begin : Wed May 15 2002
copyright : (C) 2002 by Pete Bernert
email : BlackDove@addcom.de
+
+ Portions (C) GraÅžvydas "notaz" Ignotas, 2010-2012
+
***************************************************************************/
/***************************************************************************
* *
#define _IN_SPU
#include "externals.h"
-#include "cfg.h"
+#include "registers.h"
#include "dsoundoss.h"
-#include "regs.h"
#ifdef ENABLE_NLS
#include <libintl.h>
#define N_(x) (x)
#endif
+#ifdef __ARM_ARCH_7A__
+ #define ssat32_to_16(v) \
+ asm("ssat %0,#16,%1" : "=r" (v) : "r" (v))
+#else
+ #define ssat32_to_16(v) do { \
+ if (v < -32768) v = -32768; \
+ else if (v > 32767) v = 32767; \
+ } while (0)
+#endif
+
+#define PSXCLK 33868800 /* 33.8688 MHz */
+
+/*
#if defined (USEMACOSX)
static char * libraryName = N_("Mac OS X Sound");
#elif defined (USEALSA)
#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
unsigned char * spuMemC;
unsigned char * pSpuIrq=0;
unsigned char * pSpuBuffer;
-unsigned char * pMixIrq=0;
// user settings
-int iVolume=3;
+int iVolume=768; // 1024 is 1.0
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 int dwNoiseVal; // global noise generator
+unsigned int dwNoiseCount;
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
+unsigned int dwNewChannel=0; // flags for faster testing, if new channel starts
+unsigned int dwChannelOn=0; // not silent channels
+unsigned int dwPendingChanOff=0;
+unsigned int dwChannelDead=0; // silent+not useful channels
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 },
+static const int f[8][2] = { { 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 } };
-int SSumR[NSSIZE];
-int SSumL[NSSIZE];
+int ChanBuf[NSSIZE+3];
+int SSumLR[(NSSIZE+3)*2];
int iFMod[NSSIZE];
int iCycle = 0;
short * pS;
+static int decode_dirty_ch;
+int decode_pos;
+int had_dma;
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
+static int cycles_since_update;
+
+#define CDDA_BUFFER_SIZE (16384 * sizeof(uint32_t)) // must be power of 2
////////////////////////////////////////////////////////////////////////
// CODE AREA
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];
+ //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];
#include "xa.c"
+static void do_irq(void)
+{
+ //if(!(spuStat & STAT_IRQ))
+ {
+ spuStat |= STAT_IRQ; // asserted status?
+ if(irqCallback) irqCallback();
+ }
+}
+
+static int check_irq(int ch, unsigned char *pos)
+{
+ if((spuCtrl & CTRL_IRQ) && pos == pSpuIrq)
+ {
+ //printf("ch%d irq %04x\n", ch, pos - spuMemC);
+ do_irq();
+ return 1;
+ }
+ return 0;
+}
+
////////////////////////////////////////////////////////////////////////
// START SOUND... called by main thread to setup a new sound on a channel
////////////////////////////////////////////////////////////////////////
StartADSR(ch);
StartREVERB(ch);
- s_chan[ch].pCurr=s_chan[ch].pStart; // set sample start
+ // 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].SB[26]=0; // init mixing vars
+ s_chan[ch].SB[27]=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;
// ALL KIND OF HELPERS
////////////////////////////////////////////////////////////////////////
-INLINE void VoiceChangeFrequency(int ch)
+INLINE int FModChangeFrequency(int ch,int ns)
{
- 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;
+ unsigned int NP=s_chan[ch].iRawPitch;
+ int sinc;
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;
+ sinc=NP<<4; // calc frequency
if(iUseInterpolation==1) // freq change in simple interpolation mode
- s_chan[ch].SB[32]=1;
+ 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;
-}
+ return sinc;
+}
////////////////////////////////////////////////////////////////////////
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;
- }
+ ssat32_to_16(fa);
if(iUseInterpolation>=2) // gauss/cubic interpolation
{
////////////////////////////////////////////////////////////////////////
-INLINE int iGetInterpolationVal(int ch)
+INLINE int iGetInterpolationVal(int ch, int spos)
{
int fa;
case 3: // cubic interpolation
{
long xd;int gpos;
- xd = ((s_chan[ch].spos) >> 1)+1;
+ xd = (spos >> 1)+1;
gpos = s_chan[ch].SB[28];
fa = gval(3) - 3*gval(2) + 3*gval(1) - gval0;
case 2: // gauss interpolation
{
int vl, vr;int gpos;
- vl = (s_chan[ch].spos >> 6) & ~3;
+ vl = (spos >> 6) & ~3;
gpos = s_chan[ch].SB[28];
vr=(gauss[vl]*gval0)&~2047;
vr+=(gauss[vl+1]*gval(1))&~2047;
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!
