jedi-academy/code/client/snd_dma.cpp

/*****************************************************************************
 * name:		snd_dma.c
 *
 * desc:		main control for any streaming sound output device
 *
 *
 *****************************************************************************/
// leave this as first line for PCH reasons...
//
#include "../server/exe_headers.h"

#include "snd_local.h"
#include "cl_mp3.h"
#include "snd_music.h"

static void S_Play_f(void);
static void S_SoundList_f(void);
static void S_Music_f(void);
static void S_SetDynamicMusic_f(void);

void S_Update_();
void S_StopAllSounds(void);
static void S_UpdateBackgroundTrack( void );
sfx_t *S_FindName( const char *name );
static int SND_FreeSFXMem(sfx_t *sfx);

extern qboolean Sys_LowPhysicalMemory();

//////////////////////////
//
// vars for bgrnd music track...
//
const int iMP3MusicStream_DiskBytesToRead = 10000;//4096;
const int iMP3MusicStream_DiskBufferSize = iMP3MusicStream_DiskBytesToRead*2; //*10;

typedef struct
{	
	qboolean	bIsMP3;
	//
	// MP3 specific...
	//
	sfx_t		sfxMP3_Bgrnd;
	MP3STREAM	streamMP3_Bgrnd;	// this one is pointed at by the sfx_t's ptr, and is NOT the one the decoder uses every cycle
	channel_t	chMP3_Bgrnd;		// ... but the one in this struct IS.	
	//
	// MP3 disk streamer stuff... (if music is non-dynamic)
	//
	byte		byMP3MusicStream_DiskBuffer[iMP3MusicStream_DiskBufferSize];
	int			iMP3MusicStream_DiskReadPos;
	int			iMP3MusicStream_DiskWindowPos;
	//
	// MP3 disk-load stuff (for use during dynamic music, which is mem-resident)
	//
	byte		*pLoadedData;	// Z_Malloc, Z_Free	// these two MUST be kept as valid/invalid together	
	char		sLoadedDataName[MAX_QPATH];			//  " " " " "  
	int			iLoadedDataLen;	
	//
	// remaining dynamic fields...
	//
	int			iXFadeVolumeSeekTime;
	int			iXFadeVolumeSeekTo;	// when changing this, set the above timer to Sys_Milliseconds(). 
									//	Note that this should be thought of more as an up/down bool rather than as a 
									//	number now, in other words set it only to 0 or 255. I'll probably change this
									//	to actually be a bool later.
	int			iXFadeVolume;		// 0 = silent, 255 = max mixer vol, though still modulated via overall music_volume 
	float		fSmoothedOutVolume;
	qboolean	bActive;			// whether playing or not
	qboolean	bExists;			// whether was even loaded for this level (ie don't try and start playing it)
	//
	// new dynamic fields...
	//
	qboolean	bTrackSwitchPending;
	MusicState_e eTS_NewState;
	float		 fTS_NewTime;
	//
	// Generic...
	//
	fileHandle_t s_backgroundFile;	// valid handle, else -1 if an MP3 (so that NZ compares still work)
	wavinfo_t	s_backgroundInfo;
	int			s_backgroundSamples;	

	void Rewind()
	{
		MP3Stream_Rewind( &chMP3_Bgrnd );
		s_backgroundSamples = sfxMP3_Bgrnd.iSoundLengthInSamples;
	}

	void SeekTo(float fTime)
	{
		chMP3_Bgrnd.iMP3SlidingDecodeWindowPos = 0;
		chMP3_Bgrnd.iMP3SlidingDecodeWritePos = 0;
		MP3Stream_SeekTo( &chMP3_Bgrnd, fTime );
		s_backgroundSamples = sfxMP3_Bgrnd.iSoundLengthInSamples;
	}

} MusicInfo_t;

static void S_SetDynamicMusicState( MusicState_e musicState );

#define fDYNAMIC_XFADE_SECONDS (1.0f)

static MusicInfo_t	tMusic_Info[eBGRNDTRACK_NUMBEROF]	= {0};
static qboolean		bMusic_IsDynamic					= qfalse;
static MusicState_e	eMusic_StateActual					= eBGRNDTRACK_EXPLORE;	// actual state, can be any enum
static MusicState_e	eMusic_StateRequest					= eBGRNDTRACK_EXPLORE;	// requested state, can only be explore, action, boss, or silence
static char			sMusic_BackgroundLoop[MAX_QPATH]	= {0};	// only valid for non-dynamic music
static char			sInfoOnly_CurrentDynamicMusicSet[64];	// any old reasonable size, only has to fit stuff like "kejim_post"
//
//////////////////////////


// =======================================================================
// Internal sound data & structures
// =======================================================================
 
// only begin attenuating sound volumes when outside the FULLVOLUME range
#define		SOUND_FULLVOLUME	256

#define		SOUND_ATTENUATE		0.0008f
#define		VOICE_ATTENUATE		0.004f

const float	SOUND_FMAXVOL=0.75;//1.0;
const int	SOUND_MAXVOL=255;

channel_t   s_channels[MAX_CHANNELS];

int			s_soundStarted;
qboolean	s_soundMuted;

dma_t		dma;

int			listener_number;
vec3_t		listener_origin;
vec3_t		listener_axis[3];

int			s_soundtime;		// sample PAIRS
int   		s_paintedtime; 		// sample PAIRS

// MAX_SFX may be larger than MAX_SOUNDS because
// of custom player sounds
#define		MAX_SFX			10000	//512 * 2
sfx_t		s_knownSfx[MAX_SFX];
int			s_numSfx;

#define		LOOP_HASH		128
static	sfx_t		*sfxHash[LOOP_HASH];


cvar_t		*s_volume;
cvar_t		*s_volumeVoice;
cvar_t		*s_testsound;
cvar_t		*s_khz;
cvar_t		*s_allowDynamicMusic;
cvar_t		*s_show;
cvar_t		*s_mixahead;
cvar_t		*s_mixPreStep;
cvar_t		*s_musicVolume;
cvar_t		*s_separation;
cvar_t		*s_lip_threshold_1;
cvar_t		*s_lip_threshold_2;
cvar_t		*s_lip_threshold_3;
cvar_t		*s_lip_threshold_4;
cvar_t		*s_CPUType;
cvar_t		*s_language;	// note that this is distinct from "g_language"
cvar_t		*s_dynamix;
cvar_t		*s_debugdynamic;

typedef struct 
{ 
	unsigned char	volume;
	vec3_t			origin;
//	vec3_t			velocity;
/*	const*/ sfx_t		*sfx;
	int				mergeFrame;
	int				entnum;
	soundChannel_t	entchan;

	// For Open AL
	bool	bProcessed;
	bool	bRelative;
} loopSound_t;

#define	MAX_LOOP_SOUNDS		64
int			numLoopSounds;
loopSound_t	loopSounds[MAX_LOOP_SOUNDS];


int			s_rawend;
portable_samplepair_t	s_rawsamples[MAX_RAW_SAMPLES];
vec3_t		s_entityPosition[MAX_GENTITIES];
int			s_entityWavVol[MAX_GENTITIES];
int			s_entityWavVol_back[MAX_GENTITIES];


/**************************************************************************************************\
*
*	Open AL Specific
*
\**************************************************************************************************/

#define FLT_MAX         3.402823466e+38F
#define FLT_MIN         1.175494351e-38F
#define sqr(a)			((a)*(a))
#define ENV_UPDATE_RATE	100			// Environmental audio update rate (in ms)

//#define DISPLAY_CLOSEST_ENVS		// Displays the closest env. zones (including the one the listener is in)

#define DEFAULT_REF_DISTANCE		300.0f		// Default reference distance
#define DEFAULT_VOICE_REF_DISTANCE	1500.0f		// Default voice reference distance

int			s_UseOpenAL	= true;	// Determines if using Open AL or the default software mixer

ALfloat		listener_pos[3];		// Listener Position
ALfloat		listener_ori[6];		// Listener Orientation
int			s_numChannels;			// Number of AL Sources == Num of Channels
short		s_rawdata[MAX_RAW_SAMPLES*2];	// Used for Raw Samples (Music etc...)

channel_t *S_OpenALPickChannel(int entnum, int entchannel);
int  S_MP3PreProcessLipSync(channel_t *ch, short *data);
void UpdateSingleShotSounds();
void UpdateLoopingSounds();
void AL_UpdateRawSamples();
void S_SetLipSyncs();

// EAX Related

#ifdef HAVE_EAX
typedef struct
{
	ALuint		ulNumApertures;
	ALint		lFXSlotID;
	ALboolean	bUsed;
	struct
	{
		ALfloat vPos1[3];
		ALfloat vPos2[3];
		ALfloat vCenter[3];
	} Aperture[64];
} ENVTABLE, *LPENVTABLE;

typedef struct
{
	long lEnvID;
	long lApertureNum;
	float flDist;
} REVERBDATA, *LPREVERBDATA;

typedef struct
{
	GUID	FXSlotGuid;
	ALint	lEnvID;
} FXSLOTINFO, *LPFXSLOTINFO;

ALboolean				s_bEAX;					// Is EAX 4.0 support available
bool					s_bEALFileLoaded;		// Has an .eal file been loaded for the current level
bool					s_bInWater;				// Underwater effect currently active 
int						s_EnvironmentID;		// EAGLE ID of current environment
LPEAXMANAGER			s_lpEAXManager;			// Pointer to EAXManager object
HINSTANCE				s_hEAXManInst;			// Handle of EAXManager DLL
EAXSet					s_eaxSet;				// EAXSet() function
EAXGet					s_eaxGet;				// EAXGet() function
EAXREVERBPROPERTIES		s_eaxLPCur;				// Current EAX Parameters
LPENVTABLE				s_lpEnvTable=NULL;		// Stores information about each environment zone
long					s_lLastEnvUpdate;		// Time of last EAX update
long					s_lNumEnvironments;		// Number of environment zones
long					s_NumFXSlots;			// Number of EAX 4.0 FX Slots
FXSLOTINFO				s_FXSlotInfo[EAX_MAX_FXSLOTS];	// Stores information about the EAX 4.0 FX Slots

static void InitEAXManager();
static void ReleaseEAXManager();
static bool LoadEALFile(char *szEALFilename);
static void UnloadEALFile();
static void UpdateEAXListener();
static void UpdateEAXBuffer(channel_t *ch);
static void EALFileInit(char *level);
float CalcDistance(EMPOINT A, EMPOINT B);

void Normalize(EAXVECTOR *v)
{
	float flMagnitude;

	flMagnitude = (float)sqrt(sqr(v->x) + sqr(v->y) + sqr(v->z));

	v->x = v->x / flMagnitude;
	v->y = v->y / flMagnitude;
	v->z = v->z / flMagnitude;	
}

// EAX 4.0 GUIDS ... confidential information ...

const GUID EAXPROPERTYID_EAX40_FXSlot0 = { 0xc4d79f1e, 0xf1ac, 0x436b, { 0xa8, 0x1d, 0xa7, 0x38, 0xe7, 0x4, 0x54, 0x69} };

const GUID EAXPROPERTYID_EAX40_FXSlot1 = { 0x8c00e96, 0x74be, 0x4491, { 0x93, 0xaa, 0xe8, 0xad, 0x35, 0xa4, 0x91, 0x17} };

const GUID EAXPROPERTYID_EAX40_FXSlot2 = { 0x1d433b88, 0xf0f6, 0x4637, { 0x91, 0x9f, 0x60, 0xe7, 0xe0, 0x6b, 0x5e, 0xdd} };

const GUID EAXPROPERTYID_EAX40_FXSlot3 = { 0xefff08ea, 0xc7d8, 0x44ab, { 0x93, 0xad, 0x6d, 0xbd, 0x5f, 0x91, 0x0, 0x64} };

const GUID EAXPROPERTYID_EAX40_Context = { 0x1d4870ad, 0xdef, 0x43c0, { 0xa4, 0xc, 0x52, 0x36, 0x32, 0x29, 0x63, 0x42} };

const GUID EAXPROPERTYID_EAX40_Source = { 0x1b86b823, 0x22df, 0x4eae, { 0x8b, 0x3c, 0x12, 0x78, 0xce, 0x54, 0x42, 0x27} };

const GUID EAX_NULL_GUID = { 0x00000000, 0x0000, 0x0000, { 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00 } };

const GUID EAX_PrimaryFXSlotID = { 0xf317866d, 0x924c, 0x450c, { 0x86, 0x1b, 0xe6, 0xda, 0xa2, 0x5e, 0x7c, 0x20} };

const GUID EAX_REVERB_EFFECT = { 0xcf95c8f, 0xa3cc, 0x4849, { 0xb0, 0xb6, 0x83, 0x2e, 0xcc, 0x18, 0x22, 0xdf} };

#endif // HAVE_EAX

/**************************************************************************************************\
*
*	End of Open AL Specific
*
\**************************************************************************************************/

// instead of clearing a whole channel_t struct, we're going to skip the MP3SlidingDecodeBuffer[] buffer in the middle...
//
static inline void Channel_Clear(channel_t *ch)
{
	// memset (ch, 0, sizeof(*ch));

	memset(ch,0,offsetof(channel_t,MP3SlidingDecodeBuffer));

	byte *const p = (byte *)ch + offsetof(channel_t,MP3SlidingDecodeBuffer) + sizeof(ch->MP3SlidingDecodeBuffer);

	memset(p,0,(sizeof(*ch) - offsetof(channel_t,MP3SlidingDecodeBuffer)) - sizeof(ch->MP3SlidingDecodeBuffer));
}

// ====================================================================
// User-setable variables
// ====================================================================
static void DynamicMusicInfoPrint(void)
{
	if (bMusic_IsDynamic)
	{
		// horribly lazy... ;-)
		//
		LPCSTR psRequestMusicState	= Music_BaseStateToString( eMusic_StateRequest );
		LPCSTR psActualMusicState	= Music_BaseStateToString( eMusic_StateActual, qtrue );
		if (psRequestMusicState == NULL)
		{
			psRequestMusicState = "<unknown>";
		}
		if (psActualMusicState	== NULL)
		{
			psActualMusicState	= "<unknown>";
		}
		
		Com_Printf("( Dynamic music ON, request state: '%s'(%d), actual: '%s' (%d) )\n", psRequestMusicState, eMusic_StateRequest, psActualMusicState, eMusic_StateActual);
	}
	else
	{
		Com_Printf("( Dynamic music OFF )\n");
	}
}

void S_SoundInfo_f(void) {	
	Com_Printf("----- Sound Info -----\n" );

	if (!s_soundStarted) {
		Com_Printf ("sound system not started\n");
	} else {
		if ( s_soundMuted ) {
			Com_Printf ("sound system is muted\n");
		}

		if (s_UseOpenAL)
		{
#ifdef HAVE_EAX
			Com_Printf("EAX 4.0 %s supported\n",s_bEAX?"is":"not");
			Com_Printf("Eal file %s loaded\n",s_bEALFileLoaded?"is":"not");
			Com_Printf("s_EnvironmentID = %d\n",s_EnvironmentID);
			Com_Printf("s_bInWater = %s\n",s_bInWater?"true":"false");	
#endif
		}
		else
		{
			Com_Printf("%5d stereo\n", dma.channels - 1);
			Com_Printf("%5d samples\n", dma.samples);
			Com_Printf("%5d samplebits\n", dma.samplebits);
			Com_Printf("%5d submission_chunk\n", dma.submission_chunk);
			Com_Printf("%5d speed\n", dma.speed);
			Com_Printf("0x%x dma buffer\n", dma.buffer);
		}	

		if (bMusic_IsDynamic)
		{		
			DynamicMusicInfoPrint();
			Com_Printf("( Dynamic music set name: \"%s\" )\n",sInfoOnly_CurrentDynamicMusicSet);
		}
		else
		{
			if (!s_allowDynamicMusic->integer)
			{
				Com_Printf("( Dynamic music inhibited (s_allowDynamicMusic == 0) )\n", sMusic_BackgroundLoop );
			}
			if ( tMusic_Info[eBGRNDTRACK_NONDYNAMIC].s_backgroundFile )
			{
				Com_Printf("Background file: %s\n", sMusic_BackgroundLoop );
			}
			else 
			{
				Com_Printf("No background file.\n" );
			}
		}
	}
	S_DisplayFreeMemory();
	Com_Printf("----------------------\n" );
}



/*
================
S_Init
================
*/
void S_Init( void ) {
	ALCcontext *ALCContext = NULL;
	ALCdevice *ALCDevice = NULL;
	ALfloat listenerPos[]={0.0,0.0,0.0};
	ALfloat listenerVel[]={0.0,0.0,0.0};
	ALfloat	listenerOri[]={0.0,0.0,-1.0, 0.0,1.0,0.0};
	cvar_t	*cv;
	qboolean	r;
	int i, j;
	channel_t *ch;
	char *mapname;

	Com_Printf("\n------- sound initialization -------\n");

	s_volume = Cvar_Get ("s_volume", "0.5", CVAR_ARCHIVE);
	s_volumeVoice= Cvar_Get ("s_volumeVoice", "1.0", CVAR_ARCHIVE);
	s_musicVolume = Cvar_Get ("s_musicvolume", "0.25", CVAR_ARCHIVE);
	s_separation = Cvar_Get ("s_separation", "0.5", CVAR_ARCHIVE);
	s_khz = Cvar_Get ("s_khz", "22", CVAR_ARCHIVE|CVAR_LATCH);
	s_allowDynamicMusic = Cvar_Get ("s_allowDynamicMusic", "1", CVAR_ARCHIVE);
	s_mixahead = Cvar_Get ("s_mixahead", "0.2", CVAR_ARCHIVE);

	s_mixPreStep = Cvar_Get ("s_mixPreStep", "0.05", CVAR_ARCHIVE);
	s_show = Cvar_Get ("s_show", "0", CVAR_CHEAT);
	s_testsound = Cvar_Get ("s_testsound", "0", CVAR_CHEAT);
	s_debugdynamic = Cvar_Get("s_debugdynamic","0", 0);
	s_lip_threshold_1 = Cvar_Get("s_threshold1" , "0.3",0);
	s_lip_threshold_2 = Cvar_Get("s_threshold2" , "4",0);
	s_lip_threshold_3 = Cvar_Get("s_threshold3" , "6",0);
	s_lip_threshold_4 = Cvar_Get("s_threshold4" , "8",0);

	s_language = Cvar_Get("s_language","english",CVAR_ARCHIVE | CVAR_NORESTART);

	MP3_InitCvars();

	s_CPUType = Cvar_Get("sys_cpuid","",0);

#ifdef _MSC_VER
#if	!id386
#else
	extern unsigned int uiMMXAvailable;
	uiMMXAvailable = !!(s_CPUType->integer >= CPUID_INTEL_MMX);
#endif
#endif

	cv = Cvar_Get ("s_initsound", "1", CVAR_ROM);
	if ( !cv->integer ) {
		s_soundStarted = 0;	// needed in case you set s_initsound to 0 midgame then snd_restart (div0 err otherwise later)
		Com_Printf ("not initializing.\n");
		Com_Printf("------------------------------------\n");
		return;
	}

	Cmd_AddCommand("play", S_Play_f);
	Cmd_AddCommand("music", S_Music_f);
	Cmd_AddCommand("soundlist", S_SoundList_f);
	Cmd_AddCommand("soundinfo", S_SoundInfo_f);
	Cmd_AddCommand("soundstop", S_StopAllSounds);
	Cmd_AddCommand("mp3_calcvols", S_MP3_CalcVols_f);
	Cmd_AddCommand("s_dynamic", S_SetDynamicMusic_f);

	if (s_UseOpenAL)
	{	
		ALCDevice = alcOpenDevice(NULL);
		if (!ALCDevice)
			return;

		//Create context(s)
		ALCContext = alcCreateContext(ALCDevice, NULL);
		if (!ALCContext)
			return;

		//Set active context
		alcMakeContextCurrent(ALCContext);		
		if (alcGetError(ALCDevice) != ALC_NO_ERROR)
			return;

		s_soundStarted = 1;
		s_soundMuted = 1;
		s_soundtime = 0;
		s_paintedtime = 0;
		s_rawend = 0;

		S_StopAllSounds();

		//S_SoundInfo_f();

		// Set Listener attributes
		alListenerfv(AL_POSITION,listenerPos);
		alListenerfv(AL_VELOCITY,listenerVel);
		alListenerfv(AL_ORIENTATION,listenerOri);

#ifdef HAVE_EAX
		InitEAXManager();
#endif

		memset(s_channels, 0, sizeof(s_channels));

		s_numChannels = 0;

		// Create as many AL Sources (up to Max) as possible
		for (i = 0; i < MAX_CHANNELS; i++)
		{
			alGenSources(1, &s_channels[i].alSource); // &g_Sources[i]);
			if (alGetError() != AL_NO_ERROR)
			{
				// Reached limit of sources
				break;
			}
			alSourcef(s_channels[i].alSource, AL_REFERENCE_DISTANCE, DEFAULT_REF_DISTANCE);
			if (alGetError() != AL_NO_ERROR)
			{
				break;
			}

			// Sources / Channels are not sending to any Slots (other than the Listener / Primary FX Slot)
			s_channels[i].lSlotID = -1;

#ifdef HAVE_EAX
			if (s_bEAX)
			{
				// Remove the RoomAuto flag from each Source (to remove Reverb Engine Statistical
				// model that is assuming units are in metres)
				// Without this call reverb sends from the sources will attenuate too quickly
				// with distance, especially for the non-primary reverb zones.

				unsigned long ulFlags = 0;

				if (s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_FLAGS,
							s_channels[i].alSource, &ulFlags, sizeof(ulFlags))!=AL_NO_ERROR)
							OutputDebugString("Failed to to remove Source flags\n");
			}
#endif

			s_numChannels++;
		}

		// Generate AL Buffers for streaming audio playback (used for MP3s)
		ch = s_channels + 1;
		for (i = 1; i < s_numChannels; i++, ch++)
		{
			for (j = 0; j < NUM_STREAMING_BUFFERS; j++)
			{
				alGenBuffers(1, &(ch->buffers[j].BufferID));
				ch->buffers[j].status = UNQUEUED;
				ch->buffers[j].Data = (char *)Z_Malloc(STREAMING_BUFFER_SIZE, TAG_SND_RAWDATA, qfalse);
			}
		}

		// clear out the lip synching override array
		memset(s_entityWavVol, 0, sizeof(s_entityWavVol));

		// These aren't really relevant for Open AL, but for completeness ...
		dma.speed = 22050;
		dma.channels = 2;
		dma.samplebits = 16;
		dma.samples = 0;
		dma.submission_chunk = 0;
		dma.buffer = NULL;

		// Clamp sound volumes between 0.0f and 1.0f (just in case they aren't already)
		if (s_volume->value < 0.f)
			s_volume->value = 0.f;
		if (s_volume->value > 1.f)
			s_volume->value = 1.f;

		if (s_volumeVoice->value < 0.f)
			s_volumeVoice->value = 0.f;
		if (s_volumeVoice->value > 1.f)
			s_volumeVoice->value = 1.f;

		if (s_musicVolume->value < 0.f)
			s_musicVolume->value = 0.f;
		if (s_musicVolume->value > 1.f)
			s_musicVolume->value = 1.f;
		
		// s_init could be called in game, if so there may be an .eal file 
		// for this level

		mapname = Cvar_VariableString( "mapname" );
#ifdef HAVE_EAX
		EALFileInit(mapname);
#endif

	}
	else
	{
		r = SNDDMA_Init();

		if ( r ) {
			s_soundStarted = 1;
			s_soundMuted = 1;
	//		s_numSfx = 0;	// do NOT do this here now!!!

			s_soundtime = 0;
			s_paintedtime = 0;

			S_StopAllSounds ();

			//S_SoundInfo_f();
		}
	}

//	Com_Printf("------------------------------------\n");

//	Com_Printf("\n--- ambient sound initialization ---\n");

	AS_Init();
}

// only called from snd_restart. QA request...
//
void S_ReloadAllUsedSounds(void)
{
	if (s_soundStarted && !s_soundMuted )
	{
		// new bit, reload all soundsthat are used on the current level...
		//
		for (int i=1 ; i < s_numSfx ; i++)	// start @ 1 to skip freeing default sound
		{
			sfx_t *sfx = &s_knownSfx[i];

			if (!sfx->bInMemory && !sfx->bDefaultSound && sfx->iLastLevelUsedOn == RE_RegisterMedia_GetLevel()){
				S_memoryLoad(sfx);
			}
		}
	}
}

// =======================================================================
// Shutdown sound engine
// =======================================================================

void S_Shutdown( void )
{
	ALCcontext	*ALCContext;
	ALCdevice	*ALCDevice;
	channel_t	*ch;
	int			i, j;

	if ( !s_soundStarted ) {
		return;
	}

	S_FreeAllSFXMem();
	S_UnCacheDynamicMusic();

	if (s_UseOpenAL)
	{
		// Release all the AL Sources (including Music channel (Source 0))
		for (i = 0; i < s_numChannels; i++)
		{
			alDeleteSources(1, &(s_channels[i].alSource));
		}

		// Release Streaming AL Buffers
		ch = s_channels + 1;
		for (i = 1; i < s_numChannels; i++, ch++)
		{
			for (j = 0; j < NUM_STREAMING_BUFFERS; j++)
			{
				alDeleteBuffers(1, &(ch->buffers[j].BufferID));
				ch->buffers[j].BufferID = 0;
				ch->buffers[j].status = UNQUEUED;
				if (ch->buffers[j].Data)
				{
					Z_Free(ch->buffers[j].Data);
					ch->buffers[j].Data = NULL;
				}
			}
		}

		// Get active context
		ALCContext = alcGetCurrentContext();
		// Get device for active context
		ALCDevice = alcGetContextsDevice(ALCContext);
		// Release context(s)
		alcDestroyContext(ALCContext);
		// Close device
		alcCloseDevice(ALCDevice);

#ifdef HAVE_EAX
		ReleaseEAXManager();
#endif

		s_numChannels = 0;

	}
	else
	{
		SNDDMA_Shutdown();
	}

	s_soundStarted = 0;

	Cmd_RemoveCommand("play");
	Cmd_RemoveCommand("music");
	Cmd_RemoveCommand("stopsound");
	Cmd_RemoveCommand("soundlist");
	Cmd_RemoveCommand("soundinfo");
	Cmd_RemoveCommand("soundstop");
	Cmd_RemoveCommand("mp3_calcvols");
	Cmd_RemoveCommand("s_dynamic");
	AS_Free();
}



/*
	Mutes / Unmutes all OpenAL sound
*/
void S_AL_MuteAllSounds(qboolean bMute)
{
     if (!s_soundStarted)
          return;

     if (!s_UseOpenAL)
          return;

     if (bMute)
     {
          alListenerf(AL_GAIN, 0.0f);
     }
     else
     {
          alListenerf(AL_GAIN, 1.0f);
     }
}





// =======================================================================
// Load a sound
// =======================================================================
/*
================
return a hash value for the sfx name
================
*/
static long S_HashSFXName(const char *name) {
	int		i;
	long	hash;
	char	letter;

	hash = 0;
	i = 0;
	while (name[i] != '\0') {
		letter = tolower(name[i]);
		if (letter =='.') break;				// don't include extension
		if (letter =='\\') letter = '/';		// damn path names
		hash+=(long)(letter)*(i+119);
		i++;
	}
	hash &= (LOOP_HASH-1);
	return hash;
}

/*
==================
S_FindName

Will allocate a new sfx if it isn't found
==================
*/
sfx_t *S_FindName( const char *name ) {
	int		i;
	int		hash;

	sfx_t	*sfx;

	if (!name) {
		Com_Error (ERR_FATAL, "S_FindName: NULL\n");
	}
	if (!name[0]) {
		Com_Error (ERR_FATAL, "S_FindName: empty name\n");
	}

	if (strlen(name) >= MAX_QPATH) {
		Com_Error (ERR_FATAL, "Sound name too long: %s", name);
	}

	char sSoundNameNoExt[MAX_QPATH];
	COM_StripExtension(name,sSoundNameNoExt);

	hash = S_HashSFXName(sSoundNameNoExt);

	sfx = sfxHash[hash];
	// see if already loaded
	while (sfx) {
		if (!Q_stricmp(sfx->sSoundName, sSoundNameNoExt) ) {
			return sfx;
		}
		sfx = sfx->next;
	}
/*
	// find a free sfx
	for (i=0 ; i < s_numSfx ; i++) {
		if (!s_knownSfx[i].soundName[0]) {
			break;
		}
	}
*/
	i = s_numSfx;	//we don't clear the soundName after failed loads any more, so it'll always be the last entry
		
	if (s_numSfx == MAX_SFX) 
	{
		// ok, no sfx's free, but are there any with defaultSound set? (which the registering ent will never
		//	see because he gets zero returned if it's default...)
		//
		for (i=0 ; i < s_numSfx ; i++) {
			if (s_knownSfx[i].bDefaultSound) {
				break;
			}
		}

		if (i==s_numSfx)
		{	  
			// genuinely out of handles...

