/***************************************************************************
spu.c  -  description
-------------------
begin                : Wed May 15 2002
copyright            : (C) 2002 by Pete Bernert
email                : BlackDove@addcom.de
***************************************************************************/
/***************************************************************************
*                                                                         *
*   This program is free software; you can redistribute it and/or modify  *
*   it under the terms of the GNU General Public License as published by  *
*   the Free Software Foundation; either version 2 of the License, or     *
*   (at your option) any later version. See also the license.txt file for *
*   additional informations.                                              *
*                                                                         *
***************************************************************************/

#include "stdafx.h"
#define _WINDOWS
#define _IN_SPU

#include "externals.h"
#include "cfg.h"
#include "dsoundoss.h"
#include "regs.h"

#ifdef _WINDOWS
#include "debug.h"
#include "record.h"
#endif

#ifdef ENABLE_NLS
#include <libintl.h>
#include <locale.h>
#define _(x)  gettext(x)
#define N_(x) (x)
#else
#define _(x)  (x)
#define N_(x) (x)
#endif

// globals

// psx buffer / addresses

unsigned short  regArea[10000];
unsigned short  spuMem[256*1024];
unsigned char * spuMemC;
unsigned char * pSpuIrq=0;
unsigned char * pSpuBuffer;
unsigned char * pMixIrq=0;

// user settings

int             iVolume=3;
int             iXAPitch=1;
int             iUseTimer=2;
int             iSPUIRQWait=1;
int             iRecordMode=0;
int             iUseReverb=2;
int             iUseInterpolation=2;
int             iDisStereo=0;

// MAIN infos struct for each channel

SPUCHAN         s_chan[MAXCHAN+1];                     // channel + 1 infos (1 is security for fmod handling)
REVERBInfo      rvb;

unsigned long   dwNoiseVal=1;                          // global noise generator
int             iSpuAsyncWait=0;

unsigned short  spuCtrl=0;                             // some vars to store psx reg infos
unsigned short  spuStat=0;
unsigned short  spuIrq=0;
unsigned long   spuAddr=0xffffffff;                    // address into spu mem
int             bEndThread=0;                          // thread handlers
int             bThreadEnded=0;
int             bSpuInit=0;
int             bSPUIsOpen=0;

int iSPUDebugMode=0;

#ifdef _WINDOWS
HWND    hWMain=0;                                      // window handle
HWND    hWDebug=0;
HWND    hWRecord=0;
static HANDLE   hMainThread;                           
#else
static pthread_t thread = (pthread_t)-1;               // thread id (linux)
#endif

unsigned long dwNewChannel=0;                          // flags for faster testing, if new channel starts

void (CALLBACK *irqCallback)(void)=0;                  // func of main emu, called on spu irq
void (CALLBACK *cddavCallback)(unsigned short,unsigned short)=0;

// certain globals (were local before, but with the new timeproc I need em global)

static const int f[5][2] = {   {    0,  0  },
{   60,  0  },
{  115, -52 },
{   98, -55 },
{  122, -60 } };
int SSumR[NSSIZE];
int SSumL[NSSIZE];
int iFMod[NSSIZE];
int iCycle = 0;
short * pS;

int lastch=-1;             // last channel processed on spu irq in timer mode
static int lastns=0;       // last ns pos
static int iSecureStart=0; // secure start counter

////////////////////////////////////////////////////////////////////////
// CODE AREA
////////////////////////////////////////////////////////////////////////

// dirty inline func includes

#include "reverb.c"
#include "adsr.c"


/*** 
FAST FIX
***/
#define SPUinit	PEOPS_SPUinit
#define SPUshutdown	PEOPS_SPUshutdown
#define SPUopen	PEOPS_SPUopen
#define SPUclose	PEOPS_SPUclose
#define SPUsetConfigFile	PEOPS_SPUsetConfigFile
void SPUsetConfigFile(char * pCfg){};
#define SPUabout	PEOPS_SPUabout
#define SPUtest	PEOPS_SPUtest
#define SPUwriteRegister	PEOPS_SPUwriteRegister
#define SPUreadRegister	PEOPS_SPUreadRegister
#define SPUwriteDMA	PEOPS_SPUwriteDMA
#define SPUreadDMA	PEOPS_SPUreadDMA
#define SPUwriteDMAMem	PEOPS_SPUwriteDMAMem
#define SPUreadDMAMem	PEOPS_SPUreadDMAMem
#define SPUplayADPCMchannel	PEOPS_SPUplayADPCMchannel
#define SPUfreeze	PEOPS_SPUfreeze
#define SPUregisterCallback	PEOPS_SPUregisterCallback
#define SPUregisterCDDAVolume	PEOPS_SPUregisterCDDAVolume
#define SPUasync	PEOPS_SPUasync

