1 /***************************************************************************
4 begin : Wed May 15 2002
5 copyright : (C) 2002 by Pete Bernert
6 email : BlackDove@addcom.de
7 ***************************************************************************/
8 /***************************************************************************
10 * This program is free software; you can redistribute it and/or modify *
11 * it under the terms of the GNU General Public License as published by *
12 * the Free Software Foundation; either version 2 of the License, or *
13 * (at your option) any later version. See also the license.txt file for *
14 * additional informations. *
16 ***************************************************************************/
22 #include "externals.h"
24 #include "dsoundoss.h"
30 #define _(x) gettext(x)
37 #if defined (USEMACOSX)
38 static char * libraryName = N_("Mac OS X Sound");
39 #elif defined (USEALSA)
40 static char * libraryName = N_("ALSA Sound");
41 #elif defined (USEOSS)
42 static char * libraryName = N_("OSS Sound");
43 #elif defined (USESDL)
44 static char * libraryName = N_("SDL Sound");
45 #elif defined (USEPULSEAUDIO)
46 static char * libraryName = N_("PulseAudio Sound");
48 static char * libraryName = N_("NULL Sound");
51 static char * libraryInfo = N_("P.E.Op.S. Sound Driver V1.7\nCoded by Pete Bernert and the P.E.Op.S. team\n");
55 // psx buffer / addresses
57 unsigned short regArea[10000];
58 unsigned short spuMem[256*1024];
59 unsigned char * spuMemC;
60 unsigned char * pSpuIrq=0;
61 unsigned char * pSpuBuffer;
62 unsigned char * pMixIrq=0;
73 int iUseInterpolation=2;
76 // MAIN infos struct for each channel
78 SPUCHAN s_chan[MAXCHAN+1]; // channel + 1 infos (1 is security for fmod handling)
81 unsigned long dwNoiseVal=1; // global noise generator
84 unsigned short spuCtrl=0; // some vars to store psx reg infos
85 unsigned short spuStat=0;
86 unsigned short spuIrq=0;
87 unsigned long spuAddr=0xffffffff; // address into spu mem
88 int bEndThread=0; // thread handlers
93 static pthread_t thread = (pthread_t)-1; // thread id (linux)
95 unsigned long dwNewChannel=0; // flags for faster testing, if new channel starts
97 void (CALLBACK *irqCallback)(void)=0; // func of main emu, called on spu irq
98 void (CALLBACK *cddavCallback)(unsigned short,unsigned short)=0;
100 // certain globals (were local before, but with the new timeproc I need em global)
102 static const int f[5][2] = { { 0, 0 },
113 int lastch=-1; // last channel processed on spu irq in timer mode
114 static int lastns=0; // last ns pos
115 static int iSecureStart=0; // secure start counter
117 ////////////////////////////////////////////////////////////////////////
119 ////////////////////////////////////////////////////////////////////////
121 // dirty inline func includes
126 ////////////////////////////////////////////////////////////////////////
127 // helpers for simple interpolation
130 // easy interpolation on upsampling, no special filter, just "Pete's common sense" tm
132 // instead of having n equal sample values in a row like:
136 // we compare the current delta change with the next delta change.