			// if we ever reach this, let me know and I'll either boost the array or put in a map-used-on
			//	reference to enable sfx_t recycling. TA codebase relies on being able to have structs for every sound
			//	used anywhere, ever, all at once (though audio bit-buffer gets recycled). SOF1 used about 1900 distinct
			//	events, so current MAX_SFX limit should do, or only need a small boost...	-ste
			//

			Com_Error (ERR_FATAL, "S_FindName: out of sfx_t");
		}
	}
	else
	{
		s_numSfx++;
	}
	
	sfx = &s_knownSfx[i];
	memset (sfx, 0, sizeof(*sfx));
	Q_strncpyz(sfx->sSoundName, sSoundNameNoExt, sizeof(sfx->sSoundName));
	Q_strlwr(sfx->sSoundName);//force it down low

	sfx->next = sfxHash[hash];
	sfxHash[hash] = sfx;

	return sfx;
}

/*
=================
S_DefaultSound
=================
*/
static void S_DefaultSound( sfx_t *sfx ) {
	
	int		i;

	sfx->iSoundLengthInSamples	= 512;								// #samples, ie shorts
	sfx->pSoundData				= (short *)	SND_malloc(512*2, sfx);	// ... so *2 for alloc bytes	
	sfx->bInMemory				= true;
	
	for ( i=0 ; i < sfx->iSoundLengthInSamples ; i++ ) 
	{
		sfx->pSoundData[i] = i;
	}
}


/*
===================
S_DisableSounds

Disables sounds until the next S_BeginRegistration.
This is called when the hunk is cleared and the sounds
are no longer valid.
===================
*/
void S_DisableSounds( void ) {
	S_StopAllSounds();
	s_soundMuted = qtrue;
}

/*
=====================
S_BeginRegistration

=====================
*/
void S_BeginRegistration( void )
{
	char *mapname;

	s_soundMuted = qfalse;		// we can play again

#ifdef HAVE_EAX
	// Find name of level so we can load in the appropriate EAL file
	if (s_UseOpenAL)
	{
		mapname = Cvar_VariableString( "mapname" );
		EALFileInit(mapname);
		// clear carry crap from previous map
		for (int i = 0; i < EAX_MAX_FXSLOTS; i++)
		{
			s_FXSlotInfo[i].lEnvID = -1;
		}
	}
#endif

	if (s_numSfx == 0) {
		SND_setup();

		s_numSfx = 0;
		memset( s_knownSfx, 0, sizeof( s_knownSfx ) );
		memset(sfxHash, 0, sizeof(sfx_t *)*LOOP_HASH);

#ifdef _DEBUG
		sfx_t *sfx = S_FindName( "***DEFAULT***" );
		S_DefaultSound( sfx );
#else
		S_RegisterSound("sound/null.wav");
#endif
	}
}

#ifdef HAVE_EAX
static void EALFileInit(char *level)
{
	long		lRoom;
	char		name[MAX_QPATH];
	char		szEALFilename[MAX_QPATH];
	int			i;

	// If an EAL File is already unloaded, remove it
	if (s_bEALFileLoaded)
	{
		UnloadEALFile();
	}

	// Reset variables
	s_bInWater = false;

	// Try and load an EAL file for the new level
	COM_StripExtension(level, name);
	Com_sprintf(szEALFilename, MAX_QPATH, "eagle/%s.eal", name);

	s_bEALFileLoaded = LoadEALFile(szEALFilename);

	if (!s_bEALFileLoaded)
	{
		Com_sprintf(szEALFilename, MAX_QPATH, "base/eagle/%s.eal", name);
		s_bEALFileLoaded = LoadEALFile(szEALFilename);
	}

	if (s_bEALFileLoaded)
	{
		s_lLastEnvUpdate = Com_Milliseconds();
	}
	else
	{
		// Mute reverbs if no EAL file is found
		if ((s_bEAX)&&(s_eaxSet))
		{
			lRoom = -10000;
			for (i = 0; i < s_NumFXSlots; i++)
			{
				s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXREVERB_ROOM, NULL,
					&lRoom, sizeof(long));
			}
		}
	}
}
#endif // HAVE_EAX



/*
==================
S_RegisterSound

Creates a default buzz sound if the file can't be loaded
==================
*/
sfxHandle_t	S_RegisterSound( const char *name) 
{
	sfx_t	*sfx;

	if (!s_soundStarted) {
		return 0;
	}

	if ( strlen( name ) >= MAX_QPATH ) {		
		Com_Printf( S_COLOR_RED"Sound name exceeds MAX_QPATH - %s\n", name );
		return 0;
	}

	sfx = S_FindName( name );

	SND_TouchSFX(sfx);

	if ( sfx->bDefaultSound )
		return 0;

	if (s_UseOpenAL)
	{
		if ((sfx->pSoundData) || (sfx->Buffer))
			return sfx - s_knownSfx;
	}
	else
	{
		if ( sfx->pSoundData )
		{
			return sfx - s_knownSfx;
		}
	}

	sfx->bInMemory = false;

	S_memoryLoad(sfx);

	if ( sfx->bDefaultSound ) {
#ifndef FINAL_BUILD
		Com_DPrintf( S_COLOR_YELLOW "WARNING: could not find %s - using default\n", sfx->sSoundName );
#endif


		return 0;
	}

	return sfx - s_knownSfx;
}

void S_memoryLoad(sfx_t	*sfx) 
{
	// load the sound file...
	//
	if ( !S_LoadSound( sfx ) ) 
	{
//		Com_Printf( S_COLOR_YELLOW "WARNING: couldn't load sound: %s\n", sfx->sSoundName );
		sfx->bDefaultSound = true;
	}
	sfx->bInMemory = true;
}



//=============================================================================
static qboolean S_CheckChannelStomp( int chan1, int chan2 )
{
	if (!s_UseOpenAL)
	{
		if ( chan1 == chan2 )
		{
			return qtrue;
		}
	}

	if ( ( chan1 == CHAN_VOICE || chan1 == CHAN_VOICE_ATTEN || chan1 == CHAN_VOICE_GLOBAL  ) && ( chan2 == CHAN_VOICE || chan2 == CHAN_VOICE_ATTEN || chan2 == CHAN_VOICE_GLOBAL ) )
	{
		return qtrue;
	}

	return qfalse;
}


/*
=================
S_PickChannel
=================
*/
// there were 2 versions of this, one for A3D and one normal, but the normal one wouldn't compile because
//	it hadn't been updated for some time, so rather than risk anything weird/out of date, I just removed the 
//	A3D lines from this version and deleted the other one. 
//
// If this really bothers you then feel free to play with it. -Ste.
//
channel_t *S_PickChannel(int entnum, int entchannel)
{
    int			ch_idx;
	channel_t	*ch, *firstToDie;
	qboolean	foundChan = qfalse;

	if (s_UseOpenAL)
		return S_OpenALPickChannel(entnum, entchannel);

	if ( entchannel<0 ) {
		Com_Error (ERR_DROP, "S_PickChannel: entchannel<0");
	}

	// Check for replacement sound, or find the best one to replace

    firstToDie = &s_channels[0];

	for ( int pass = 0; (pass < ((entchannel == CHAN_AUTO || entchannel == CHAN_LESS_ATTEN)?1:2)) && !foundChan; pass++ )
	{
		for (ch_idx = 0, ch = &s_channels[0]; ch_idx < MAX_CHANNELS ; ch_idx++, ch++ ) 
		{
			if ( entchannel == CHAN_AUTO || entchannel == CHAN_LESS_ATTEN || pass > 0 )
			{//if we're on the second pass, just find the first open chan
				if ( !ch->thesfx )
				{//grab the first open channel
					firstToDie = ch;
					break;
				}

			}
			else if ( ch->entnum == entnum && S_CheckChannelStomp( ch->entchannel, entchannel ) ) 
			{
				// always override sound from same entity
				if ( s_show->integer == 1 && ch->thesfx ) {
					Com_Printf( S_COLOR_YELLOW"...overrides %s\n", ch->thesfx->sSoundName );
					ch->thesfx = 0;	//just to clear the next error msg
				}
				firstToDie = ch;
				foundChan = qtrue;
				break;
			}

			// don't let anything else override local player sounds
			if ( ch->entnum == listener_number 	&& entnum != listener_number && ch->thesfx) {
				continue;
			}

			// don't override loop sounds
			if ( ch->loopSound ) {
				continue;
			}

			if ( ch->startSample < firstToDie->startSample ) {
				firstToDie = ch;
			}
		}
	}

	if ( s_show->integer == 1 && firstToDie->thesfx ) {
		Com_Printf( S_COLOR_RED"***kicking %s\n", firstToDie->thesfx->sSoundName );
	}

	Channel_Clear(firstToDie);	// memset(firstToDie, 0, sizeof(*firstToDie));
    
	return firstToDie;
}



/*
	For use with Open AL

	Allows more than one sound of the same type to emanate from the same entity - sounds much better
	on hardware this way esp. rapid fire modes of weapons!
*/
channel_t *S_OpenALPickChannel(int entnum, int entchannel)
{
    int			ch_idx;
	channel_t	*ch, *ch_firstToDie;
	bool	foundChan = false;
	float	source_pos[3];

	if ( entchannel < 0 ) 
	{
		Com_Error (ERR_DROP, "S_PickChannel: entchannel<0");
	}

	// Check for replacement sound, or find the best one to replace

	ch_firstToDie = s_channels + 1;	// channel 0 is reserved for Music

	for (ch_idx = 1, ch = s_channels + ch_idx; ch_idx < s_numChannels; ch_idx++, ch++)
	{
		if ( ch->entnum == entnum && S_CheckChannelStomp( ch->entchannel, entchannel ) ) 
		{
			// always override sound from same entity
			if ( s_show->integer == 1 && ch->thesfx ) {
				Com_Printf( S_COLOR_YELLOW"...overrides %s\n", ch->thesfx->sSoundName );
				ch->thesfx = 0;	//just to clear the next error msg
			}
			ch_firstToDie = ch;
			foundChan = true;
			break;
		}
	}

	if (!foundChan)
	for (ch_idx = 1, ch = s_channels + ch_idx; ch_idx < s_numChannels; ch_idx++, ch++)
	{
		// See if the channel is free
		if (!ch->thesfx)
		{
			ch_firstToDie = ch;
			foundChan = true;
			break;
		}
	}

	if (!foundChan)
	{
		for (ch_idx = 1, ch = s_channels + ch_idx; ch_idx < s_numChannels; ch_idx++, ch++)
		{
			if (	(ch->entnum == entnum) && (ch->entchannel == entchannel) && (ch->entchannel != CHAN_AMBIENT)
				 && (ch->entnum != listener_number) ) 
			{
				// Same entity and same type of sound effect (entchannel)
				ch_firstToDie = ch;
				foundChan = true;
				break;
			}
		}
	}

	int longestDist;
	int dist;

	if (!foundChan)
	{
		// Find sound effect furthest from listener
		ch = s_channels + 1;

		if (ch->fixed_origin)
		{
			// Convert to Open AL co-ordinates
			source_pos[0] = ch->origin[0];
			source_pos[1] = ch->origin[2];
			source_pos[2] = -ch->origin[1];

			longestDist = ((listener_pos[0] - source_pos[0]) * (listener_pos[0] - source_pos[0])) +
						  ((listener_pos[1] - source_pos[1]) * (listener_pos[1] - source_pos[1])) +
						  ((listener_pos[2] - source_pos[2]) * (listener_pos[2] - source_pos[2]));
		}
		else
		{
			if (ch->entnum == listener_number)
				longestDist = 0;
			else
			{
				if (ch->bLooping)
				{
					// Convert to Open AL co-ordinates
					source_pos[0] = loopSounds[ch->entnum].origin[0];
					source_pos[1] = loopSounds[ch->entnum].origin[2];
					source_pos[2] = -loopSounds[ch->entnum].origin[1];
				}
				else
				{
					// Convert to Open AL co-ordinates
					source_pos[0] = s_entityPosition[ch->entnum][0];
					source_pos[1] = s_entityPosition[ch->entnum][2];
					source_pos[2] = -s_entityPosition[ch->entnum][1];
				}

				longestDist = ((listener_pos[0] - source_pos[0]) * (listener_pos[0] - source_pos[0])) +
							  ((listener_pos[1] - source_pos[1]) * (listener_pos[1] - source_pos[1])) +
							  ((listener_pos[2] - source_pos[2]) * (listener_pos[2] - source_pos[2]));
			}
		}

		for (ch_idx = 2, ch = s_channels + ch_idx; ch_idx < s_numChannels; ch_idx++, ch++)
		{
			if (ch->fixed_origin)
			{
				// Convert to Open AL co-ordinates
				source_pos[0] = ch->origin[0];
				source_pos[1] = ch->origin[2];
				source_pos[2] = -ch->origin[1];
			
				dist = ((listener_pos[0] - source_pos[0]) * (listener_pos[0] - source_pos[0])) +
					   ((listener_pos[1] - source_pos[1]) * (listener_pos[1] - source_pos[1])) +
					   ((listener_pos[2] - source_pos[2]) * (listener_pos[2] - source_pos[2]));
			}
			else
			{
				if (ch->entnum == listener_number)
					dist = 0;
				else
				{
					if (ch->bLooping)
					{
						// Convert to Open AL co-ordinates
						source_pos[0] = loopSounds[ch->entnum].origin[0];
						source_pos[1] = loopSounds[ch->entnum].origin[2];
						source_pos[2] = -loopSounds[ch->entnum].origin[1];
					}
					else
					{
						// Convert to Open AL co-ordinates
						source_pos[0] = s_entityPosition[ch->entnum][0];
						source_pos[1] = s_entityPosition[ch->entnum][2];
						source_pos[2] = -s_entityPosition[ch->entnum][1];
					}

					dist = ((listener_pos[0] - source_pos[0]) * (listener_pos[0] - source_pos[0])) +
					       ((listener_pos[1] - source_pos[1]) * (listener_pos[1] - source_pos[1])) +
						   ((listener_pos[2] - source_pos[2]) * (listener_pos[2] - source_pos[2]));
				}
			}

			if (dist > longestDist)
			{
				longestDist = dist;
				ch_firstToDie = ch;
			}
		}
	}

	if (ch_firstToDie->bPlaying)
	{
		if (s_show->integer == 1 && ch_firstToDie->thesfx ) 
		{
			Com_Printf(S_COLOR_RED"***kicking %s\n", ch_firstToDie->thesfx->sSoundName );
		}

		// Stop sound
		alSourceStop(ch_firstToDie->alSource);
		ch_firstToDie->bPlaying = false;
	}

	// Reset channel variables
	memset(&ch_firstToDie->MP3StreamHeader, 0, sizeof(MP3STREAM));
	ch_firstToDie->bLooping = false;
	ch_firstToDie->bProcessed = false;
	ch_firstToDie->bStreaming = false;
	
    return ch_firstToDie;
}


/*
=================
S_SpatializeOrigin

Used for spatializing s_channels
=================
*/
static void S_SpatializeOrigin (const vec3_t origin, float master_vol, int *left_vol, int *right_vol, soundChannel_t channel)
{
    vec_t		dot;
    vec_t		dist;
    vec_t		lscale, rscale, scale;
    vec3_t		source_vec;
	float		dist_mult = SOUND_ATTENUATE;

	// calculate stereo seperation and distance attenuation
	VectorSubtract(origin, listener_origin, source_vec);

	dist = VectorNormalize(source_vec);
	if ( channel == CHAN_VOICE )
	{
		dist -= SOUND_FULLVOLUME * 3.0f;
//		dist_mult = VOICE_ATTENUATE;	// tweak added (this fixes an NPC dialogue "in your ears" bug, but we're not sure if it'll make a bunch of others fade too early. Too close to shipping...)
	}
	else if ( channel == CHAN_LESS_ATTEN )
	{
		dist -= SOUND_FULLVOLUME * 5.0f; // maybe is too large
	}
	else if ( channel == CHAN_VOICE_ATTEN )
	{
		dist -= SOUND_FULLVOLUME * 1.35f; // used to be 0.15f, dropped off too sharply - dmv
		dist_mult = VOICE_ATTENUATE;
	}
	else if ( channel == CHAN_VOICE_GLOBAL )
	{
		dist = -1;
	}
	else	// use normal attenuation.
	{
		dist -= SOUND_FULLVOLUME;
	}
	
	if (dist < 0)
	{
		dist = 0;			// close enough to be at full volume
	}
	dist *= dist_mult;		// different attenuation levels

	dot = -DotProduct(listener_axis[1], source_vec);

	if (dma.channels == 1)	// || !dist_mult)
	{ // no attenuation = no spatialization
		rscale = SOUND_FMAXVOL;
		lscale = SOUND_FMAXVOL;
	}
	else
	{
		//rscale = 0.5 * (1.0 + dot);
		//lscale = 0.5 * (1.0 - dot);
		rscale = s_separation->value + ( 1.0f - s_separation->value ) * dot;
		lscale = s_separation->value - ( 1.0f - s_separation->value ) * dot;
		if ( rscale < 0 ) 
		{
			rscale = 0;
		}
		if ( lscale < 0 ) 
		{
			lscale = 0;
		}
	}

	// add in distance effect
	scale = (1.0f - dist) * rscale;
	*right_vol = (int) (master_vol * scale);
	if (*right_vol < 0)
	{
		*right_vol = 0;
	}

	scale = (1.0f - dist) * lscale;
	*left_vol = (int) (master_vol * scale);
	if (*left_vol < 0)
	{
		*left_vol = 0;
	}
}


// =======================================================================
// Start a sound effect
// =======================================================================

/*
====================
S_StartAmbientSound

Starts an ambient, 'one-shot" sound.
====================
*/

void S_StartAmbientSound( const vec3_t origin, int entityNum, unsigned char volume, sfxHandle_t sfxHandle )
{
	channel_t	*ch;
	/*const*/ sfx_t *sfx;

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}
	if ( !origin && ( entityNum < 0 || entityNum > MAX_GENTITIES ) )
		Com_Error( ERR_DROP, "S_StartAmbientSound: bad entitynum %i", entityNum );

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx )
		Com_Error( ERR_DROP, "S_StartAmbientSound: handle %i out of range", sfxHandle );

	sfx = &s_knownSfx[ sfxHandle ];
	if (sfx->bInMemory == qfalse){
		S_memoryLoad(sfx);
	}	
	SND_TouchSFX(sfx);

	if (s_UseOpenAL)
	{
		if (volume==0)
			return;
	}

	if ( s_show->integer == 1 ) {
		Com_Printf( "%i : %s on (%d) Ambient\n", s_paintedtime, sfx->sSoundName, entityNum );
	}

	// pick a channel to play on
	ch = S_PickChannel( entityNum, CHAN_AMBIENT );
	if (!ch) {
		return;
	}
	
	if (origin) 
	{
		VectorCopy (origin, ch->origin);
		ch->fixed_origin = qtrue;
	} 
	else 
	{
		ch->fixed_origin = qfalse;
	}

	ch->master_vol = volume;
	ch->entnum = entityNum;
	ch->entchannel = CHAN_AMBIENT;
	ch->thesfx = sfx;
	ch->startSample = START_SAMPLE_IMMEDIATE;

	ch->leftvol = ch->master_vol;		// these will get calced at next spatialize
	ch->rightvol = ch->master_vol;		// unless the game isn't running

	if (sfx->pMP3StreamHeader)
	{
		memcpy(&ch->MP3StreamHeader,sfx->pMP3StreamHeader,	sizeof(ch->MP3StreamHeader));
		//ch->iMP3SlidingDecodeWritePos = 0; // These will be zero from the memset in S_PickChannel(), but keep them here for reference...
		//ch->iMP3SlidingDecodeWindowPos= 0; //
	}
	else
	{
		memset(&ch->MP3StreamHeader,0,						sizeof(ch->MP3StreamHeader));
	}
}

/*
====================
S_StartSound

Validates the parms and ques the sound up
if pos is NULL, the sound will be dynamically sourced from the entity
Entchannel 0 will never override a playing sound
====================
*/
void S_StartSound(const vec3_t origin, int entityNum, soundChannel_t entchannel, sfxHandle_t sfxHandle ) 
{
	int i;
	channel_t	*ch;
	/*const*/ sfx_t *sfx;

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}

	if ( !origin && ( entityNum < 0 || entityNum > MAX_GENTITIES ) ) {
		Com_Error( ERR_DROP, "S_StartSound: bad entitynum %i", entityNum );
	}

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx ) {
		Com_Error( ERR_DROP, "S_StartSound: handle %i out of range", sfxHandle );
	}

	sfx = &s_knownSfx[ sfxHandle ];
	if (sfx->bInMemory == qfalse){
		S_memoryLoad(sfx);
	}
	SND_TouchSFX(sfx);

	if ( s_show->integer == 1 ) {
		Com_Printf( "%i : %s for ent %d, chan=%d\n", s_paintedtime, sfx->sSoundName, entityNum, entchannel );
	}

	if (s_UseOpenAL)
	{
		if (entchannel == CHAN_VOICE)
		{
			// Make howlers and sand_creature VOICE effects use the normal fall-off (they will still be affected
			// by the Voice Volume)
			if ((strstr(sfx->sSoundName, "sand_creature")!=NULL) || (strstr(sfx->sSoundName, "howler")!=NULL))
			{
				entchannel = CHAN_VOICE_ATTEN;
			}
		}
		if (entchannel == CHAN_WEAPON)
		{
			// Check if we are playing a 'charging' sound, if so, stop it now ..
			ch = s_channels + 1;
			for (i = 1; i < s_numChannels; i++, ch++)
			{
				if ((ch->entnum == entityNum) && (ch->entchannel == CHAN_WEAPON) && (ch->thesfx) && (strstr(ch->thesfx->sSoundName, "altcharge") != NULL))
				{
					// Stop this sound
					alSourceStop(ch->alSource);
					alSourcei(ch->alSource, AL_BUFFER, NULL);
					ch->bPlaying = false;
					ch->thesfx = NULL;
					break;
				}
			}
		}
		else
		{
			ch = s_channels + 1;
			for (i = 1; i < s_numChannels; i++, ch++)
			{
				if ((ch->entnum == entityNum) && (ch->thesfx) && (strstr(ch->thesfx->sSoundName, "falling") != NULL))
				{
					// Stop this sound
					alSourceStop(ch->alSource);
					alSourcei(ch->alSource, AL_BUFFER, NULL);
					ch->bPlaying = false;
					ch->thesfx = NULL;
					break;
				}
			}
		}
	}


	// pick a channel to play on
	
	ch = S_PickChannel( entityNum, entchannel );
	if (!ch) {
		return;
	}

	if (origin) {
		VectorCopy (origin, ch->origin);
		ch->fixed_origin = qtrue;
	} else {
		ch->fixed_origin = qfalse;
	}

	ch->master_vol = SOUND_MAXVOL;	//FIXME: Um.. control?
	ch->entnum = entityNum;
	ch->entchannel = entchannel;
	ch->thesfx = sfx;
	ch->startSample = START_SAMPLE_IMMEDIATE;

	ch->leftvol = ch->master_vol;		// these will get calced at next spatialize
	ch->rightvol = ch->master_vol;		// unless the game isn't running

	if (entchannel < CHAN_AMBIENT && entityNum == listener_number) {	//only do it for body sounds not local sounds
		ch->master_vol = SOUND_MAXVOL * SOUND_FMAXVOL;	//this won't be attenuated so let it scale down
	}
	if ( entchannel == CHAN_VOICE || entchannel == CHAN_VOICE_ATTEN || entchannel == CHAN_VOICE_GLOBAL ) 
	{
		s_entityWavVol[ ch->entnum ] = -1;	//we've started the sound but it's silent for now
	}

	if (sfx->pMP3StreamHeader)
	{
		memcpy(&ch->MP3StreamHeader,sfx->pMP3StreamHeader,	sizeof(ch->MP3StreamHeader));
		//ch->iMP3SlidingDecodeWritePos = 0; // These will be zero from the memset in S_PickChannel(), but keep them here for reference...
		//ch->iMP3SlidingDecodeWindowPos= 0; //
	}
	else
	{
		memset(&ch->MP3StreamHeader,0,						sizeof(ch->MP3StreamHeader));
	}
}

/*
==================
S_StartLocalSound
==================
*/
void S_StartLocalSound( sfxHandle_t sfxHandle, int channelNum ) {
	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx ) {
		Com_Error( ERR_DROP, "S_StartLocalSound: handle %i out of range", sfxHandle );
	}

	S_StartSound (NULL, listener_number, (soundChannel_t)channelNum, sfxHandle );
}


/*
==================
S_StartLocalLoopingSound
==================
*/
void S_StartLocalLoopingSound( sfxHandle_t sfxHandle) {
	vec3_t nullVec = {0,0,0};

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx ) {
		Com_Error( ERR_DROP, "S_StartLocalLoopingSound: handle %i out of range", sfxHandle );
	}