-////////////////////////////////////////////////////////////////////////
+static void decode_block_data(int *dest, const unsigned char *src, int predict_nr, int shift_factor)
+{
+ int nSample;
+ int fa, s_1, s_2, d, s;
-// 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
+ s_1 = dest[27];
+ s_2 = dest[26];
-#define PAUSE_W 5
-#define PAUSE_L 5000
+ for (nSample = 0; nSample < 28; src++)
+ {
+ d = (int)*src;
+ s = (int)(signed short)((d & 0x0f) << 12);
-////////////////////////////////////////////////////////////////////////
+ fa = s >> shift_factor;
+ fa += ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6);
+ s_2=s_1;s_1=fa;
+
+ dest[nSample++] = fa;
+
+ s = (int)(signed short)((d & 0xf0) << 8);
+ fa = s >> shift_factor;
+ fa += ((s_1 * f[predict_nr][0])>>6) + ((s_2 * f[predict_nr][1])>>6);
+ s_2=s_1;s_1=fa;
-static void *MAINThread(void *arg)
+ dest[nSample++] = fa;
+ }
+}
+
+static int decode_block(int ch)
+{
+ unsigned char *start;
+ int predict_nr,shift_factor,flags;
+ int ret = 0;
+
+ start=s_chan[ch].pCurr; // set up the current pos
+
+ if(s_chan[ch].prevflags&1) // 1: stop/loop
+ {
+ if(!(s_chan[ch].prevflags&2))
+ {
+ dwChannelOn&=~(1<<ch); // -> turn everything off
+ s_chan[ch].bStop=1;
+ s_chan[ch].ADSRX.EnvelopeVol=0;
+ }
+
+ start = s_chan[ch].pLoop;
+ }
+ else
+ ret = check_irq(ch, start); // hack, see check_irq below..
+
+ predict_nr=(int)start[0];
+ shift_factor=predict_nr&0xf;
+ predict_nr >>= 4;
+
+ decode_block_data(s_chan[ch].SB, start + 2, predict_nr, shift_factor);
+
+ flags=(int)start[1];
+ if(flags&4)
+ s_chan[ch].pLoop=start; // loop adress
+
+ start+=16;
+
+ if(flags&1) { // 1: stop/loop
+ start = s_chan[ch].pLoop;
+ ret |= check_irq(ch, start); // hack.. :(
+ }
+
+ if (start - spuMemC >= 0x80000)
+ start = spuMemC;
+
+ s_chan[ch].pCurr = start; // store values for next cycle
+ s_chan[ch].prevflags = flags;
+
+ return ret;
+}
+
+// do block, but ignore sample data
+static int skip_block(int ch)
{
- int s_1,s_2,fa,ns;
-#ifndef _MACOSX
- int voldiv = iVolume;
+ unsigned char *start = s_chan[ch].pCurr;
+ int flags = start[1];
+ int ret = check_irq(ch, start);
+
+ if(s_chan[ch].prevflags & 1)
+ start = s_chan[ch].pLoop;
+
+ if(flags & 4)
+ s_chan[ch].pLoop = start;
+
+ start += 16;
+
+ if(flags & 1)
+ start = s_chan[ch].pLoop;
+
+ s_chan[ch].pCurr = start;
+ s_chan[ch].prevflags = flags;
+
+ return ret;
+}
+
+#define make_do_samples(name, fmod_code, interp_start, interp1_code, interp2_code, interp_end) \
+static int do_samples_##name(int ch, int ns, int ns_to) \
+{ \
+ int sinc = s_chan[ch].sinc; \
+ int spos = s_chan[ch].spos; \
+ int sbpos = s_chan[ch].iSBPos; \
+ int *SB = s_chan[ch].SB; \
+ int ret = -1; \
+ int d, fa; \
+ interp_start; \
+ \
+ for (; ns < ns_to; ns++) \
+ { \
+ fmod_code; \
+ \
+ while (spos >= 0x10000) \
+ { \
+ if(sbpos == 28) \
+ { \
+ sbpos = 0; \
+ d = decode_block(ch); \
+ if(d) \
+ ret = ns_to = ns + 1; \
+ } \
+ \
+ fa = SB[sbpos++]; \
+ interp1_code; \
+ spos -= 0x10000; \
+ } \
+ \
+ interp2_code; \
+ spos += sinc; \
+ } \
+ \
+ s_chan[ch].sinc = sinc; \
+ s_chan[ch].spos = spos; \
