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Cog/Frameworks/AudioOverload/aosdk/eng_psf/peops2/spu.c

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Added AudioOverload plugin.
2009-03-01 03:04:03 -03:00
/***************************************************************************
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. *
* *
***************************************************************************/
//*************************************************************************//
// History of changes:
//
// 2005/08/29 - Pete
// - changed to 48Khz output
//
// 2004/12/25 - Pete
// - inc'd version for pcsx2-0.7
//
// 2004/04/18 - Pete
// - changed all kind of things in the plugin
//
// 2004/04/04 - Pete
// - changed plugin to emulate PS2 spu
//
// 2003/04/07 - Eric
// - adjusted cubic interpolation algorithm
//
// 2003/03/16 - Eric
// - added cubic interpolation
//
// 2003/03/01 - linuzappz
// - libraryName changes using ALSA
//
// 2003/02/28 - Pete
// - added option for type of interpolation
// - adjusted spu irqs again (Thousant Arms, Valkyrie Profile)
// - added MONO support for MSWindows DirectSound
//
// 2003/02/20 - kode54
// - amended interpolation code, goto GOON could skip initialization of gpos and cause segfault
//
// 2003/02/19 - kode54
// - moved SPU IRQ handler and changed sample flag processing
//
// 2003/02/18 - kode54
// - moved ADSR calculation outside of the sample decode loop, somehow I doubt that
// ADSR timing is relative to the frequency at which a sample is played... I guess
// this remains to be seen, and I don't know whether ADSR is applied to noise channels...
//
// 2003/02/09 - kode54
// - one-shot samples now process the end block before stopping
// - in light of removing fmod hack, now processing ADSR on frequency channel as well
//
// 2003/02/08 - kode54
// - replaced easy interpolation with gaussian
// - removed fmod averaging hack
// - changed .sinc to be updated from .iRawPitch, no idea why it wasn't done this way already (<- Pete: because I sometimes fail to see the obvious, haharhar :)
//
// 2003/02/08 - linuzappz
// - small bugfix for one usleep that was 1 instead of 1000
// - added iDisStereo for no stereo (Linux)
//
// 2003/01/22 - Pete
// - added easy interpolation & small noise adjustments
//
// 2003/01/19 - Pete
// - added Neill's reverb
//
// 2003/01/12 - Pete
// - added recording window handlers
//
// 2003/01/06 - Pete
// - added Neill's ADSR timings
//
// 2002/12/28 - Pete
// - adjusted spu irq handling, fmod handling and loop handling
//
// 2002/08/14 - Pete
// - added extra reverb
//
// 2002/06/08 - linuzappz
// - SPUupdate changed for SPUasync
//
// 2002/05/15 - Pete
// - generic cleanup for the Peops release
//
//*************************************************************************//
#include "stdafx.h"
#define _IN_SPU
#include "../peops2/externals.h"
#include "../peops2/regs.h"
#include "../peops2/dma.h"
////////////////////////////////////////////////////////////////////////
// globals
////////////////////////////////////////////////////////////////////////
// psx buffer / addresses
unsigned short regArea[32*1024];
unsigned short spuMem[1*1024*1024];
unsigned char * spuMemC;
unsigned char * pSpuIrq[2];
unsigned char * pSpuBuffer;
// user settings
int iUseXA=0;
int iVolume=3;
int iXAPitch=1;
int iUseTimer=2;
int iSPUIRQWait=1;
int iDebugMode=0;
int iRecordMode=0;
int iUseReverb=1;
int iUseInterpolation=2;
// MAIN infos struct for each channel
SPUCHAN s_chan[MAXCHAN+1]; // channel + 1 infos (1 is security for fmod handling)
REVERBInfo rvb[2];
unsigned long dwNoiseVal=1; // global noise generator
unsigned short spuCtrl2[2]; // some vars to store psx reg infos
unsigned short spuStat2[2];
unsigned long spuIrq2[2];
unsigned long spuAddr2[2]; // address into spu mem
unsigned long spuRvbAddr2[2];
unsigned long spuRvbAEnd2[2];
int bEndThread=0; // thread handlers
int bThreadEnded=0;
int bSpuInit=0;
int bSPUIsOpen=0;
unsigned long dwNewChannel2[2]; // flags for faster testing, if new channel starts
unsigned long dwEndChannel2[2];
// UNUSED IN PS2 YET
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)
const int f[5][2] = { { 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 } };
int SSumR[NSSIZE];
int SSumL[NSSIZE];
int iCycle=0;
short * pS;
static 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
extern void ps2_update(unsigned char *samples, long lBytes);
////////////////////////////////////////////////////////////////////////
// CODE AREA
////////////////////////////////////////////////////////////////////////
// dirty inline func includes
#include "reverb.c"
#include "adsr.c"
////////////////////////////////////////////////////////////////////////
// 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:
// /
// /
//
static 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
//
static 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
////////////////////////////////////////////////////////////////////////
static INLINE void StartSound(int ch)
{
dwNewChannel2[ch/24]&=~(1<<(ch%24)); // clear new channel bit
dwEndChannel2[ch/24]&=~(1<<(ch%24)); // clear end channel bit
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
}
////////////////////////////////////////////////////////////////////////
// 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!