	S_AddLoopingSound( listener_number, nullVec, nullVec, sfxHandle );

}

// returns length in milliseconds of supplied sound effect...  (else 0 for bad handle now)
//
float S_GetSampleLengthInMilliSeconds( sfxHandle_t sfxHandle)
{
	sfx_t *sfx;

	if (!s_soundStarted)
	{	//we have no sound, so let's just make a reasonable guess
		return 512 * 1000;
	}

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx )
		return 0.0f;

	sfx = &s_knownSfx[ sfxHandle ];

	float f = (float)sfx->iSoundLengthInSamples / (float)dma.speed;

	return (f * 1000);
}


/*
==================
S_ClearSoundBuffer

If we are about to perform file access, clear the buffer
so sound doesn't stutter.
==================
*/
void S_ClearSoundBuffer( void ) {
	int		clear;
		
	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}
#if 0	//this causes scripts to freak when the sounds get cut...
	// clear all the sounds so they don't
	// start back up after the load finishes
	memset( s_channels, 0, sizeof( s_channels ) );
	// clear out the lip synching override array
	memset(s_entityWavVol, 0,sizeof(s_entityWavVol));
#endif
	s_rawend = 0;

	if (!s_UseOpenAL)
	{
		if (dma.samplebits == 8)
			clear = 0x80;
		else
			clear = 0;

		SNDDMA_BeginPainting ();
		if (dma.buffer)
			memset(dma.buffer, clear, dma.samples * dma.samplebits/8);
		SNDDMA_Submit ();
	}
	else
	{
		s_paintedtime = 0;
		s_soundtime = 0;
	}
}


// kinda kludgy way to stop a special-use sfx_t playing...
//
void S_CIN_StopSound(sfxHandle_t sfxHandle)
{
	if ( sfxHandle < 0 || sfxHandle >= s_numSfx ) {
		Com_Error( ERR_DROP, "S_CIN_StopSound: handle %i out of range", sfxHandle );
	}

	sfx_t *sfx = &s_knownSfx[ sfxHandle ];
	channel_t *ch = s_channels;
	int i;

	for ( i = 0; i < MAX_CHANNELS ; i++, ch++ ) 
	{
		if ( !ch->thesfx || (ch->leftvol<0.25 && ch->rightvol<0.25 )) {
			continue;
		}
		if (ch->thesfx == sfx)
		{
			if (s_UseOpenAL)
			{
				alSourceStop(s_channels[i].alSource);
			}
			SND_FreeSFXMem(ch->thesfx);	// heh, may as well...
			ch->thesfx = NULL;
			memset(&ch->MP3StreamHeader, 0, sizeof(MP3STREAM));
			ch->bLooping = false;
			ch->bProcessed = false;
			ch->bPlaying = false;
			ch->bStreaming = false;
			break;
		}
	}
}


/*
==================
S_StopAllSounds
==================
*/
void S_StopSounds(void)
{
	int i; //, j;
	channel_t *ch;

	if ( !s_soundStarted ) {
		return;
	}

	// stop looping sounds
	S_ClearLoopingSounds();

	// clear all the s_channels
	if (s_UseOpenAL)
	{
		ch = s_channels;
		for (i = 0; i < s_numChannels; i++, ch++)
		{
			alSourceStop(s_channels[i].alSource);
			alSourcei(s_channels[i].alSource, AL_BUFFER, NULL);
			ch->thesfx = NULL;
			memset(&ch->MP3StreamHeader, 0, sizeof(MP3STREAM));
			ch->bLooping = false;
			ch->bProcessed = false;
			ch->bPlaying = false;
			ch->bStreaming = false;
		}
	}
	else
	{
		memset(s_channels, 0, sizeof(s_channels));
	}

	// clear out the lip synching override array
	memset(s_entityWavVol, 0,sizeof(s_entityWavVol));

	S_ClearSoundBuffer ();
}

/*
==================
S_StopAllSounds
 and music
==================
*/
void S_StopAllSounds(void) {
	if ( !s_soundStarted ) {
		return;
	}
	// stop the background music
	S_StopBackgroundTrack();

	S_StopSounds();
}

/*
==============================================================

continuous looping sounds are added each frame

==============================================================
*/

/*
==================
S_ClearLoopingSounds

==================
*/
void S_ClearLoopingSounds( void )
{
	int i;

	if (s_UseOpenAL)
	{
		for (i = 0; i < MAX_LOOP_SOUNDS; i++)
			loopSounds[i].bProcessed = false;
	}
	numLoopSounds = 0;

}

/*
==================
S_AddLoopingSound

Called during entity generation for a frame
Include velocity in case I get around to doing doppler...
==================
*/
void S_AddLoopingSound( int entityNum, const vec3_t origin, const vec3_t velocity, sfxHandle_t sfxHandle, soundChannel_t chan ) {
	/*const*/ sfx_t *sfx;

  	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}
	if ( numLoopSounds >= MAX_LOOP_SOUNDS ) {
		//assert(numLoopSounds<MAX_LOOP_SOUNDS);
#ifndef FINAL_BUILD
		Com_Printf( "S_AddLoopingSound: MAX_LOOP_SOUNDS\n");
#endif
		return;
	}

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx ) {
		Com_Error( ERR_DROP, "S_AddLoopingSound: handle %i out of range", sfxHandle );
	}

	sfx = &s_knownSfx[ sfxHandle ];
	if (sfx->bInMemory == qfalse){
		S_memoryLoad(sfx);
	}
	SND_TouchSFX(sfx);

	if ( !sfx->iSoundLengthInSamples ) {
		Com_Error( ERR_DROP, "%s has length 0", sfx->sSoundName );
	}
	assert(!sfx->pMP3StreamHeader);
	VectorCopy( origin, loopSounds[numLoopSounds].origin );
//	VectorCopy( velocity, loopSounds[numLoopSounds].velocity );
	loopSounds[numLoopSounds].sfx = sfx;
	loopSounds[numLoopSounds].volume = SOUND_MAXVOL;
	loopSounds[numLoopSounds].entnum = entityNum;
	loopSounds[numLoopSounds].entchan = chan;
	numLoopSounds++;
}


/*
==================
S_AddAmbientLoopingSound
==================
*/
void S_AddAmbientLoopingSound( const vec3_t origin, unsigned char volume, sfxHandle_t sfxHandle ) 
{
	/*const*/ sfx_t *sfx;

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}
	if ( numLoopSounds >= MAX_LOOP_SOUNDS ) {
		return;
	}

	if (s_UseOpenAL)
	{
		if (volume == 0)
			return;
	}

	if ( sfxHandle < 0 || sfxHandle >= s_numSfx ) {
		Com_Error( ERR_DROP, "S_StartSound: handle %i out of range", sfxHandle );
	}

	sfx = &s_knownSfx[ sfxHandle ];
	if (sfx->bInMemory == qfalse){
		S_memoryLoad(sfx);
	}
	SND_TouchSFX(sfx);

	if ( !sfx->iSoundLengthInSamples ) {
		Com_Error( ERR_DROP, "%s has length 0", sfx->sSoundName );
	}
	VectorCopy( origin, loopSounds[numLoopSounds].origin );
	loopSounds[numLoopSounds].sfx = sfx;
	assert(!sfx->pMP3StreamHeader);
	
	//TODO: Calculate the distance falloff
	loopSounds[numLoopSounds].volume = volume;
	numLoopSounds++;
}



/*
==================
S_AddLoopSounds

Spatialize all of the looping sounds.
All sounds are on the same cycle, so any duplicates can just
sum up the channel multipliers.
==================
*/
void S_AddLoopSounds (void) 
{
	int			i, j;
	int			left, right, left_total, right_total;
	channel_t	*ch;
	loopSound_t	*loop, *loop2;
	static int	loopFrame;

	loopFrame++;
	for ( i = 0 ; i < numLoopSounds ; i++) {
		loop = &loopSounds[i];
		if ( loop->mergeFrame == loopFrame ) {
			continue;	// already merged into an earlier sound
		}

		// find the total contribution of all sounds of this type
		left_total = right_total = 0;

		for ( j = i ; j < numLoopSounds ; j++) {
			loop2 = &loopSounds[j];
			if ( loop2->sfx != loop->sfx ) {
				continue;
			}
			loop2->mergeFrame = loopFrame;	// don't check this again later

			S_SpatializeOrigin( loop2->origin, loop2->volume, &left, &right, loop2->entchan);

			left_total += left;
			right_total += right;
		}

		if (left_total == 0 && right_total == 0)
			continue;		// not audible

		// allocate a channel
		ch = S_PickChannel(0, 0);
		if (!ch)
			return;

		if (left_total > SOUND_MAXVOL)
			left_total = SOUND_MAXVOL;
		if (right_total > SOUND_MAXVOL)
			right_total = SOUND_MAXVOL;
		ch->leftvol = left_total;
		ch->rightvol = right_total;
		ch->loopSound = qtrue;	// remove next frame
		ch->thesfx = loop->sfx;

		// you cannot use MP3 files here because they offer only streaming access, not random
		//
		if (loop->sfx->pMP3StreamHeader)
		{				
			Com_Error( ERR_DROP, "S_AddLoopSounds(): Cannot use streamed MP3 files here for random access (%s)\n",loop->sfx->sSoundName );
		}
		else
		{
			memset(&ch->MP3StreamHeader,0,						sizeof(ch->MP3StreamHeader));
		}
	}
}

//=============================================================================

/*
=================
S_ByteSwapRawSamples

If raw data has been loaded in little endien binary form, this must be done.
If raw data was calculated, as with ADPCM, this should not be called.
=================
*/
static void S_ByteSwapRawSamples( int samples, int width, int s_channels, const byte *data ) {
	int		i;

	if ( width != 2 ) {
		return;
	}
	if ( LittleShort( 256 ) == 256 ) {
		return;
	}

	if ( s_channels == 2 ) {
		samples <<= 1;
	}
	for ( i = 0 ; i < samples ; i++ ) {
		((short *)data)[i] = LittleShort( ((short *)data)[i] );
	}
}


portable_samplepair_t *S_GetRawSamplePointer() {
	return s_rawsamples;
}


/*
============
S_RawSamples

Music streaming
============
*/
void S_RawSamples( int samples, int rate, int width, int s_channels, const byte *data, float volume, qboolean bFirstOrOnlyUpdateThisFrame )
{
	int		i;
	int		src, dst;
	float	scale;
	int		intVolume;

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}

	intVolume = 256 * volume;

	if ( s_rawend < s_soundtime ) {
		Com_DPrintf( "S_RawSamples: resetting minimum: %i < %i\n", s_rawend, s_soundtime );
		s_rawend = s_soundtime;
	}

	scale = (float)rate / dma.speed;

//Com_Printf ("%i < %i < %i\n", s_soundtime, s_paintedtime, s_rawend);
	if (s_channels == 2 && width == 2)
	{
		if (scale == 1.0)
		{	// optimized case
			if (bFirstOrOnlyUpdateThisFrame)
			{
				for (i=0 ; i<samples ; i++)
				{
					dst = s_rawend&(MAX_RAW_SAMPLES-1);
					s_rawend++;
					s_rawsamples[dst].left = ((short *)data)[i*2] * intVolume;
					s_rawsamples[dst].right = ((short *)data)[i*2+1] * intVolume;
				}
			}
			else
			{
				for (i=0 ; i<samples ; i++)
				{
					dst = s_rawend&(MAX_RAW_SAMPLES-1);
					s_rawend++;
					s_rawsamples[dst].left  += ((short *)data)[i*2] * intVolume;
					s_rawsamples[dst].right += ((short *)data)[i*2+1] * intVolume;
				}
			}
		}
		else
		{
			if (bFirstOrOnlyUpdateThisFrame)
			{
				for (i=0 ; ; i++)
				{
					src = i*scale;
					if (src >= samples)
						break;
					dst = s_rawend&(MAX_RAW_SAMPLES-1);
					s_rawend++;
					s_rawsamples[dst].left = ((short *)data)[src*2] * intVolume;
					s_rawsamples[dst].right = ((short *)data)[src*2+1] * intVolume;
				}
			}
			else
			{
				for (i=0 ; ; i++)
				{
					src = i*scale;
					if (src >= samples)
						break;
					dst = s_rawend&(MAX_RAW_SAMPLES-1);
					s_rawend++;
					s_rawsamples[dst].left  += ((short *)data)[src*2] * intVolume;
					s_rawsamples[dst].right += ((short *)data)[src*2+1] * intVolume;
				}
			}
		}
	}
	else if (s_channels == 1 && width == 2)
	{
		if (bFirstOrOnlyUpdateThisFrame)
		{
			for (i=0 ; ; i++)
			{
				src = i*scale;
				if (src >= samples)
					break;
				dst = s_rawend&(MAX_RAW_SAMPLES-1);
				s_rawend++;
				s_rawsamples[dst].left = ((short *)data)[src] * intVolume;
				s_rawsamples[dst].right = ((short *)data)[src] * intVolume;
			}
		}
		else
		{
			for (i=0 ; ; i++)
			{
				src = i*scale;
				if (src >= samples)
					break;
				dst = s_rawend&(MAX_RAW_SAMPLES-1);
				s_rawend++;
				s_rawsamples[dst].left  += ((short *)data)[src] * intVolume;
				s_rawsamples[dst].right += ((short *)data)[src] * intVolume;
			}
		}
	}
	else if (s_channels == 2 && width == 1)
	{
		intVolume *= 256;

		if (bFirstOrOnlyUpdateThisFrame)
		{
			for (i=0 ; ; i++)
			{
				src = i*scale;
				if (src >= samples)
					break;
				dst = s_rawend&(MAX_RAW_SAMPLES-1);
				s_rawend++;
				s_rawsamples[dst].left = ((char *)data)[src*2] * intVolume;
				s_rawsamples[dst].right = ((char *)data)[src*2+1] * intVolume;
			}
		}
		else
		{
			for (i=0 ; ; i++)
			{
				src = i*scale;
				if (src >= samples)
					break;
				dst = s_rawend&(MAX_RAW_SAMPLES-1);
				s_rawend++;
				s_rawsamples[dst].left  += ((char *)data)[src*2] * intVolume;
				s_rawsamples[dst].right += ((char *)data)[src*2+1] * intVolume;
			}
		}
	}
	else if (s_channels == 1 && width == 1)
	{
		intVolume *= 256;

		if (bFirstOrOnlyUpdateThisFrame)
		{
			for (i=0 ; ; i++)
			{
				src = i*scale;
				if (src >= samples)
					break;
				dst = s_rawend&(MAX_RAW_SAMPLES-1);
				s_rawend++;
				s_rawsamples[dst].left = (((byte *)data)[src]-128) * intVolume;
				s_rawsamples[dst].right = (((byte *)data)[src]-128) * intVolume;
			}
		}
		else
		{
			for (i=0 ; ; i++)
			{
				src = i*scale;
				if (src >= samples)
					break;
				dst = s_rawend&(MAX_RAW_SAMPLES-1);
				s_rawend++;
				s_rawsamples[dst].left  += (((byte *)data)[src]-128) * intVolume;
				s_rawsamples[dst].right += (((byte *)data)[src]-128) * intVolume;
			}
		}
	}

	if ( s_rawend > s_soundtime + MAX_RAW_SAMPLES ) {
		Com_DPrintf( "S_RawSamples: overflowed %i > %i\n", s_rawend, s_soundtime );
	}
}

//=============================================================================

/*
=====================
S_UpdateEntityPosition

let the sound system know where an entity currently is
======================
*/
void S_UpdateEntityPosition( int entityNum, const vec3_t origin )
{
	ALfloat pos[3];
	channel_t *ch;
	int i;

	if ( entityNum < 0 || entityNum > MAX_GENTITIES ) {
		Com_Error( ERR_DROP, "S_UpdateEntityPosition: bad entitynum %i", entityNum );
	}

	if (s_UseOpenAL)
	{
		if (entityNum == 0)
			return;

		ch = s_channels + 1;
		for (i = 1; i < s_numChannels; i++, ch++)
		{
			if ((s_channels[i].bPlaying) & (s_channels[i].entnum == entityNum) & (!s_channels[i].bLooping))
			{
				// Ignore position updates for CHAN_VOICE_GLOBAL
				if (ch->entchannel != CHAN_VOICE_GLOBAL && ch->entchannel != CHAN_ANNOUNCER)
				{
					pos[0] = origin[0];
					pos[1] = origin[2];
					pos[2] = -origin[1];
					alSourcefv(s_channels[i].alSource, AL_POSITION, pos);

#ifdef HAVE_EAX
					UpdateEAXBuffer(ch);
#endif
				}

/*				pos[0] = origin[0];
				pos[1] = origin[2];
				pos[2] = -origin[1];
				alSourcefv(s_channels[i].alSource, AL_POSITION, pos);
	
#ifdef HAVE_EAX
				if ((s_bEALFileLoaded) && !( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL ) )
				{
					UpdateEAXBuffer(ch);
				}
#endif
*/
			}
		}
	}

	VectorCopy( origin, s_entityPosition[entityNum] );
}


// Given a current wav we are playing, and our position within it, lets figure out its volume...
//
// (this is mostly Jake's code from EF1, which explains a lot...:-)
//
static int next_amplitude = 0;
static int S_CheckAmplitude(channel_t	*ch, const int s_oldpaintedtime )
{
	// now, is this a cycle - or have we just started a new sample - where we should update the backup table, and write this value
	// into the new table? or should we just take the value FROM the back up table and feed it out.
	assert( ch->startSample != START_SAMPLE_IMMEDIATE );
	if ( ch->startSample == s_oldpaintedtime || (next_amplitude < s_soundtime) )//(ch->startSample == START_SAMPLE_IMMEDIATE)//!s_entityWavVol_back[ch->entnum] 
	{
		int	sample;
		int	sample_total = 0;
		int	count = 0;
		short *current_pos_s;
//		char *current_pos_c;
		int	offset = 0;

		// if we haven't started the sample yet, we must be at the beginning
		current_pos_s = ((short*)ch->thesfx->pSoundData);
//		current_pos_c = ((char*)ch->thesfx->data);

		//if (ch->startSample != START_SAMPLE_IMMEDIATE)
		//{
			// figure out where we are in the sample right now.
			offset = s_oldpaintedtime - ch->startSample;//s_paintedtime
			current_pos_s += offset;
//			current_pos_c += offset;
		//}
		
		// scan through 10 samples 100( at 11hz or 200 at 22hz) samples apart.
		//
		for (int i=0; i<10; i++)
		{				
			//
			// have we run off the end?
			if ((offset + (i*100)) > ch->thesfx->iSoundLengthInSamples)
			{
				break;
			}
//			if (ch->thesfx->width == 1)
//			{
//				sample = current_pos_c[i*100];
//			}
//			else
			{					
				switch (ch->thesfx->eSoundCompressionMethod)
				{					
					case ct_16:
					{
						sample = LittleShort(current_pos_s[i*100]);
					}
					break;

					case ct_MP3:
					{
						const int iIndex = (i*100) + ((offset * /*ch->thesfx->width*/2) - ch->iMP3SlidingDecodeWindowPos);
						const short* pwSamples = (short*) (ch->MP3SlidingDecodeBuffer + iIndex);

						sample = LittleShort(*pwSamples);						
					}
					break;

					default:			
					{
						assert(0);
						sample = 0;
					}
					break;
				}
				
//				if (sample < 0)
//					sample = -sample;
				sample = sample>>8;
			}
			// square it for better accuracy
			sample_total += (sample * sample);
			count++;
		}
		
		// if we are already done with this sample, then its silence
		if (!count)
		{
			return(0);
		}
		sample_total /= count;

		// I hate doing this, but its the simplest way
		if (sample_total < ch->thesfx->fVolRange * s_lip_threshold_1->value)
		{
		// tell the scripts that are relying on this that we are still going, but actually silent right now.
			sample = -1;
		}
		else
		if (sample_total < ch->thesfx->fVolRange * s_lip_threshold_2->value)
		{
			sample = 1;
		}
		else
		if (sample_total < ch->thesfx->fVolRange * s_lip_threshold_3->value)
		{
			sample = 2;
		}
		else
		if (sample_total < ch->thesfx->fVolRange * s_lip_threshold_4->value)
		{
			sample = 3;
		}
		else
		{
			sample = 4;
		}

//		OutputDebugString(va("Returning sample %d\n",sample));

		// store away the value we got into the back up table
		s_entityWavVol_back[ ch->entnum ] = sample;
		return (sample);
	}
	// no, just get last value calculated from backup table
	assert( s_entityWavVol_back[ch->entnum] );
	return (s_entityWavVol_back[ ch->entnum]);
}
/*
============
S_Respatialize

Change the volumes of all the playing sounds for changes in their positions
============
*/
void S_Respatialize( int entityNum, const vec3_t head, vec3_t axis[3], qboolean inwater )
{
#ifdef HAVE_EAX
	EAXOCCLUSIONPROPERTIES eaxOCProp;
	EAXACTIVEFXSLOTS eaxActiveSlots;
#endif
	unsigned int ulEnvironment;
	int			i;
	channel_t	*ch;

	if ( !s_soundStarted || s_soundMuted ) {
		return; 
	}

	if (s_UseOpenAL)
	{
		listener_pos[0] = head[0];
		listener_pos[1] = head[2];
		listener_pos[2] = -head[1];
		alListenerfv(AL_POSITION, listener_pos);

		listener_ori[0] = axis[0][0];
		listener_ori[1] = axis[0][2];
		listener_ori[2] = -axis[0][1];
		listener_ori[3] = axis[2][0];
		listener_ori[4] = axis[2][2];
		listener_ori[5] = -axis[2][1];
		alListenerfv(AL_ORIENTATION, listener_ori);
#ifdef HAVE_EAX
		// Update EAX effects here
		if (s_bEALFileLoaded)
		{
			// Check if the Listener is underwater
			if (inwater)
			{
				// Check if we have already applied Underwater effect
				if (!s_bInWater)
				{
					// New underwater fix
					for (i = 0; i < EAX_MAX_FXSLOTS; i++)
					{
						s_FXSlotInfo[i].lEnvID = -1;
					}

					// Load underwater reverb effect into FX Slot 0, and set this as the Primary FX Slot
					ulEnvironment = EAX_ENVIRONMENT_UNDERWATER;
					s_eaxSet(&EAXPROPERTYID_EAX40_FXSlot0, EAXREVERB_ENVIRONMENT,
						NULL, &ulEnvironment, sizeof(unsigned int));
					s_EnvironmentID = 999;

					s_eaxSet(&EAXPROPERTYID_EAX40_Context, EAXCONTEXT_PRIMARYFXSLOTID, NULL, (ALvoid*)&EAXPROPERTYID_EAX40_FXSlot0,
						sizeof(GUID));

					// Occlude all sounds into this environment, and mute all their sends to other reverbs
					eaxOCProp.lOcclusion = -3000;
					eaxOCProp.flOcclusionLFRatio = 0.0f;
					eaxOCProp.flOcclusionRoomRatio = 1.37f;
					eaxOCProp.flOcclusionDirectRatio = 1.0f;

					eaxActiveSlots.guidActiveFXSlots[0] = EAX_NULL_GUID;
					eaxActiveSlots.guidActiveFXSlots[1] = EAX_PrimaryFXSlotID;

					ch = s_channels + 1;
					for (i = 1; i < s_numChannels; i++, ch++)
					{
						// New underwater fix
						s_channels[i].lSlotID = -1;

						s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_OCCLUSIONPARAMETERS,
							ch->alSource, &eaxOCProp, sizeof(EAXOCCLUSIONPROPERTIES));

						s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_ACTIVEFXSLOTID, ch->alSource,
							&eaxActiveSlots, 2*sizeof(GUID));
					}

					s_bInWater = true;
				}
			}
			else
			{
				// Not underwater ... check if the underwater effect is still present
				if (s_bInWater)
				{
					s_bInWater = false;

					// Remove underwater Reverb effect, and reset Occlusion / Obstruction amount on all Sources
					UpdateEAXListener();

					ch = s_channels + 1;
					for (i = 1; i < s_numChannels; i++, ch++)
					{
						UpdateEAXBuffer(ch);
					}
				}
				else
				{
					UpdateEAXListener();
				}
			}
		}
#endif // HAVE_EAX
	}
	else
	{
		listener_number = entityNum;
		VectorCopy(head, listener_origin);
		VectorCopy(axis[0], listener_axis[0]);
		VectorCopy(axis[1], listener_axis[1]);
		VectorCopy(axis[2], listener_axis[2]);

		// update spatialization for dynamic sounds	
		ch = s_channels;
		for ( i = 0 ; i < MAX_CHANNELS ; i++, ch++ ) {
			if ( !ch->thesfx ) {
				continue;
			}
			if ( ch->loopSound ) {	// loopSounds are regenerated fresh each frame
				Channel_Clear(ch);	// memset (ch, 0, sizeof(*ch));
				continue;
			}

			// anything coming from the view entity will always be full volume
			if (ch->entnum == listener_number || ch->entchannel == CHAN_VOICE_GLOBAL || ch->entchannel == CHAN_ANNOUNCER) {
				ch->leftvol = ch->master_vol;
				ch->rightvol = ch->master_vol;
			} else {
				const vec3_t	*origin;
				if (ch->fixed_origin) {
					origin = &ch->origin;
				} else {
					origin = &s_entityPosition[ ch->entnum ];
				}

				S_SpatializeOrigin (*origin, (float)ch->master_vol, &ch->leftvol, &ch->rightvol, ch->entchannel);
			}

			//NOTE: Made it so that voice sounds keep playing, even out of range
			//		so that tasks waiting for sound completion keep proper timing
			if ( !( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL ) && !ch->leftvol && !ch->rightvol ) {
				Channel_Clear(ch);	// memset (ch, 0, sizeof(*ch));
				continue;
			}
		}

		// add loopsounds
		S_AddLoopSounds ();
	}

	return;
}


/*
========================
S_ScanChannelStarts

Returns qtrue if any new sounds were started since the last mix
========================
*/
qboolean S_ScanChannelStarts( void ) {
	channel_t		*ch;
	int				i;
	qboolean		newSamples;

	newSamples = qfalse;
	ch = s_channels;
	for (i=0; i<MAX_CHANNELS ; i++, ch++) {
		if ( !ch->thesfx ) {
			continue;
		}
		if ( ch->loopSound ) {
			continue;
		}

		// if this channel was just started this frame,
		// set the sample count to it begins mixing
		// into the very first sample
		if ( ch->startSample == START_SAMPLE_IMMEDIATE ) {
			ch->startSample = s_paintedtime;
			newSamples = qtrue;
			continue;
		}

		// if it is completely finished by now, clear it
		if ( ch->startSample + ch->thesfx->iSoundLengthInSamples <= s_paintedtime ) {
			ch->thesfx = NULL;
			continue;
		}
	}

	return newSamples;
}

// this is now called AFTER the DMA painting, since it's only the painter calls that cause the MP3s to be unpacked,
//	and therefore to have data readable by the lip-sync volume calc code.
//
void S_DoLipSynchs( const int s_oldpaintedtime )
{
	channel_t		*ch;
	int				i;
	qboolean		newSamples;

