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pcsx2/plugins/PeopsSPU2/spunew.c

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Re-Added eol-style:native properties to the repository. The settings got lost when we merged from Playground to Official. Added interface.cpp (plugin/pcsx2 interface) and savestate.cpp to SPU2ghz, to help clean up SPU2.cpp. git-svn-id: http://pcsx2.googlecode.com/svn/trunk@463 96395faa-99c1-11dd-bbfe-3dabce05a288
2009-02-09 21:15:56 +00:00
/***************************************************************************
spu.c - description
-------------------
begin : Wed May 15 2002
copyright : (C) 2002 by Pete Bernert
email : BlackDove@addcom.de
***************************************************************************/
/***************************************************************************
* *
* This program is free software; you can redistribute it and/or modify *
* it under the terms of the GNU General Public License as published by *
* the Free Software Foundation; either version 2 of the License, or *
* (at your option) any later version. See also the license.txt file for *
* additional informations. *
* *
***************************************************************************/
//*************************************************************************//
// History of changes:
//
// 2005/08/29 - Pete
// - changed to 48Khz output
//
// 2004/12/25 - Pete
// - inc'd version for pcsx2-0.7
//
// 2004/04/18 - Pete
// - changed all kind of things in the plugin
//
// 2004/04/04 - Pete
// - changed plugin to emulate PS2 spu
//
// 2003/04/07 - Eric
// - adjusted cubic interpolation algorithm
//
// 2003/03/16 - Eric
// - added cubic interpolation
//
// 2003/03/01 - linuzappz
// - libraryName changes using ALSA
//
// 2003/02/28 - Pete
// - added option for type of interpolation
// - adjusted spu irqs again (Thousant Arms, Valkyrie Profile)
// - added MONO support for MSWindows DirectSound
//
// 2003/02/20 - kode54
// - amended interpolation code, goto GOON could skip initialization of gpos and cause segfault
//
// 2003/02/19 - kode54
// - moved SPU IRQ handler and changed sample flag processing
//
// 2003/02/18 - kode54
// - moved ADSR calculation outside of the sample decode loop, somehow I doubt that
// ADSR timing is relative to the frequency at which a sample is played... I guess
// this remains to be seen, and I don't know whether ADSR is applied to noise channels...
//
// 2003/02/09 - kode54
// - one-shot samples now process the end block before stopping
// - in light of removing fmod hack, now processing ADSR on frequency channel as well
//
// 2003/02/08 - kode54
// - replaced easy interpolation with gaussian
// - removed fmod averaging hack
// - changed .sinc to be updated from .iRawPitch, no idea why it wasn't done this way already (<- Pete: because I sometimes fail to see the obvious, haharhar :)
//
// 2003/02/08 - linuzappz
// - small bugfix for one usleep that was 1 instead of 1000
// - added iDisStereo for no stereo (Linux)
//
// 2003/01/22 - Pete
// - added easy interpolation & small noise adjustments
//
// 2003/01/19 - Pete
// - added Neill's reverb
//
// 2003/01/12 - Pete
// - added recording window handlers
//
// 2003/01/06 - Pete
// - added Neill's ADSR timings
//
// 2002/12/28 - Pete
// - adjusted spu irq handling, fmod handling and loop handling
//
// 2002/08/14 - Pete
// - added extra reverb
//
// 2002/06/08 - linuzappz
// - SPUupdate changed for SPUasync
//
// 2002/05/15 - Pete
// - generic cleanup for the Peops release
//
//*************************************************************************//
#include "stdafx.h"
#define _IN_SPU
#include "externals.h"
#include "cfg.h"
#include "dsoundoss.h"
#include "regs.h"
