/*
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
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
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
#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
* 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
*/
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
*/
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.
*/
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);
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