	// clear out the lip synching override array for this frame
	memset(s_entityWavVol, 0,(MAX_GENTITIES * 4));

	newSamples = qfalse;
	ch = s_channels;
	for (i=0; i<MAX_CHANNELS ; i++, ch++) {
		if ( !ch->thesfx ) {
			continue;
		}
		if ( ch->loopSound ) {
			continue;
		}

		// if we are playing a sample that should override the lip texture on its owning model, lets figure out
		// what the amplitude is, stick it in a table, then return it 
		if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
		{
			// go away and work out amplitude for this sound we are playing right now.
			s_entityWavVol[ ch->entnum ] = S_CheckAmplitude( ch, s_oldpaintedtime );
			if ( s_show->integer == 3 ) {
				Com_Printf( "(%i)%i %s vol = %i\n", ch->entnum, i, ch->thesfx->sSoundName, s_entityWavVol[ ch->entnum ] );
			}
		}
	}

	if (next_amplitude < s_soundtime)	{
		next_amplitude = s_soundtime + 800;
	}
}

/*
============
S_Update

Called once each time through the main loop
============
*/
void S_Update( void ) {
	int			i;
	channel_t	*ch;

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}

	//
	// debugging output
	//
	if ( s_show->integer == 2 ) {
		//int total = 0;
		//int totalMeg =0;
		ch = s_channels;
		for (i=0 ; i<MAX_CHANNELS; i++, ch++) {
			if ( ch->thesfx && (ch->leftvol || ch->rightvol) ) {				
				Com_Printf ("(%i) %3i %3i %s\n", ch->entnum, ch->leftvol, ch->rightvol, ch->thesfx->sSoundName);
				//total++;
				//totalMeg += Z_Size(ch->thesfx->pSoundData);
				//if (ch->thesfx->pMP3StreamHeader)
				//{
				//	totalMeg += sizeof(*ch->thesfx->pMP3StreamHeader);
				//}
			}
		}
		
		//if (total)
		//	Com_Printf ("----(%i)---- painted: %i, SND %.2fMB\n", total, s_paintedtime, totalMeg/1024.0f/1024.0f);
	}

	// The Open AL code, handles background music in the S_UpdateRawSamples function
	if (!s_UseOpenAL)
	{
		// add raw data from streamed samples
		S_UpdateBackgroundTrack();
	}

	// mix some sound
	S_Update_();
}

void S_GetSoundtime(void)
{
	int		samplepos;
	static	int		buffers;
	static	int		oldsamplepos;
	int		fullsamples;
	
	fullsamples = dma.samples / dma.channels;

	// it is possible to miscount buffers if it has wrapped twice between
	// calls to S_Update.  Oh well.
	samplepos = SNDDMA_GetDMAPos();
	if (samplepos < oldsamplepos)
	{
		buffers++;					// buffer wrapped
		
		if (s_paintedtime > 0x40000000)
		{	// time to chop things off to avoid 32 bit limits
			buffers = 0;
			s_paintedtime = fullsamples;
			S_StopAllSounds ();
		}
	}
	oldsamplepos = samplepos;

	s_soundtime = buffers*fullsamples + samplepos/dma.channels;

#if 0
// check to make sure that we haven't overshot
	if (s_paintedtime < s_soundtime)
	{
		Com_DPrintf ("S_Update_ : overflow\n");
		s_paintedtime = s_soundtime;
	}
#endif

	if ( dma.submission_chunk < 256 ) {
		s_paintedtime = (int)(s_soundtime + s_mixPreStep->value * dma.speed);
	} else {
		s_paintedtime = s_soundtime + dma.submission_chunk;
	}
}


void S_Update_(void) {
	unsigned        endtime;
	int				samps;
	channel_t		*ch;
	int i,j;
	int			source;
	float		pos[3];
#ifdef _DEBUG
	char szString[256];
#endif

	if ( !s_soundStarted || s_soundMuted ) {
		return;
	}

	if (s_UseOpenAL)
	{
		UpdateSingleShotSounds();

		ch = s_channels + 1;
		for ( i = 1; i < MAX_CHANNELS ; i++, ch++ )
		{	
			if ( !ch->thesfx || (ch->bPlaying))
				continue;

			source = ch - s_channels;

			if (ch->entchannel == CHAN_VOICE_GLOBAL || ch->entchannel == CHAN_ANNOUNCER)
			{
				// Always play these sounds at 0,0,-1 (in front of listener)
				pos[0] = 0.0f;
				pos[1] = 0.0f;
				pos[2] = -1.0f;

				alSourcefv(s_channels[source].alSource, AL_POSITION, pos);
				alSourcei(s_channels[source].alSource, AL_LOOPING, AL_FALSE);
				alSourcei(s_channels[source].alSource, AL_SOURCE_RELATIVE, AL_TRUE);
				if (ch->entchannel == CHAN_ANNOUNCER)
				{
					alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.0f);
				}
				else
				{
					alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volumeVoice->value) / 255.0f);
				}
			}
			else
			{
				// Get position of source
				if (ch->fixed_origin)
				{
					pos[0] = ch->origin[0];
					pos[1] = ch->origin[2];
					pos[2] = -ch->origin[1];
					alSourcei(s_channels[source].alSource, AL_SOURCE_RELATIVE, AL_FALSE);
				}
				else
				{
					if (ch->entnum == listener_number)
					{
						pos[0] = 0.0f;
						pos[1] = 0.0f;
						pos[2] = 0.0f;
						alSourcei(s_channels[source].alSource, AL_SOURCE_RELATIVE, AL_TRUE);
					}
					else
					{
						// Get position of Entity
						if (ch->bLooping)
						{
							pos[0] = loopSounds[ ch->entnum ].origin[0];
							pos[1] = loopSounds[ ch->entnum ].origin[2];
							pos[2] = -loopSounds[ ch->entnum ].origin[1];
						}
						else
						{
							pos[0] = s_entityPosition[ch->entnum][0];
							pos[1] = s_entityPosition[ch->entnum][2];
							pos[2] = -s_entityPosition[ch->entnum][1];
						}
						alSourcei(s_channels[source].alSource, AL_SOURCE_RELATIVE, AL_FALSE);
					}
				}
		
				alSourcefv(s_channels[source].alSource, AL_POSITION, pos);
				alSourcei(s_channels[source].alSource, AL_LOOPING, AL_FALSE);

				if (ch->entchannel == CHAN_VOICE)
				{
					// Reduced fall-off (Large Reference Distance), affected by Voice Volume
					alSourcef(s_channels[source].alSource, AL_REFERENCE_DISTANCE, DEFAULT_VOICE_REF_DISTANCE);
					alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volumeVoice->value) / 255.0f);
				}
				else if (ch->entchannel == CHAN_VOICE_ATTEN)
				{
					// Normal fall-off, affected by Voice Volume
					alSourcef(s_channels[source].alSource, AL_REFERENCE_DISTANCE, DEFAULT_REF_DISTANCE);
					alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volumeVoice->value) / 255.0f);
				}
				else if (ch->entchannel == CHAN_LESS_ATTEN)
				{
					// Reduced fall-off, affected by Sound Effect Volume
					alSourcef(s_channels[source].alSource, AL_REFERENCE_DISTANCE, DEFAULT_VOICE_REF_DISTANCE);
					alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.f);
				}
				else
				{
					// Normal fall-off, affect by Sound Effect Volume
					alSourcef(s_channels[source].alSource, AL_REFERENCE_DISTANCE, DEFAULT_REF_DISTANCE);
					alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.f);
				}
			}

#ifdef HAVE_EAX
			if (s_bEALFileLoaded)
				UpdateEAXBuffer(ch);
#endif

			int nBytesDecoded = 0;
			int nTotalBytesDecoded = 0;
			int nBuffersToAdd = 0;

			if (ch->thesfx->pMP3StreamHeader)
			{
				memcpy(&ch->MP3StreamHeader, ch->thesfx->pMP3StreamHeader,	sizeof(ch->MP3StreamHeader));
				ch->iMP3SlidingDecodeWritePos = 0;
				ch->iMP3SlidingDecodeWindowPos= 0;

				// Reset streaming buffers status's
				for (i = 0; i < NUM_STREAMING_BUFFERS; i++)
					ch->buffers[i].status = UNQUEUED;

				// Decode (STREAMING_BUFFER_SIZE / 1152) MP3 frames for each of the NUM_STREAMING_BUFFERS AL Buffers
				for (i = 0; i < NUM_STREAMING_BUFFERS; i++)
				{
					nTotalBytesDecoded = 0;
					
					for (j = 0; j < (STREAMING_BUFFER_SIZE / 1152); j++)
					{
						nBytesDecoded = C_MP3Stream_Decode(&ch->MP3StreamHeader, 0);	// added ,0 ?
						memcpy(ch->buffers[i].Data + nTotalBytesDecoded, ch->MP3StreamHeader.bDecodeBuffer, nBytesDecoded);
						if (ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
						{
							if (ch->thesfx->lipSyncData)
							{
								ch->thesfx->lipSyncData[(i*NUM_STREAMING_BUFFERS)+j] = S_MP3PreProcessLipSync(ch, (short *)(ch->MP3StreamHeader.bDecodeBuffer));
							}
							else
							{
#ifdef _DEBUG
								sprintf(szString, "Missing lip-sync info. for %s\n", ch->thesfx->sSoundName);
								OutputDebugString(szString);
#endif
							}
						}
						nTotalBytesDecoded += nBytesDecoded;
					}

					if (nTotalBytesDecoded != STREAMING_BUFFER_SIZE)
					{
						memset(ch->buffers[i].Data + nTotalBytesDecoded, 0, (STREAMING_BUFFER_SIZE - nTotalBytesDecoded));
						break;
					}
				}

				if (i >= NUM_STREAMING_BUFFERS)
					nBuffersToAdd = NUM_STREAMING_BUFFERS;
				else
					nBuffersToAdd = i + 1;

				// Make sure queue is empty first
				alSourcei(s_channels[source].alSource, AL_BUFFER, NULL);

				for (i = 0; i < nBuffersToAdd; i++)
				{
					// Copy decoded data to AL Buffer
					alBufferData(ch->buffers[i].BufferID, AL_FORMAT_MONO16, ch->buffers[i].Data, STREAMING_BUFFER_SIZE, 22050);
		
					// Queue AL Buffer on Source
					alSourceQueueBuffers(s_channels[source].alSource, 1, &(ch->buffers[i].BufferID));
					if (alGetError() == AL_NO_ERROR)
					{
						ch->buffers[i].status = QUEUED;
					}
				}

				// Clear error state, and check for successful Play call
				alGetError();
				alSourcePlay(s_channels[source].alSource);
				if (alGetError() == AL_NO_ERROR)
					s_channels[source].bPlaying = true;

				ch->bStreaming = true;

				if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
				{
					if (ch->thesfx->lipSyncData)
					{
						// Record start time for Lip-syncing
						s_channels[source].iStartTime = Com_Milliseconds();

						// Prepare lipsync value(s)
						s_entityWavVol[ ch->entnum ] = ch->thesfx->lipSyncData[0];
					}
					else
					{
#ifdef _DEBUG
						sprintf(szString, "Missing lip-sync info. for %s\n", ch->thesfx->sSoundName);
						OutputDebugString(szString);
#endif
					}
				}

				return;
			}
			else
			{
				// Attach buffer to source
				alSourcei(s_channels[source].alSource, AL_BUFFER, ch->thesfx->Buffer);

				ch->bStreaming = false;

				// Clear error state, and check for successful Play call
				alGetError();
				alSourcePlay(s_channels[source].alSource);
				if (alGetError() == AL_NO_ERROR)
					s_channels[source].bPlaying = true;

				if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
				{
					if (ch->thesfx->lipSyncData)
					{
						// Record start time for Lip-syncing
						s_channels[source].iStartTime = Com_Milliseconds();

						// Prepare lipsync value(s)
						s_entityWavVol[ ch->entnum ] = ch->thesfx->lipSyncData[0];
					}
					else
					{
#ifdef _DEBUG
						sprintf(szString, "Missing lip-sync info. for %s\n", ch->thesfx->sSoundName);
						OutputDebugString(szString);
#endif
					}
				}
			}
		}

		S_SetLipSyncs();

		UpdateLoopingSounds();

		AL_UpdateRawSamples();
	}
	else
	{
		// Updates s_soundtime
		S_GetSoundtime();

		const int s_oldpaintedtime = s_paintedtime;

		// clear any sound effects that end before the current time,
		// and start any new sounds
		S_ScanChannelStarts();

		// mix ahead of current position
		endtime = (int)(s_soundtime + s_mixahead->value * dma.speed);

		// mix to an even submission block size
		endtime = (endtime + dma.submission_chunk-1)
			& ~(dma.submission_chunk-1);

		// never mix more than the complete buffer
		samps = dma.samples >> (dma.channels-1);
		if (endtime - s_soundtime > samps)
			endtime = s_soundtime + samps;


		SNDDMA_BeginPainting ();

		S_PaintChannels (endtime);

		SNDDMA_Submit ();

		S_DoLipSynchs( s_oldpaintedtime );
	}
}


void UpdateSingleShotSounds()
{
	int i, j, k;
	ALint state;
	ALint processed;
	channel_t *ch;
#ifdef _DEBUG
	char szString[256];
#endif

	// Firstly, check if any single-shot sounds have completed, or if they need more data (for streaming Sources),
	// and/or if any of the currently playing (non-Ambient) looping sounds need to be stopped
	ch = s_channels + 1;
	for (i = 1; i < s_numChannels; i++, ch++)
	{
		ch->bProcessed = false;
		if ((s_channels[i].bPlaying) && (!ch->bLooping))
		{	
			// Single-shot
			if (s_channels[i].bStreaming == false)
			{
				alGetSourcei(s_channels[i].alSource, AL_SOURCE_STATE, &state);
				if (state == AL_STOPPED)
				{
					s_channels[i].thesfx = NULL;
					s_channels[i].bPlaying = false;
				}
			}
			else
			{	
				// Process streaming sample

				// Procedure :-
				// if more data to play
				//		if any UNQUEUED Buffers
				//			fill them with data
				//		(else ?)
				//			get number of buffers processed
				//			fill them with data
				//		restart playback if it has stopped (buffer underrun)
				// else
				//		free channel
				
				int nBytesDecoded;

				if (ch->thesfx->pMP3StreamHeader)
				{
					if (ch->MP3StreamHeader.iSourceBytesRemaining == 0)
					{
						// Finished decoding data - if the source has finished playing then we're done
						alGetSourcei(ch->alSource, AL_SOURCE_STATE, &state);
						if (state == AL_STOPPED)
						{
							// Attach NULL buffer to Source to remove any buffers left in the queue
							alSourcei(ch->alSource, AL_BUFFER, NULL);
							ch->thesfx = NULL;
							ch->bPlaying = false;
						}
						// Move on to next channel ...
						continue;
					}

					// Check to see if any Buffers have been processed
					alGetSourcei(ch->alSource, AL_BUFFERS_PROCESSED, &processed);

					ALuint buffer;
					while (processed)
					{
						alSourceUnqueueBuffers(ch->alSource, 1, &buffer);			
						for (j = 0; j < NUM_STREAMING_BUFFERS; j++)
						{
							if (ch->buffers[j].BufferID == buffer)
							{
								ch->buffers[j].status = UNQUEUED;
								break;
							}
						}
						processed--;
					}

					int nTotalBytesDecoded = 0;

					for (j = 0; j < NUM_STREAMING_BUFFERS; j++)
					{
						if ((ch->buffers[j].status == UNQUEUED) & (ch->MP3StreamHeader.iSourceBytesRemaining > 0))
						{
							nTotalBytesDecoded = 0;
		
							for (k = 0; k < (STREAMING_BUFFER_SIZE / 1152); k++)
							{
								nBytesDecoded = C_MP3Stream_Decode(&ch->MP3StreamHeader, 0); // added ,0

								if (nBytesDecoded > 0)
								{
									memcpy(ch->buffers[j].Data + nTotalBytesDecoded, ch->MP3StreamHeader.bDecodeBuffer, nBytesDecoded);

									if (ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
									{
										if (ch->thesfx->lipSyncData)
										{
											ch->thesfx->lipSyncData[(j*4)+k] = S_MP3PreProcessLipSync(ch, (short *)(ch->buffers[j].Data));
										}
										else
										{
#ifdef _DEBUG
											sprintf(szString, "Missing lip-sync info. for %s\n", ch->thesfx->sSoundName);
											OutputDebugString(szString);
#endif
										}
									}
									nTotalBytesDecoded += nBytesDecoded;
								}
								else
								{
									// Make sure that iSourceBytesRemaining is 0
									if (ch->MP3StreamHeader.iSourceBytesRemaining != 0)
									{
										ch->MP3StreamHeader.iSourceBytesRemaining = 0;
										break;
									}
								}
							}

							if (nTotalBytesDecoded != STREAMING_BUFFER_SIZE)
							{
								memset(ch->buffers[j].Data + nTotalBytesDecoded, 0, (STREAMING_BUFFER_SIZE - nTotalBytesDecoded));

								// Move data to buffer
								alBufferData(ch->buffers[j].BufferID, AL_FORMAT_MONO16, ch->buffers[j].Data, STREAMING_BUFFER_SIZE, 22050);

								// Queue Buffer on Source
								alSourceQueueBuffers(ch->alSource, 1, &(ch->buffers[j].BufferID));

								// Update status of Buffer
								ch->buffers[j].status = QUEUED;

								break;
							}
							else
							{
								// Move data to buffer
								alBufferData(ch->buffers[j].BufferID, AL_FORMAT_MONO16, ch->buffers[j].Data, STREAMING_BUFFER_SIZE, 22050);

								// Queue Buffer on Source
								alSourceQueueBuffers(ch->alSource, 1, &(ch->buffers[j].BufferID));
								
								// Update status of Buffer
								ch->buffers[j].status = QUEUED;
							}
						}
					}

					// Get state of Buffer
					alGetSourcei(ch->alSource, AL_SOURCE_STATE, &state);
					if (state != AL_PLAYING)
					{
						alSourcePlay(ch->alSource);
#ifdef _DEBUG
						char szString[256];
						sprintf(szString,"[%d] Restarting playback of single-shot streaming MP3 sample - still have %d bytes to decode\n", i, ch->MP3StreamHeader.iSourceBytesRemaining);
						OutputDebugString(szString);
#endif
					}
				}
			}
		}
	}
}




void UpdateLoopingSounds()
{
	int i,j;
	ALuint source;
	channel_t *ch;
	loopSound_t	*loop;
	float pos[3];

	// First check to see if any of the looping sounds are already playing at the correct positions
	ch = s_channels + 1;
	for (i = 1; i < s_numChannels; i++, ch++)
	{
		if (ch->bLooping && s_channels[i].bPlaying)
		{
			for (j = 0; j < numLoopSounds; j++)
			{
				loop = &loopSounds[j];

				// If this channel is playing the right sound effect at the right position then mark this channel and looping sound
				// as processed
				if ((loop->bProcessed == false) && (ch->thesfx == loop->sfx) )
				{
					if ( (loop->origin[0] == listener_pos[0]) && (loop->origin[1] == -listener_pos[2])
						&& (loop->origin[2] == listener_pos[1]) )
					{
						// Assume that this sound is head relative
						if (!loop->bRelative)
						{
							// Set position to 0,0,0 and turn on Head Relative Mode
							float pos[3];
							pos[0] = 0.f;
							pos[1] = 0.f;
							pos[2] = 0.f;

							alSourcefv(s_channels[i].alSource, AL_POSITION, pos);
							alSourcei(s_channels[i].alSource, AL_SOURCE_RELATIVE, AL_TRUE);
							loop->bRelative = true;
						}

						// Make sure Gain is set correctly
						if (ch->master_vol != loop->volume)
						{
							ch->master_vol = loop->volume;
							alSourcef(s_channels[i].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.f);
						}

						ch->bProcessed = true;
						loop->bProcessed = true;
					}	
					else if ((loop->bProcessed == false) && (ch->thesfx == loop->sfx) && (!memcmp(ch->origin, loop->origin, sizeof(ch->origin))))
					{
						// Match !
						ch->bProcessed = true;
						loop->bProcessed = true;

						// Make sure Gain is set correctly
						if (ch->master_vol != loop->volume)
						{
							ch->master_vol = loop->volume;
							alSourcef(s_channels[i].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.f);
						}

						break;
					}
				}
			}
		}
	}

	// Next check if the correct looping sound is playing, but at the wrong position
	ch = s_channels + 1;
	for (i = 1; i < s_numChannels; i++, ch++)
	{
		if ((ch->bLooping) && (ch->bProcessed == false) && s_channels[i].bPlaying)
		{
			for (j = 0; j < numLoopSounds; j++)
			{
				loop = &loopSounds[j];

				if ((loop->bProcessed == false) && (ch->thesfx == loop->sfx))
				{
					// Same sound - wrong position
					ch->origin[0] = loop->origin[0];
					ch->origin[1] = loop->origin[1];
					ch->origin[2] = loop->origin[2];

					pos[0] = loop->origin[0];
					pos[1] = loop->origin[2];
					pos[2] = -loop->origin[1];
					alSourcefv(s_channels[i].alSource, AL_POSITION, pos);

					ch->master_vol = loop->volume;
					alSourcef(s_channels[i].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.f);

#ifdef HAVE_EAX
					if (s_bEALFileLoaded)
						UpdateEAXBuffer(ch);
#endif

					ch->bProcessed = true;
					loop->bProcessed = true;
					break;
				}
			}
		}
	}

	// If any non-procesed looping sounds are still playing on a channel, they can be removed as they are no longer
	// required
	ch = s_channels + 1;
	for (i = 1; i < s_numChannels; i++, ch++)
	{
		if (s_channels[i].bPlaying && ch->bLooping && !ch->bProcessed)
		{
			// Sound no longer needed
			alSourceStop(s_channels[i].alSource);
			ch->thesfx = NULL;
			ch->bPlaying = false;
		}
	}

#ifdef _DEBUG
	alGetError();
#endif
	// Finally if there are any non-processed sounds left, we need to try and play them
	for (j = 0; j < numLoopSounds; j++)
	{
		loop = &loopSounds[j];

		if (loop->bProcessed == false)
		{
			ch = S_PickChannel(0,0);

			ch->master_vol = loop->volume;
			ch->entnum = loop->entnum;
			ch->entchannel = loop->entchan;
			ch->thesfx = loop->sfx;
			ch->bLooping = true;

			// Check if the Source is positioned at exactly the same location as the listener
			if ( (loop->origin[0] == listener_pos[0]) && (loop->origin[1] == -listener_pos[2])
						&& (loop->origin[2] == listener_pos[1]) )
			{
				// Assume that this sound is head relative
				loop->bRelative = true;
				ch->origin[0] = 0.f;
				ch->origin[1] = 0.f;
				ch->origin[2] = 0.f;
			}
			else
			{
				ch->origin[0] = loop->origin[0];
				ch->origin[1] = loop->origin[1];
				ch->origin[2] = loop->origin[2];
				loop->bRelative = false;
			}
			
			ch->fixed_origin = loop->bRelative;
			pos[0] = ch->origin[0];
			pos[1] = ch->origin[2];
			pos[2] = -ch->origin[1];

			source = ch - s_channels;
			alSourcei(s_channels[source].alSource, AL_BUFFER, ch->thesfx->Buffer);
			alSourcefv(s_channels[source].alSource, AL_POSITION, pos);
			alSourcei(s_channels[source].alSource, AL_LOOPING, AL_TRUE);
			alSourcef(s_channels[source].alSource, AL_GAIN, ((float)(ch->master_vol) * s_volume->value) / 255.0f);
			alSourcei(s_channels[source].alSource, AL_SOURCE_RELATIVE, ch->fixed_origin ? AL_TRUE : AL_FALSE);
			if (ch->entchannel == CHAN_LESS_ATTEN)
			{	// Reduced fall-off
				alSourcef(s_channels[source].alSource, AL_REFERENCE_DISTANCE, DEFAULT_VOICE_REF_DISTANCE);
			}
			else
			{
				alSourcef(s_channels[source].alSource, AL_REFERENCE_DISTANCE, DEFAULT_REF_DISTANCE);
			}
#ifdef HAVE_EAX
			if (s_bEALFileLoaded)
				UpdateEAXBuffer(ch);
#endif

			alGetError();
			alSourcePlay(s_channels[source].alSource);
			if (alGetError() == AL_NO_ERROR)
				s_channels[source].bPlaying = true;
		}
	}
}





void AL_UpdateRawSamples()
{
	ALuint buffer;
	ALint size;
	ALint processed;
	ALint state;
	int i,j,src;

#ifdef _DEBUG
	// Clear Open AL Error
	alGetError();
#endif

	S_UpdateBackgroundTrack();

	// Find out how many buffers have been processed (played) by the Source
	alGetSourcei(s_channels[0].alSource, AL_BUFFERS_PROCESSED, &processed);

	while (processed)
	{
		// Unqueue each buffer, determine the length of the buffer, and then delete it
		alSourceUnqueueBuffers(s_channels[0].alSource, 1, &buffer);
		alGetBufferi(buffer, AL_SIZE, &size);
		alDeleteBuffers(1, &buffer);
		
		// Update sg.soundtime (+= number of samples played (number of bytes / 4))
		s_soundtime += (size >> 2);

		processed--;
	}

	// Add new data to a new Buffer and queue it on the Source
	if (s_rawend > s_paintedtime)
	{
		size = (s_rawend - s_paintedtime)<<2;
		if (size > (MAX_RAW_SAMPLES<<2))
		{
#ifdef _DEBUG
			OutputDebugString("UpdateRawSamples :- Raw Sample buffer has overflowed !!!\n");
#endif
			size = MAX_RAW_SAMPLES<<2;
			s_paintedtime = s_rawend - MAX_RAW_SAMPLES;
		}

		// Copy samples from RawSamples to audio buffer (sg.rawdata)
		for (i = s_paintedtime, j = 0; i < s_rawend; i++, j+=2)
		{
			src = i & (MAX_RAW_SAMPLES - 1);
			s_rawdata[j] = (short)(s_rawsamples[src].left>>8);
			s_rawdata[j+1] = (short)(s_rawsamples[src].right>>8);
		}