////////////////////////////////////////////////////////////////////////
// helpers for simple interpolation

//
// easy interpolation on upsampling, no special filter, just "Pete's common sense" tm
//
// instead of having n equal sample values in a row like:
//       ____
//           |____
//
// we compare the current delta change with the next delta change.
//
// if curr_delta is positive,
//
//  - and next delta is smaller (or changing direction):
//         \.
//          -__
//
//  - and next delta significant (at least twice) bigger:
//         --_
//            \.
//
//  - and next delta is nearly same:
//          \.
//           \.
//
//
// if curr_delta is negative,
//
//  - and next delta is smaller (or changing direction):
//          _--
//         /
//
//  - and next delta significant (at least twice) bigger:
//            /
//         __- 
//
//  - and next delta is nearly same:
//           /
//          /
//


INLINE void InterpolateUp(int ch)
{
	if(s_chan[ch].SB[32]==1)                              // flag == 1? calc step and set flag... and don't change the value in this pass
	{
		const int id1=s_chan[ch].SB[30]-s_chan[ch].SB[29];  // curr delta to next val
		const int id2=s_chan[ch].SB[31]-s_chan[ch].SB[30];  // and next delta to next-next val :)

		s_chan[ch].SB[32]=0;

		if(id1>0)                                           // curr delta positive
		{
			if(id2<id1)
			{s_chan[ch].SB[28]=id1;s_chan[ch].SB[32]=2;}
			else
				if(id2<(id1<<1))
					s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x10000L;
				else
					s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x20000L; 
		}
		else                                                // curr delta negative
		{
			if(id2>id1)
			{s_chan[ch].SB[28]=id1;s_chan[ch].SB[32]=2;}
			else
				if(id2>(id1<<1))
					s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x10000L;
				else
					s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x20000L; 
		}
	}
	else
		if(s_chan[ch].SB[32]==2)                              // flag 1: calc step and set flag... and don't change the value in this pass
		{
			s_chan[ch].SB[32]=0;

			s_chan[ch].SB[28]=(s_chan[ch].SB[28]*s_chan[ch].sinc)/0x20000L;
			if(s_chan[ch].sinc<=0x8000)
				s_chan[ch].SB[29]=s_chan[ch].SB[30]-(s_chan[ch].SB[28]*((0x10000/s_chan[ch].sinc)-1));
			else s_chan[ch].SB[29]+=s_chan[ch].SB[28];
		}
		else                                                  // no flags? add bigger val (if possible), calc smaller step, set flag1
			s_chan[ch].SB[29]+=s_chan[ch].SB[28];
}

//
// even easier interpolation on downsampling, also no special filter, again just "Pete's common sense" tm
//

INLINE void InterpolateDown(int ch)
{
	if(s_chan[ch].sinc>=0x20000L)                                 // we would skip at least one val?
	{
		s_chan[ch].SB[29]+=(s_chan[ch].SB[30]-s_chan[ch].SB[29])/2; // add easy weight
		if(s_chan[ch].sinc>=0x30000L)                               // we would skip even more vals?
			s_chan[ch].SB[29]+=(s_chan[ch].SB[31]-s_chan[ch].SB[30])/2;// add additional next weight
	}
}

////////////////////////////////////////////////////////////////////////
// helpers for gauss interpolation

#define gval0 (((short*)(&s_chan[ch].SB[29]))[gpos])
#define gval(x) (((short*)(&s_chan[ch].SB[29]))[(gpos+x)&3])

#include "gauss_i.h"

////////////////////////////////////////////////////////////////////////

#include "xa.c"

////////////////////////////////////////////////////////////////////////
// START SOUND... called by main thread to setup a new sound on a channel
////////////////////////////////////////////////////////////////////////

INLINE void StartSound(int ch)
{
	StartADSR(ch);
	StartREVERB(ch);

	s_chan[ch].pCurr=s_chan[ch].pStart;                   // set sample start

	s_chan[ch].s_1=0;                                     // init mixing vars
	s_chan[ch].s_2=0;
	s_chan[ch].iSBPos=28;

	s_chan[ch].bNew=0;                                    // init channel flags
	s_chan[ch].bStop=0;
	s_chan[ch].bOn=1;

	s_chan[ch].SB[29]=0;                                  // init our interpolation helpers
	s_chan[ch].SB[30]=0;

	if(iUseInterpolation>=2)                              // gauss interpolation?
	{s_chan[ch].spos=0x30000L;s_chan[ch].SB[28]=0;}  // -> start with more decoding
	else {s_chan[ch].spos=0x10000L;s_chan[ch].SB[31]=0;}  // -> no/simple interpolation starts with one 44100 decoding

	dwNewChannel&=~(1<<ch);                               // clear new channel bit
}

////////////////////////////////////////////////////////////////////////
// ALL KIND OF HELPERS
////////////////////////////////////////////////////////////////////////