138 // if curr_delta is positive,
140 // - and next delta is smaller (or changing direction):
144 // - and next delta significant (at least twice) bigger:
148 // - and next delta is nearly same:
153 // if curr_delta is negative,
155 // - and next delta is smaller (or changing direction):
159 // - and next delta significant (at least twice) bigger:
163 // - and next delta is nearly same:
169 INLINE void InterpolateUp(int ch)
171 if(s_chan[ch].SB[32]==1) // flag == 1? calc step and set flag... and don't change the value in this pass
173 const int id1=s_chan[ch].SB[30]-s_chan[ch].SB[29]; // curr delta to next val
174 const int id2=s_chan[ch].SB[31]-s_chan[ch].SB[30]; // and next delta to next-next val :)
178 if(id1>0) // curr delta positive
181 {s_chan[ch].SB[28]=id1;s_chan[ch].SB[32]=2;}
184 s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x10000L;
186 s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x20000L;
188 else // curr delta negative
191 {s_chan[ch].SB[28]=id1;s_chan[ch].SB[32]=2;}
194 s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x10000L;
196 s_chan[ch].SB[28]=(id1*s_chan[ch].sinc)/0x20000L;
200 if(s_chan[ch].SB[32]==2) // flag 1: calc step and set flag... and don't change the value in this pass
204 s_chan[ch].SB[28]=(s_chan[ch].SB[28]*s_chan[ch].sinc)/0x20000L;
205 if(s_chan[ch].sinc<=0x8000)
206 s_chan[ch].SB[29]=s_chan[ch].SB[30]-(s_chan[ch].SB[28]*((0x10000/s_chan[ch].sinc)-1));
207 else s_chan[ch].SB[29]+=s_chan[ch].SB[28];
209 else // no flags? add bigger val (if possible), calc smaller step, set flag1
210 s_chan[ch].SB[29]+=s_chan[ch].SB[28];
214 // even easier interpolation on downsampling, also no special filter, again just "Pete's common sense" tm
217 INLINE void InterpolateDown(int ch)
219 if(s_chan[ch].sinc>=0x20000L) // we would skip at least one val?
221 s_chan[ch].SB[29]+=(s_chan[ch].SB[30]-s_chan[ch].SB[29])/2; // add easy weight
222 if(s_chan[ch].sinc>=0x30000L) // we would skip even more vals?
223 s_chan[ch].SB[29]+=(s_chan[ch].SB[31]-s_chan[ch].SB[30])/2;// add additional next weight
227 ////////////////////////////////////////////////////////////////////////
228 // helpers for gauss interpolation
230 #define gval0 (((short*)(&s_chan[ch].SB[29]))[gpos])
231 #define gval(x) (((short*)(&s_chan[ch].SB[29]))[(gpos+x)&3])
235 ////////////////////////////////////////////////////////////////////////
239 ////////////////////////////////////////////////////////////////////////
240 // START SOUND... called by main thread to setup a new sound on a channel
241 ////////////////////////////////////////////////////////////////////////
243 INLINE void StartSound(int ch)
248 // fussy timing issues - do in VoiceOn
249 //s_chan[ch].pCurr=s_chan[ch].pStart; // set sample start
250 //s_chan[ch].bStop=0;
253 s_chan[ch].s_1=0; // init mixing vars
255 s_chan[ch].iSBPos=28;
257 s_chan[ch].bNew=0; // init channel flags
259 s_chan[ch].SB[29]=0; // init our interpolation helpers
262 if(iUseInterpolation>=2) // gauss interpolation?
263 {s_chan[ch].spos=0x30000L;s_chan[ch].SB[28]=0;} // -> start with more decoding
264 else {s_chan[ch].spos=0x10000L;s_chan[ch].SB[31]=0;} // -> no/simple interpolation starts with one 44100 decoding
266 dwNewChannel&=~(1<<ch); // clear new channel bit
269 ////////////////////////////////////////////////////////////////////////
270 // ALL KIND OF HELPERS
271 ////////////////////////////////////////////////////////////////////////
273 INLINE void VoiceChangeFrequency(int ch)
275 s_chan[ch].iUsedFreq=s_chan[ch].iActFreq; // -> take it and calc steps
276 s_chan[ch].sinc=s_chan[ch].iRawPitch<<4;
277 if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
278 if(iUseInterpolation==1) s_chan[ch].SB[32]=1; // -> freq change in simle imterpolation mode: set flag
281 ////////////////////////////////////////////////////////////////////////
283 INLINE void FModChangeFrequency(int ch,int ns)
285 int NP=s_chan[ch].iRawPitch;
287 NP=((32768L+iFMod[ns])*NP)/32768L;
289 if(NP>0x3fff) NP=0x3fff;
292 NP=(44100L*NP)/(4096L); // calc frequency
294 s_chan[ch].iActFreq=NP;
295 s_chan[ch].iUsedFreq=NP;
296 s_chan[ch].sinc=(((NP/10)<<16)/4410);
297 if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
298 if(iUseInterpolation==1) // freq change in simple interpolation mode
303 ////////////////////////////////////////////////////////////////////////
305 // noise handler... just produces some noise data
306 // surely wrong... and no noise frequency (spuCtrl&0x3f00) will be used...