////////////////////////////////////////////////////////////////////////
static u32 sampcount;
static u32 decaybegin;
static u32 decayend;
// Counting to 65536 results in full volume offage.
void setlength2(s32 stop, s32 fade)
{
if(stop==~0)
{
decaybegin=~0;
}
else
{
stop=(stop*441)/10;
fade=(fade*441)/10;
decaybegin=stop;
decayend=stop+fade;
}
}
// 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
////////////////////////////////////////////////////////////////////////
int iSpuAsyncWait=0;
static void *MAINThread(int samp2run)
{
int s_1,s_2,fa,voldiv=iVolume;
unsigned char * start;unsigned int nSample;
int ch,predict_nr,shift_factor,flags,d,d2,s;
int gpos,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(dwNewChannel2[0] || dwNewChannel2[1]) // 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
/* if (!iSecureStart)
{
iSecureStart=0; // reset secure
return;
}*/
#if 0
while(!iSecureStart && !bEndThread) // && // no new start? no thread end?
// (SoundGetBytesBuffered()>TESTSIZE)) // and still enuff data in sound buffer?
{
iSecureStart=0; // reset secure
if(iUseTimer) return 0; // linux no-thread mode? bye
if(dwNewChannel2[0] || dwNewChannel2[1])
iSecureStart=1; // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop
}
#endif
//--------------------------------------------------// continue from irq handling in timer mode?
if(lastch>=0) // will be -1 if no continue is pending
{
ch=lastch; 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?
{
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
}
// ns=0;
// while(ns<NSSIZE) // loop until 1 ms of data is reached
{
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(spuCtrl2[ch/24]&0x40) // some irq active?
{
if((pSpuIrq[ch/24] > start-16 && // irq address reached?
pSpuIrq[ch/24] <= start) ||
((flags&1) && // special: irq on looping addr, when stop/loop flag is set
(pSpuIrq[ch/24] > s_chan[ch].pLoop-16 &&
pSpuIrq[ch/24] <= s_chan[ch].pLoop)))
{
s_chan[ch].iIrqDone=1; // -> debug flag
if(irqCallback) irqCallback(); // -> call main emu (not supported in SPU2 right now)
else
{
if(ch<24) InterruptDMA4(); // -> let's see what is happening if we call our irqs instead ;)
else InterruptDMA7();
}
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
{
dwEndChannel2[ch/24]|=(1<<(ch%24));
// We play this block out first...
//if(!(flags&2)|| s_chan[ch].pLoop==NULL)
// 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;
{
lastch=ch;
// lastns=ns; // changemeback
Updated DUMB
2013-09-28 00:24:23 -03:00
return 0;
Added AudioOverload plugin.
2009-03-01 03:04:03 -03:00
}
}
////////////////////////////////////////////
GOON: ;
}
fa=s_chan[ch].SB[s_chan[ch].iSBPos++]; // get sample data
// if((spuCtrl2[ch/24]&0x4000)==0) fa=0; // muted?
// else // else adjust
{
if(fa>32767L) fa=32767L;
if(fa<-32767L) fa=-32767L;
}
if(iUseInterpolation>=2) // gauss/cubic interpolation
{
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
s_chan[ch].spos -= 0x10000L;
}
////////////////////////////////////////////////
// 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
if(s_chan[ch].bNoise)
{
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-((spuCtrl2[ch/24]&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
} //----------------------------------------
else // NO NOISE (NORMAL SAMPLE DATA) HERE
{//------------------------------------------//
if(iUseInterpolation==3) // cubic interpolation
{
long xd;
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;
}
//------------------------------------------//
else
if(iUseInterpolation==2) // gauss interpolation
{
int vl, vr;
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;
/*
vr=(gauss[vl]*gval0)>>9;
vr+=(gauss[vl+1]*gval(1))>>9;
vr+=(gauss[vl+2]*gval(2))>>9;
vr+=(gauss[vl+3]*gval(3))>>9;
fa = vr>>2;
*/
}
//------------------------------------------//
else
if(iUseInterpolation==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];
}
//------------------------------------------//
else fa=s_chan[ch].SB[29]; // no interpolation
}
s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // add adsr
if(s_chan[ch].bFMod==2) // fmod freq channel
{
int NP=s_chan[ch+1].iRawPitch;
double intr;
NP=((32768L+s_chan[ch].sval)*NP)/32768L; // mmm... I still need to adjust that to 1/48 khz... we will wait for the first game/demo using it to decide how to do it :)
if(NP>0x3fff) NP=0x3fff;
if(NP<0x1) NP=0x1;
intr = (double)48000.0f / (double)44100.0f * (double)NP;
NP = (UINT32)intr;
NP=(44100L*NP)/(4096L); // calc frequency
s_chan[ch+1].iActFreq=NP;
s_chan[ch+1].iUsedFreq=NP;
s_chan[ch+1].sinc=(((NP/10)<<16)/4410);
if(!s_chan[ch+1].sinc) s_chan[ch+1].sinc=1;
if(iUseInterpolation==1) // freq change in sipmle interpolation mode
s_chan[ch+1].SB[32]=1;
// mmmm... set up freq decoding positions?