INLINE void VoiceChangeFrequency(int ch)
{
	s_chan[ch].iUsedFreq=s_chan[ch].iActFreq;             // -> take it and calc steps
	s_chan[ch].sinc=s_chan[ch].iRawPitch<<4;
	if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
	if(iUseInterpolation==1) s_chan[ch].SB[32]=1;         // -> freq change in simle imterpolation mode: set flag
}

////////////////////////////////////////////////////////////////////////

INLINE void FModChangeFrequency(int ch,int ns)
{
	int NP=s_chan[ch].iRawPitch;

	NP=((32768L+iFMod[ns])*NP)/32768L;

	if(NP>0x3fff) NP=0x3fff;
	if(NP<0x1)    NP=0x1;

	NP=(44100L*NP)/(4096L);                               // calc frequency

	s_chan[ch].iActFreq=NP;
	s_chan[ch].iUsedFreq=NP;
	s_chan[ch].sinc=(((NP/10)<<16)/4410);
	if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
	if(iUseInterpolation==1)                              // freq change in simple interpolation mode
		s_chan[ch].SB[32]=1;
	iFMod[ns]=0;
}                    

////////////////////////////////////////////////////////////////////////

// noise handler... just produces some noise data
// surely wrong... and no noise frequency (spuCtrl&0x3f00) will be used...
// and sometimes the noise will be used as fmod modulation... pfff

INLINE int iGetNoiseVal(int ch)
{
	int fa;

	if((dwNoiseVal<<=1)&0x80000000L)
	{
		dwNoiseVal^=0x0040001L;
		fa=((dwNoiseVal>>2)&0x7fff);
		fa=-fa;
	}
	else fa=(dwNoiseVal>>2)&0x7fff;

	// mmm... depending on the noise freq we allow bigger/smaller changes to the previous val
	fa=s_chan[ch].iOldNoise+((fa-s_chan[ch].iOldNoise)/((0x001f-((spuCtrl&0x3f00)>>9))+1));
	if(fa>32767L)  fa=32767L;
	if(fa<-32767L) fa=-32767L;              
	s_chan[ch].iOldNoise=fa;

	if(iUseInterpolation<2)                               // no gauss/cubic interpolation?
		s_chan[ch].SB[29] = fa;                               // -> store noise val in "current sample" slot
	return fa;
}                                 

////////////////////////////////////////////////////////////////////////

INLINE void StoreInterpolationVal(int ch,int fa)
{
	if(s_chan[ch].bFMod==2)                               // fmod freq channel
		s_chan[ch].SB[29]=fa;
	else
	{
		if((spuCtrl&0x4000)==0) fa=0;                       // muted?
		else                                                // else adjust
		{
			if(fa>32767L)  fa=32767L;
			if(fa<-32767L) fa=-32767L;              
		}

		if(iUseInterpolation>=2)                            // gauss/cubic interpolation
		{     
			int gpos = s_chan[ch].SB[28];
			gval0 = fa;          
			gpos = (gpos+1) & 3;
			s_chan[ch].SB[28] = gpos;
		}
		else
			if(iUseInterpolation==1)                            // simple interpolation
			{
				s_chan[ch].SB[28] = 0;                    
				s_chan[ch].SB[29] = s_chan[ch].SB[30];            // -> helpers for simple linear interpolation: delay real val for two slots, and calc the two deltas, for a 'look at the future behaviour'
				s_chan[ch].SB[30] = s_chan[ch].SB[31];
				s_chan[ch].SB[31] = fa;
				s_chan[ch].SB[32] = 1;                            // -> flag: calc new interolation
			}
			else s_chan[ch].SB[29]=fa;                          // no interpolation
	}
}

////////////////////////////////////////////////////////////////////////

INLINE int iGetInterpolationVal(int ch)
{
	int fa;

	if(s_chan[ch].bFMod==2) return s_chan[ch].SB[29];

	switch(iUseInterpolation)
	{   
		//--------------------------------------------------//
	case 3:                                             // cubic interpolation
		{
			long xd;int gpos;
			xd = ((s_chan[ch].spos) >> 1)+1;
			gpos = s_chan[ch].SB[28];

			fa  = gval(3) - 3*gval(2) + 3*gval(1) - gval0;
			fa *= (xd - (2<<15)) / 6;
			fa >>= 15;
			fa += gval(2) - gval(1) - gval(1) + gval0;
			fa *= (xd - (1<<15)) >> 1;
			fa >>= 15;
			fa += gval(1) - gval0;
			fa *= xd;
			fa >>= 15;
			fa = fa + gval0;