#include "debug.h"
#include "record.h"
#include "resource.h"
#include "dma.h"
#include "registers.h"
////////////////////////////////////////////////////////////////////////
// spu version infos/name
////////////////////////////////////////////////////////////////////////
const unsigned char version = 5;
const unsigned char revision = 1;
const unsigned char build = 6;
static char * libraryName = "P.E.Op.S. SPU2 DSound Driver";
static char * libraryInfo = "P.E.Op.S. SPU2 Driver V1.6\nCoded by Pete Bernert, Saqib and the P.E.Op.S. team\n";
////////////////////////////////////////////////////////////////////////
// globals
////////////////////////////////////////////////////////////////////////
// psx buffer / addresses
unsigned short regArea[32*1024];
short spuMem[2*1024*1024];
char * spuMemC;
unsigned char * pSpuIrq[2];
unsigned char * pSpuBuffer;
unsigned char * pSpuStreamBuffer[2];
// user settings
int iVolume=3;
int iDebugMode=0;
int iRecordMode=0;
int iUseReverb=0;
int iUseInterpolation=2;
// MAIN infos struct for each channel
SPUCHAN s_chan[MAXCHAN+1]; // channel + 1 infos (1 is security for fmod handling)
REVERBInfo rvb[2];
unsigned long dwNoiseVal=1; // global noise generator
int iSPUIRQWait=1;
unsigned short spuCtrl2[2]; // some vars to store psx reg infos
unsigned short spuStat2[2];
unsigned long spuIrq2[2];
unsigned long spuAddr2[2]; // address into spu mem
unsigned long spuRvbAddr2[2];
unsigned long spuRvbAEnd2[2];
int bEndThread=0; // thread handlers
int bSpuInit=0;
int bSPUIsOpen=0;
int bThreadEnded=0;
int iUseTimer=2;
int aSyncMode=0;
unsigned long aSyncCounter=0;
unsigned long aSyncWait=0;
DWORD aSyncTimerNew;
DWORD aSyncTimerOld;
HWND hWMain=0; // window handle
HWND hWDebug=0;
HWND hWRecord=0;
static HANDLE hMainThread;
unsigned long dwNewChannel2[2]; // flags for faster testing, if new channel starts
unsigned long dwEndChannel2[2];
void (CALLBACK *irqCallbackDMA4)()=0; // func of main emu, called on spu irq
void (CALLBACK *irqCallbackDMA7)()=0; // func of main emu, called on spu irq
void (CALLBACK *irqCallbackSPU2)()=0; // func of main emu, called on spu irq
// certain globals (were local before, but with the new timeproc I need em global)
const int f[5][2] = { { 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 } };
int SSumR[NSSIZE];
int SSumL[NSSIZE];
extern ADMA Adma4;
extern ADMA Adma7;
DINPUT DirectInputC0, DirectInputC1;
extern unsigned short interrupt;
int SPUCycles;
extern int SPUStartCycle[2];
extern int SPUTargetCycle[2];
int iCycle=0;
short * pS;
short * pS1;
static 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>=0x20000) // 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>=0x30000) // 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"
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
// START SOUND... called by main thread to setup a new sound on a channel
////////////////////////////////////////////////////////////////////////
INLINE void StartSound(int ch)
{
dwNewChannel2[ch/24]&=~(1<<(ch%24)); // clear new channel bit
dwEndChannel2[ch/24]&=~(1<<(ch%24)); // clear end channel bit
StartADSR(ch);
StartREVERB(ch);
s_chan[ch].pCurr=s_chan[ch].pStart; // set sample start
s_chan[ch].s_1=0; // init mixing vars
s_chan[ch].s_2=0;
s_chan[ch].iSBPos=28;
s_chan[ch].bNew=0; // init channel flags
s_chan[ch].bStop=0;
s_chan[ch].bOn=1;
s_chan[ch].SB[29]=0; // init our interpolation helpers
s_chan[ch].SB[30]=0;
if(iUseInterpolation>=2) // gauss interpolation?
{s_chan[ch].spos=0x30000L;s_chan[ch].SB[28]=0;} // -> start with more decoding
else {s_chan[ch].spos=0x10000L;s_chan[ch].SB[31]=0;} // -> no/simple interpolation starts with one 44100 decoding
}
void UpdateMainVolL() // LEFT VOLUME
{
short vol = regArea[PS2_C0_MVOLL];
if(vol&0x8000) // sweep?
