copied refactor to trunk

1 files changed, 110 insertions, 79 deletions

diff --git a/other/sdl/include/SDL_audio.h b/other/sdl/include/SDL_audio.h
index 68ec4759..3a8e7fa8 100644
--- a/other/sdl/include/SDL_audio.h
+++ b/other/sdl/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);