+ s_chan[ch].iSBPos = sbpos; \
+ interp_end; \
+ \
+ return ret; \
+}
+
+#define fmod_recv_check \
+ if(s_chan[ch].bFMod==1 && iFMod[ns]) \
+ sinc = FModChangeFrequency(ch,ns)
+
+make_do_samples(default, fmod_recv_check, ,
+ StoreInterpolationVal(ch, fa),
+ ChanBuf[ns] = iGetInterpolationVal(ch, spos), )
+make_do_samples(noint, , fa = s_chan[ch].SB[29], , ChanBuf[ns] = fa, s_chan[ch].SB[29] = fa)
+
+#define simple_interp_store \
+ s_chan[ch].SB[28] = 0; \
+ s_chan[ch].SB[29] = s_chan[ch].SB[30]; \
+ s_chan[ch].SB[30] = s_chan[ch].SB[31]; \
+ s_chan[ch].SB[31] = fa; \
+ s_chan[ch].SB[32] = 1
+
+#define simple_interp_get \
+ if(sinc<0x10000) /* -> upsampling? */ \
+ InterpolateUp(ch); /* --> interpolate up */ \
+ else InterpolateDown(ch); /* --> else down */ \
+ ChanBuf[ns] = s_chan[ch].SB[29]
+
+make_do_samples(simple, , ,
+ simple_interp_store, simple_interp_get, )
+
+static int do_samples_noise(int ch, int ns, int ns_to)
+{
+ int level, shift, bit;
+ int ret = -1, d;
+
+ s_chan[ch].spos += s_chan[ch].sinc * (ns_to - ns);
+ while (s_chan[ch].spos >= 28*0x10000)
+ {
+ d = skip_block(ch);
+ if (d)
+ ret = ns_to;
+ s_chan[ch].spos -= 28*0x10000;
+ }
+
+ // modified from DrHell/shalma, no fraction
+ level = (spuCtrl >> 10) & 0x0f;
+ level = 0x8000 >> level;
+
+ for (; ns < ns_to; ns++)
+ {
+ dwNoiseCount += 2;
+ if (dwNoiseCount >= level)
+ {
+ dwNoiseCount -= level;
+ shift = (dwNoiseVal >> 10) & 0x1f;
+ bit = (0x69696969 >> shift) & 1;
+ if (dwNoiseVal & 0x8000)
+ bit ^= 1;
+ dwNoiseVal = (dwNoiseVal << 1) | bit;
+ }
+
+ ChanBuf[ns] = (signed short)dwNoiseVal;
+ }
+
+ return ret;
+}
+
+#ifdef __arm__
+// asm code; lv and rv must be 0-3fff
+extern void mix_chan(int start, int count, int lv, int rv);
+extern void mix_chan_rvb(int start, int count, int lv, int rv);
#else
- const int voldiv = 2;
+static void mix_chan(int start, int count, int lv, int rv)
+{
+ int *dst = SSumLR + start * 2;
+ const int *src = ChanBuf + start;
+ int l, r;
+
+ while (count--)
+ {
+ int sval = *src++;
+
+ l = (sval * lv) >> 14;
+ r = (sval * rv) >> 14;
+ *dst++ += l;
+ *dst++ += r;
+ }
+}
+
+static void mix_chan_rvb(int start, int count, int lv, int rv)
+{
+ int *dst = SSumLR + start * 2;
+ int *drvb = sRVBStart + start * 2;
+ const int *src = ChanBuf + start;
+ int l, r;
+
+ while (count--)
+ {
+ int sval = *src++;
+
+ l = (sval * lv) >> 14;
+ r = (sval * rv) >> 14;
+ *dst++ += l;
+ *dst++ += r;
+ *drvb++ += l;
+ *drvb++ += r;
+ }
+}
#endif
- unsigned char * start;unsigned int nSample;
- int ch,predict_nr,shift_factor,flags,d,s;
+
+// 0x0800-0x0bff Voice 1
+// 0x0c00-0x0fff Voice 3
+static void noinline do_decode_bufs(int which, int start, int count)
+{
+ const int *src = ChanBuf + start;
+ unsigned short *dst = &spuMem[0x800/2 + which*0x400/2];
+ int cursor = decode_pos;
+
+ while (count-- > 0)
+ {
+ dst[cursor] = *src++;
+ cursor = (cursor + 1) & 0x1ff;
+ }
+
+ // decode_pos is updated and irqs are checked later, after voice loop
+}
+
+////////////////////////////////////////////////////////////////////////
+// MAIN SPU FUNCTION
+// here is the main job handler...