		} break;
		//--------------------------------------------------//
	case 2:                                             // gauss interpolation
		{
			int vl, vr;int gpos;
			vl = (s_chan[ch].spos >> 6) & ~3;
			gpos = s_chan[ch].SB[28];
			vr=(gauss[vl]*gval0)&~2047;
			vr+=(gauss[vl+1]*gval(1))&~2047;
			vr+=(gauss[vl+2]*gval(2))&~2047;
			vr+=(gauss[vl+3]*gval(3))&~2047;
			fa = vr>>11;
		} break;
		//--------------------------------------------------//
	case 1:                                             // simple interpolation
		{
			if(s_chan[ch].sinc<0x10000L)                      // -> upsampling?
				InterpolateUp(ch);                           // --> interpolate up
			else InterpolateDown(ch);                         // --> else down
			fa=s_chan[ch].SB[29];
		} break;
		//--------------------------------------------------//
	default:                                            // no interpolation
		{
			fa=s_chan[ch].SB[29];                  
		} break;
		//--------------------------------------------------//
	}

	return fa;
}

////////////////////////////////////////////////////////////////////////
// MAIN SPU FUNCTION
// here is the main job handler... thread, timer or direct func call
// basically the whole sound processing is done in this fat func!
////////////////////////////////////////////////////////////////////////

// 5 ms waiting phase, if buffer is full and no new sound has to get started
// .. can be made smaller (smallest val: 1 ms), but bigger waits give
// better performance

#define PAUSE_W 5
#define PAUSE_L 5000

////////////////////////////////////////////////////////////////////////

#ifdef _WINDOWS
static VOID CALLBACK MAINProc(UINT nTimerId, UINT msg, DWORD dwUser, DWORD dwParam1, DWORD dwParam2)
#else
static void *MAINThread(void *arg)
#endif
{
	int s_1,s_2,fa,ns;
#ifndef _MACOSX
	int voldiv = iVolume;
#else
	const int voldiv = 2;
#endif
	unsigned char * start;unsigned int nSample;
	int ch,predict_nr,shift_factor,flags,d,s;
	int bIRQReturn=0;

	while(!bEndThread)                                    // until we are shutting down
	{
		// ok, at the beginning we are looking if there is
		// enuff free place in the dsound/oss buffer to
		// fill in new data, or if there is a new channel to start.
		// if not, we wait (thread) or return (timer/spuasync)
		// until enuff free place is available/a new channel gets
		// started

		if(dwNewChannel)                                    // new channel should start immedately?
		{                                                  // (at least one bit 0 ... MAXCHANNEL is set?)
			iSecureStart++;                                   // -> set iSecure
			if(iSecureStart>5) iSecureStart=0;                //    (if it is set 5 times - that means on 5 tries a new samples has been started - in a row, we will reset it, to give the sound update a chance)
		}
		else iSecureStart=0;                                // 0: no new channel should start

		while(!iSecureStart && !bEndThread &&               // no new start? no thread end?
			(SoundGetBytesBuffered()>TESTSIZE))           // and still enuff data in sound buffer?
		{
			iSecureStart=0;                                   // reset secure

#ifdef _WINDOWS
			/*
			if(iUseTimer)                                     // no-thread mode?
			{
			if(iUseTimer==1)                                // -> ok, timer mode 1: setup a oneshot timer of x ms to wait
			timeSetEvent(PAUSE_W,1,MAINProc,0,TIME_ONESHOT);
			return;                                         // -> and done this time (timer mode 1 or 2)
			}
			*/                                                  // win thread mode:
			Sleep(PAUSE_W);                                   // sleep for x ms (win)
#else
			if(iUseTimer) return 0;                           // linux no-thread mode? bye
			usleep(PAUSE_L);                                  // else sleep for x ms (linux)
#endif

			if(dwNewChannel) iSecureStart=1;                  // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop
		}

		//--------------------------------------------------// continue from irq handling in timer mode? 

		if(lastch>=0)                                       // will be -1 if no continue is pending
		{
			ch=lastch; ns=lastns; lastch=-1;                  // -> setup all kind of vars to continue
			goto GOON;                                        // -> directly jump to the continue point
		}