307 // and sometimes the noise will be used as fmod modulation... pfff
309 INLINE int iGetNoiseVal(int ch)
313 if((dwNoiseVal<<=1)&0x80000000L)
315 dwNoiseVal^=0x0040001L;
316 fa=((dwNoiseVal>>2)&0x7fff);
319 else fa=(dwNoiseVal>>2)&0x7fff;
321 // mmm... depending on the noise freq we allow bigger/smaller changes to the previous val
322 fa=s_chan[ch].iOldNoise+((fa-s_chan[ch].iOldNoise)/((0x001f-((spuCtrl&0x3f00)>>9))+1));
323 if(fa>32767L) fa=32767L;
324 if(fa<-32767L) fa=-32767L;
325 s_chan[ch].iOldNoise=fa;
327 if(iUseInterpolation<2) // no gauss/cubic interpolation?
328 s_chan[ch].SB[29] = fa; // -> store noise val in "current sample" slot
332 ////////////////////////////////////////////////////////////////////////
334 INLINE void StoreInterpolationVal(int ch,int fa)
336 if(s_chan[ch].bFMod==2) // fmod freq channel
337 s_chan[ch].SB[29]=fa;
340 if((spuCtrl&0x4000)==0) fa=0; // muted?
343 if(fa>32767L) fa=32767L;
344 if(fa<-32767L) fa=-32767L;
347 if(iUseInterpolation>=2) // gauss/cubic interpolation
349 int gpos = s_chan[ch].SB[28];
352 s_chan[ch].SB[28] = gpos;
355 if(iUseInterpolation==1) // simple interpolation
357 s_chan[ch].SB[28] = 0;
358 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'
359 s_chan[ch].SB[30] = s_chan[ch].SB[31];
360 s_chan[ch].SB[31] = fa;
361 s_chan[ch].SB[32] = 1; // -> flag: calc new interolation
363 else s_chan[ch].SB[29]=fa; // no interpolation
367 ////////////////////////////////////////////////////////////////////////
369 INLINE int iGetInterpolationVal(int ch)
373 if(s_chan[ch].bFMod==2) return s_chan[ch].SB[29];
375 switch(iUseInterpolation)
377 //--------------------------------------------------//
378 case 3: // cubic interpolation
381 xd = ((s_chan[ch].spos) >> 1)+1;
382 gpos = s_chan[ch].SB[28];
384 fa = gval(3) - 3*gval(2) + 3*gval(1) - gval0;
385 fa *= (xd - (2<<15)) / 6;
387 fa += gval(2) - gval(1) - gval(1) + gval0;
388 fa *= (xd - (1<<15)) >> 1;
390 fa += gval(1) - gval0;
396 //--------------------------------------------------//
397 case 2: // gauss interpolation
400 vl = (s_chan[ch].spos >> 6) & ~3;
401 gpos = s_chan[ch].SB[28];
402 vr=(gauss[vl]*gval0)&~2047;
403 vr+=(gauss[vl+1]*gval(1))&~2047;
404 vr+=(gauss[vl+2]*gval(2))&~2047;
405 vr+=(gauss[vl+3]*gval(3))&~2047;
408 //--------------------------------------------------//
409 case 1: // simple interpolation
411 if(s_chan[ch].sinc<0x10000L) // -> upsampling?
412 InterpolateUp(ch); // --> interpolate up
413 else InterpolateDown(ch); // --> else down
414 fa=s_chan[ch].SB[29];
416 //--------------------------------------------------//
417 default: // no interpolation
419 fa=s_chan[ch].SB[29];
421 //--------------------------------------------------//
427 ////////////////////////////////////////////////////////////////////////
429 // here is the main job handler... thread, timer or direct func call
430 // basically the whole sound processing is done in this fat func!