{
short sInc=1; // -> sweep up?
if(vol&0x2000) sInc=-1; // -> or down?
if(vol&0x1000) vol^=0xffff; // -> mmm... phase inverted? have to investigate this
vol=((vol&0x7f)+1)/2; // -> sweep: 0..127 -> 0..64
vol+=vol/(2*sInc); // -> HACK: we don't sweep right now, so we just raise/lower the volume by the half!
vol*=128;
}
else // no sweep:
{
if(vol&0x4000) // -> mmm... phase inverted? have to investigate this
//vol^=0xffff;
vol=0x3fff-(vol&0x3fff);
}
vol&=0x3fff;
regArea[PS2_C0_MVOLL]=vol; // store volume
}
////////////////////////////////////////////////////////////////////////
// RIGHT VOLUME register write
////////////////////////////////////////////////////////////////////////
void UpdateMainVolR() // RIGHT VOLUME
{
short vol = regArea[PS2_C0_MVOLR];
if(vol&0x8000) // comments... see above :)
{
short sInc=1;
if(vol&0x2000) sInc=-1;
if(vol&0x1000) vol^=0xffff;
vol=((vol&0x7f)+1)/2;
vol+=vol/(2*sInc);
vol*=128;
}
else
{
if(vol&0x4000) //vol=vol^=0xffff;
vol=0x3fff-(vol&0x3fff);
}
vol&=0x3fff;
regArea[PS2_C0_MVOLR]=vol;
}
////////////////////////////////////////////////////////////////////////
// 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
extern unsigned long MemAddr[2];
////////////////////////////////////////////////////////////////////////
int iSpuAsyncWait=0;
extern int MMIXC0, MMIXC1;
VOID CALLBACK MAINProc(UINT nTimerId,UINT msg,DWORD dwUser,DWORD dwParam1, DWORD dwParam2)
{
int s_1,s_2,fa,ns;
int core = 0;
unsigned char * start;
unsigned int nSample;
int d;
int ch,predict_nr,shift_factor,flags,s;
int gpos,bIRQReturn=0;
while(!bEndThread) // until we are shutting down
{
//--------------------------------------------------//
// ok, at the beginning we are looking if there is
// enuff free place in the dsound/oss buffer to
// fill in new data, or if there is a new channel to start.
// if not, we wait (thread) or return (timer/spuasync)
// until enuff free place is available/a new channel gets
// started
if (aSyncMode==1) // Async supported? and enabled?
{
if (aSyncCounter<=737280) // If we have 10ms in the buffer, don't wait
{
if (aSyncWait<1000) Sleep(aSyncWait); // Wait a little to be more Synced (No more than 1 sec)
else Sleep (1000);
}
while (aSyncCounter<=368640 && !bEndThread && aSyncMode==1)
Sleep (1); // bEndThread/aSyncMode are needed, to avoid close problems
aSyncCounter -= 36864; // 1ms more done (48Hz*768cycles/Hz)
}
else
if(dwNewChannel2[0] || dwNewChannel2[1]) // new channel should start immedately?
{ // (at least one bit 0 ... MAXCHANNEL is set?)
iSecureStart++; // -> set iSecure
if(iSecureStart>5) iSecureStart=0; // (if it is set 5 times - that means on 5 tries a new samples has been started - in a row, we will reset it, to give the sound update a chance)
}
else iSecureStart=0; // 0: no new channel should start
while(!iSecureStart && !bEndThread && // no new start? no thread end?
(SoundGetBytesBuffered()>TESTSIZE)) // and still enuff data in sound buffer?
{
iSecureStart=0; // reset secure
if(iUseTimer) // no-thread mode?
{
return; // -> and done this time (timer mode 1 or 2)
}
// win thread mode:
Sleep(PAUSE_W); // sleep for x ms (win)
if(dwNewChannel2[0] || dwNewChannel2[1])
iSecureStart=1; // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop
}
//--------------------------------------------------// 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
if( s_chan[ch].iSBPos < 28 ) {
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
core = ch / 24; // Choose which core
if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq)
{
s_chan[ch].iUsedFreq = s_chan[ch].iActFreq; // -> take it and calc steps
s_chan[ch].sinc = s_chan[ch].iRawPitch<<4;
if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
if(iUseInterpolation==1) s_chan[ch].SB[32]=1; // -> freq change in simle imterpolation mode: set flag
}
ns=0;
while(ns<NSSIZE) // loop until 1 ms of data is reached
{
while(s_chan[ch].spos>=0x10000L)
{
if(s_chan[ch].iSBPos==28) // 28 reached?
