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Diffstat (limited to 'plugins/dfsound/spu.c')
| -rw-r--r-- | plugins/dfsound/spu.c | 1029 |
1 files changed, 1029 insertions, 0 deletions
diff --git a/plugins/dfsound/spu.c b/plugins/dfsound/spu.c new file mode 100644 index 0000000..c086c06 --- /dev/null +++ b/plugins/dfsound/spu.c @@ -0,0 +1,1029 @@ +/*************************************************************************** + 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 <libintl.h> +#include <locale.h> +#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(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 + +//////////////////////////////////////////////////////////////////////// + +static void *MAINThread(void *arg) +{ + 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 + + 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? + + 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) + usleep(1000L); + } + else + { + lastch=ch; + lastns=ns; + + return 0; + } + } + +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; + + 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<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 +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); +} |