--- a
+++ b/MSVC8/GemRB/includes/sdl/SDL_audio.h
@@ -0,0 +1,259 @@
+/*
+    SDL - Simple DirectMedia Layer
+    Copyright (C) 1997, 1998, 1999, 2000, 2001, 2002  Sam Lantinga
+
+    This library is free software; you can redistribute it and/or
+    modify it under the terms of the GNU Library General Public
+    License as published by the Free Software Foundation; either
+    version 2 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
+    Library General Public License for more details.
+
+    You should have received a copy of the GNU Library General Public
+    License along with this library; if not, write to the Free
+    Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+
+    Sam Lantinga
+    slouken@libsdl.org
+*/
+
+#ifdef SAVE_RCSID
+static char rcsid =
+ "@(#) $Id: SDL_audio.h,v 1.1 2003/10/14 22:31:01 balrog994 Exp $";
+#endif
+
+/* Access to the raw audio mixing buffer for the SDL library */
+
+#ifndef _SDL_audio_h
+#define _SDL_audio_h
+
+#include <stdio.h>
+
+#include "SDL_main.h"
+#include "SDL_types.h"
+#include "SDL_error.h"
+#include "SDL_rwops.h"
+#include "SDL_byteorder.h"
+
+#include "begin_code.h"
+/* Set up for C function definitions, even when using C++ */
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* The calculated values in this structure are calculated by SDL_OpenAudio() */
+typedef struct {
+	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.
+	*/
+	void (*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 */
+#define AUDIO_U16	AUDIO_U16LSB
+#define AUDIO_S16	AUDIO_S16LSB
+
+/* 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 (*filters[10])(struct SDL_AudioCVT *cvt, Uint16 format);
+	int filter_index;		/* Current audio conversion function */
+} SDL_AudioCVT;
+
+
+/* Function prototypes */
+
+/* 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.
+ *
+ * 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.
+ */
+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;
+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:
+ *	SDL_LoadWAV_RW(SDL_RWFromFile("sample.wav", "rb"), 1, ...);
+ *
+ * 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.
+ * 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);
+
+/*
+ * 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);
+
+/*
+ * 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 */