		// Need to generate more than 1 buffer for music playback
		// iterations = 0;
		// largestBufferSize = (MAX_RAW_SAMPLES / 4) * 4
		// while (size)
		//	generate a buffer
		//	if size > largestBufferSize
		//		copy sg.rawdata + ((iterations * largestBufferSize)>>1) to buffer
		//		size -= largestBufferSize
		//	else
		//		copy remainder
		//		size = 0
		//	queue the buffer
		//  iterations++;

		int iterations = 0;
		int largestBufferSize = MAX_RAW_SAMPLES;	// in bytes (== quarter of Raw Samples data)
		while (size)
		{
			alGenBuffers(1, &buffer);

			if (size > largestBufferSize)
			{
				alBufferData(buffer, AL_FORMAT_STEREO16, (char*)(s_rawdata + ((iterations * largestBufferSize)>>1)), largestBufferSize, 22050);
				size -= largestBufferSize;
			}
			else
			{
				alBufferData(buffer, AL_FORMAT_STEREO16, (char*)(s_rawdata + ((iterations * largestBufferSize)>>1)), size, 22050);
				size = 0;
			}

			alSourceQueueBuffers(s_channels[0].alSource, 1, &buffer);
			iterations++;
		}

		// Update paintedtime
		s_paintedtime = s_rawend;
		
		// Check that the Source is actually playing
		alGetSourcei(s_channels[0].alSource, AL_SOURCE_STATE, &state);
		if (state != AL_PLAYING)
		{
			// Stopped playing ... due to buffer underrun
			// Unqueue any buffers still on the Source (they will be PROCESSED), and restart playback
			alGetSourcei(s_channels[0].alSource, AL_BUFFERS_PROCESSED, &processed);
			while (processed)
			{
				alSourceUnqueueBuffers(s_channels[0].alSource, 1, &buffer);
				processed--;
				alGetBufferi(buffer, AL_SIZE, &size);
				alDeleteBuffers(1, &buffer);
		
				// Update sg.soundtime (+= number of samples played (number of bytes / 4))
				s_soundtime += (size >> 2);
			}

#ifdef _DEBUG
			OutputDebugString("Restarting / Starting playback of Raw Samples\n");
#endif
			alSourcePlay(s_channels[0].alSource);
		}
	}

#ifdef _DEBUG
	if (alGetError() != AL_NO_ERROR)
		OutputDebugString("OAL Error : UpdateRawSamples\n");
#endif
}


int S_MP3PreProcessLipSync(channel_t *ch, short *data)
{
	int i;
	int sample;
	int sampleTotal = 0;

	for (i = 0; i < 576; i += 100)
	{
		sample = LittleShort(data[i]);
		sample = sample >> 8;
		sampleTotal += sample * sample;
	}

	sampleTotal /= 6;

	if (sampleTotal < ch->thesfx->fVolRange * s_lip_threshold_1->value)
		sample = -1;
	else if (sampleTotal < ch->thesfx->fVolRange * s_lip_threshold_2->value)
		sample = 1;
	else if (sampleTotal < ch->thesfx->fVolRange * s_lip_threshold_3->value)
		sample = 2;
	else if (sampleTotal < ch->thesfx->fVolRange * s_lip_threshold_4->value)
		sample = 3;
	else
		sample = 4;

	return sample;
}


void S_SetLipSyncs()
{
	int i;
	unsigned int samples;
	int currentTime, timePlayed;
	channel_t *ch;
#ifdef _DEBUG
	char szString[256];
#endif

	currentTime = Com_Milliseconds();

	memset(s_entityWavVol, 0, sizeof(s_entityWavVol));

	ch = s_channels + 1;
	for (i = 1; i < s_numChannels; i++, ch++)
	{
		if ((!ch->thesfx)||(!ch->bPlaying))
			continue;
		
		if ( ch->entchannel == CHAN_VOICE || ch->entchannel == CHAN_VOICE_ATTEN || ch->entchannel == CHAN_VOICE_GLOBAL )
		{
			// Calculate how much time has passed since the sample was started
			timePlayed = currentTime - ch->iStartTime;

			if (ch->thesfx->eSoundCompressionMethod==ct_16)
			{
				// There is a new computed lip-sync value every 1000 samples - so find out how many samples
				// have been played and lookup the value in the lip-sync table
				samples = (timePlayed * 22050) / 1000;

				if (ch->thesfx->lipSyncData == NULL)
				{
#ifdef _DEBUG
					sprintf(szString, "Missing lip-sync info. for %s\n", ch->thesfx->sSoundName);
					OutputDebugString(szString);
#endif
				}
				
				if ((ch->thesfx->lipSyncData) && (samples < ch->thesfx->iSoundLengthInSamples))
				{
					s_entityWavVol[ ch->entnum ] = ch->thesfx->lipSyncData[samples / 1000];

//					Com_Printf("%s,  total samples = %d, current sample = %d, lip type = %d \n", ch->thesfx->sSoundName, ch->thesfx->iSoundLengthInSamples, samples, s_entityWavVol[ ch->entnum ] );
					if ( s_show->integer == 3 ) 
					{
						Com_Printf( "(%i)%i %s vol = %i\n", ch->entnum, i, ch->thesfx->sSoundName, s_entityWavVol[ ch->entnum ] );
					}
				}
			}
			else
			{
				// MP3

				// There is a new computed lip-sync value every 576 samples - so find out how many samples
				// have been played and lookup the value in the lip-sync table
				samples = (timePlayed * 22050) / 1000;

				if (ch->thesfx->lipSyncData == NULL)
				{
#ifdef _DEBUG
					sprintf(szString, "Missing lip-sync info. for %s\n", ch->thesfx->sSoundName);
					OutputDebugString(szString);
#endif
				}

				if ((ch->thesfx->lipSyncData) && (samples < ch->thesfx->iSoundLengthInSamples))
				{
					s_entityWavVol[ ch->entnum ] = ch->thesfx->lipSyncData[(samples / 576) % 16];

					if ( s_show->integer == 3 ) 
					{
						Com_Printf( "(%i)%i %s vol = %i\n", ch->entnum, i, ch->thesfx->sSoundName, s_entityWavVol[ ch->entnum ] );
					}
				}
			}
		}
	}
}


/*
===============================================================================

console functions

===============================================================================
*/

static void S_Play_f( void ) {
	int 	i;
	sfxHandle_t	h;
	char name[256];
	
	i = 1;
	while ( i<Cmd_Argc() ) {
		if ( !strrchr(Cmd_Argv(i), '.') ) {
			Com_sprintf( name, sizeof(name), "%s.wav", Cmd_Argv(1) );
		} else {
			Q_strncpyz( name, Cmd_Argv(i), sizeof(name) );
		}
		h = S_RegisterSound( name );
		if( h ) {
			S_StartLocalSound( h, CHAN_LOCAL_SOUND );
		}
		i++;
	}
}

static void S_Music_f( void ) {
	int		c;

	c = Cmd_Argc();

	if ( c == 2 ) {
		S_StartBackgroundTrack( Cmd_Argv(1), Cmd_Argv(1), qfalse );
	} else if ( c == 3 ) {
		S_StartBackgroundTrack( Cmd_Argv(1), Cmd_Argv(2), qfalse );		
	} else {
		Com_Printf ("music <musicfile> [loopfile]\n");
		return;
	}
}

// a debug function, but no harm to leave in...
//
static void S_SetDynamicMusic_f(void)
{
	int c = Cmd_Argc();

	if ( c == 2 ) 
	{
		if (bMusic_IsDynamic)
		{
			// don't need to check existance of 'explore' or 'action' music, since music wouldn't
			//	be counted as dynamic if either were missing, but other types are optional...
			//
			if (!Q_stricmp(Cmd_Argv(1),"explore"))
			{
				S_SetDynamicMusicState( eBGRNDTRACK_EXPLORE );
				return;
			}
			else
			if (!Q_stricmp(Cmd_Argv(1),"action"))
			{
				S_SetDynamicMusicState( eBGRNDTRACK_ACTION );
				return;
			}
			else
			if (!Q_stricmp(Cmd_Argv(1),"silence"))
			{
				S_SetDynamicMusicState( eBGRNDTRACK_SILENCE );
				return;
			}
			else
			if (!Q_stricmp(Cmd_Argv(1),"boss"))
			{
				if (tMusic_Info[ eBGRNDTRACK_BOSS ].bExists)
				{
					S_SetDynamicMusicState( eBGRNDTRACK_BOSS );
				}
				else
				{
					Com_Printf("No 'boss' music defined in current dynamic set\n");
				}
				return;
			}
			else
			if (!Q_stricmp(Cmd_Argv(1),"death"))
			{
				if (tMusic_Info[ eBGRNDTRACK_DEATH ].bExists)
				{
					S_SetDynamicMusicState( eBGRNDTRACK_DEATH );
				}
				else
				{
					Com_Printf("No 'death' music defined in current dynamic set\n");
				}
				return;
			}
		}
		else
		{
			DynamicMusicInfoPrint();	// print "inactive" string
			return;
		}
	}

	// show usage...
	//
	Com_Printf("Usage: s_dynamic <explore/action/silence/boss/death>\n");
	DynamicMusicInfoPrint();
}


// this table needs to be in-sync with the typedef'd enum "SoundCompressionMethod_t"...	-ste
//
static const char *sSoundCompressionMethodStrings[ct_NUMBEROF] = 
{
	"16b",	// ct_16
	"mp3"	// ct_MP3
};
void S_SoundList_f( void ) {
	int		i;
	sfx_t	*sfx;
	int		size, total;
	int		iVariantCap = -1;	// for %d-inquiry stuff
	int		iTotalBytes = 0;

	qboolean bWavOnly = qfalse;
	qboolean bShouldBeMP3 = qfalse;

	if ( Cmd_Argc() == 2 ) 
	{
		if (!Q_stricmp(Cmd_Argv(1), "shouldbeMP3"))
		{
			bShouldBeMP3 = qtrue;
		}
		else
		if (!Q_stricmp(Cmd_Argv(1), "wavonly"))
		{
			bWavOnly = qtrue;
		}
		else
		{
			if (!Q_stricmp(Cmd_Argv(1), "1"))
			{
				iVariantCap = 1;
			}
			else
			if (!Q_stricmp(Cmd_Argv(1), "2"))
			{
				iVariantCap = 2;
			}
			else
			if (!Q_stricmp(Cmd_Argv(1), "3"))
			{
				iVariantCap = 3;
			}
		}		
	}
	else
	{
		Com_Printf("( additional (mutually exclusive) options available:\n'wavonly', 'ShouldBeMP3', '1'/'2'/'3' for %%d-variant capping )\n" );
	}



	total = 0;

	Com_Printf("\n");
	Com_Printf("                    InMemory?\n");
	Com_Printf("                    |\n");
	Com_Printf("                    |  LevelLastUsedOn\n");
	Com_Printf("                    |  |\n");
	Com_Printf("                    |  |\n");
	Com_Printf(" Slot   Smpls Type  |  |   Name\n");
//	Com_Printf(" Slot   Smpls Type  InMem?   Name\n");

	for (sfx=s_knownSfx, i=0 ; i<s_numSfx ; i++, sfx++) 
	{
		extern cvar_t *cv_MP3overhead;
		qboolean bMP3DumpOverride = bShouldBeMP3 && cv_MP3overhead && !sfx->bDefaultSound && !sfx->pMP3StreamHeader && sfx->pSoundData && (Z_Size(sfx->pSoundData) > cv_MP3overhead->integer);

		if (bMP3DumpOverride || (!bShouldBeMP3 && (!bWavOnly || sfx->eSoundCompressionMethod == ct_16)))
		{
			qboolean bDumpThisOne = qtrue;
			if (iVariantCap >= 1 && iVariantCap <= 3)
			{
				int iStrLen = strlen(sfx->sSoundName);
				if (iStrLen > 2)	// crash-safety, jic.
				{
					char c  = sfx->sSoundName[iStrLen-1];
					char c2 = sfx->sSoundName[iStrLen-2];
					if (!isdigit(c2) // quick-avoid of stuff like "pain75"
						&& isdigit(c) && atoi(va("%c",c)) > iVariantCap)
					{
						// need to see if this %d-variant should be omitted, in other words if there's a %1 version then skip this...
						//
						char sFindName[MAX_QPATH];
						Q_strncpyz(sFindName,sfx->sSoundName,sizeof(sFindName));
						sFindName[iStrLen-1] = '1';
						int i2;
						sfx_t *sfx2;
						for (sfx2 = s_knownSfx, i2=0 ; i2<s_numSfx ; i2++, sfx2++) 
						{
							if (!Q_stricmp(sFindName,sfx2->sSoundName))
							{
								bDumpThisOne = qfalse;	// found a %1-variant of this, so use variant capping and ignore this sfx_t
								break;
							}
						}
					}
				}
			}

			size = sfx->iSoundLengthInSamples;
			if (sfx->bDefaultSound)
			{
				Com_Printf("%5d Missing file: \"%s\"\n", i, sfx->sSoundName );
			}
			else
			{
				if (bDumpThisOne)
				{
					iTotalBytes += (sfx->bInMemory && sfx->pSoundData) ? Z_Size(sfx->pSoundData) : 0;
					iTotalBytes += (sfx->bInMemory && sfx->pMP3StreamHeader) ? sizeof(*sfx->pMP3StreamHeader) : 0;
					total		+=  sfx->bInMemory ? size : 0;				
				}
				Com_Printf("%5d %7i [%s] %s %2d %s", i, size, sSoundCompressionMethodStrings[sfx->eSoundCompressionMethod], sfx->bInMemory?"y":"n", sfx->iLastLevelUsedOn, sfx->sSoundName );

				if (!bDumpThisOne)
				{
					Com_Printf("   ( Skipping, variant capped )");
					//OutputDebugString(va("Variant capped: %s\n",sfx->sSoundName));
				}
				Com_Printf("\n");
			}
		}
	}
	Com_Printf(" Slot   Smpls Type  In? Lev  Name\n");


	Com_Printf ("Total resident samples: %i %s ( not mem usage, see 'meminfo' ).\n", total, bWavOnly?"(WAV only)":"");
	Com_Printf ("%d out of %d sfx_t slots used\n", s_numSfx, MAX_SFX);
	Com_Printf ("%.2fMB bytes used when counting sfx_t->pSoundData + MP3 headers (if any)\n", (float)iTotalBytes / 1024.0f / 1024.0f);
	S_DisplayFreeMemory();
}


/*
===============================================================================

background music functions

===============================================================================
*/

int	FGetLittleLong( fileHandle_t f ) {
	int		v;

	FS_Read( &v, sizeof(v), f );

	return LittleLong( v);
}

int	FGetLittleShort( fileHandle_t f ) {
	short	v;

	FS_Read( &v, sizeof(v), f );

	return LittleShort( v);
}

// returns the length of the data in the chunk, or 0 if not found
int S_FindWavChunk( fileHandle_t f, char *chunk ) {
	char	name[5];
	int		len;
	int		r;

	name[4] = 0;
	len = 0;
	r = FS_Read( name, 4, f );
	if ( r != 4 ) {
		return 0;
	}
	len = FGetLittleLong( f );
	if ( len < 0 || len > 0xfffffff ) {
		len = 0;
		return 0;
	}
	len = (len + 1 ) & ~1;		// pad to word boundary
//	s_nextWavChunk += len + 8;

	if ( strcmp( name, chunk ) ) {
		return 0;
	}
	return len;
}

// fixme: need to move this into qcommon sometime?, but too much stuff altered by other people and I won't be able
//	to compile again for ages if I check that out...
//
// DO NOT replace this with a call to FS_FileExists, that's for checking about writing out, and doesn't work for this.
//
qboolean S_FileExists( const char *psFilename )
{
	fileHandle_t fhTemp;

	FS_FOpenFileRead (psFilename, &fhTemp, qtrue);	// qtrue so I can fclose the handle without closing a PAK
	if (!fhTemp) 
		return qfalse;
	
	FS_FCloseFile(fhTemp);
	return qtrue;
}

// some stuff for streaming MP3 files from disk (not pleasant, but nothing about MP3 is, other than compression ratios...)
//
static void MP3MusicStream_Reset(MusicInfo_t *pMusicInfo)
{
	pMusicInfo->iMP3MusicStream_DiskReadPos		= 0;
	pMusicInfo->iMP3MusicStream_DiskWindowPos	= 0;
}

//
// return is where the decoder should read from...
//
static byte *MP3MusicStream_ReadFromDisk(MusicInfo_t *pMusicInfo, int iReadOffset, int iReadBytesNeeded)
{
	if (iReadOffset < pMusicInfo->iMP3MusicStream_DiskWindowPos)
	{
		assert(0);											// should never happen
		return pMusicInfo->byMP3MusicStream_DiskBuffer;		// ...but return something safe anyway
	}

	while (iReadOffset + iReadBytesNeeded > pMusicInfo->iMP3MusicStream_DiskReadPos)
	{
		int iBytesRead = FS_Read( pMusicInfo->byMP3MusicStream_DiskBuffer + (pMusicInfo->iMP3MusicStream_DiskReadPos - pMusicInfo->iMP3MusicStream_DiskWindowPos), iMP3MusicStream_DiskBytesToRead, pMusicInfo->s_backgroundFile );

		pMusicInfo->iMP3MusicStream_DiskReadPos += iBytesRead;

		if (iBytesRead != iMP3MusicStream_DiskBytesToRead)	// quietly ignore any requests to read past file end
		{
			break;		// we need to do this because the disk read code can't know how much source data we need to
						//	read for a given number of requested output bytes, so we'll always be asking for too many
		}
	}

	// if reached halfway point in buffer (approx 20k), backscroll it...
	//
	if (pMusicInfo->iMP3MusicStream_DiskReadPos - pMusicInfo->iMP3MusicStream_DiskWindowPos > iMP3MusicStream_DiskBufferSize/2)
	{
		int iMoveSrcOffset = iReadOffset - pMusicInfo->iMP3MusicStream_DiskWindowPos;		
		int iMoveCount     = (pMusicInfo->iMP3MusicStream_DiskReadPos - pMusicInfo->iMP3MusicStream_DiskWindowPos ) - iMoveSrcOffset;
		memmove( &pMusicInfo->byMP3MusicStream_DiskBuffer, &pMusicInfo->byMP3MusicStream_DiskBuffer[iMoveSrcOffset], iMoveCount);
		pMusicInfo->iMP3MusicStream_DiskWindowPos += iMoveSrcOffset;
	}

	return pMusicInfo->byMP3MusicStream_DiskBuffer + (iReadOffset - pMusicInfo->iMP3MusicStream_DiskWindowPos);
}


// does NOT set s_rawend!...
//
static void S_StopBackgroundTrack_Actual( MusicInfo_t *pMusicInfo ) 
{
	if ( pMusicInfo->s_backgroundFile ) 
	{
		if ( pMusicInfo->s_backgroundFile != -1)
		{
			Sys_EndStreamedFile( pMusicInfo->s_backgroundFile );
			FS_FCloseFile( pMusicInfo->s_backgroundFile );
		}
		pMusicInfo->s_backgroundFile = 0;	
	}
}

static void FreeMusic( MusicInfo_t *pMusicInfo )
{
	if (pMusicInfo->pLoadedData)
	{
		Z_Free(pMusicInfo->pLoadedData);
		pMusicInfo->pLoadedData		= NULL;		// these two MUST be kept as valid/invalid together
		pMusicInfo->sLoadedDataName[0]= '\0';	// 
		pMusicInfo->iLoadedDataLen	= 0;
	}
}

// called only by snd_shutdown (from snd_restart or app exit)
//
void S_UnCacheDynamicMusic( void )
{
	for (int i = eBGRNDTRACK_DATABEGIN; i != eBGRNDTRACK_DATAEND; i++)
	{
		FreeMusic( &tMusic_Info[i]);
	}
}

static qboolean S_StartBackgroundTrack_Actual( MusicInfo_t *pMusicInfo, qboolean qbDynamic, const char *intro, const char *loop )
{
	int		len;
	char	dump[16];
	char	name[MAX_QPATH];

	Q_strncpyz( sMusic_BackgroundLoop, loop, sizeof( sMusic_BackgroundLoop ));	

	Q_strncpyz( name, intro, sizeof( name ) - 4 );	// this seems to be so that if the filename hasn't got an extension
													//	but doesn't have the room to append on either then you'll just
													//	get the "soft" fopen() error, rather than the ERR_DROP you'd get
													//	if COM_DefaultExtension didn't have room to add it on.
	COM_DefaultExtension( name, sizeof( name ), ".mp3" );

	// close the background track, but DON'T reset s_rawend (or remaining music bits that haven't been output yet will be cut off)
	//
	S_StopBackgroundTrack_Actual( pMusicInfo );

	pMusicInfo->bIsMP3 = qfalse;

	if ( !intro[0] ) {
		return qfalse;
	}

	// new bit, if file requested is not same any loaded one (if prev was in-mem), ditch it...
	//
	if (Q_stricmp(name, pMusicInfo->sLoadedDataName))
	{
		FreeMusic( pMusicInfo );
	}

	if (!Q_stricmpn(name+(strlen(name)-4),".mp3",4))
	{
		if (pMusicInfo->pLoadedData)
		{				
			pMusicInfo->s_backgroundFile = -1;
		}
		else
		{
			pMusicInfo->iLoadedDataLen = FS_FOpenFileRead( name, &pMusicInfo->s_backgroundFile, qtrue );
		}

		if (!pMusicInfo->s_backgroundFile)
		{
			Com_Printf( S_COLOR_RED"Couldn't open music file %s\n", name );
			return qfalse;
		}

		MP3MusicStream_Reset( pMusicInfo );

		byte *pbMP3DataSegment	= NULL;
		int iInitialMP3ReadSize = 8192;		// fairly arbitrary, whatever size this is then the decoder is allowed to
											// scan up to halfway of it to find floating headers, so don't make it 
											// too small. 8k works fine.
		qboolean bMusicSucceeded = qfalse;
		if (qbDynamic)
		{	
			if (!pMusicInfo->pLoadedData)
			{
				pMusicInfo->pLoadedData = (byte *) Z_Malloc(pMusicInfo->iLoadedDataLen, TAG_SND_DYNAMICMUSIC, qfalse);

				S_ClearSoundBuffer();
				FS_Read(pMusicInfo->pLoadedData, pMusicInfo->iLoadedDataLen, pMusicInfo->s_backgroundFile);
				Q_strncpyz(pMusicInfo->sLoadedDataName, name, sizeof(pMusicInfo->sLoadedDataName));
			}

			// enable the rest of the code to work as before...
			//
			pbMP3DataSegment	= pMusicInfo->pLoadedData;
			iInitialMP3ReadSize = pMusicInfo->iLoadedDataLen;
		}
		else
		{
			pbMP3DataSegment = MP3MusicStream_ReadFromDisk(pMusicInfo, 0, iInitialMP3ReadSize);
		}

		if (MP3_IsValid(name, pbMP3DataSegment, iInitialMP3ReadSize, qtrue /*bStereoDesired*/))
		{
			// init stream struct...
			//
			memset(&pMusicInfo->streamMP3_Bgrnd,0,sizeof(pMusicInfo->streamMP3_Bgrnd));
			char *psError = C_MP3Stream_DecodeInit( &pMusicInfo->streamMP3_Bgrnd, pbMP3DataSegment, pMusicInfo->iLoadedDataLen,
													dma.speed,
													16,		// sfx->width * 8,
													qtrue	// bStereoDesired
													);


			if (psError == NULL)
			{
				// init sfx struct & setup the few fields I actually need...
				//
				memset(	   &pMusicInfo->sfxMP3_Bgrnd,0,sizeof(pMusicInfo->sfxMP3_Bgrnd));
				//			pMusicInfo->sfxMP3_Bgrnd.width					= 2;			// read by MP3_GetSamples()
							pMusicInfo->sfxMP3_Bgrnd.iSoundLengthInSamples	= 0x7FFFFFFF;	// max possible +ve int, since music finishes when decoder stops
							pMusicInfo->sfxMP3_Bgrnd.pMP3StreamHeader		= &pMusicInfo->streamMP3_Bgrnd;
				Q_strncpyz( pMusicInfo->sfxMP3_Bgrnd.sSoundName, name, sizeof(pMusicInfo->sfxMP3_Bgrnd.sSoundName) );

				if (qbDynamic)
				{
					MP3Stream_InitPlayingTimeFields ( &pMusicInfo->streamMP3_Bgrnd, name, pbMP3DataSegment, pMusicInfo->iLoadedDataLen, qtrue);
				}

				pMusicInfo->s_backgroundInfo.format		= WAV_FORMAT_MP3;	// not actually used this way, but just ensures we don't match one of the legit formats
				pMusicInfo->s_backgroundInfo.channels	= 2;		// always, for our MP3s when used for music (else 1 for FX)
				pMusicInfo->s_backgroundInfo.rate		= dma.speed;
				pMusicInfo->s_backgroundInfo.width		= 2;		// always, for our MP3s
				pMusicInfo->s_backgroundInfo.samples	= pMusicInfo->sfxMP3_Bgrnd.iSoundLengthInSamples;
				pMusicInfo->s_backgroundSamples			= pMusicInfo->sfxMP3_Bgrnd.iSoundLengthInSamples;

				memset(&pMusicInfo->chMP3_Bgrnd,0,sizeof(pMusicInfo->chMP3_Bgrnd));
						pMusicInfo->chMP3_Bgrnd.thesfx = &pMusicInfo->sfxMP3_Bgrnd;
				memcpy(&pMusicInfo->chMP3_Bgrnd.MP3StreamHeader, pMusicInfo->sfxMP3_Bgrnd.pMP3StreamHeader, sizeof(*pMusicInfo->sfxMP3_Bgrnd.pMP3StreamHeader));				

				if (qbDynamic)
				{					
					if (pMusicInfo->s_backgroundFile != -1)
					{
						FS_FCloseFile( pMusicInfo->s_backgroundFile );
						pMusicInfo->s_backgroundFile = -1;	// special mp3 value for "valid, but not a real file"
					}
				}
				
				pMusicInfo->bIsMP3 = qtrue;
				bMusicSucceeded = qtrue;
			}
			else
			{
				Com_Printf(S_COLOR_RED"Error streaming file %s: %s\n", name, psError);
				if (pMusicInfo->s_backgroundFile != -1)
				{
					FS_FCloseFile( pMusicInfo->s_backgroundFile );
				}
				pMusicInfo->s_backgroundFile = 0;
			}
		}
		else
		{
			// MP3_IsValid() will already have printed any errors via Com_Printf at this point...
			//			
			if (pMusicInfo->s_backgroundFile != -1)
			{
				FS_FCloseFile( pMusicInfo->s_backgroundFile );
			}
			pMusicInfo->s_backgroundFile = 0;
		}
		
		return bMusicSucceeded;
	}
	else	// not an mp3 file
	{
		//
		// open up a wav file and get all the info
		//
		FS_FOpenFileRead( name, &pMusicInfo->s_backgroundFile, qtrue );
		if ( !pMusicInfo->s_backgroundFile ) {
			Com_Printf( S_COLOR_YELLOW "WARNING: couldn't open music file %s\n", name );
			return qfalse;
		}