// s_chan[ch+1].iSBPos=28;
// s_chan[ch+1].spos=0x10000L;
}
else
{
//////////////////////////////////////////////
// ok, left/right sound volume (psx volume goes from 0 ... 0x3fff)
if(s_chan[ch].iMute)
s_chan[ch].sval=0; // debug mute
else
{
if(s_chan[ch].bVolumeL)
SSumL[0]+=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000L;
if(s_chan[ch].bVolumeR)
SSumR[0]+=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000L;
}
//////////////////////////////////////////////
// now let us store sound data for reverb
if(s_chan[ch].bRVBActive) StoreREVERB(ch,0);
}
////////////////////////////////////////////////
// ok, go on until 1 ms data of this channel is collected
s_chan[ch].spos += s_chan[ch].sinc;
}
ENDX: ;
}
}
//---------------------------------------------------//
//- here we have another 1 ms of sound data
//---------------------------------------------------//
///////////////////////////////////////////////////////
// mix all channels (including reverb) into one buffer
SSumL[0]+=MixREVERBLeft(0,0);
SSumL[0]+=MixREVERBLeft(0,1);
SSumR[0]+=MixREVERBRight(0);
SSumR[0]+=MixREVERBRight(1);
d=SSumL[0]/voldiv;SSumL[0]=0;
d2=SSumR[0]/voldiv;SSumR[0]=0;
if(d<-32767) d=-32767;if(d>32767) d=32767;
if(d2<-32767) d2=-32767;if(d2>32767) d2=32767;
if(sampcount>=decaybegin)
{
s32 dmul;
if(decaybegin!=~0) // Is anyone REALLY going to be playing a song
// for 13 hours?
{
if(sampcount>=decayend)
{
// ao_song_done = 1;
return(0);
}
dmul=256-(256*(sampcount-decaybegin)/(decayend-decaybegin));
d=(d*dmul)>>8;
d2=(d2*dmul)>>8;
}
}
sampcount++;
*pS++=d;
*pS++=d2;
InitREVERB();
//////////////////////////////////////////////////////
// feed the sound
// wanna have around 1/60 sec (16.666 ms) updates
if ((((unsigned char *)pS)-((unsigned char *)pSpuBuffer)) == (735*4))
{
ps2_update((u8*)pSpuBuffer,(u8*)pS-(u8*)pSpuBuffer);
pS=(short *)pSpuBuffer;
}
}
// end of big main loop...
bThreadEnded=1;
return 0;
}
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
// SPU ASYNC... even newer epsxe func
// 1 time every 'cycle' cycles... harhar
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2async(unsigned long cycle)
{
if(iSpuAsyncWait)
{
iSpuAsyncWait++;
if(iSpuAsyncWait<=64) return;
iSpuAsyncWait=0;
}
MAINThread(0); // -> linux high-compat mode
}
////////////////////////////////////////////////////////////////////////
// INIT/EXIT STUFF
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
// SPUINIT: this func will be called first by the main emu
////////////////////////////////////////////////////////////////////////
EXPORT_GCC long CALLBACK SPU2init(void)
{
spuMemC=(unsigned char *)spuMem; // just small setup
memset((void *)s_chan,0,MAXCHAN*sizeof(SPUCHAN));
memset(rvb,0,2*sizeof(REVERBInfo));
sampcount = 0;
InitADSR();
return 0;
}
////////////////////////////////////////////////////////////////////////
// SETUPTIMER: init of certain buffers and threads/timers
////////////////////////////////////////////////////////////////////////
static void SetupTimer(void)
{
memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers
memset(SSumL,0,NSSIZE*sizeof(int));
pS=(short *)pSpuBuffer; // setup soundbuffer pointer
bEndThread=0; // init thread vars
bThreadEnded=0;
bSpuInit=1; // flag: we are inited
}
////////////////////////////////////////////////////////////////////////
// REMOVETIMER: kill threads/timers
////////////////////////////////////////////////////////////////////////
static void RemoveTimer(void)
{
bEndThread=1; // raise flag to end thread
bThreadEnded=0; // no more spu is running
bSpuInit=0;
}
////////////////////////////////////////////////////////////////////////
// SETUPSTREAMS: init most of the spu buffers
