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|
/***************************************************************************
spu.c - description
-------------------
begin : Wed May 15 2002
copyright : (C) 2002 by Pete Bernert
email : BlackDove@addcom.de
Portions (C) Gražvydas "notaz" Ignotas, 2010-2012
***************************************************************************/
/***************************************************************************
* *
* 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 "registers.h"
#include "out.h"
#include "arm_features.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
#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)
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;
// user settings
int iVolume=768; // 1024 is 1.0
int iXAPitch=1;
int iUseReverb=2;
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;
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 bSpuInit=0;
int bSPUIsOpen=0;
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[8][2] = { { 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 } };
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 cycles_since_update;
#define CDDA_BUFFER_SIZE (16384 * sizeof(uint32_t)) // must be power of 2
////////////////////////////////////////////////////////////////////////
// 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) ((int)((short*)(&s_chan[ch].SB[29]))[(gpos+x)&3])
#include "gauss_i.h"
////////////////////////////////////////////////////////////////////////
#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
////////////////////////////////////////////////////////////////////////
INLINE void StartSound(int ch)
{
StartADSR(ch);
StartREVERB(ch);
// fussy timing issues - do in VoiceOn
//s_chan[ch].pCurr=s_chan[ch].pStart; // set sample start
//s_chan[ch].bStop=0;
//s_chan[ch].bOn=1;
s_chan[ch].SB[26]=0; // init mixing vars
s_chan[ch].SB[27]=0;
s_chan[ch].iSBPos=28;
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 int FModChangeFrequency(int ch,int ns)
{
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;
sinc=NP<<4; // calc frequency
if(iUseInterpolation==1) // freq change in simple interpolation mode
s_chan[ch].SB[32]=1;
iFMod[ns]=0;
return sinc;
}
////////////////////////////////////////////////////////////////////////
INLINE void StoreInterpolationVal(int ch,int fa)
{
if(s_chan[ch].bFMod==2) // fmod freq channel
s_chan[ch].SB[29]=fa;
else
{
ssat32_to_16(fa);
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 spos)
{
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 = (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 = (spos >> 6) & ~3;
gpos = s_chan[ch].SB[28];
vr=(gauss[vl]*(int)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;
}
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;
s_1 = dest[27];
s_2 = dest[26];
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;
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)
{
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 HAVE_ARMV5
// 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
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
// 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(!bIRQReturn)
{
// 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(!forced_updates && out_current->busy()) // still enuff data in sound buffer?
{
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_from=lastns; lastch=-1; // -> setup all kind of vars to continue
}
silentch=~(dwChannelOn|dwNewChannel);
//--------------------------------------------------//
//- main channel loop -//
//--------------------------------------------------//
{
for(;ch<MAXCHAN;ch++) // loop em all... we will collect 1 ms of sound of each playing channel
{
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;
}
MixADSR(ch, ns_from, ns_to);
if(ch==1 || ch==3)
{
do_decode_bufs(ch/2, ns_from, ns_to-ns_from);
decode_dirty_ch |= 1<<ch;
}
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);
}
}
// 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].spos += s_chan[ch].sinc * (ns_to - ns_from);
while(s_chan[ch].spos >= 28 * 0x10000)
{
unsigned char *start = s_chan[ch].pCurr;
// no need for bIRQReturn since the channel is silent
skip_block(ch);
if(start == s_chan[ch].pCurr)
{
// looping on self
dwChannelDead |= 1<<ch;
s_chan[ch].spos = 0;
break;
}
s_chan[ch].spos -= 28 * 0x10000;
}
}
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 XA infos (if any)
MixXA();
///////////////////////////////////////////////////////
// mix all channels (including reverb) into one buffer
if(iUseReverb)
REVERBDo();
if((spuCtrl&0x4000)==0) // muted? (rare, don't optimize for this)
{
memset(pS, 0, NSSIZE * 2 * sizeof(pS[0]));
pS += NSSIZE*2;
}
else
for (ns = 0; ns < NSSIZE*2; )
{
d = SSumLR[ns]; SSumLR[ns] = 0;
d = d * volmult >> 10;
ssat32_to_16(d);
*pS++ = d;
ns++;
d = SSumLR[ns]; SSumLR[ns] = 0;
d = d * volmult >> 10;
ssat32_to_16(d);
*pS++ = d;
ns++;
}
//////////////////////////////////////////////////////
// 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(unlikely((spuCtrl&CTRL_IRQ) && pSpuIrq && pSpuIrq<spuMemC+0x1000))
{
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)))
{
//printf("decoder irq %x\n", decode_pos);
do_irq();
bIRQReturn = 1;
}
}
decode_pos = (decode_pos + NSSIZE) & 0x1ff;
InitREVERB();
// feed the sound
// wanna have around 1/60 sec (16.666 ms) updates
if (iCycle++ > 16/FRAG_MSECS)
{
out_current->feed(pSpuBuffer,
((unsigned char *)pS) - ((unsigned char *)pSpuBuffer));
pS = (short *)pSpuBuffer;
iCycle = 0;
}
}
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)
{
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)
{
forced_updates = 1;
do_update = 1;
had_dma = 0;
}
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;
old_ctrl = spuCtrl;
if(do_update)
do_samples(forced_updates);
}
// 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
int CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
{
if (!pcm) return -1;
if (nbytes<=0) return -1;
return FeedCDDA((unsigned char *)pcm, nbytes);
}
// to be called after state load
void ClearWorkingState(void)
{
memset(SSumLR,0,sizeof(SSumLR)); // init some mixing buffers
memset(iFMod,0,sizeof(iFMod));
pS=(short *)pSpuBuffer; // setup soundbuffer pointer
}
// 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(CDDA_BUFFER_SIZE);
CDDAEnd = CDDAStart + 16384;
CDDAPlay = CDDAStart;
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 = 0xf; // -> init sustain
s_chan[i].pLoop=spuMemC;
s_chan[i].pCurr=spuMemC;
}
ClearWorkingState();
bSpuInit=1; // flag: we are inited
}
// 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();
spuIrq = 0;
spuAddr = 0xffffffff;
spuMemC = (unsigned char *)spuMem;
decode_pos = 0;
memset((void *)s_chan, 0, (MAXCHAN + 1) * sizeof(SPUCHAN));
pSpuIrq = 0;
lastch = -1;
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!)
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
out_current->finish(); // no more sound handling
return 0;
}
// SPUSHUTDOWN: called by main emu on final exit
long CALLBACK SPUshutdown(void)
{
SPUclose();
RemoveStreams(); // no more streaming
bSpuInit=0;
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);
}
*/
// 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
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