		// skip the riff wav header

		FS_Read(dump, 12, pMusicInfo->s_backgroundFile);

		if ( !S_FindWavChunk( pMusicInfo->s_backgroundFile, "fmt " ) ) {
			Com_Printf( S_COLOR_YELLOW "WARNING: No fmt chunk in %s\n", name );
			FS_FCloseFile( pMusicInfo->s_backgroundFile );
			pMusicInfo->s_backgroundFile = 0;
			return qfalse;
		}

		// save name for soundinfo
		pMusicInfo->s_backgroundInfo.format = FGetLittleShort( pMusicInfo->s_backgroundFile );
		pMusicInfo->s_backgroundInfo.channels = FGetLittleShort( pMusicInfo->s_backgroundFile );
		pMusicInfo->s_backgroundInfo.rate = FGetLittleLong( pMusicInfo->s_backgroundFile );
		FGetLittleLong(  pMusicInfo->s_backgroundFile );
		FGetLittleShort(  pMusicInfo->s_backgroundFile );
		pMusicInfo->s_backgroundInfo.width = FGetLittleShort( pMusicInfo->s_backgroundFile ) / 8;

		if ( pMusicInfo->s_backgroundInfo.format != WAV_FORMAT_PCM ) {
			FS_FCloseFile( pMusicInfo->s_backgroundFile );
			pMusicInfo->s_backgroundFile = 0;
			Com_Printf(S_COLOR_YELLOW "WARNING: Not a microsoft PCM format wav: %s\n", name);
			return qfalse;
		}

		if ( pMusicInfo->s_backgroundInfo.channels != 2 || pMusicInfo->s_backgroundInfo.rate != 22050 ) {
			Com_Printf(S_COLOR_YELLOW "WARNING: music file %s is not 22k stereo\n", name );
		}

		if ( ( len = S_FindWavChunk( pMusicInfo->s_backgroundFile, "data" ) ) == 0 ) {
			FS_FCloseFile( pMusicInfo->s_backgroundFile );
			pMusicInfo->s_backgroundFile = 0;
			Com_Printf(S_COLOR_YELLOW "WARNING: No data chunk in %s\n", name);
			return qfalse;
		}

		pMusicInfo->s_backgroundInfo.samples = len / (pMusicInfo->s_backgroundInfo.width * pMusicInfo->s_backgroundInfo.channels);

		pMusicInfo->s_backgroundSamples = pMusicInfo->s_backgroundInfo.samples;

		//
		// start the background streaming
		//
		Sys_BeginStreamedFile( pMusicInfo->s_backgroundFile, 0x10000 );
	}

	return qtrue;
}


static void S_SwitchDynamicTracks( MusicState_e eOldState, MusicState_e eNewState, qboolean bNewTrackStartsFullVolume )
{
	// copy old track into fader...
	//
	tMusic_Info[ eBGRNDTRACK_FADE ] = tMusic_Info[ eOldState ];
//	tMusic_Info[ eBGRNDTRACK_FADE ].bActive = qtrue;	// inherent
//	tMusic_Info[ eBGRNDTRACK_FADE ].bExists = qtrue;	// inherent
	tMusic_Info[ eBGRNDTRACK_FADE ].iXFadeVolumeSeekTime= Sys_Milliseconds();
	tMusic_Info[ eBGRNDTRACK_FADE ].iXFadeVolumeSeekTo	= 0;
	//
	// ... and deactivate...
	//
	tMusic_Info[ eOldState ].bActive = qfalse;
	//
	// set new track to either full volume or fade up...
	//
	tMusic_Info[eNewState].bActive				= qtrue;
	tMusic_Info[eNewState].iXFadeVolumeSeekTime	= Sys_Milliseconds();
	tMusic_Info[eNewState].iXFadeVolumeSeekTo	= 255;
	tMusic_Info[eNewState].iXFadeVolume			= bNewTrackStartsFullVolume ? 255 : 0;

	eMusic_StateActual = eNewState;


	if (s_debugdynamic->integer)
	{
		LPCSTR	psNewStateString = Music_BaseStateToString( eNewState, qtrue );
				psNewStateString = psNewStateString?psNewStateString:"<unknown>";

		Com_Printf( S_COLOR_MAGENTA "S_SwitchDynamicTracks( \"%s\" )\n", psNewStateString );
	}
}

// called by both the config-string parser and the console-command state-changer...
//
// This either changes the music right now (copying track structures etc), or leaves the new state as pending
//	so it gets picked up by the general music player if in a transition that can't be overridden...
//
static void S_SetDynamicMusicState( MusicState_e eNewState )
{
	if (eMusic_StateRequest != eNewState)
	{
		eMusic_StateRequest  = eNewState;

		if (s_debugdynamic->integer)
		{
			LPCSTR	psNewStateString = Music_BaseStateToString( eNewState, qtrue );
					psNewStateString = psNewStateString?psNewStateString:"<unknown>";

			Com_Printf( S_COLOR_MAGENTA "S_SetDynamicMusicState( Request: \"%s\" )\n", psNewStateString );
		}		
	}
}


static void S_HandleDynamicMusicStateChange( void )
{
	if (eMusic_StateRequest != eMusic_StateActual)
	{
		// check whether or not the new request can be honoured, given what's currently playing...
		//
		if (Music_StateCanBeInterrupted( eMusic_StateActual, eMusic_StateRequest ))
		{
			LP_MP3STREAM pMP3StreamActual = &tMusic_Info[ eMusic_StateActual ].chMP3_Bgrnd.MP3StreamHeader;

			switch (eMusic_StateRequest)
			{
				case eBGRNDTRACK_EXPLORE:	// ... from action or silence
				{
					switch (eMusic_StateActual)
					{
						case eBGRNDTRACK_ACTION:	// action->explore
						{
							// find the transition track to play, and the entry point for explore when we get there,
							//	and also see if we're at a permitted exit point to switch at all...
							//
							float fPlayingTimeElapsed = MP3Stream_GetPlayingTimeInSeconds( pMP3StreamActual ) - MP3Stream_GetRemainingTimeInSeconds( pMP3StreamActual );

							// supply:
							//
							// playing point in float seconds
							// enum of track being queried
							//
							// get:
							//
							// enum of transition track to switch to
							// float time of entry point of new track *after* transition

							MusicState_e	eTransition;
							float			fNewTrackEntryTime = 0.0f;
							if (Music_AllowedToTransition( fPlayingTimeElapsed, eBGRNDTRACK_ACTION, &eTransition, &fNewTrackEntryTime))
							{
								S_SwitchDynamicTracks( eMusic_StateActual, eTransition, qfalse );	// qboolean bNewTrackStartsFullVolume

								tMusic_Info[eTransition].Rewind();
								tMusic_Info[eTransition].bTrackSwitchPending	= qtrue;
								tMusic_Info[eTransition].eTS_NewState			= eMusic_StateRequest;
								tMusic_Info[eTransition].fTS_NewTime			= fNewTrackEntryTime;
							}
						}
						break;						

						case eBGRNDTRACK_SILENCE:	// silence->explore
						{
							S_SwitchDynamicTracks( eMusic_StateActual, eMusic_StateRequest, qfalse );	// qboolean bNewTrackStartsFullVolume

//							float fEntryTime = Music_GetRandomEntryTime( eMusic_StateRequest );
//							tMusic_Info[ eMusic_StateRequest ].SeekTo(fEntryTime);
							tMusic_Info[ eMusic_StateRequest ].Rewind();
						}
						break;

						default:	// trying to transition from some state I wasn't aware you could transition from (shouldn't happen), so ignore
						{
							assert(0); 
							S_SwitchDynamicTracks( eMusic_StateActual, eBGRNDTRACK_SILENCE, qfalse );	// qboolean bNewTrackStartsFullVolume
						}
						break;
					}
				}
				break;

				case eBGRNDTRACK_SILENCE:	// from explore or action
				{
					switch (eMusic_StateActual)
					{
						case eBGRNDTRACK_ACTION:	// action->silence
						case eBGRNDTRACK_EXPLORE:	// explore->silence
						{
							// find the transition track to play, and the entry point for explore when we get there,
							//	and also see if we're at a permitted exit point to switch at all...
							//
							float fPlayingTimeElapsed = MP3Stream_GetPlayingTimeInSeconds( pMP3StreamActual ) - MP3Stream_GetRemainingTimeInSeconds( pMP3StreamActual );

							MusicState_e	eTransition;
							float			fNewTrackEntryTime = 0.0f;
							if (Music_AllowedToTransition( fPlayingTimeElapsed, eMusic_StateActual, &eTransition, &fNewTrackEntryTime))
							{
								S_SwitchDynamicTracks( eMusic_StateActual, eTransition, qfalse );	// qboolean bNewTrackStartsFullVolume

								tMusic_Info[eTransition].Rewind();
								tMusic_Info[eTransition].bTrackSwitchPending	= qtrue;
								tMusic_Info[eTransition].eTS_NewState			= eMusic_StateRequest;
								tMusic_Info[eTransition].fTS_NewTime			= 0.0f;	//fNewTrackEntryTime;  irrelevant when switching to silence
							}
						}
						break;

						default:		// some unhandled type switching to silence
							assert(0);	// fall through since boss case just does silence->switch anyway

						case eBGRNDTRACK_BOSS:	// boss->silence
						{
							S_SwitchDynamicTracks( eMusic_StateActual, eBGRNDTRACK_SILENCE, qfalse );	// qboolean bNewTrackStartsFullVolume
						}
						break;
					}
				}
				break;

				case eBGRNDTRACK_ACTION:	// anything->action
				{
					switch (eMusic_StateActual)
					{
						case eBGRNDTRACK_SILENCE:	// silence->action
						{
							S_SwitchDynamicTracks( eMusic_StateActual, eMusic_StateRequest, qfalse );	// qboolean bNewTrackStartsFullVolume
							tMusic_Info[ eMusic_StateRequest ].Rewind();
						}
						break;

						default:	// !silence->action
						{
							S_SwitchDynamicTracks( eMusic_StateActual, eMusic_StateRequest, qtrue );	// qboolean bNewTrackStartsFullVolume
							float fEntryTime = Music_GetRandomEntryTime( eMusic_StateRequest );
							tMusic_Info[ eMusic_StateRequest ].SeekTo(fEntryTime);
						}
						break;
					}
				}
				break;

				case eBGRNDTRACK_BOSS:
				{	
					S_SwitchDynamicTracks( eMusic_StateActual, eMusic_StateRequest, qfalse );	// qboolean bNewTrackStartsFullVolume
					//
					// ( no need to fast forward or rewind, boss track is only entered into once, at start, and can't exit )
					//
				}
				break;

				case eBGRNDTRACK_DEATH:
				{	
					S_SwitchDynamicTracks( eMusic_StateActual, eMusic_StateRequest, qtrue );	// qboolean bNewTrackStartsFullVolume
					//
					// ( no need to fast forward or rewind, death track is only entered into once, at start, and can't exit or loop)
					//
				}
				break;

				default: assert(0); break;	// unknown new mode request, so just ignore it
			}
		}
	}
}



static char gsIntroMusic[MAX_QPATH]={0};
static char gsLoopMusic [MAX_QPATH]={0};

void S_RestartMusic( void ) 
{
	if (s_soundStarted && !s_soundMuted )
	{
		//if (gsIntroMusic[0] || gsLoopMusic[0])	// dont test this anymore (but still *use* them), they're blank for JK2 dynamic-music levels anyway
		{
			MusicState_e ePrevState	= eMusic_StateRequest;
			S_StartBackgroundTrack( gsIntroMusic, gsLoopMusic, qfalse );	// ( default music start will set the state to EXPLORE )
			S_SetDynamicMusicState( ePrevState );					// restore to prev state
		}
	}
}

// Basic logic here is to see if the intro file specified actually exists, and if so, then it's not dynamic music,
//	When called by the cgame start it loads up, then stops the playback (because of stutter issues), so that when the
//	actual snapshot is received and the real play request is processed the data has already been loaded so will be quicker.
//
// to be honest, although the code still plays WAVs some of the file-check logic only works for MP3s, so if you ever want
//	to use WAV music you'll have to do some tweaking below (but I've got other things to do so it'll have to wait - Ste)
//
void S_StartBackgroundTrack( const char *intro, const char *loop, qboolean bCalledByCGameStart )
{
	bMusic_IsDynamic = qfalse;

	if (!s_soundStarted)
	{	//we have no sound, so don't even bother trying
		return;
	}

	if ( !intro ) {
		intro = "";
	}
	if ( !loop || !loop[0] ) {
		loop = intro;
	}

	Q_strncpyz(gsIntroMusic,intro, sizeof(gsIntroMusic));
	Q_strncpyz(gsLoopMusic, loop,  sizeof(gsLoopMusic));

	char sNameIntro[MAX_QPATH];
	char sNameLoop [MAX_QPATH];
	Q_strncpyz(sNameIntro,	intro,	sizeof(sNameIntro));
	Q_strncpyz(sNameLoop,	loop,	sizeof(sNameLoop));

	COM_DefaultExtension( sNameIntro, sizeof( sNameIntro ), ".mp3" );
	COM_DefaultExtension( sNameLoop,  sizeof( sNameLoop),	".mp3" );

	// if dynamic music not allowed, then just stream the explore music instead of playing dynamic...
	//
	if (!s_allowDynamicMusic->integer && Music_DynamicDataAvailable(intro))	// "intro", NOT "sName" (i.e. don't use version with ".mp3" extension)
	{
		LPCSTR psMusicName = Music_GetFileNameForState( eBGRNDTRACK_DATABEGIN );
		if (psMusicName && S_FileExists( psMusicName ))
		{
			Q_strncpyz(sNameIntro,psMusicName,sizeof(sNameIntro));
			Q_strncpyz(sNameLoop, psMusicName,sizeof(sNameLoop ));
		}
	}

	// conceptually we always play the 'intro'[/sName] track, intro-to-loop transition is handled in UpdateBackGroundTrack().
	//
	if ( (strstr(sNameIntro,"/") && S_FileExists( sNameIntro )) )	// strstr() check avoids extra file-exists check at runtime if reverting from streamed music to dynamic since literal files all need at least one slash in their name (eg "music/blah")
	{
		LPCSTR psLoopName = S_FileExists( sNameLoop ) ? sNameLoop : sNameIntro;
		Com_DPrintf("S_StartBackgroundTrack: Found/using non-dynamic music track '%s' (loop: '%s')\n", sNameIntro, psLoopName);
		S_StartBackgroundTrack_Actual( &tMusic_Info[eBGRNDTRACK_NONDYNAMIC], bMusic_IsDynamic, sNameIntro, psLoopName );
	}
	else
	{
		if (Music_DynamicDataAvailable(intro))	// "intro", NOT "sName" (i.e. don't use version with ".mp3" extension)
		{
			extern const char *Music_GetLevelSetName(void);
			Q_strncpyz(sInfoOnly_CurrentDynamicMusicSet, Music_GetLevelSetName(), sizeof(sInfoOnly_CurrentDynamicMusicSet));
			int i;
			for (i = eBGRNDTRACK_DATABEGIN; i != eBGRNDTRACK_DATAEND; i++)
			{
				qboolean bOk = qfalse;
				LPCSTR psMusicName = Music_GetFileNameForState( (MusicState_e) i);
				if (psMusicName && (!Q_stricmp(tMusic_Info[i].sLoadedDataName, psMusicName) || S_FileExists( psMusicName )) )
				{
					bOk = S_StartBackgroundTrack_Actual( &tMusic_Info[i], qtrue, psMusicName, loop );
				}
				
				tMusic_Info[i].bExists = bOk;

				if (!tMusic_Info[i].bExists)
				{
					FreeMusic( &tMusic_Info[i] );
				}
			}

			//
			// default all tracks to OFF first (and set any other vars)
			//
			for (i=0; i<eBGRNDTRACK_NUMBEROF; i++)
			{
				tMusic_Info[i].bActive				= qfalse;
				tMusic_Info[i].bTrackSwitchPending	= qfalse;
				tMusic_Info[i].fSmoothedOutVolume	= 0.25f;
			}

			if (tMusic_Info[eBGRNDTRACK_EXPLORE].bExists &&
				tMusic_Info[eBGRNDTRACK_ACTION ].bExists
				)
			{
				Com_DPrintf("S_StartBackgroundTrack: Found dynamic music tracks\n");
				bMusic_IsDynamic = qtrue;

				//
				// ... then start the default music state...
				//
				eMusic_StateActual = eMusic_StateRequest = eBGRNDTRACK_EXPLORE;

				MusicInfo_t *pMusicInfo = &tMusic_Info[ eMusic_StateActual ];

				pMusicInfo->bActive				= qtrue;
				pMusicInfo->iXFadeVolumeSeekTime= Sys_Milliseconds();
				pMusicInfo->iXFadeVolumeSeekTo	= 255;
				pMusicInfo->iXFadeVolume		= 0;			
				
	//#ifdef _DEBUG
	//			float fRemaining = MP3Stream_GetPlayingTimeInSeconds( &pMusicInfo->chMP3_Bgrnd.MP3StreamHeader);
	//#endif
			}
			else
			{
				Com_Printf( S_COLOR_RED "Dynamic music did not have both 'action' and 'explore' versions, inhibiting...\n");
				S_StopBackgroundTrack();
			}
		}
		else
		{
			if (sNameIntro[0]!='.')	// blank name with ".mp3" or whatever attached - no error print out
			{
				Com_Printf( S_COLOR_RED "Unable to find music \"%s\" as explicit track or dynamic music entry!\n",sNameIntro);
				S_StopBackgroundTrack();
			}
		}
	}	

	if (bCalledByCGameStart)
	{
		S_StopBackgroundTrack();
	}
}

void S_StopBackgroundTrack( void )
{
	for (int i=0; i<eBGRNDTRACK_NUMBEROF; i++)
	{
		S_StopBackgroundTrack_Actual( &tMusic_Info[i] );
	}

	s_rawend = 0;
}



// qboolean return is true only if we're changing from a streamed intro to a dynamic loop...
//
static qboolean S_UpdateBackgroundTrack_Actual( MusicInfo_t *pMusicInfo, qboolean bFirstOrOnlyMusicTrack, float fDefaultVolume) 
{
	int		bufferSamples;
	int		fileSamples;
	byte	raw[30000];		// just enough to fit in a mac stack frame  (note that MP3 doesn't use full size of it)
	int		fileBytes;
	int		r;

	float fMasterVol = fDefaultVolume; // s_musicVolume->value;

	if (bMusic_IsDynamic)
	{
		// step xfade volume...
		//
		if ( pMusicInfo->iXFadeVolume != pMusicInfo->iXFadeVolumeSeekTo )
		{
			int iFadeMillisecondsElapsed = Sys_Milliseconds() - pMusicInfo->iXFadeVolumeSeekTime;

			if (iFadeMillisecondsElapsed > (fDYNAMIC_XFADE_SECONDS * 1000))
			{
				pMusicInfo->iXFadeVolume = pMusicInfo->iXFadeVolumeSeekTo;
			}
			else
			{
				pMusicInfo->iXFadeVolume = (int) (255.0f * ((float)iFadeMillisecondsElapsed/(fDYNAMIC_XFADE_SECONDS * 1000.0f)));
				if (pMusicInfo->iXFadeVolumeSeekTo == 0)	// bleurgh
					pMusicInfo->iXFadeVolume = 255 - pMusicInfo->iXFadeVolume;
			}
		}
		fMasterVol *= (float)((float)pMusicInfo->iXFadeVolume / 255.0f);
	}

// this is to work around an obscure issue to do with sliding decoder windows and amounts being requested, since the
//	original MP3 stream-decoder wrapper was designed to work with audio-paintbuffer sized pieces... Basically 30000
//	is far too big for the window decoder to handle in one request because of the time-travel issue associated with
//	normal sfx buffer painting, and allowing sufficient sliding room, even though the music file never goes back in time.
//
#define SIZEOF_RAW_BUFFER_FOR_MP3 4096
#define RAWSIZE (pMusicInfo->bIsMP3?SIZEOF_RAW_BUFFER_FOR_MP3:sizeof(raw))

	if ( !pMusicInfo->s_backgroundFile ) {
		return qfalse;
	}	

	pMusicInfo->fSmoothedOutVolume = (pMusicInfo->fSmoothedOutVolume + fMasterVol)/2.0f;
//	OutputDebugString(va("%f\n",pMusicInfo->fSmoothedOutVolume));

	// don't bother playing anything if musicvolume is 0
	if ( pMusicInfo->fSmoothedOutVolume <= 0 ) {
		return qfalse;
	}

	// see how many samples should be copied into the raw buffer
	if ( s_rawend < s_soundtime ) {
		s_rawend = s_soundtime;
	}

	while ( s_rawend < s_soundtime + MAX_RAW_SAMPLES ) 
	{
		bufferSamples = MAX_RAW_SAMPLES - (s_rawend - s_soundtime);

		// decide how much data needs to be read from the file
		fileSamples = bufferSamples * pMusicInfo->s_backgroundInfo.rate / dma.speed;

		// don't try and read past the end of the file
		if ( fileSamples > pMusicInfo->s_backgroundSamples ) {
			fileSamples = pMusicInfo->s_backgroundSamples;
		}

		// our max buffer size
		fileBytes = fileSamples * (pMusicInfo->s_backgroundInfo.width * pMusicInfo->s_backgroundInfo.channels);
		if (fileBytes > RAWSIZE ) {
			fileBytes = RAWSIZE;
			fileSamples = fileBytes / (pMusicInfo->s_backgroundInfo.width * pMusicInfo->s_backgroundInfo.channels);
		}

		qboolean qbForceFinish = qfalse;
		if (pMusicInfo->bIsMP3)
		{
			int iStartingSampleNum = pMusicInfo->chMP3_Bgrnd.thesfx->iSoundLengthInSamples - pMusicInfo->s_backgroundSamples;	// but this IS relevant
			// Com_Printf(S_COLOR_YELLOW "Requesting MP3 samples: sample %d\n",iStartingSampleNum);


			if (pMusicInfo->s_backgroundFile == -1)
			{
				// in-mem...
				//
				qbForceFinish = (MP3Stream_GetSamples( &pMusicInfo->chMP3_Bgrnd, iStartingSampleNum, fileBytes/2, (short*) raw, qtrue ))?qfalse:qtrue;				

				//Com_Printf(S_COLOR_YELLOW "Music time remaining: %f seconds\n", MP3Stream_GetRemainingTimeInSeconds( &pMusicInfo->chMP3_Bgrnd.MP3StreamHeader ));				
			}
			else
			{
				// streaming an MP3 file instead... (note that the 'fileBytes' request size isn't that relevant for MP3s, 
				//										since code here can't know how much the MP3 needs to decompress)
				//
				byte *pbScrolledStreamData = MP3MusicStream_ReadFromDisk(pMusicInfo, pMusicInfo->chMP3_Bgrnd.MP3StreamHeader.iSourceReadIndex, fileBytes);

				pMusicInfo->chMP3_Bgrnd.MP3StreamHeader.pbSourceData = pbScrolledStreamData - pMusicInfo->chMP3_Bgrnd.MP3StreamHeader.iSourceReadIndex;

				qbForceFinish = (MP3Stream_GetSamples( &pMusicInfo->chMP3_Bgrnd, iStartingSampleNum, fileBytes/2, (short*) raw, qtrue ))?qfalse:qtrue;
			}
		}
		else
		{
			// streaming a WAV off disk...
			//
			r = Sys_StreamedRead( raw, 1, fileBytes, pMusicInfo->s_backgroundFile );
			if ( r != fileBytes ) {
				Com_Printf(S_COLOR_RED"StreamedRead failure on music track\n");
				S_StopBackgroundTrack();
				return qfalse;
			}

			// byte swap if needed (do NOT do for MP3 decoder, that has an internal big/little endian handler)
			//
			S_ByteSwapRawSamples( fileSamples, pMusicInfo->s_backgroundInfo.width, pMusicInfo->s_backgroundInfo.channels, raw );
		}

		// add to raw buffer
		S_RawSamples(	fileSamples, pMusicInfo->s_backgroundInfo.rate, 
						pMusicInfo->s_backgroundInfo.width, pMusicInfo->s_backgroundInfo.channels, raw, pMusicInfo->fSmoothedOutVolume,
						bFirstOrOnlyMusicTrack
					);

		pMusicInfo->s_backgroundSamples -= fileSamples;
		if ( !pMusicInfo->s_backgroundSamples || qbForceFinish ) 
		{
			// loop the music, or play the next piece if we were on the intro...
			//	(but not for dynamic, that can only be used for loop music)
			//
			if (bMusic_IsDynamic)	// needs special logic for this, different call
			{
				pMusicInfo->Rewind();
			}
			else
			{
				// for non-dynamic music we need to check if "sMusic_BackgroundLoop" is an actual filename,
				//	or if it's a dynamic music specifier (which can't literally exist), in which case it should set
				//	a return flag then exit...
				//
				char sTestName[MAX_QPATH*2];// *2 so COM_DefaultExtension doesn't do an ERR_DROP if there was no space
											//	for an extension, since this is a "soft" test				
				Q_strncpyz( sTestName, sMusic_BackgroundLoop, sizeof(sTestName));
				COM_DefaultExtension(sTestName, sizeof(sTestName), ".mp3");

				if (S_FileExists( sTestName ))
				{
					S_StartBackgroundTrack_Actual( pMusicInfo, qfalse, sMusic_BackgroundLoop, sMusic_BackgroundLoop );
				}
				else
				{
					// proposed file doesn't exist, but this may be a dynamic track we're wanting to loop, 
					//	so exit with a special flag...
					//
					return qtrue;
				}
			}
			if ( !pMusicInfo->s_backgroundFile ) 
			{
				return qfalse;		// loop failed to restart
			}
		}
	}

#undef SIZEOF_RAW_BUFFER_FOR_MP3
#undef RAWSIZE

	return qfalse;
}


// used to be just for dynamic, but now even non-dynamic music has to know whether it should be silent or not...
//
static LPCSTR S_Music_GetRequestedState(void)
{
	int iStringOffset = cl.gameState.stringOffsets[CS_DYNAMIC_MUSIC_STATE];
	if (iStringOffset)
	{
		LPCSTR psCommand = cl.gameState.stringData+iStringOffset; 

		return psCommand;
	}
	
	return NULL;
}


// scan the configstring to see if there's been a state-change requested...
// (note that even if the state doesn't change it still gets here, so do a same-state check for applying)
//
// then go on to do transition handling etc...
//
static void S_CheckDynamicMusicState(void)
{
	LPCSTR psCommand = S_Music_GetRequestedState();

	if (psCommand)
	{
		MusicState_e eNewState;

		if ( !Q_stricmpn( psCommand, "silence", 7) )
		{				
			eNewState = eBGRNDTRACK_SILENCE;
		}
		else if ( !Q_stricmpn( psCommand, "action", 6) )
		{
			eNewState = eBGRNDTRACK_ACTION;
		}
		else if ( !Q_stricmpn( psCommand, "boss", 4) )
		{
			// special case, boss music is optional and may not be defined...
			//
			if (tMusic_Info[ eBGRNDTRACK_BOSS ].bExists)
			{
				eNewState = eBGRNDTRACK_BOSS;
			}
			else
			{
				// ( leave it playing current track )
				//
				eNewState = eMusic_StateActual;
			}
		}
		else if ( !Q_stricmpn( psCommand, "death", 5) )
		{
			// special case, death music is optional and may not be defined...
			//
			if (tMusic_Info[ eBGRNDTRACK_DEATH ].bExists)
			{
				eNewState = eBGRNDTRACK_DEATH;
			}
			else
			{
				// ( leave it playing current track, typically either boss or action )
				//
				eNewState = eMusic_StateActual;
			}
		}
		else
		{
			// seems a reasonable default...
			//
			eNewState = eBGRNDTRACK_EXPLORE;
		}
		