+// basically the whole sound processing is done in this fat func!
+////////////////////////////////////////////////////////////////////////
+
+static int do_samples(int forced_updates)
+{
+ int volmult = iVolume;
+ int ns,ns_from,ns_to;
+ int ch,d,silentch;
int bIRQReturn=0;
- while(!bEndThread) // until we are shutting down
+ while(1)
{
// ok, at the beginning we are looking if there is
// enuff free place in the dsound/oss buffer to
// 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?
+ if(!forced_updates && SoundGetBytesBuffered()) // 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
+ return 0;
}
+ cycles_since_update = 0;
+ if(forced_updates > 0)
+ forced_updates--;
+
//--------------------------------------------------// 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=lastns; lastch=-1; // -> setup all kind of vars to continue
- goto GOON; // -> directly jump to the continue point
+ ch=lastch; ns_from=lastns; lastch=-1; // -> setup all kind of vars to continue
}
+ silentch=~(dwChannelOn|dwNewChannel);
+
//--------------------------------------------------//
//- main channel loop -//
//--------------------------------------------------//
{
- for(ch=0;ch<MAXCHAN;ch++) // loop em all... we will collect 1 ms of sound of each playing channel
+ for(;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(dwNewChannel&(1<<ch)) StartSound(ch); // start new sound
+ if(!(dwChannelOn&(1<<ch))) continue; // channel not playing? next
+
+ if(s_chan[ch].bNoise)
+ d=do_samples_noise(ch, ns_from, ns_to);
+ else if(s_chan[ch].bFMod==2 || (s_chan[ch].bFMod==0 && iUseInterpolation==0))
+ d=do_samples_noint(ch, ns_from, ns_to);
+ else if(s_chan[ch].bFMod==0 && iUseInterpolation==1)
+ d=do_samples_simple(ch, ns_from, ns_to);
+ else
+ d=do_samples_default(ch, ns_from, ns_to);
+ if(d>=0)
+ {
+ bIRQReturn=1;
+ lastch=ch;
+ lastns=ns_to=d;
+ }
- if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq) // new psx frequency?
- VoiceChangeFrequency(ch);
+ MixADSR(ch, ns_from, ns_to);
- ns=0;
- while(ns<NSSIZE) // loop until 1 ms of data is reached
+ if(ch==1 || ch==3)
{
- if(s_chan[ch].bFMod==1 && iFMod[ns]) // fmod freq channel
- FModChangeFrequency(ch,ns);
+ do_decode_bufs(ch/2, ns_from, ns_to-ns_from);
+ decode_dirty_ch |= 1<<ch;
+ }
- 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].bFMod==2) // fmod freq channel
+ memcpy(iFMod, ChanBuf, sizeof(iFMod));
+ else if(s_chan[ch].bRVBActive)
+ mix_chan_rvb(ns_from,ns_to-ns_from,s_chan[ch].iLeftVolume,s_chan[ch].iRightVolume);
+ else
+ mix_chan(ns_from,ns_to-ns_from,s_chan[ch].iLeftVolume,s_chan[ch].iRightVolume);
+ }
+ }
- if(s_chan[ch].bNoise)
- fa=iGetNoiseVal(ch); // get noise val
- else fa=iGetInterpolationVal(ch); // get sample val
+ // advance "stopped" channels that can cause irqs
+ // (all chans are always playing on the real thing..)
