/*************************************************************************** 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 _IN_SPU #include "externals.h" #include "cfg.h" #include "dsoundoss.h" #include "regs.h" #ifdef ENABLE_NLS #include #include #define _(x) gettext(x) #define N_(x) (x) #else #define _(x) (x) #define N_(x) (x) #endif #if defined (USEMACOSX) static char * libraryName = N_("Mac OS X Sound"); #elif defined (USEALSA) static char * libraryName = N_("ALSA Sound"); #elif defined (USEOSS) static char * libraryName = N_("OSS Sound"); #elif defined (USESDL) static char * libraryName = N_("SDL Sound"); #elif defined (USEPULSEAUDIO) static char * libraryName = N_("PulseAudio Sound"); #else static char * libraryName = N_("NULL Sound"); #endif static char * libraryInfo = N_("P.E.Op.S. Sound Driver V1.7\nCoded by Pete Bernert and the P.E.Op.S. team\n"); // 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 iDebugMode=0; 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; static pthread_t thread = (pthread_t)-1; // thread id (linux) 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" //////////////////////////////////////////////////////////////////////// // 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(id2id1) {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); // fussy timing issues - do in VoiceOn //s_chan[ch].pCurr=s_chan[ch].pStart; // set sample start //s_chan[ch].bStop=0; //s_chan[ch].bOn=1; 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].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< 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 //////////////////////////////////////////////////////////////////////// static void *MAINThread(void *arg) { int s_1,s_2,fa,ns,ns_from,ns_to; #if !defined(_MACOSX) && !defined(__arm__) 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 if(iUseTimer) return 0; // linux no-thread mode? bye usleep(PAUSE_L); // else sleep for x ms (linux) 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? ns_from=0; ns_to=NSSIZE; ch=0; if(lastch>=0) // will be -1 if no continue is pending { ch=lastch; ns_from=lastns+1; lastch=-1; // -> setup all kind of vars to continue } //--------------------------------------------------// //- main channel loop -// //--------------------------------------------------// { for(;ch=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; lastch=ch; lastns=ns; ns_to=ns+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; 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 s_chan[ch].spos += s_chan[ch].sinc; } ENDX: ; } } 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() 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=pMixIrq+(ch*0x400) && pSpuIrqspuMemC+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; return 0; } // 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 MAINThread(0); // -> linux high-compat mode } } // 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 if(!iUseTimer) // linux: use thread { pthread_create(&thread, NULL, MAINThread, NULL); } } // REMOVETIMER: kill threads/timers void RemoveTimer(void) { bEndThread=1; // raise flag to end thread 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 } 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 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 = 0; lastch = -1; //ReadConfigSPU(); // read user stuff SetupStreams(); // prepare streaming return 0; } // SPUOPEN: called by main emu after init long CALLBACK SPUopen(void) { if (bSPUIsOpen) return 0; // security for some stupid main emus 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) { #ifdef _MACOSX DoConfiguration(); #else // StartCfgTool("CFG"); #endif return 0; } // SPUABOUT: show about window void CALLBACK SPUabout(void) { #ifdef _MACOSX DoAbout(); #else // StartCfgTool("ABOUT"); #endif } // 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; } // COMMON PLUGIN INFO FUNCS /* char * CALLBACK PSEgetLibName(void) { return _(libraryName); } unsigned long CALLBACK PSEgetLibType(void) { return PSE_LT_SPU; } unsigned long CALLBACK PSEgetLibVersion(void) { return (1 << 16) | (6 << 8); } char * SPUgetLibInfos(void) { return _(libraryInfo); } */