431 ////////////////////////////////////////////////////////////////////////
433 // 5 ms waiting phase, if buffer is full and no new sound has to get started
434 // .. can be made smaller (smallest val: 1 ms), but bigger waits give
435 // better performance
440 ////////////////////////////////////////////////////////////////////////
442 static void *MAINThread(void *arg)
444 int s_1,s_2,fa,ns,ns_from,ns_to;
445 #if !defined(_MACOSX) && !defined(__arm__)
446 int voldiv = iVolume;
448 const int voldiv = 2;
450 unsigned char * start;unsigned int nSample;
451 int ch,predict_nr,shift_factor,flags,d,s;
454 while(!bEndThread) // until we are shutting down
456 // ok, at the beginning we are looking if there is
457 // enuff free place in the dsound/oss buffer to
458 // fill in new data, or if there is a new channel to start.
459 // if not, we wait (thread) or return (timer/spuasync)
460 // until enuff free place is available/a new channel gets
463 if(dwNewChannel) // new channel should start immedately?
464 { // (at least one bit 0 ... MAXCHANNEL is set?)
465 iSecureStart++; // -> set iSecure
466 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)
468 else iSecureStart=0; // 0: no new channel should start
470 while(!iSecureStart && !bEndThread && // no new start? no thread end?
471 (SoundGetBytesBuffered()>TESTSIZE)) // and still enuff data in sound buffer?
473 iSecureStart=0; // reset secure
475 if(iUseTimer) return 0; // linux no-thread mode? bye
476 usleep(PAUSE_L); // else sleep for x ms (linux)
478 if(dwNewChannel) iSecureStart=1; // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop
481 //--------------------------------------------------// continue from irq handling in timer mode?
486 if(lastch>=0) // will be -1 if no continue is pending
488 ch=lastch; ns_from=lastns+1; lastch=-1; // -> setup all kind of vars to continue
491 //--------------------------------------------------//
492 //- main channel loop -//
493 //--------------------------------------------------//
495 for(;ch<MAXCHAN;ch++) // loop em all... we will collect 1 ms of sound of each playing channel
497 if(s_chan[ch].bNew) StartSound(ch); // start new sound
498 if(!s_chan[ch].bOn) continue; // channel not playing? next
500 if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq) // new psx frequency?
501 VoiceChangeFrequency(ch);
503 for(ns=ns_from;ns<ns_to;ns++) // loop until 1 ms of data is reached
505 if(s_chan[ch].bFMod==1 && iFMod[ns]) // fmod freq channel
506 FModChangeFrequency(ch,ns);
508 while(s_chan[ch].spos>=0x10000L)
510 if(s_chan[ch].iSBPos==28) // 28 reached?
512 start=s_chan[ch].pCurr; // set up the current pos
514 if (start == (unsigned char*)-1) // special "stop" sign
516 s_chan[ch].bOn=0; // -> turn everything off
517 s_chan[ch].ADSRX.lVolume=0;
518 s_chan[ch].ADSRX.EnvelopeVol=0;
519 goto ENDX; // -> and done for this channel
524 //////////////////////////////////////////// spu irq handler here? mmm... do it later
529 predict_nr=(int)*start;start++;
530 shift_factor=predict_nr&0xf;
532 flags=(int)*start;start++;
534 // -------------------------------------- //
536 for (nSample=0;nSample<28;start++)
540 if(s&0x8000) s|=0xffff0000;
542 fa=(s >> shift_factor);
543 fa=fa + ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6);
547 s_chan[ch].SB[nSample++]=fa;
549 if(s&0x8000) s|=0xffff0000;
550 fa=(s>>shift_factor);
551 fa=fa + ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6);
554 s_chan[ch].SB[nSample++]=fa;
557 //////////////////////////////////////////// irq check
559 if(irqCallback && (spuCtrl&0x40)) // some callback and irq active?
561 if((pSpuIrq > start-16 && // irq address reached?