+ if(spuCtrl&CTRL_IRQ)
+ for(ch=0;ch<MAXCHAN;ch++)
+ {
+ if(!(silentch&(1<<ch))) continue; // already handled
+ if(dwChannelDead&(1<<ch)) continue;
+ if(s_chan[ch].pCurr > pSpuIrq && s_chan[ch].pLoop > pSpuIrq)
+ continue;
- s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // mix adsr
+ s_chan[ch].spos += s_chan[ch].sinc * (ns_to - ns_from);
+ while(s_chan[ch].spos >= 28 * 0x10000)
+ {
+ unsigned char *start = s_chan[ch].pCurr;
- 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
+ // no need for bIRQReturn since the channel is silent
+ skip_block(ch);
+ if(start == s_chan[ch].pCurr)
{
- //////////////////////////////////////////////
- // 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);
+ // looping on self
+ dwChannelDead |= 1<<ch;
+ s_chan[ch].spos = 0;
+ break;
}
- ////////////////////////////////////////////////
- // ok, go on until 1 ms data of this channel is collected
-
- ns++;
- s_chan[ch].spos += s_chan[ch].sinc;
-
+ s_chan[ch].spos -= 28 * 0x10000;
}
-ENDX: ;
}
- }
+
+ if(bIRQReturn) // special return for "spu irq - wait for cpu action"
+ return 0;
+
+ if(unlikely(silentch & decode_dirty_ch & (1<<1))) // must clear silent channel decode buffers
+ {
+ memset(&spuMem[0x800/2], 0, 0x400);
+ decode_dirty_ch &= ~(1<<1);
+ }
+ if(unlikely(silentch & decode_dirty_ch & (1<<3)))
+ {
+ memset(&spuMem[0xc00/2], 0, 0x400);
+ decode_dirty_ch &= ~(1<<3);
+ }
//---------------------------------------------------//
//- here we have another 1 ms of sound data
///////////////////////////////////////////////////////
// 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();
+ if(iUseReverb)
+ REVERBDo();
- dr = SSumR[ns] / voldiv; SSumR[ns] = 0;
- if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767;
- *pS++ = (dl + dr) / 2;
- }
+ if((spuCtrl&0x4000)==0) // muted? (rare, don't optimize for this)
+ {
+ memset(pS, 0, NSSIZE * 2 * sizeof(pS[0]));
+ pS += NSSIZE*2;
}
- else // stereo:
- for (ns = 0; ns < NSSIZE; ns++)
+ else
+ for (ns = 0; ns < NSSIZE*2; )
{
- SSumL[ns] += MixREVERBLeft(ns);
-
- d = SSumL[ns] / voldiv; SSumL[ns] = 0;
- if (d < -32767) d = -32767; if (d > 32767) d = 32767;
+ d = SSumLR[ns]; SSumLR[ns] = 0;
+ d = d * volmult >> 10;
+ ssat32_to_16(d);
*pS++ = d;
+ ns++;
- SSumR[ns] += MixREVERBRight();
-
- d = SSumR[ns] / voldiv; SSumR[ns] = 0;
- if(d < -32767) d = -32767; if(d > 32767) d = 32767;
+ d = SSumLR[ns]; SSumLR[ns] = 0;
+ d = d * volmult >> 10;
+ ssat32_to_16(d);
*pS++ = d;
+ ns++;
}
//////////////////////////////////////////////////////
// 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)
+ if(unlikely((spuCtrl&CTRL_IRQ) && pSpuIrq && pSpuIrq<spuMemC+0x1000))
{
- for(ns=0;ns<NSSIZE;ns++)
+ int irq_pos=(pSpuIrq-spuMemC)/2 & 0x1ff;
+ if((decode_pos <= irq_pos && irq_pos < decode_pos+NSSIZE)
+ || (decode_pos+NSSIZE > 0x200 && irq_pos < ((decode_pos+NSSIZE) & 0x1ff)))
{
- 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;
+ //printf("decoder irq %x\n", decode_pos);
+ do_irq();
}
}
+ decode_pos = (decode_pos + NSSIZE) & 0x1ff;
InitREVERB();
// feed the sound
// wanna have around 1/60 sec (16.666 ms) updates
- if (iCycle++ > 16)
+ if (iCycle++ > 16/FRAG_MSECS)
{
SoundFeedStreamData((unsigned char *)pSpuBuffer,
((unsigned char *)pS) - ((unsigned char *)pSpuBuffer));
}
}
- // end of big main loop...