		S_SetDynamicMusicState( eNewState );
	}

	S_HandleDynamicMusicStateChange();
}

static void S_UpdateBackgroundTrack( void )
{
	if (bMusic_IsDynamic)
	{
		if (s_debugdynamic->integer == 2)
		{
			DynamicMusicInfoPrint();
		}
				
		S_CheckDynamicMusicState();
		
		if (eMusic_StateActual != eBGRNDTRACK_SILENCE)
		{
			MusicInfo_t *pMusicInfoCurrent = &tMusic_Info[ (eMusic_StateActual == eBGRNDTRACK_FADE)?eBGRNDTRACK_EXPLORE:eMusic_StateActual ];
			MusicInfo_t *pMusicInfoFadeOut = &tMusic_Info[ eBGRNDTRACK_FADE ];

			if ( pMusicInfoCurrent->s_backgroundFile == -1)
			{
				int iRawEnd = s_rawend;
				S_UpdateBackgroundTrack_Actual( pMusicInfoCurrent, qtrue, s_musicVolume->value );

	/*			static int iPrevFrontVol = 0;
				if (iPrevFrontVol != pMusicInfoCurrent->iXFadeVolume)
				{
					iPrevFrontVol  = pMusicInfoCurrent->iXFadeVolume;
					Com_Printf("front vol = %d\n",pMusicInfoCurrent->iXFadeVolume);
				}
	*/
				if (pMusicInfoFadeOut->bActive)
				{
					s_rawend = iRawEnd;
					S_UpdateBackgroundTrack_Actual( pMusicInfoFadeOut, qfalse, s_musicVolume->value );	// inactive-checked internally
	/*
					static int iPrevFadeVol = 0;
					if (iPrevFadeVol != pMusicInfoFadeOut->iXFadeVolume)
					{
						iPrevFadeVol  = pMusicInfoFadeOut->iXFadeVolume;
						Com_Printf("fade vol = %d\n",pMusicInfoFadeOut->iXFadeVolume);
					}
	*/
					//
					// only do this for the fader!...
					//
					if (pMusicInfoFadeOut->iXFadeVolume == 0)
					{
						pMusicInfoFadeOut->bActive = qfalse;
					}
				}
				
				float fRemainingTimeInSeconds = MP3Stream_GetRemainingTimeInSeconds( &pMusicInfoCurrent->chMP3_Bgrnd.MP3StreamHeader );
				// Com_Printf("Remaining: %3.3f\n",fRemainingTimeInSeconds);

				if ( fRemainingTimeInSeconds < fDYNAMIC_XFADE_SECONDS*2 )
				{
					// now either loop current track, switch if finishing a transition, or stop if finished a death...
					//
					if (pMusicInfoCurrent->bTrackSwitchPending)
					{
						pMusicInfoCurrent->bTrackSwitchPending = qfalse;	// ack
						S_SwitchDynamicTracks( eMusic_StateActual, pMusicInfoCurrent->eTS_NewState, qfalse);	// qboolean bNewTrackStartsFullVolume
						if (tMusic_Info[ pMusicInfoCurrent->eTS_NewState ].bExists)	// don't do this if switching to silence
						{							
							tMusic_Info[ pMusicInfoCurrent->eTS_NewState ].SeekTo(pMusicInfoCurrent->fTS_NewTime);
						}					
					}
					else
					{
						// normal looping, so set rewind current track, set volume to 0 and fade up to full (unless death track playing, then stays quiet)
						//	(while fader copy of end-section fades down)
						//
						// copy current track to fader...
						//
						*pMusicInfoFadeOut = *pMusicInfoCurrent;	// struct copy
						pMusicInfoFadeOut->iXFadeVolumeSeekTime	= Sys_Milliseconds();
						pMusicInfoFadeOut->iXFadeVolumeSeekTo	= 0;
						//
						pMusicInfoCurrent->Rewind();
						pMusicInfoCurrent->iXFadeVolumeSeekTime	= Sys_Milliseconds();
						pMusicInfoCurrent->iXFadeVolumeSeekTo	= (eMusic_StateActual == eBGRNDTRACK_DEATH) ? 0: 255;
						pMusicInfoCurrent->iXFadeVolume			= 0;
					}
				}
			}
		}
		else
		{
			// special case, when foreground music is shut off but fader still running to fade off previous track...
			//
			MusicInfo_t *pMusicInfoFadeOut = &tMusic_Info[ eBGRNDTRACK_FADE ];
			if (pMusicInfoFadeOut->bActive)
			{
				S_UpdateBackgroundTrack_Actual( pMusicInfoFadeOut, qtrue, s_musicVolume->value );
				if (pMusicInfoFadeOut->iXFadeVolume == 0)
				{
					pMusicInfoFadeOut->bActive = qfalse;
				}
			}	
		}
	}
	else
	{
		// standard / non-dynamic one-track music...
		//
		LPCSTR psCommand = S_Music_GetRequestedState();	// special check just for "silence" case...
		qboolean bShouldBeSilent = (psCommand && !Q_stricmp(psCommand,"silence"));
		float fDesiredVolume = bShouldBeSilent ? 0.0f : s_musicVolume->value;
		//
		// internal to this code is a volume-smoother...
		//
		qboolean bNewTrackDesired = S_UpdateBackgroundTrack_Actual(&tMusic_Info[eBGRNDTRACK_NONDYNAMIC], qtrue, fDesiredVolume);

		if (bNewTrackDesired)
		{
			S_StartBackgroundTrack( sMusic_BackgroundLoop, sMusic_BackgroundLoop, qfalse );
		}
	}
}


cvar_t *s_soundpoolmegs = NULL;


// currently passing in sfx as a param in case I want to do something with it later.
//
byte *SND_malloc(int iSize, sfx_t *sfx) 
{
	byte *pData = (byte *) Z_Malloc(iSize, TAG_SND_RAWDATA, qfalse);	// don't bother asking for zeroed mem

	// if "s_soundpoolmegs" is < 0, then the -ve of the value is the maximum amount of sounds we're allowed to have loaded...
	//
	if (s_soundpoolmegs && s_soundpoolmegs->integer < 0)
	{
		while ( (Z_MemSize(TAG_SND_RAWDATA) + Z_MemSize(TAG_SND_MP3STREAMHDR)) > ((-s_soundpoolmegs->integer) * 1024 * 1024))		
		{
			int iBytesFreed = SND_FreeOldestSound(sfx);
			if (iBytesFreed == 0)
				break;	// sanity
		}
	}

	return pData;
}


// called once-only in EXE lifetime...
//
void SND_setup() 
{		 
	s_soundpoolmegs = Cvar_Get("s_soundpoolmegs", "25", CVAR_ARCHIVE);
	if (Sys_LowPhysicalMemory() )
	{
		Cvar_Set("s_soundpoolmegs", "0");
	}
	
//	Com_Printf("Sound memory manager started\n");
}


// ask how much mem an sfx has allocated...
//
static int SND_MemUsed(sfx_t *sfx)
{
	int iSize = 0;
	if (sfx->pSoundData){
		iSize += Z_Size(sfx->pSoundData);
	}

	if (sfx->pMP3StreamHeader) {
		iSize += Z_Size(sfx->pMP3StreamHeader);
	}

	return iSize;
}

// free any allocated sfx mem...
//
// now returns # bytes freed to help with z_malloc()-fail recovery
//
static int SND_FreeSFXMem(sfx_t *sfx)
{
	int iBytesFreed = 0;

	if (s_UseOpenAL)
	{
		alGetError();
		if (sfx->Buffer)
		{
			alDeleteBuffers(1, &(sfx->Buffer));
#ifdef _DEBUG
			char szString[256];
			if (alGetError() != AL_NO_ERROR)
			{
				sprintf(szString, "Failed to delete AL Buffer (%s) ... !\n", sfx->sSoundName);
				OutputDebugString(szString);
			}
#endif
			sfx->Buffer = 0;
		}

		if (sfx->lipSyncData)
		{
			iBytesFreed +=	Z_Free(	sfx->lipSyncData);
									sfx->lipSyncData = NULL;
		}
	}

	if (						sfx->pSoundData) {
		iBytesFreed +=	Z_Free(	sfx->pSoundData );
								sfx->pSoundData = NULL;
	}

	sfx->bInMemory = false;	

	if (						sfx->pMP3StreamHeader) {
		iBytesFreed +=	Z_Free(	sfx->pMP3StreamHeader );
								sfx->pMP3StreamHeader = NULL;
	}

	return iBytesFreed;
}

void S_DisplayFreeMemory() 
{
	int iSoundDataSize = Z_MemSize ( TAG_SND_RAWDATA ) + Z_MemSize( TAG_SND_MP3STREAMHDR );
	int iMusicDataSize = Z_MemSize ( TAG_SND_DYNAMICMUSIC );

	if (iSoundDataSize || iMusicDataSize)
	{
		Com_Printf("\n%.2fMB audio data:  ( %.2fMB WAV/MP3 ) + ( %.2fMB Music )\n",
					((float)(iSoundDataSize+iMusicDataSize))/1024.0f/1024.0f,
										((float)(iSoundDataSize))/1024.0f/1024.0f,
																((float)(iMusicDataSize))/1024.0f/1024.0f
					);

		// now count up amount used on this level...
		//
		iSoundDataSize = 0;
		for (int i=1; i<s_numSfx; i++)
		{
			sfx_t *sfx = &s_knownSfx[i];

			if (sfx->iLastLevelUsedOn == RE_RegisterMedia_GetLevel()){
				iSoundDataSize += SND_MemUsed(sfx);
			}
		}

		Com_Printf("%.2fMB in sfx_t alloc data (WAV/MP3) loaded this level\n",(float)iSoundDataSize/1024.0f/1024.0f);
	}
}

void SND_TouchSFX(sfx_t *sfx)
{
	sfx->iLastTimeUsed		= Com_Milliseconds()+1;
	sfx->iLastLevelUsedOn	= RE_RegisterMedia_GetLevel();
}


// currently this is only called during snd_shutdown or snd_restart
//
void S_FreeAllSFXMem(void)
{
	for (int i=1 ; i < s_numSfx ; i++)	// start @ 1 to skip freeing default sound
	{
		SND_FreeSFXMem(&s_knownSfx[i]);
	}
}

// returns number of bytes freed up...
//
// new param is so we can be usre of not freeing ourselves (without having to rely on possible uninitialised timers etc)
//
int SND_FreeOldestSound(sfx_t *pButNotThisOne /* = NULL */) 
{	
	int iBytesFreed = 0;
	sfx_t *sfx;

	int	iOldest = Com_Milliseconds();
	int	iUsed	= 0;

	// start on 1 so we never dump the default sound...
	//
	for (int i=1 ; i < s_numSfx ; i++) 
	{
		sfx = &s_knownSfx[i];

		if (sfx != pButNotThisOne)
		{
			if (!sfx->bDefaultSound && sfx->bInMemory && sfx->iLastTimeUsed < iOldest) 
			{
				// new bit, we can't throw away any sfx_t struct in use by a channel, else the paint code will crash...
				//
				int iChannel;
				for (iChannel=0; iChannel<MAX_CHANNELS; iChannel++)
				{
					channel_t *ch = & s_channels[iChannel];

					if (ch->thesfx == sfx)
						break;	// damn, being used
				}
				if (iChannel == MAX_CHANNELS)
				{
					// this sfx_t struct wasn't used by any channels, so we can lose it...
					//			
					iUsed = i;
					iOldest = sfx->iLastTimeUsed;
				}
			}
		}
	}

	if (iUsed)
	{
		sfx = &s_knownSfx[ iUsed ];
	
		Com_DPrintf("SND_FreeOldestSound: freeing sound %s\n", sfx->sSoundName);
	
		iBytesFreed = SND_FreeSFXMem(sfx);
	}

	return iBytesFreed;
}
int SND_FreeOldestSound(void)
{
	return SND_FreeOldestSound(NULL);	// I had to add a void-arg version of this because of link issues, sigh
}


// just before we drop into a level, ensure the audio pool is under whatever the maximum
//	pool size is (but not by dropping out sounds used by the current level)...
//
// returns qtrue if at least one sound was dropped out, so z_malloc-fail recovery code knows if anything changed
//
extern qboolean gbInsideLoadSound;
qboolean SND_RegisterAudio_LevelLoadEnd(qboolean bDeleteEverythingNotUsedThisLevel /* 99% qfalse */)
{
	qboolean bAtLeastOneSoundDropped = qfalse;

	Com_DPrintf( "SND_RegisterAudio_LevelLoadEnd():\n");

	if (gbInsideLoadSound)
	{
		Com_DPrintf( "(Inside S_LoadSound (z_malloc recovery?), exiting...\n");
	}
	else
	{
		int iLoadedAudioBytes	 = Z_MemSize ( TAG_SND_RAWDATA ) + Z_MemSize( TAG_SND_MP3STREAMHDR );
		const int iMaxAudioBytes = s_soundpoolmegs->integer * 1024 * 1024;

		for (int i=1; i<s_numSfx && ( iLoadedAudioBytes > iMaxAudioBytes || bDeleteEverythingNotUsedThisLevel) ; i++) // i=1 so we never page out default sound
		{
			sfx_t *sfx = &s_knownSfx[i];

			if (sfx->bInMemory)
			{
				qboolean bDeleteThis = qfalse;

				if (bDeleteEverythingNotUsedThisLevel)
				{
					bDeleteThis = (sfx->iLastLevelUsedOn != RE_RegisterMedia_GetLevel());
				}
				else
				{
					bDeleteThis = (sfx->iLastLevelUsedOn < RE_RegisterMedia_GetLevel());
				}

				if (bDeleteThis)
				{
					Com_DPrintf( "Dumping sfx_t \"%s\"\n",sfx->sSoundName);			

					if (SND_FreeSFXMem(sfx))
					{
						bAtLeastOneSoundDropped = qtrue;
					}

					iLoadedAudioBytes = Z_MemSize ( TAG_SND_RAWDATA ) + Z_MemSize( TAG_SND_MP3STREAMHDR );					
				}
			}
		}
	}

	Com_DPrintf( "SND_RegisterAudio_LevelLoadEnd(): Ok\n");	

	return bAtLeastOneSoundDropped;
}


#ifdef HAVE_EAX
/****************************************************************************************************\
*
*	EAX Related
*
\****************************************************************************************************/

/*
	Initialize the EAX Manager
*/
void InitEAXManager()
{
	LPEAXMANAGERCREATE lpEAXManagerCreateFn;
	EAXFXSLOTPROPERTIES FXSlotProp;
	GUID	Effect;
	GUID	FXSlotGuids[4];
	int i;

	s_bEALFileLoaded = false;

	// Check for EAX 4.0 support
	s_bEAX = alIsExtensionPresent((ALubyte*)"EAX4.0");
	
	if (s_bEAX)
	{
		Com_Printf("Found EAX 4.0 native support\n");
	}
	else
	{
		// Support for EAXUnified (automatic translation of EAX 4.0 calls into EAX 3.0)
		if ((alIsExtensionPresent((ALubyte*)"EAX3.0")) && (alIsExtensionPresent((ALubyte*)"EAX4.0Emulated")))
		{
			s_bEAX = AL_TRUE;
			Com_Printf("Found EAX 4.0 EMULATION support\n");
		}
	}

	if (s_bEAX)
	{
		s_eaxSet = (EAXSet)alGetProcAddress((ALubyte*)"EAXSet");
		if (s_eaxSet == NULL)
			s_bEAX = false;
		s_eaxGet = (EAXGet)alGetProcAddress((ALubyte*)"EAXGet");
		if (s_eaxGet == NULL)
			s_bEAX = false;
	}

	// If we have detected EAX support, then try and load the EAX Manager DLL
	if (s_bEAX)
	{
		s_hEAXManInst = LoadLibrary("EAXMan.dll");
		if (s_hEAXManInst)
		{
			lpEAXManagerCreateFn = (LPEAXMANAGERCREATE)GetProcAddress(s_hEAXManInst, "EaxManagerCreate");
			if (lpEAXManagerCreateFn)
			{
				if (lpEAXManagerCreateFn(&s_lpEAXManager)==EM_OK)
				{
					// Configure our EAX 4.0 Effect Slots

					s_NumFXSlots = 0;
					for (i = 0; i < EAX_MAX_FXSLOTS; i++)
					{
						s_FXSlotInfo[i].FXSlotGuid = EAX_NULL_GUID;
						s_FXSlotInfo[i].lEnvID = -1;
					}

					FXSlotGuids[0] = EAXPROPERTYID_EAX40_FXSlot0;
					FXSlotGuids[1] = EAXPROPERTYID_EAX40_FXSlot1;
					FXSlotGuids[2] = EAXPROPERTYID_EAX40_FXSlot2;
					FXSlotGuids[3] = EAXPROPERTYID_EAX40_FXSlot3;

					// For each effect slot, try and load a reverb and lock the slot
					FXSlotProp.guidLoadEffect = EAX_REVERB_EFFECT;
					FXSlotProp.lVolume = 0;
					FXSlotProp.lLock = EAXFXSLOT_LOCKED;
					FXSlotProp.ulFlags = EAXFXSLOTFLAGS_ENVIRONMENT;

					for (i = 0; i < EAX_MAX_FXSLOTS; i++)
					{
						if (s_eaxSet(&FXSlotGuids[i], EAXFXSLOT_ALLPARAMETERS, NULL, &FXSlotProp, sizeof(EAXFXSLOTPROPERTIES))==AL_NO_ERROR)
						{
							// We can use this slot
							s_FXSlotInfo[s_NumFXSlots].FXSlotGuid = FXSlotGuids[i];
							s_NumFXSlots++;
						}
						else
						{
							// If this slot already contains a reverb, then we will use it anyway (Slot 0 will
							// be in this category).  (It probably means that Slot 0 is locked)
							if (s_eaxGet(&FXSlotGuids[i], EAXFXSLOT_LOADEFFECT, NULL, &Effect, sizeof(GUID))==AL_NO_ERROR)
							{
								if (Effect == EAX_REVERB_EFFECT)
								{
									// We can use this slot
									// Make sure the environment flag is on
									s_eaxSet(&FXSlotGuids[i], EAXFXSLOT_FLAGS, NULL, &FXSlotProp.ulFlags, sizeof(unsigned long));
									s_FXSlotInfo[s_NumFXSlots].FXSlotGuid = FXSlotGuids[i];
									s_NumFXSlots++;
								}
							}
						}
					}

					return;
				}
			}
		}
	}

	// If the EAXManager library was loaded (and there was a problem), then unload it
	if (s_hEAXManInst)
	{
		FreeLibrary(s_hEAXManInst);
		s_hEAXManInst = NULL;
	}

	s_lpEAXManager = NULL;
	s_bEAX = false;

	return;
}

/*
	Release the EAX Manager
*/
void ReleaseEAXManager()
{
	s_bEAX = false;

	UnloadEALFile();

	if (s_lpEAXManager)
	{
		s_lpEAXManager->Release();
		s_lpEAXManager = NULL;
	}
	if (s_hEAXManInst)
	{
		FreeLibrary(s_hEAXManInst);
		s_hEAXManInst = NULL;
	}
}


/*
	Try to load the given .eal file
*/
static bool LoadEALFile(char *szEALFilename)
{
	char		*ealData = NULL;
	HRESULT		hr;
	long		i, j, lID, lEnvID;
	EMPOINT		EMPoint;
	char		szAperture[128];
	char		szFullEALFilename[MAX_QPATH];
	long		lNumInst, lNumInstA, lNumInstB;
	bool		bLoaded = false;
	bool		bValid = true;
	int			result;
	char		szString[256];
	
	if ((!s_lpEAXManager) || (!s_bEAX))
		return false;

	if (strstr(szEALFilename, "nomap"))
		return false;

	s_EnvironmentID = 0xFFFFFFFF;

	// Assume there is no aperture information in the .eal file
	s_lpEnvTable = NULL;

	// Load EAL file from PAK file
	result = FS_ReadFile(szEALFilename, (void **)&ealData);

	if ((ealData) && (result != -1))
	{
		hr = s_lpEAXManager->LoadDataSet(ealData, EMFLAG_LOADFROMMEMORY);
	
		// Unload EAL file
		FS_FreeFile (ealData);
	
		if (hr == EM_OK)
		{
			Com_DPrintf("Loaded %s by Quake loader\n", szEALFilename);
			bLoaded = true;
		}
	}
	else
	{
		// Failed to load via Quake loader, try manually
		Com_sprintf(szFullEALFilename, MAX_QPATH, "base/%s", szEALFilename);
		if (SUCCEEDED(s_lpEAXManager->LoadDataSet(szFullEALFilename, 0)))
		{
			Com_DPrintf("Loaded %s by EAXManager\n", szEALFilename);
			bLoaded = true;
		}
	}

	if (bLoaded)
	{
		// For a valid eal file ... need to find 'Center' tag, record num of instances,  and then find
		// the right number of instances of 'Aperture0a' and 'Aperture0b'.

		if (s_lpEAXManager->GetSourceID("Center", &lID)==EM_OK)
		{
			if (s_lpEAXManager->GetSourceNumInstances(lID, &s_lNumEnvironments)==EM_OK)
			{
				if (s_lpEAXManager->GetSourceID("Aperture0a", &lID)==EM_OK)
				{
					if (s_lpEAXManager->GetSourceNumInstances(lID, &lNumInst)==EM_OK)
					{
						if (lNumInst == s_lNumEnvironments)
						{
							if (s_lpEAXManager->GetSourceID("Aperture0b", &lID)==EM_OK)
							{
								if (s_lpEAXManager->GetSourceNumInstances(lID, &lNumInst)==EM_OK)
								{
									if (lNumInst == s_lNumEnvironments)
									{
										// Check equal numbers of ApertureXa and ApertureXb
										i = 1;
										while (true)
										{
											lNumInstA = lNumInstB = 0;

											sprintf(szAperture,"Aperture%da",i);
											if ((s_lpEAXManager->GetSourceID(szAperture, &lID)==EM_OK) && (s_lpEAXManager->GetSourceNumInstances(lID, &lNumInstA)==EM_OK))
											{
												sprintf(szAperture,"Aperture%db",i);
												s_lpEAXManager->GetSourceID(szAperture, &lID);
												s_lpEAXManager->GetSourceNumInstances(lID, &lNumInstB);

												if (lNumInstA!=lNumInstB)
												{
													Com_DPrintf( S_COLOR_YELLOW "Invalid EAL file - %d Aperture%da tags, and %d Aperture%db tags\n", lNumInstA, i, lNumInstB, i);
													bValid = false;
												}
											}
											else
											{
												break;
											}

											i++;
										}

										if (bValid)
										{
											s_lpEnvTable = (LPENVTABLE)Z_Malloc(s_lNumEnvironments * sizeof(ENVTABLE), TAG_NEWDEL, qtrue);
										}
									}
									else
										Com_DPrintf( S_COLOR_YELLOW "Invalid EAL File - expected %d instances of Aperture0b, found %d\n", s_lNumEnvironments, lNumInst);
								}
								else
									Com_DPrintf( S_COLOR_YELLOW "EAXManager- failed GetSourceNumInstances()\n");
							}
							else
								Com_DPrintf( S_COLOR_YELLOW "Invalid EAL File - no instances of 'Aperture0b' source-tag\n");
						}
						else
							Com_DPrintf( S_COLOR_YELLOW "Invalid EAL File - found %d instances of the 'Center' tag, but only %d instances of 'Aperture0a'\n", s_lNumEnvironments, lNumInst);
					}
					else
						Com_DPrintf( S_COLOR_YELLOW "EAXManager- failed GetSourceNumInstances()\n");
				}
				else
					Com_DPrintf( S_COLOR_YELLOW "Invalid EAL File - no instances of 'Aperture0a' source-tag\n");
			}
			else
				Com_DPrintf( S_COLOR_YELLOW "EAXManager- failed GetSourceNumInstances()\n");
		}
		else
			Com_DPrintf( S_COLOR_YELLOW "Invalid EAL File - no instances of 'Center' source-tag\n");


		if (s_lpEnvTable)
		{
			i = 0;
			while (true)
			{
				sprintf(szAperture, "Aperture%da", i);
				if (s_lpEAXManager->GetSourceID(szAperture, &lID)==EM_OK)
				{
					if (s_lpEAXManager->GetSourceNumInstances(lID, &lNumInst)==EM_OK)
					{
						for (j = 0; j < s_lNumEnvironments; j++)
						{
							s_lpEnvTable[j].bUsed = false;
						}

						for (j = 0; j < lNumInst; j++)
						{
							if (s_lpEAXManager->GetSourceInstancePos(lID, j, &EMPoint)==EM_OK)
							{
								if (s_lpEAXManager->GetListenerDynamicAttributes(0, &EMPoint, &lEnvID, 0)==EM_OK)
								{
									if ((lEnvID >= 0) && (lEnvID < s_lNumEnvironments))
									{
										assert(s_lpEnvTable[lEnvID].ulNumApertures < 64);
										if (!s_lpEnvTable[lEnvID].bUsed)
										{
											s_lpEnvTable[lEnvID].bUsed = true;
											s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[0] = EMPoint.fX;
											s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[1] = EMPoint.fY;
											s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[2] = EMPoint.fZ;
										}
										else
										{
											s_lpEAXManager->GetEnvironmentName(lEnvID, szString, 256);
											Com_DPrintf( S_COLOR_YELLOW "Found more than one occurance of Aperture%da in %s sub-space\n", i, szString);
											Com_DPrintf( S_COLOR_YELLOW "One tag at %.3f,%.3f,%.3f, other at %.3f,%.3f,%.3f\n", EMPoint.fX, EMPoint.fY, EMPoint.fZ,
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[0], s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[1],
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[2]);
											bValid = false;
										}
									}
									else
									{
										if (lEnvID==-1)
											Com_DPrintf( S_COLOR_YELLOW "%s (%.3f,%.3f,%.3f) in Default Environment - please remove\n", szAperture, EMPoint.fX, EMPoint.fY, EMPoint.fZ);
										else
											Com_DPrintf( S_COLOR_YELLOW "Detected more reverb presets than zones - please delete unused presets\n");
										bValid = false;
									}
								}
							}
						}
					}
				}
				else
				{
					break;
				}

				if (bValid)
				{
					sprintf(szAperture, "Aperture%db", i);
					if (s_lpEAXManager->GetSourceID(szAperture, &lID)==EM_OK)
					{
						if (s_lpEAXManager->GetSourceNumInstances(lID, &lNumInst)==EM_OK)
						{
							for (j = 0; j < s_lNumEnvironments; j++)
							{
								s_lpEnvTable[j].bUsed = false;
							}

							for (j = 0; j < lNumInst; j++)
							{
								if (s_lpEAXManager->GetSourceInstancePos(lID, j, &EMPoint)==EM_OK)
								{
									if (s_lpEAXManager->GetListenerDynamicAttributes(0, &EMPoint, &lEnvID, 0)==EM_OK)
									{
										if ((lEnvID >= 0) && (lEnvID < s_lNumEnvironments))
										{
											if (!s_lpEnvTable[lEnvID].bUsed)
											{
												s_lpEnvTable[lEnvID].bUsed = true;
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos2[0] = EMPoint.fX;
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos2[1] = EMPoint.fY;
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos2[2] = EMPoint.fZ;
											}
											else
											{
												s_lpEAXManager->GetEnvironmentName(lEnvID, szString, 256);
												Com_DPrintf( S_COLOR_YELLOW "Found more than one occurance of Aperture%db in %s sub-space\n", i, szString);
												bValid = false;
											}