////////////////////////////////////////////////////////////////////////
static void SetupStreams(void)
{
int i;
pSpuBuffer=(unsigned char *)malloc(32768); // alloc mixing buffer
i=NSSIZE*2;
sRVBStart[0] = (int *)malloc(i*4); // alloc reverb buffer
memset(sRVBStart[0],0,i*4);
sRVBEnd[0] = sRVBStart[0] + i;
sRVBPlay[0] = sRVBStart[0];
sRVBStart[1] = (int *)malloc(i*4); // alloc reverb buffer
memset(sRVBStart[1],0,i*4);
sRVBEnd[1] = sRVBStart[1] + i;
sRVBPlay[1] = sRVBStart[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;
}
}
////////////////////////////////////////////////////////////////////////
// REMOVESTREAMS: free most buffer
////////////////////////////////////////////////////////////////////////
static void RemoveStreams(void)
{
free(pSpuBuffer); // free mixing buffer
pSpuBuffer=NULL;
free(sRVBStart[0]); // free reverb buffer
sRVBStart[0]=0;
free(sRVBStart[1]); // free reverb buffer
sRVBStart[1]=0;
/*
int i;
for(i=0;i<MAXCHAN;i++)
{
WaitForSingleObject(s_chan[i].hMutex,2000);
ReleaseMutex(s_chan[i].hMutex);
if(s_chan[i].hMutex)
{CloseHandle(s_chan[i].hMutex);s_chan[i].hMutex=0;}
}
*/
}
////////////////////////////////////////////////////////////////////////
// SPUOPEN: called by main emu after init
////////////////////////////////////////////////////////////////////////
EXPORT_GCC long CALLBACK SPU2open(void *pDsp)
{
if(bSPUIsOpen) return 0; // security for some stupid main emus
iUseXA=0; // just small setup
iVolume=3;
bEndThread=0;
bThreadEnded=0;
spuMemC=(unsigned char *)spuMem;
memset((void *)s_chan,0,(MAXCHAN+1)*sizeof(SPUCHAN));
pSpuIrq[0]=0;
pSpuIrq[1]=0;
iSPUIRQWait=1;
dwNewChannel2[0]=0;
dwNewChannel2[1]=0;
dwEndChannel2[0]=0;
dwEndChannel2[1]=0;
spuCtrl2[0]=0;
spuCtrl2[1]=0;
spuStat2[0]=0;
spuStat2[1]=0;
spuIrq2[0]=0;
spuIrq2[1]=0;
spuAddr2[0]=0xffffffff;
spuAddr2[1]=0xffffffff;
spuRvbAddr2[0]=0;
spuRvbAddr2[1]=0;
spuRvbAEnd2[0]=0;
spuRvbAEnd2[1]=0;
// ReadConfig(); // read user stuff
// SetupSound(); // setup midas (before init!)
SetupStreams(); // prepare streaming
SetupTimer(); // timer for feeding data
bSPUIsOpen=1;
return 0;
}
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
// SPUCLOSE: called before shutdown
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2close(void)
{
if(!bSPUIsOpen) return; // some security
bSPUIsOpen=0; // no more open
RemoveTimer(); // no more feeding
// RemoveSound(); // no more sound handling
RemoveStreams(); // no more streaming
}
////////////////////////////////////////////////////////////////////////
// SPUSHUTDOWN: called by main emu on final exit
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2shutdown(void)
{
return;
}
////////////////////////////////////////////////////////////////////////
// SPUTEST: we don't test, we are always fine ;)
////////////////////////////////////////////////////////////////////////
EXPORT_GCC long CALLBACK SPU2test(void)
{
return 0;
}
////////////////////////////////////////////////////////////////////////
// SETUP CALLBACKS
// this functions will be called once,
// passes a callback that should be called on SPU-IRQ/cdda volume change
////////////////////////////////////////////////////////////////////////
// not used yet
EXPORT_GCC void CALLBACK SPU2irqCallback(void (CALLBACK *callback)(void))
{
irqCallback = callback;
}
// not used yet
EXPORT_GCC void CALLBACK SPU2registerCallback(void (CALLBACK *callback)(void))
{
irqCallback = callback;
}
// not used yet
EXPORT_GCC void CALLBACK SPU2registerCDDAVolume(void (CALLBACK *CDDAVcallback)(unsigned short,unsigned short))
{
cddavCallback = CDDAVcallback;
}
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