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view src/psf2/peops2/spu.c @ 3203:f5456241bff9 default tip
changed include path from audacious to audlegacy.
| author | Yoshiki Yazawa <yaz@honeyplanet.jp> |
|---|---|
| date | Tue, 10 Nov 2009 05:19:25 +0900 |
| parents | 77f8f7f1b0d8 |
| children |
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/*************************************************************************** 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 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 iSecureStart=0; // secure start counter extern void psf2_update(unsigned char *samples, long lBytes, void *data); //////////////////////////////////////////////////////////////////////// // 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(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) { 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; static u32 seektime; int psf2_seek(u32 t) { seektime=t*441/10; if(seektime>sampcount) return(1); return(0); } // Counting to 65536 results in full volume offage. void setlength2(s32 stop, s32 fade) { seektime = 0; 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, void *data) { int s_1,s_2,fa; 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 return NULL; } } //////////////////////////////////////////// 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];SSumL[0]=0; d2=SSumR[0];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) { psf2_update(NULL, 0, data); 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 (seektime != 0 && sampcount < seektime) { pS=(short *)pSpuBuffer; } else if ((((unsigned char *)pS)-((unsigned char *)pSpuBuffer)) == (735*4)) { short *pSilenceIter = (short *)pSpuBuffer; int iSilenceCount = 0; for (; pSilenceIter < pS; pSilenceIter++) { if (*pSilenceIter == 0) iSilenceCount++; if (iSilenceCount > 20) break; } if (iSilenceCount < 20) psf2_update((u8*)pSpuBuffer,(u8*)pS-(u8*)pSpuBuffer, data); 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, void *data) { if(iSpuAsyncWait) { iSpuAsyncWait++; if(iSpuAsyncWait<=64) return; iSpuAsyncWait=0; } MAINThread(0, data); // -> 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 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; }