563 ((flags&1) && // special: irq on looping addr, when stop/loop flag is set
564 (pSpuIrq > s_chan[ch].pLoop-16 &&
565 pSpuIrq <= s_chan[ch].pLoop)))
567 s_chan[ch].iIrqDone=1; // -> debug flag
568 irqCallback(); // -> call main emu
570 if(iSPUIRQWait) // -> option: wait after irq for main emu
581 //////////////////////////////////////////// flag handler
583 if((flags&4) && (!s_chan[ch].bIgnoreLoop))
584 s_chan[ch].pLoop=start-16; // loop adress
586 if(flags&1) // 1: stop/loop
588 // We play this block out first...
589 //if(!(flags&2)) // 1+2: do loop... otherwise: stop
590 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)
591 { // and checking if pLoop is set avoids crashes, yeah
592 start = (unsigned char*)-1;
596 start = s_chan[ch].pLoop;
600 s_chan[ch].pCurr=start; // store values for next cycle
607 fa=s_chan[ch].SB[s_chan[ch].iSBPos++]; // get sample data
609 StoreInterpolationVal(ch,fa); // store val for later interpolation
611 s_chan[ch].spos -= 0x10000L;
614 if(s_chan[ch].bNoise)
615 fa=iGetNoiseVal(ch); // get noise val
616 else fa=iGetInterpolationVal(ch); // get sample val
618 s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // mix adsr
620 if(s_chan[ch].bFMod==2) // fmod freq channel
621 iFMod[ns]=s_chan[ch].sval; // -> store 1T sample data, use that to do fmod on next channel
622 else // no fmod freq channel
624 //////////////////////////////////////////////
625 // ok, left/right sound volume (psx volume goes from 0 ... 0x3fff)
628 s_chan[ch].sval=0; // debug mute
631 SSumL[ns]+=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000L;
632 SSumR[ns]+=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000L;
635 //////////////////////////////////////////////
636 // now let us store sound data for reverb
638 if(s_chan[ch].bRVBActive) StoreREVERB(ch,ns);
641 ////////////////////////////////////////////////
642 // ok, go on until 1 ms data of this channel is collected
644 s_chan[ch].spos += s_chan[ch].sinc;
650 if(bIRQReturn) // special return for "spu irq - wait for cpu action"
655 DWORD dwWatchTime=timeGetTime_spu()+2500;
657 while(iSpuAsyncWait && !bEndThread &&
658 timeGetTime_spu()<dwWatchTime)
669 //---------------------------------------------------//
670 //- here we have another 1 ms of sound data
671 //---------------------------------------------------//
672 // mix XA infos (if any)
676 ///////////////////////////////////////////////////////
677 // mix all channels (including reverb) into one buffer
679 if(iDisStereo) // no stereo?
682 for (ns = 0; ns < NSSIZE; ns++)
684 SSumL[ns] += MixREVERBLeft(ns);
686 dl = SSumL[ns] / voldiv; SSumL[ns] = 0;
687 if (dl < -32767) dl = -32767; if (dl > 32767) dl = 32767;
689 SSumR[ns] += MixREVERBRight();
691 dr = SSumR[ns] / voldiv; SSumR[ns] = 0;
692 if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767;
693 *pS++ = (dl + dr) / 2;
697 for (ns = 0; ns < NSSIZE; ns++)
699 SSumL[ns] += MixREVERBLeft(ns);
701 d = SSumL[ns] / voldiv; SSumL[ns] = 0;
702 if (d < -32767) d = -32767; if (d > 32767) d = 32767;
705 SSumR[ns] += MixREVERBRight();
707 d = SSumR[ns] / voldiv; SSumR[ns] = 0;
708 if(d < -32767) d = -32767; if(d > 32767) d = 32767;
712 //////////////////////////////////////////////////////
713 // special irq handling in the decode buffers (0x0000-0x1000)
715 // the decode buffers are located in spu memory in the following way:
716 // 0x0000-0x03ff CD audio left
717 // 0x0400-0x07ff CD audio right
718 // 0x0800-0x0bff Voice 1
719 // 0x0c00-0x0fff Voice 3
720 // and decoded data is 16 bit for one sample
722 // even if voices 1/3 are off or no cd audio is playing, the internal
723 // play positions will move on and wrap after 0x400 bytes.
