * @name Native audio byte ordering + */ +/*@{*/ #if SDL_BYTEORDER == SDL_LIL_ENDIAN #define AUDIO_U16SYS AUDIO_U16LSB #define AUDIO_S16SYS AUDIO_S16LSB @@ -75,40 +117,48 @@ typedef struct SDL_AudioSpec { #define AUDIO_U16SYS AUDIO_U16MSB #define AUDIO_S16SYS AUDIO_S16MSB #endif +/*@}*/ + +/*@}*/ -/* A structure to hold a set of audio conversion filters and buffers */ +/** 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 */ + 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 */ + int filter_index; /**< Current audio conversion function */ } SDL_AudioCVT; /* Function prototypes */ -/* These functions are used internally, and should not be used unless you +/** + * @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 +/** + * 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 @@ -117,51 +167,26 @@ extern DECLSPEC char * SDLCALL SDL_AudioDriverName(char *namebuf, int maxlen); * 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. - * 'desired->userdata' is passed as the first parameter to your callback - * function. - * * 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); -/* - * Get the current audio state: - */ 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 @@ -170,11 +195,11 @@ extern DECLSPEC SDL_audiostatus SDLCALL SDL_GetAudioStatus(void); */ 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: - * SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...); + * @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 @@ -189,27 +214,29 @@ extern DECLSPEC void SDLCALL SDL_PauseAudio(int pause_on); */ 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 */ +/** 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. - * This function returns 0, or -1 if there was an error. + * + * @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(), +/** + * 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. @@ -219,26 +246,30 @@ extern DECLSPEC int SDLCALL SDL_BuildAudioCVT(SDL_AudioCVT *cvt, */ 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. */ -#define SDL_MIX_MAXVOLUME 128 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); |