											// Calculate center position of aperture (average of 2 points)

											s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vCenter[0] =
												(s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[0] +
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos2[0]) / 2;

											s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vCenter[1] =
												(s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[1] +
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos2[1]) / 2;

											s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vCenter[2] =
												(s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos1[2] +
												s_lpEnvTable[lEnvID].Aperture[s_lpEnvTable[lEnvID].ulNumApertures].vPos2[2]) / 2;

											s_lpEnvTable[lEnvID].ulNumApertures++;
											assert(s_lpEnvTable[lEnvID].ulNumApertures < 64);
										}
										else
										{
											if (lEnvID==-1)
												Com_DPrintf( S_COLOR_YELLOW "%s (%.3f,%.3f,%.3f) in Default Environment - please remove\n", szAperture, EMPoint.fX, EMPoint.fY, EMPoint.fZ);
											else
												Com_DPrintf( S_COLOR_YELLOW "Detected more reverb presets than zones - please delete unused presets\n");
											bValid = false;
										}
									}
								}
							}
						}
					}
				}

				if (!bValid)
				{
					// Found a problem
					Com_DPrintf( S_COLOR_YELLOW "EAX legacy behaviour invoked (one reverb)\n");

					Z_Free(s_lpEnvTable);
					s_lpEnvTable = NULL;
					break;
				}

				i++;
			}
		}
		else
		{
			Com_DPrintf( S_COLOR_YELLOW "EAX legacy behaviour invoked (one reverb)\n");
		}

		return true;
	}

	
	Com_DPrintf( S_COLOR_YELLOW "Failed to load %s\n", szEALFilename);
	return false;
}

/*
	Unload current .eal file
*/
static void UnloadEALFile()
{
	HRESULT hr;

	if ((!s_lpEAXManager) || (!s_bEAX))
		return;

	hr = s_lpEAXManager->FreeDataSet(0);
	s_bEALFileLoaded = false;

	if (s_lpEnvTable)
	{
		Z_Free(s_lpEnvTable);
		s_lpEnvTable = NULL;
	}

	return;
}

/*
	Updates the current EAX Reverb setting, based on the location of the listener
*/
static void UpdateEAXListener()
{
	EMPOINT ListPos, ListOri;
	EMPOINT EMAperture;
	EMPOINT EMSourcePoint;
	long lID, lSourceID, lApertureNum;
	int i, j, k;
	float flDistance, flNearest;
	EAXREVERBPROPERTIES Reverb;
	bool bFound;
	long lVolume;
	long lCurTime;
	channel_t	*ch;
	EAXVECTOR	LR, LP1, LP2, Pan;
	REVERBDATA ReverbData[3]; // Hardcoded to three (maximum no of reverbs)
#ifdef DISPLAY_CLOSEST_ENVS
	char szEnvName[256];
#endif

	if ((!s_lpEAXManager) || (!s_bEAX))
		return;
	
	lCurTime = Com_Milliseconds();

	if ((s_lLastEnvUpdate + ENV_UPDATE_RATE) < lCurTime)
	{
		// Update closest reverbs
		s_lLastEnvUpdate = lCurTime;

		// No panning information in .eal file, or we only have 1 FX Slot to use, revert to legacy
		// behaviour (i.e only one reverb)
		if ((!s_lpEnvTable) || (s_NumFXSlots==1))
		{
			// Convert Listener co-ordinate to left-handed system
			ListPos.fX = listener_pos[0];
			ListPos.fY = listener_pos[1];
			ListPos.fZ = -listener_pos[2];

			if (SUCCEEDED(s_lpEAXManager->GetListenerDynamicAttributes(0, &ListPos, &lID, EMFLAG_LOCKPOSITION)))
			{
				if (lID != s_EnvironmentID)
				{
#ifdef DISPLAY_CLOSEST_ENVS
					if (SUCCEEDED(s_lpEAXManager->GetEnvironmentName(lID, szEnvName, 256)))
						Com_Printf("Changing to '%s' zone !\n", szEnvName);
#endif
					// Get EAX Preset info.
					if (SUCCEEDED(s_lpEAXManager->GetEnvironmentAttributes(lID, &s_eaxLPCur)))
					{
						// Override
						s_eaxLPCur.flAirAbsorptionHF = 0.0f;

						// Set Environment
						s_eaxSet(&EAXPROPERTYID_EAX40_FXSlot0, EAXREVERB_ALLPARAMETERS,
							NULL, &s_eaxLPCur, sizeof(EAXREVERBPROPERTIES));

						s_EnvironmentID = lID;
					}
				}
			}

			return;
		}

		// Convert Listener position and orientation to left-handed system
		ListPos.fX = listener_pos[0];
		ListPos.fY = listener_pos[1];
		ListPos.fZ = -listener_pos[2];

		ListOri.fX = listener_ori[0];
		ListOri.fY = listener_ori[1];
		ListOri.fZ = -listener_ori[2];

		// Need to find closest s_NumFXSlots (including the Listener's slot)

		if (s_lpEAXManager->GetListenerDynamicAttributes(0, &ListPos, &lID, EMFLAG_LOCKPOSITION)==EM_OK)
		{
			if (lID == -1)
			{
				// Found default environment
//				Com_Printf( S_COLOR_YELLOW "Listener in default environment - ignoring zone !\n");
				return;
			}

			ReverbData[0].lEnvID = -1;
			ReverbData[0].lApertureNum = -1;
			ReverbData[0].flDist = FLT_MAX;

			ReverbData[1].lEnvID = -1;
			ReverbData[1].lApertureNum = -1;
			ReverbData[1].flDist = FLT_MAX;

			ReverbData[2].lEnvID = lID;
			ReverbData[2].lApertureNum = -1;
			ReverbData[2].flDist = 0.0f;

			for (i = 0; i < s_lNumEnvironments; i++)
			{
				// Ignore Environment id lID as this one will always be used
				if (i != lID)
				{
					flNearest = FLT_MAX;
					lApertureNum = 0;	//shut up compile warning

					for (j = 0; j < s_lpEnvTable[i].ulNumApertures; j++)
					{
						EMAperture.fX = s_lpEnvTable[i].Aperture[j].vCenter[0];
						EMAperture.fY = s_lpEnvTable[i].Aperture[j].vCenter[1];
						EMAperture.fZ = s_lpEnvTable[i].Aperture[j].vCenter[2];

						flDistance = CalcDistance(EMAperture, ListPos);

						if (flDistance < flNearest)
						{
							flNearest = flDistance;
							lApertureNum = j;
						}
					}

					// Now have closest point for this Environment - see if this is closer than any others

					if (flNearest < ReverbData[1].flDist)
					{
						if (flNearest < ReverbData[0].flDist)
						{
							ReverbData[1] = ReverbData[0];
							ReverbData[0].flDist = flNearest;
							ReverbData[0].lApertureNum = lApertureNum;
							ReverbData[0].lEnvID = i;
						}
						else
						{
							ReverbData[1].flDist = flNearest;
							ReverbData[1].lApertureNum = lApertureNum;
							ReverbData[1].lEnvID = i;
						}
					}
				}
			}

		}

#ifdef DISPLAY_CLOSEST_ENVS
		char szEnvName1[256] = {0};
		char szEnvName2[256] = {0};
		char szEnvName3[256] = {0};

		s_lpEAXManager->GetEnvironmentName(ReverbData[0].lEnvID, szEnvName1, 256);
		s_lpEAXManager->GetEnvironmentName(ReverbData[1].lEnvID, szEnvName2, 256);
		s_lpEAXManager->GetEnvironmentName(ReverbData[2].lEnvID, szEnvName3, 256);

		Com_Printf("Closest zones are %s, %s (Listener in %s)\n", szEnvName1,
			szEnvName2, szEnvName3);
#endif

		// Mute any reverbs no longer required ...

		for (i = 0; i < s_NumFXSlots; i++)
		{
			if ((s_FXSlotInfo[i].lEnvID != -1) && (s_FXSlotInfo[i].lEnvID != ReverbData[0].lEnvID) && (s_FXSlotInfo[i].lEnvID != ReverbData[1].lEnvID)
				&& (s_FXSlotInfo[i].lEnvID != ReverbData[2].lEnvID))
			{
				// This environment is no longer needed

				// Mute it
				lVolume = -10000;
				if (s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXFXSLOT_VOLUME, NULL, &lVolume, sizeof(long))!=AL_NO_ERROR)
					OutputDebugString("Failed to Mute FX Slot\n");

				// If any source is sending to this Slot ID then we need to stop them sending to the slot
				for (j = 1; j < s_numChannels; j++)
				{
					if (s_channels[j].lSlotID == i)
					{
						if (s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_ACTIVEFXSLOTID, s_channels[j].alSource, (void*)&EAX_NULL_GUID, sizeof(GUID))!=AL_NO_ERROR)
						{
							OutputDebugString("Failed to set Source ActiveFXSlotID to NULL\n");
						}

						s_channels[j].lSlotID = -1;
					}
				}

				assert(s_FXSlotInfo[i].lEnvID < s_lNumEnvironments && s_FXSlotInfo[i].lEnvID >= 0);
				if (s_FXSlotInfo[i].lEnvID < s_lNumEnvironments && s_FXSlotInfo[i].lEnvID >= 0)
				{
					s_lpEnvTable[s_FXSlotInfo[i].lEnvID].lFXSlotID = -1;
				}
				s_FXSlotInfo[i].lEnvID = -1;
			}
		}


		// Make sure all the reverbs we want are being rendered, if not, find an empty slot
		// and apply appropriate reverb settings
		for (j = 0; j < 3; j++)
		{
			bFound = false;

			for (i = 0; i < s_NumFXSlots; i++)
			{
				if (s_FXSlotInfo[i].lEnvID == ReverbData[j].lEnvID)
				{
					bFound = true;
					break;
				}
			}

			if (!bFound)
			{
				// Find the first available slot and use that one
				for (i = 0; i < s_NumFXSlots; i++)
				{
					if (s_FXSlotInfo[i].lEnvID == -1)
					{
						// Found slot 

						// load reverb here

						// Retrieve reverb properties from EAX Manager
						if (s_lpEAXManager->GetEnvironmentAttributes(ReverbData[j].lEnvID, &Reverb)==EM_OK)
						{
							// Override Air Absorption HF
							Reverb.flAirAbsorptionHF = 0.0f;

							s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXREVERB_ALLPARAMETERS, NULL, &Reverb, sizeof(EAXREVERBPROPERTIES));

							// See if any Sources are in this environment, if they are, enable their sends
							ch = s_channels + 1;
							for (k = 1; k < s_numChannels; k++, ch++)
							{
								if (ch->fixed_origin)
								{
									// Converting from Quake -> DS3D (for EAGLE) ... swap Y and Z
									EMSourcePoint.fX = ch->origin[0];
									EMSourcePoint.fY = ch->origin[2];
									EMSourcePoint.fZ = ch->origin[1];
								}
								else
								{
									if (ch->entnum == listener_number)
									{
										// Source at same position as listener
										// Probably won't be any Occlusion / Obstruction effect -- unless the listener is underwater
										// Converting from Open AL -> DS3D (for EAGLE) ... invert Z
										EMSourcePoint.fX = listener_pos[0];
										EMSourcePoint.fY = listener_pos[1];
										EMSourcePoint.fZ = -listener_pos[2];
									}
									else
									{
										// Get position of Entity
										// Converting from Quake -> DS3D (for EAGLE) ... swap Y and Z
										EMSourcePoint.fX = loopSounds[ ch->entnum ].origin[0];
										EMSourcePoint.fY = loopSounds[ ch->entnum ].origin[2];
										EMSourcePoint.fZ = loopSounds[ ch->entnum ].origin[1];
									}
								}

								// Get Source Environment point
								if (s_lpEAXManager->GetListenerDynamicAttributes(0, &EMSourcePoint, &lSourceID, 0)!=EM_OK)
									OutputDebugString("Failed to get environment zone for Source\n");

								if (lSourceID == i)
								{
									if (s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_ACTIVEFXSLOTID, ch->alSource, (void*)&(s_FXSlotInfo[i].FXSlotGuid), sizeof(GUID))!=AL_NO_ERROR)
									{
										OutputDebugString("Failed to set Source ActiveFXSlotID to new environment\n");
									}

									ch->lSlotID = i;
								}
							}

							assert(ReverbData[j].lEnvID < s_lNumEnvironments && ReverbData[j].lEnvID >= 0);
							if (ReverbData[j].lEnvID < s_lNumEnvironments && ReverbData[j].lEnvID >= 0)
							{
								s_FXSlotInfo[i].lEnvID = ReverbData[j].lEnvID;
								s_lpEnvTable[ReverbData[j].lEnvID].lFXSlotID = i;
							}

							break;
						}
					}
				}
			}
		}

		// Make sure Primary FX Slot ID is set correctly
		if (s_EnvironmentID != ReverbData[2].lEnvID)
		{
			s_eaxSet(&EAXPROPERTYID_EAX40_Context, EAXCONTEXT_PRIMARYFXSLOTID, NULL, &(s_FXSlotInfo[s_lpEnvTable[ReverbData[2].lEnvID].lFXSlotID].FXSlotGuid), sizeof(GUID));
			s_EnvironmentID = ReverbData[2].lEnvID;
		}

		// Have right reverbs loaded ... now to pan them and adjust volume


		// We need to rotate the vector from the Listener to the reverb Aperture by minus the listener
		// orientation

		// Need dot product of Listener Orientation and the straight ahead vector (0, 0, 1)

		// Since both vectors are already normalized, and two terms cancel out (0's), the angle
		// is the arc cosine of the z component of the Listener Orientation

		float flTheta = (float)acos(ListOri.fZ);

		// If the Listener Orientation is to the left of straight ahead, then invert the angle
		if (ListOri.fX < 0)
			flTheta = -flTheta;

		float flSin = (float)sin(-flTheta);
		float flCos = (float)cos(-flTheta);

		for (i = 0; i < min(s_NumFXSlots,s_lNumEnvironments); i++)
		{
			if (s_FXSlotInfo[i].lEnvID == s_EnvironmentID)
			{
				// Listener's environment

				// Find the closest Aperture in *this* environment

				flNearest = FLT_MAX;
				lApertureNum = 0;	//shut up compile warning

				for (j = 0; j < s_lpEnvTable[s_EnvironmentID].ulNumApertures; j++)
				{
					EMAperture.fX = s_lpEnvTable[s_EnvironmentID].Aperture[j].vCenter[0];
					EMAperture.fY = s_lpEnvTable[s_EnvironmentID].Aperture[j].vCenter[1];
					EMAperture.fZ = s_lpEnvTable[s_EnvironmentID].Aperture[j].vCenter[2];

					flDistance = CalcDistance(EMAperture, ListPos);

					if (flDistance < flNearest)
					{
						flNearest = flDistance;
						lApertureNum = j;
					}
				}

				// Have closest environment, work out pan vector direction

				LR.x = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vCenter[0] - ListPos.fX;
				LR.y = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vCenter[1] - ListPos.fY;
				LR.z = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vCenter[2] - ListPos.fZ;

				Pan.x = (LR.x * flCos) + (LR.z * flSin);
				Pan.y = 0.0f;
				Pan.z = (LR.x * -flSin) + (LR.z * flCos);

				Normalize(&Pan);


				// Adjust magnitude ...

				// Magnitude is based on the angle subtended by the aperture, so compute the angle between
				// the vector from the Listener to Pos1 of the aperture, and the vector from the
				// Listener to Pos2 of the aperture.

				
				LP1.x = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vPos1[0] - ListPos.fX;
				LP1.y = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vPos1[1] - ListPos.fY;
				LP1.z = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vPos1[2] - ListPos.fZ;

				LP2.x = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vPos2[0] - ListPos.fX;
				LP2.y = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vPos2[1] - ListPos.fY;
				LP2.z = s_lpEnvTable[s_EnvironmentID].Aperture[lApertureNum].vPos2[2] - ListPos.fZ;

				Normalize(&LP1);
				Normalize(&LP2);

				float flGamma = acos((LP1.x * LP2.x) + (LP1.y * LP2.y) + (LP1.z * LP2.z));

				// We want opposite magnitude (because we are 'in' this environment)
				float flMagnitude = 1.0f - ((2.0f * (float)sin(flGamma/2.0f)) / flGamma);

				// Negative (because pan should be 180 degrees)
				Pan.x *= -flMagnitude;
				Pan.y *= -flMagnitude;
				Pan.z *= -flMagnitude;

				if (s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXREVERB_REVERBPAN, NULL, &Pan, sizeof(EAXVECTOR))!=AL_NO_ERROR)
					OutputDebugString("Failed to set Listener Reverb Pan\n");

				if (s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXREVERB_REFLECTIONSPAN, NULL, &Pan, sizeof(EAXVECTOR))!=AL_NO_ERROR)
					OutputDebugString("Failed to set Listener Reflections Pan\n");
			}
			else
			{
				// Find out which Reverb this is
				if (ReverbData[0].lEnvID == s_FXSlotInfo[i].lEnvID)
					k = 0;
				else
					k = 1;

				LR.x = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vCenter[0] - ListPos.fX;
				LR.y = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vCenter[1] - ListPos.fY;
				LR.z = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vCenter[2] - ListPos.fZ;

				// Rotate the vector

				Pan.x = (LR.x * flCos) + (LR.z * flSin);
				Pan.y = 0.0f;
				Pan.z = (LR.x * -flSin) + (LR.z * flCos);

				Normalize(&Pan);

				// Adjust magnitude ...

				// Magnitude is based on the angle subtended by the aperture, so compute the angle between
				// the vector from the Listener to Pos1 of the aperture, and the vector from the
				// Listener to Pos2 of the aperture.

				
				LP1.x = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vPos1[0] - ListPos.fX;
				LP1.y = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vPos1[1] - ListPos.fY;
				LP1.z = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vPos1[2] - ListPos.fZ;

				LP2.x = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vPos2[0] - ListPos.fX;
				LP2.y = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vPos2[1] - ListPos.fY;
				LP2.z = s_lpEnvTable[ReverbData[k].lEnvID].Aperture[ReverbData[k].lApertureNum].vPos2[2] - ListPos.fZ;

				Normalize(&LP1);
				Normalize(&LP2);

				float flGamma = acos((LP1.x * LP2.x) + (LP1.y * LP2.y) + (LP1.z * LP2.z));
				float flMagnitude = (2.0f * (float)sin(flGamma/2.0f)) / flGamma;

				Pan.x *= flMagnitude;
				Pan.y *= flMagnitude;
				Pan.z *= flMagnitude;

				if (s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXREVERB_REVERBPAN, NULL, &Pan, sizeof(EAXVECTOR))!=AL_NO_ERROR)
					OutputDebugString("Failed to set Reverb Pan\n");

				if (s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXREVERB_REFLECTIONSPAN, NULL, &Pan, sizeof(EAXVECTOR))!=AL_NO_ERROR)
					OutputDebugString("Failed to set Reflections Pan\n");
			}
		}

		lVolume = 0;
		for (i = 0; i < s_NumFXSlots; i++)
		{
			if (s_eaxSet(&s_FXSlotInfo[i].FXSlotGuid, EAXFXSLOT_VOLUME, NULL, &lVolume, sizeof(long))!=AL_NO_ERROR)
				OutputDebugString("Failed to set FX Slot Volume to 0\n");
		}
	}

	return;
}

/*
	Updates the EAX Buffer related effects on the given Source
*/
static void UpdateEAXBuffer(channel_t *ch)
{
	HRESULT hr;
	EMPOINT EMSourcePoint;
	EMPOINT EMVirtualSourcePoint;
	EAXOBSTRUCTIONPROPERTIES eaxOBProp;
	EAXOCCLUSIONPROPERTIES eaxOCProp;
	int i;
	long lSourceID;

	// If EAX Manager is not initialized, or there is no EAX support, or the listener
	// is underwater, return
	if ((!s_lpEAXManager) || (!s_bEAX) || (s_bInWater))
		return;
	
	// Set Occlusion Direct Ratio to the default value (it won't get set by the current version of
	// EAX Manager)
	eaxOCProp.flOcclusionDirectRatio = EAXSOURCE_DEFAULTOCCLUSIONDIRECTRATIO;

	// Convert Source co-ordinate to left-handed system
	if (ch->fixed_origin)
	{
		// Converting from Quake -> DS3D (for EAGLE) ... swap Y and Z
		EMSourcePoint.fX = ch->origin[0];
		EMSourcePoint.fY = ch->origin[2];
		EMSourcePoint.fZ = ch->origin[1];
	}
	else
	{
		if (ch->entnum == listener_number)
		{
			// Source at same position as listener
			// Probably won't be any Occlusion / Obstruction effect -- unless the listener is underwater
			// Converting from Open AL -> DS3D (for EAGLE) ... invert Z
			EMSourcePoint.fX = listener_pos[0];
			EMSourcePoint.fY = listener_pos[1];
			EMSourcePoint.fZ = -listener_pos[2];
		}
		else
		{
			// Get position of Entity
			// Converting from Quake -> DS3D (for EAGLE) ... swap Y and Z
			if (ch->bLooping)
			{
				EMSourcePoint.fX = loopSounds[ ch->entnum ].origin[0];
				EMSourcePoint.fY = loopSounds[ ch->entnum ].origin[2];
				EMSourcePoint.fZ = loopSounds[ ch->entnum ].origin[1];
			}
			else
			{
				EMSourcePoint.fX = s_entityPosition[ch->entnum][0];
				EMSourcePoint.fY = s_entityPosition[ch->entnum][2];
				EMSourcePoint.fZ = s_entityPosition[ch->entnum][1];
			}
		}
	}

	long lExclusion;

	// Just determine what environment the source is in
	if (s_lpEAXManager->GetListenerDynamicAttributes(0, &EMSourcePoint, &lSourceID, 0)==EM_OK)
	{
		// See if a Slot is rendering this environment
		for (i = 0; i < s_NumFXSlots; i++)
		{
			if (s_FXSlotInfo[i].lEnvID == lSourceID)
			{
				// If the Source is not sending to this slot, then enable the send now
				if (ch->lSlotID != i)
				{
					// Set this
					if (s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_ACTIVEFXSLOTID, ch->alSource, &s_FXSlotInfo[i].FXSlotGuid, sizeof(GUID))!=AL_NO_ERROR)
						OutputDebugString("UpdateEAXBuffer = failed to set ActiveFXSlotID\n");

					ch->lSlotID = i;
				}

				break;
			}
		}
	}
	else
	{
		OutputDebugString("UpdateEAXBuffer::Failed to get Source environment zone\n");
	}

	// Add some Exclusion to sounds that are not located in the Listener's environment
	if (s_FXSlotInfo[ch->lSlotID].lEnvID == s_EnvironmentID)
	{
		lExclusion = 0;
		if (s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_EXCLUSION, ch->alSource, &lExclusion, sizeof(long))!=AL_NO_ERROR)
			OutputDebugString("UpdateEAXBuffer : Failed to set exclusion to 0\n");
	}
	else
	{
		lExclusion = -1000;
		if (s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_EXCLUSION, ch->alSource, &lExclusion, sizeof(long))!=AL_NO_ERROR)
			OutputDebugString("UpdateEAXBuffer : Failed to set exclusion to -1000\n");
	}

	if ((ch->entchannel == CHAN_VOICE) || (ch->entchannel == CHAN_VOICE_ATTEN) || (ch->entchannel == CHAN_VOICE_GLOBAL))
	{
		// Remove any Occlusion + Obstruction
		eaxOBProp.lObstruction = EAXSOURCE_DEFAULTOBSTRUCTION;
		eaxOBProp.flObstructionLFRatio = EAXSOURCE_DEFAULTOBSTRUCTIONLFRATIO;

		eaxOCProp.lOcclusion = EAXSOURCE_DEFAULTOCCLUSION;
		eaxOCProp.flOcclusionLFRatio = EAXSOURCE_DEFAULTOCCLUSIONLFRATIO;
		eaxOCProp.flOcclusionRoomRatio = EAXSOURCE_DEFAULTOCCLUSIONROOMRATIO;
		eaxOCProp.flOcclusionDirectRatio = EAXSOURCE_DEFAULTOCCLUSIONDIRECTRATIO;

		s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_OBSTRUCTIONPARAMETERS,
			ch->alSource, &eaxOBProp, sizeof(EAXOBSTRUCTIONPROPERTIES));

		s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_OCCLUSIONPARAMETERS,
			ch->alSource, &eaxOCProp, sizeof(EAXOCCLUSIONPROPERTIES));
	}
	else
	{
		// Check for Occlusion + Obstruction
		hr = s_lpEAXManager->GetSourceDynamicAttributes(0, &EMSourcePoint, &eaxOBProp.lObstruction, &eaxOBProp.flObstructionLFRatio,
			&eaxOCProp.lOcclusion, &eaxOCProp.flOcclusionLFRatio, &eaxOCProp.flOcclusionRoomRatio, &EMVirtualSourcePoint, 0);
		if (hr == EM_OK)
		{	
			// Set EAX effect !
			s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_OBSTRUCTIONPARAMETERS,
				ch->alSource, &eaxOBProp, sizeof(EAXOBSTRUCTIONPROPERTIES));

			s_eaxSet(&EAXPROPERTYID_EAX40_Source, EAXSOURCE_OCCLUSIONPARAMETERS,
				ch->alSource, &eaxOCProp, sizeof(EAXOCCLUSIONPROPERTIES));
		}
	}

	return;
}

float CalcDistance(EMPOINT A, EMPOINT B)
{
	return (float)sqrt(sqr(A.fX - B.fX)+sqr(A.fY - B.fY) + sqr(A.fZ - B.fZ));
}
#endif // HAVE_EAX