724 // Therefore: we just need a pointer from spumem+0 to spumem+3ff, and
725 // increase this pointer on each sample by 2 bytes. If this pointer
726 // (or 0x400 offsets of this pointer) hits the spuirq address, we generate
727 // an IRQ. Only problem: the "wait for cpu" option is kinda hard to do here
728 // in some of Peops timer modes. So: we ignore this option here (for now).
730 if(pMixIrq && irqCallback)
732 for(ns=0;ns<NSSIZE;ns++)
734 if((spuCtrl&0x40) && pSpuIrq && pSpuIrq<spuMemC+0x1000)
738 if(pSpuIrq>=pMixIrq+(ch*0x400) && pSpuIrq<pMixIrq+(ch*0x400)+2)
739 {irqCallback();s_chan[ch].iIrqDone=1;}
742 pMixIrq+=2;if(pMixIrq>spuMemC+0x3ff) pMixIrq=spuMemC;
749 // wanna have around 1/60 sec (16.666 ms) updates
752 SoundFeedStreamData((unsigned char *)pSpuBuffer,
753 ((unsigned char *)pS) - ((unsigned char *)pSpuBuffer));
754 pS = (short *)pSpuBuffer;
759 // end of big main loop...
766 // SPU ASYNC... even newer epsxe func
767 // 1 time every 'cycle' cycles... harhar
769 void CALLBACK SPUasync(unsigned long cycle)
774 if(iSpuAsyncWait<=64) return;
778 if(iUseTimer==2) // special mode, only used in Linux by this spu (or if you enable the experimental Windows mode)
780 if(!bSpuInit) return; // -> no init, no call
782 MAINThread(0); // -> linux high-compat mode
786 // SPU UPDATE... new epsxe func
787 // 1 time every 32 hsync lines
788 // (312/32)x50 in pal
789 // (262/32)x60 in ntsc
791 // since epsxe 1.5.2 (linux) uses SPUupdate, not SPUasync, I will
792 // leave that func in the linux port, until epsxe linux is using
793 // the async function as well
795 void CALLBACK SPUupdate(void)
802 void CALLBACK SPUplayADPCMchannel(xa_decode_t *xap)
805 if(!xap->freq) return; // no xa freq ? bye
807 FeedXA(xap); // call main XA feeder
811 void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
814 if (nbytes<=0) return;
816 FeedCDDA((unsigned char *)pcm, nbytes);
819 // SETUPTIMER: init of certain buffers and threads/timers
820 void SetupTimer(void)
822 memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers
823 memset(SSumL,0,NSSIZE*sizeof(int));
824 memset(iFMod,0,NSSIZE*sizeof(int));
825 pS=(short *)pSpuBuffer; // setup soundbuffer pointer
827 bEndThread=0; // init thread vars
829 bSpuInit=1; // flag: we are inited
831 if(!iUseTimer) // linux: use thread
833 pthread_create(&thread, NULL, MAINThread, NULL);
837 // REMOVETIMER: kill threads/timers
838 void RemoveTimer(void)
840 bEndThread=1; // raise flag to end thread
842 if(!iUseTimer) // linux tread?