		//--------------------------------------------------//
		//- main channel loop                              -// 
		//--------------------------------------------------//
		{
			for(ch=0;ch<MAXCHAN;ch++)                         // loop em all... we will collect 1 ms of sound of each playing channel
			{
				if(s_chan[ch].bNew) StartSound(ch);             // start new sound
				if(!s_chan[ch].bOn) continue;                   // channel not playing? next

				if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq)   // new psx frequency?
					VoiceChangeFrequency(ch);

				ns=0;
				while(ns<NSSIZE)                                // loop until 1 ms of data is reached
				{
					if(s_chan[ch].bFMod==1 && iFMod[ns])          // fmod freq channel
						FModChangeFrequency(ch,ns);

					while(s_chan[ch].spos>=0x10000L)
					{
						if(s_chan[ch].iSBPos==28)                   // 28 reached?
						{
							start=s_chan[ch].pCurr;                   // set up the current pos

							if (start == (unsigned char*)-1)          // special "stop" sign
							{
								s_chan[ch].bOn=0;                       // -> turn everything off
								s_chan[ch].ADSRX.lVolume=0;
								s_chan[ch].ADSRX.EnvelopeVol=0;
								goto ENDX;                              // -> and done for this channel
							}

							s_chan[ch].iSBPos=0;

							//////////////////////////////////////////// spu irq handler here? mmm... do it later

							s_1=s_chan[ch].s_1;
							s_2=s_chan[ch].s_2;

							predict_nr=(int)*start;start++;
							shift_factor=predict_nr&0xf;
							predict_nr >>= 4;
							flags=(int)*start;start++;

							// -------------------------------------- // 

							for (nSample=0;nSample<28;start++)      
							{
								d=(int)*start;
								s=((d&0xf)<<12);
								if(s&0x8000) s|=0xffff0000;

								fa=(s >> shift_factor);
								fa=fa + ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6);
								s_2=s_1;s_1=fa;
								s=((d & 0xf0) << 8);

								s_chan[ch].SB[nSample++]=fa;

								if(s&0x8000) s|=0xffff0000;
								fa=(s>>shift_factor);
								fa=fa + ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6);
								s_2=s_1;s_1=fa;

								s_chan[ch].SB[nSample++]=fa;
							}

							//////////////////////////////////////////// irq check

							if(irqCallback && (spuCtrl&0x40))         // some callback and irq active?
							{
								if((pSpuIrq >  start-16 &&              // irq address reached?
									pSpuIrq <= start) ||
									((flags&1) &&                        // special: irq on looping addr, when stop/loop flag is set 
									(pSpuIrq >  s_chan[ch].pLoop-16 &&
									pSpuIrq <= s_chan[ch].pLoop)))
								{
									s_chan[ch].iIrqDone=1;                // -> debug flag
									irqCallback();                        // -> call main emu

									if(iSPUIRQWait)                       // -> option: wait after irq for main emu
									{
										iSpuAsyncWait=1;
										bIRQReturn=1;
									}
								}
							}

							//////////////////////////////////////////// flag handler

							if((flags&4) && (!s_chan[ch].bIgnoreLoop))
								s_chan[ch].pLoop=start-16;               // loop adress

							if(flags&1)                               // 1: stop/loop
							{
								// We play this block out first...
								//if(!(flags&2))                          // 1+2: do loop... otherwise: stop
								if(flags!=3 || s_chan[ch].pLoop==NULL)  // PETE: if we don't check exactly for 3, loop hang ups will happen (DQ4, for example)
								{                                      // and checking if pLoop is set avoids crashes, yeah
									start = (unsigned char*)-1;
								}
								else
								{
									start = s_chan[ch].pLoop;
								}
							}

							s_chan[ch].pCurr=start;                   // store values for next cycle
							s_chan[ch].s_1=s_1;
							s_chan[ch].s_2=s_2;

							if(bIRQReturn)                            // special return for "spu irq - wait for cpu action"
							{
								bIRQReturn=0;
								if(iUseTimer!=2)
								{ 
									DWORD dwWatchTime=timeGetTime_spu()+2500;

									while(iSpuAsyncWait && !bEndThread && 
										timeGetTime_spu()<dwWatchTime)
#ifdef _WINDOWS
										Sleep(1);
#else
										usleep(1000L);
#endif
								}
								else
								{
									lastch=ch; 
									lastns=ns;

#ifdef _WINDOWS
									return;
#else
									return 0;
#endif
								}
							}