-
- bThreadEnded = 1;
-
return 0;
}
// SPU ASYNC... even newer epsxe func
// 1 time every 'cycle' cycles... harhar
+// rearmed: called every 2ms now
+
void CALLBACK SPUasync(unsigned long cycle)
{
- if(iSpuAsyncWait)
+ static int old_ctrl;
+ int forced_updates = 0;
+ int do_update = 0;
+
+ if(!bSpuInit) return; // -> no init, no call
+
+ cycles_since_update += cycle;
+
+ if(dwNewChannel || had_dma)
{
- iSpuAsyncWait++;
- if(iSpuAsyncWait<=64) return;
- iSpuAsyncWait=0;
+ forced_updates = 1;
+ do_update = 1;
+ had_dma = 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
+ if((spuCtrl&CTRL_IRQ) && (((spuCtrl^old_ctrl)&CTRL_IRQ) // irq was enabled
+ || cycles_since_update > PSXCLK/60 / 4)) {
+ do_update = 1;
+ forced_updates = cycles_since_update / (PSXCLK/44100) / NSSIZE;
+ }
+ // with no irqs, once per frame should be fine (using a bit more because of BIAS)
+ else if(cycles_since_update > PSXCLK/60 * 5/4)
+ do_update = 1;
- MAINThread(0); // -> linux high-compat mode
- }
+ old_ctrl = spuCtrl;
+
+ if(do_update)
+ do_samples(forced_updates);
}
// SPU UPDATE... new epsxe func
}
// CDDA AUDIO
-void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
+int CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
{
- if (!pcm) return;
- if (nbytes<=0) return;
+ if (!pcm) return -1;
+ if (nbytes<=0) return -1;
- FeedCDDA((unsigned char *)pcm, nbytes);
+ return FeedCDDA((unsigned char *)pcm, nbytes);
}
-// SETUPTIMER: init of certain buffers and threads/timers
-void SetupTimer(void)
+// to be called after state load
+void ClearWorkingState(void)
{
- memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers
- memset(SSumL,0,NSSIZE*sizeof(int));
- memset(iFMod,0,NSSIZE*sizeof(int));
+ memset(SSumLR,0,sizeof(SSumLR)); // init some mixing buffers
+ memset(iFMod,0,sizeof(iFMod));
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
XAFeed = XAStart;
CDDAStart = // alloc cdda buffer
- (uint32_t *)malloc(16384 * sizeof(uint32_t));
+ (uint32_t *)malloc(CDDA_BUFFER_SIZE);
CDDAEnd = CDDAStart + 16384;
CDDAPlay = CDDAStart;
- CDDAFeed = CDDAStart + 1;
+ CDDAFeed = CDDAStart;
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].ADSRX.SustainLevel = 0xf; // -> init sustain
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
+ ClearWorkingState();
+
+ bSpuInit=1; // flag: we are inited
}
// REMOVESTREAMS: free most buffer
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;
+ decode_pos = 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;
if (bSPUIsOpen) return 0; // security for some stupid main emus
SetupSound(); // setup sound (before init!)
- SetupTimer(); // timer for feeding data
bSPUIsOpen = 1;
bSPUIsOpen = 0; // no more open
- RemoveTimer(); // no more feeding
RemoveSound(); // no more sound handling
return 0;
{
SPUclose();
RemoveStreams(); // no more streaming
+ bSpuInit=0;
return 0;
}
#ifdef _MACOSX
DoConfiguration();
#else
- StartCfgTool("CFG");
+// StartCfgTool("CFG");
#endif
return 0;
}
#ifdef _MACOSX
DoAbout();
#else
- StartCfgTool("ABOUT");
+// StartCfgTool("ABOUT");
#endif
}
}
// COMMON PLUGIN INFO FUNCS
+/*
char * CALLBACK PSEgetLibName(void)
{
return _(libraryName);
{
return _(libraryInfo);
}
+*/
+
+// debug
+void spu_get_debug_info(int *chans_out, int *run_chans, int *fmod_chans_out, int *noise_chans_out)
+{
+ int ch = 0, fmod_chans = 0, noise_chans = 0, irq_chans = 0;
+
+ for(;ch<MAXCHAN;ch++)
+ {
+ if (!(dwChannelOn & (1<<ch)))
+ continue;
+ if (s_chan[ch].bFMod == 2)
+ fmod_chans |= 1 << ch;
+ if (s_chan[ch].bNoise)
+ noise_chans |= 1 << ch;
+ if((spuCtrl&CTRL_IRQ) && s_chan[ch].pCurr <= pSpuIrq && s_chan[ch].pLoop <= pSpuIrq)
+ irq_chans |= 1 << ch;
+ }
+
+ *chans_out = dwChannelOn;
+ *run_chans = ~dwChannelOn & ~dwChannelDead & irq_chans;
+ *fmod_chans_out = fmod_chans;
+ *noise_chans_out = noise_chans;
+}
+
+// vim:shiftwidth=1:expandtab
notaz.gp2x.de / pcsx_rearmed.git / blobdiff