845 while(!bThreadEnded && i<2000) {usleep(1000L);i++;} // -> wait until thread has ended
846 if(thread!=(pthread_t)-1) {pthread_cancel(thread);thread=(pthread_t)-1;} // -> cancel thread anyway
849 bThreadEnded=0; // no more spu is running
853 // SETUPSTREAMS: init most of the spu buffers
854 void SetupStreams(void)
858 pSpuBuffer=(unsigned char *)malloc(32768); // alloc mixing buffer
860 if(iUseReverb==1) i=88200*2;
863 sRVBStart = (int *)malloc(i*4); // alloc reverb buffer
864 memset(sRVBStart,0,i*4);
865 sRVBEnd = sRVBStart + i;
866 sRVBPlay = sRVBStart;
868 XAStart = // alloc xa buffer
869 (uint32_t *)malloc(44100 * sizeof(uint32_t));
870 XAEnd = XAStart + 44100;
874 CDDAStart = // alloc cdda buffer
875 (uint32_t *)malloc(16384 * sizeof(uint32_t));
876 CDDAEnd = CDDAStart + 16384;
877 CDDAPlay = CDDAStart;
878 CDDAFeed = CDDAStart + 1;
880 for(i=0;i<MAXCHAN;i++) // loop sound channels
882 // we don't use mutex sync... not needed, would only
884 // s_chan[i].hMutex=CreateMutex(NULL,FALSE,NULL);
885 s_chan[i].ADSRX.SustainLevel = 1024; // -> init sustain
887 s_chan[i].iIrqDone=0;
888 s_chan[i].pLoop=spuMemC;
889 s_chan[i].pStart=spuMemC;
890 s_chan[i].pCurr=spuMemC;
893 pMixIrq=spuMemC; // enable decoded buffer irqs by setting the address
896 // REMOVESTREAMS: free most buffer
897 void RemoveStreams(void)
899 free(pSpuBuffer); // free mixing buffer
901 free(sRVBStart); // free reverb buffer
903 free(XAStart); // free XA buffer
905 free(CDDAStart); // free CDDA buffer
911 // SPUINIT: this func will be called first by the main emu
912 long CALLBACK SPUinit(void)
914 spuMemC = (unsigned char *)spuMem; // just small setup
915 memset((void *)&rvb, 0, sizeof(REVERBInfo));
921 spuAddr = 0xffffffff;
924 spuMemC = (unsigned char *)spuMem;
926 memset((void *)s_chan, 0, (MAXCHAN + 1) * sizeof(SPUCHAN));
931 //ReadConfigSPU(); // read user stuff
932 SetupStreams(); // prepare streaming
937 // SPUOPEN: called by main emu after init
938 long CALLBACK SPUopen(void)
940 if (bSPUIsOpen) return 0; // security for some stupid main emus
942 SetupSound(); // setup sound (before init!)
943 SetupTimer(); // timer for feeding data
947 return PSE_SPU_ERR_SUCCESS;
950 // SPUCLOSE: called before shutdown
951 long CALLBACK SPUclose(void)
953 if (!bSPUIsOpen) return 0; // some security
955 bSPUIsOpen = 0; // no more open
957 RemoveTimer(); // no more feeding
958 RemoveSound(); // no more sound handling
963 // SPUSHUTDOWN: called by main emu on final exit
964 long CALLBACK SPUshutdown(void)
967 RemoveStreams(); // no more streaming
972 // SPUTEST: we don't test, we are always fine ;)
973 long CALLBACK SPUtest(void)
978 // SPUCONFIGURE: call config dialog
979 long CALLBACK SPUconfigure(void)
984 // StartCfgTool("CFG");
989 // SPUABOUT: show about window
990 void CALLBACK SPUabout(void)
995 // StartCfgTool("ABOUT");
1000 // this functions will be called once,
1001 // passes a callback that should be called on SPU-IRQ/cdda volume change
1002 void CALLBACK SPUregisterCallback(void (CALLBACK *callback)(void))
1004 irqCallback = callback;
1007 void CALLBACK SPUregisterCDDAVolume(void (CALLBACK *CDDAVcallback)(unsigned short,unsigned short))
1009 cddavCallback = CDDAVcallback;
1012 // COMMON PLUGIN INFO FUNCS
1014 char * CALLBACK PSEgetLibName(void)
1016 return _(libraryName);
1019 unsigned long CALLBACK PSEgetLibType(void)
1024 unsigned long CALLBACK PSEgetLibVersion(void)
1026 return (1 << 16) | (6 << 8);
1029 char * SPUgetLibInfos(void)
1031 return _(libraryInfo);