GOON: ;
						}

						fa=s_chan[ch].SB[s_chan[ch].iSBPos++];      // get sample data

						StoreInterpolationVal(ch,fa);               // store val for later interpolation

						s_chan[ch].spos -= 0x10000L;
					}

					if(s_chan[ch].bNoise)
						fa=iGetNoiseVal(ch);                     // get noise val
					else fa=iGetInterpolationVal(ch);             // get sample val

					s_chan[ch].sval = (MixADSR(ch) * fa) / 1023;  // mix adsr

					if(s_chan[ch].bFMod==2)                       // fmod freq channel
						iFMod[ns]=s_chan[ch].sval;                   // -> store 1T sample data, use that to do fmod on next channel
					else                                          // no fmod freq channel
					{
						//////////////////////////////////////////////
						// ok, left/right sound volume (psx volume goes from 0 ... 0x3fff)

						if(s_chan[ch].iMute) 
							s_chan[ch].sval=0;                         // debug mute
						else
						{
							SSumL[ns]+=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000L;
							SSumR[ns]+=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000L;
						}

						//////////////////////////////////////////////
						// now let us store sound data for reverb    

						if(s_chan[ch].bRVBActive) StoreREVERB(ch,ns);
					}

					////////////////////////////////////////////////
					// ok, go on until 1 ms data of this channel is collected

					ns++;
					s_chan[ch].spos += s_chan[ch].sinc;

				}
ENDX:   ;
			}
		}

		//---------------------------------------------------//
		//- here we have another 1 ms of sound data
		//---------------------------------------------------//
		// mix XA infos (if any)

		MixXA();

		///////////////////////////////////////////////////////
		// mix all channels (including reverb) into one buffer

		if(iDisStereo)                                       // no stereo?
		{
			int dl, dr;
			for (ns = 0; ns < NSSIZE; ns++)
			{
				SSumL[ns] += MixREVERBLeft(ns);

				dl = SSumL[ns] / voldiv; SSumL[ns] = 0;
				if (dl < -32767) dl = -32767; if (dl > 32767) dl = 32767;

				SSumR[ns] += MixREVERBRight();

				dr = SSumR[ns] / voldiv; SSumR[ns] = 0;
				if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767;
				*pS++ = (dl + dr) / 2;
			}
		}
		else                                                 // stereo:
			for (ns = 0; ns < NSSIZE; ns++)
			{
				SSumL[ns] += MixREVERBLeft(ns);

				d = SSumL[ns] / voldiv; SSumL[ns] = 0;
				if (d < -32767) d = -32767; if (d > 32767) d = 32767;
				*pS++ = d;

				SSumR[ns] += MixREVERBRight();

				d = SSumR[ns] / voldiv; SSumR[ns] = 0;
				if(d < -32767) d = -32767; if(d > 32767) d = 32767;
				*pS++ = d;
			}

			//////////////////////////////////////////////////////                   
			// special irq handling in the decode buffers (0x0000-0x1000)
			// we know: 
			// the decode buffers are located in spu memory in the following way:
			// 0x0000-0x03ff  CD audio left
			// 0x0400-0x07ff  CD audio right
			// 0x0800-0x0bff  Voice 1
			// 0x0c00-0x0fff  Voice 3
			// and decoded data is 16 bit for one sample
			// we assume: 
			// even if voices 1/3 are off or no cd audio is playing, the internal
			// play positions will move on and wrap after 0x400 bytes.
			// Therefore: we just need a pointer from spumem+0 to spumem+3ff, and 
			// increase this pointer on each sample by 2 bytes. If this pointer
			// (or 0x400 offsets of this pointer) hits the spuirq address, we generate
			// an IRQ. Only problem: the "wait for cpu" option is kinda hard to do here
			// in some of Peops timer modes. So: we ignore this option here (for now).

			if(pMixIrq && irqCallback)
			{
				for(ns=0;ns<NSSIZE;ns++)
				{
					if((spuCtrl&0x40) && pSpuIrq && pSpuIrq<spuMemC+0x1000)                 
					{
						for(ch=0;ch<4;ch++)
						{
							if(pSpuIrq>=pMixIrq+(ch*0x400) && pSpuIrq<pMixIrq+(ch*0x400)+2)
							{irqCallback();s_chan[ch].iIrqDone=1;}
						}
					}
					pMixIrq+=2;if(pMixIrq>spuMemC+0x3ff) pMixIrq=spuMemC;
				}
			}

			InitREVERB();

			// feed the sound
			// wanna have around 1/60 sec (16.666 ms) updates
			if (iCycle++ > 16)
			{
				SoundFeedStreamData((unsigned char *)pSpuBuffer,
					((unsigned char *)pS) - ((unsigned char *)pSpuBuffer));
				pS = (short *)pSpuBuffer;
				iCycle = 0;
			}
	}

	// end of big main loop...

	bThreadEnded = 1;

#ifndef _WINDOWS
	return 0;
#endif
}

////////////////////////////////////////////////////////////////////////
// WINDOWS THREAD... simply calls the timer func and stays forever :)
////////////////////////////////////////////////////////////////////////

#ifdef _WINDOWS

DWORD WINAPI MAINThreadEx(LPVOID lpParameter)
{
	MAINProc(0,0,0,0,0);
	return 0;
}

#endif

// SPU ASYNC... even newer epsxe func
//  1 time every 'cycle' cycles... harhar

void CALLBACK SPUasync(unsigned long cycle)
{
	if(iSpuAsyncWait)
	{
		iSpuAsyncWait++;
		if(iSpuAsyncWait<=64) return;
		iSpuAsyncWait=0;
	}

	if(iUseTimer==2)                                      // special mode, only used in Linux by this spu (or if you enable the experimental Windows mode)
	{
		if(!bSpuInit) return;                               // -> no init, no call

#ifdef _WINDOWS
		MAINProc(0,0,0,0,0);                                // -> experimental win mode... not really tested... don't like the drawbacks
#else
		MAINThread(0);                                      // -> linux high-compat mode
#endif
	}
}

// SPU UPDATE... new epsxe func
//  1 time every 32 hsync lines
//  (312/32)x50 in pal
//  (262/32)x60 in ntsc

// since epsxe 1.5.2 (linux) uses SPUupdate, not SPUasync, I will
// leave that func in the linux port, until epsxe linux is using
// the async function as well

void CALLBACK SPUupdate(void)
{
	SPUasync(0);
}

// XA AUDIO

void CALLBACK SPUplayADPCMchannel(xa_decode_t *xap)
{
	if(!xap)       return;
	if(!xap->freq) return;                                // no xa freq ? bye

	FeedXA(xap);                                          // call main XA feeder
}

// CDDA AUDIO
void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
{
	if (!pcm)      return;
	if (nbytes<=0) return;

	FeedCDDA((unsigned char *)pcm, nbytes);
}

// SETUPTIMER: init of certain buffers and threads/timers
void SetupTimer(void)
{
	memset(SSumR,0,NSSIZE*sizeof(int));                   // init some mixing buffers
	memset(SSumL,0,NSSIZE*sizeof(int));
	memset(iFMod,0,NSSIZE*sizeof(int));
	pS=(short *)pSpuBuffer;                               // setup soundbuffer pointer

	bEndThread=0;                                         // init thread vars
	bThreadEnded=0; 
	bSpuInit=1;                                           // flag: we are inited

#ifdef _WINDOWS
	/*
	if(iUseTimer==1)                                      // windows: use timer
	{
	timeBeginPeriod(1);
	timeSetEvent(1,1,MAINProc,0,TIME_ONESHOT);
	}
	else */
	if(iUseTimer==0)                                      // windows: use thread
	{
		//_beginthread(MAINThread,0,NULL);
		DWORD dw;
		hMainThread=CreateThread(NULL,0,MAINThreadEx,0,0,&dw);
		SetThreadPriority(hMainThread,
			//THREAD_PRIORITY_TIME_CRITICAL);
			THREAD_PRIORITY_HIGHEST);
	}

#else

	if(!iUseTimer)                                        // linux: use thread
	{
		pthread_create(&thread, NULL, MAINThread, NULL);
	}

#endif
}

// REMOVETIMER: kill threads/timers
void RemoveTimer(void)
{
	bEndThread=1;                                         // raise flag to end thread

#ifdef _WINDOWS

	if(iUseTimer!=2)                                      // windows thread?
	{
		while(!bThreadEnded) {Sleep(5L);}                   // -> wait till thread has ended
		Sleep(5L);
	}
	/*
	if(iUseTimer==1) timeEndPeriod(1);                    // windows timer? stop it
	*/

#else

	if(!iUseTimer)                                        // linux tread?
	{
		int i=0;
		while(!bThreadEnded && i<2000) {usleep(1000L);i++;} // -> wait until thread has ended
		if(thread!=(pthread_t)-1) {pthread_cancel(thread);thread=(pthread_t)-1;}  // -> cancel thread anyway
	}

#endif

	bThreadEnded=0;                                       // no more spu is running
	bSpuInit=0;
}

// SETUPSTREAMS: init most of the spu buffers
void SetupStreams(void)
{ 
	int i;

	pSpuBuffer=(unsigned char *)malloc(32768);            // alloc mixing buffer

	if(iUseReverb==1) i=88200*2;
	else              i=NSSIZE*2;

	sRVBStart = (int *)malloc(i*4);                       // alloc reverb buffer
	memset(sRVBStart,0,i*4);
	sRVBEnd  = sRVBStart + i;
	sRVBPlay = sRVBStart;

	XAStart =                                             // alloc xa buffer
		(uint32_t *)malloc(44100 * sizeof(uint32_t));
	XAEnd   = XAStart + 44100;
	XAPlay  = XAStart;
	XAFeed  = XAStart;

	CDDAStart =                                           // alloc cdda buffer
		(uint32_t *)malloc(16384 * sizeof(uint32_t));
	CDDAEnd   = CDDAStart + 16384;
	CDDAPlay  = CDDAStart;
	CDDAFeed  = CDDAStart + 1;

	for(i=0;i<MAXCHAN;i++)                                // loop sound channels
	{
		// we don't use mutex sync... not needed, would only 
		// slow us down:
		//   s_chan[i].hMutex=CreateMutex(NULL,FALSE,NULL);
		s_chan[i].ADSRX.SustainLevel = 1024;                // -> init sustain
		s_chan[i].iMute=0;
		s_chan[i].iIrqDone=0;
		s_chan[i].pLoop=spuMemC;
		s_chan[i].pStart=spuMemC;
		s_chan[i].pCurr=spuMemC;
	}

	pMixIrq=spuMemC;                                     // enable decoded buffer irqs by setting the address
}

// REMOVESTREAMS: free most buffer
void RemoveStreams(void)
{ 
	free(pSpuBuffer);                                     // free mixing buffer
	pSpuBuffer = NULL;
	free(sRVBStart);                                      // free reverb buffer
	sRVBStart = NULL;
	free(XAStart);                                        // free XA buffer
	XAStart = NULL;
	free(CDDAStart);                                      // free CDDA buffer
	CDDAStart = NULL;
}

// INIT/EXIT STUFF

// SPUINIT: this func will be called first by the main emu
long CALLBACK SPUinit(void)
{
	spuMemC = (unsigned char *)spuMem;                    // just small setup
	memset((void *)&rvb, 0, sizeof(REVERBInfo));
	InitADSR();

	iVolume = 3;
	iReverbOff = -1;
	spuIrq = 0;
	spuAddr = 0xffffffff;
	bEndThread = 0;
	bThreadEnded = 0;
	spuMemC = (unsigned char *)spuMem;
	pMixIrq = 0;
	memset((void *)s_chan, 0, (MAXCHAN + 1) * sizeof(SPUCHAN));
	pSpuIrq = 0;
	iSPUIRQWait = 1;
	lastch = -1;

	ReadConfig();                                         // read user stuff
	SetupStreams();                                       // prepare streaming

	return 0;
}

// SPUOPEN: called by main emu after init
#ifdef _WINDOWS
long CALLBACK SPUopen(HWND hW)                          
#else
long SPUopen(void)
#endif
{
	if (bSPUIsOpen) return 0;                             // security for some stupid main emus

#ifdef _WINDOWS
	LastWrite=0xffffffff;LastPlay=0;                      // init some play vars
#endif

	SetupSound();                                         // setup sound (before init!)
	SetupTimer();                                         // timer for feeding data

	bSPUIsOpen = 1;

	return PSE_SPU_ERR_SUCCESS;
}

// SPUCLOSE: called before shutdown
long CALLBACK SPUclose(void)
{
	if (!bSPUIsOpen) return 0;                            // some security

	bSPUIsOpen = 0;                                       // no more open

	RemoveTimer();                                        // no more feeding
	RemoveSound();                                        // no more sound handling

	return 0;
}

// SPUSHUTDOWN: called by main emu on final exit
long CALLBACK SPUshutdown(void)
{
	SPUclose();
	RemoveStreams();                                      // no more streaming

	return 0;
}

// SPUTEST: we don't test, we are always fine ;)
long CALLBACK SPUtest(void)
{
	return 0;
}

// SPUCONFIGURE: call config dialog
long CALLBACK SPUconfigure(void)
{
	return 0;
}

// SPUABOUT: show about window
void CALLBACK SPUabout(void)
{

}

// SETUP CALLBACKS
// this functions will be called once, 
// passes a callback that should be called on SPU-IRQ/cdda volume change
void CALLBACK SPUregisterCallback(void (CALLBACK *callback)(void))
{
	irqCallback = callback;
}

void CALLBACK SPUregisterCDDAVolume(void (CALLBACK *CDDAVcallback)(unsigned short,unsigned short))
{
	cddavCallback = CDDAVcallback;
}

char * SPUgetLibInfos(void)
{
	return NULL;
}