{
start=s_chan[ch].pCurr; // set up the current pos
if (s_chan[ch].bOn==0) goto ENDX; // special "stop" sign
s_chan[ch].iSBPos=0;
s_1=s_chan[ch].s_1;
s_2=s_chan[ch].s_2;
predict_nr = (int)*start;
start++;
flags = (int)*start;
start++;
shift_factor = predict_nr&0xf;
predict_nr >>= 4;
// -------------------------------------- //
for (nSample=0;nSample<28;nSample+=2,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+1]=fa;
}
//////////////////////////////////////////// irq check
if(spuCtrl2[core]&0x40) // some irq active?
{
if(iDebugMode==1) logprintf("IRQ Active ch %x, C%x\r\n", ch, ch/24);
if((pSpuIrq[core] >= start-16 && // irq address reached?
pSpuIrq[core] <= start) ||
((flags&1) && // special: irq on looping addr, when stop/loop flag is set
(pSpuIrq[core] >= s_chan[ch].pLoop-16 &&
pSpuIrq[core] <= s_chan[ch].pLoop)))
{
s_chan[ch].iIrqDone=1; // -> debug flag
if(iDebugMode==1) logprintf("Sample End ch %x, C%x\r\n", ch, ch/24);
regArea[0x7C0] |= 0x4<<core;
irqCallbackSPU2(); // -> let's see what is happening if we call our irqs instead ;)
if(iSPUIRQWait) // -> option: wait after irq for main emu
{
iSpuAsyncWait=1;
bIRQReturn=1;
}
}
}
//////////////////////////////////////////// flag handler
if(flags & 0x2) s_chan[ch].bIgnoreLoop=1; // LOOP bit
if(flags&0x4) s_chan[ch].pLoop=start-16; // LOOP/START bit
if(flags&0x1) // 1: LOOP/END bit
{
dwEndChannel2[core]|=(1<<(ch%24));
if((flags&0xF) != 0x3) // Check if no loop is present
{
s_chan[ch].bIgnoreLoop=0;
s_chan[ch].bStop = 1;
logprintf("Stopping\r\n");
}
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()+2500;
while(iSpuAsyncWait && !bEndThread &&
timeGetTime()<dwWatchTime)
Sleep(1);
}
else
{
lastch=ch;
lastns=ns;
return;
}
}
////////////////////////////////////////////
GOON: ;
}
fa=s_chan[ch].SB[s_chan[ch].iSBPos++]; // get sample data
if((spuCtrl2[core]&0x4000)==0) fa=0; // muted?
else // else adjust
{
if(fa>32767L) fa=32767L;
if(fa<-32767L) fa=-32767L;
}
if(iUseInterpolation>=2) // gauss/cubic interpolation
{
gpos = s_chan[ch].SB[28];
gval0 = fa;
gpos = (gpos+1) & 3;
s_chan[ch].SB[28] = gpos;
}
else
if(iUseInterpolation==1) // simple interpolation
{
s_chan[ch].SB[28] = 0;
s_chan[ch].SB[29] = s_chan[ch].SB[30]; // -> helpers for simple linear interpolation: delay real val for two slots, and calc the two deltas, for a 'look at the future behaviour'
s_chan[ch].SB[30] = s_chan[ch].SB[31];
s_chan[ch].SB[31] = fa;
s_chan[ch].SB[32] = 1; // -> flag: calc new interolation
}
else s_chan[ch].SB[29]=fa; // no interpolation
s_chan[ch].spos -= 0x10000L;
}
////////////////////////////////////////////////
// noise handler... just produces some noise data
// surely wrong... and no noise frequency (spuCtrl&0x3f00) will be used...
// and sometimes the noise will be used as fmod modulation... pfff
if(s_chan[ch].bNoise)
{
if((dwNoiseVal<<=1)&0x80000000L)
{
dwNoiseVal^=0x0040001L;
fa=((dwNoiseVal>>2)&0x7fff);
fa=-fa;
}
else fa=(dwNoiseVal>>2)&0x7fff;
// mmm... depending on the noise freq we allow bigger/smaller changes to the previous val
fa=s_chan[ch].iOldNoise+((fa-s_chan[ch].iOldNoise)/((0x001f-((spuCtrl2[core]&0x3f00)>>9))+1));
if(fa>32767L) fa=32767L;
if(fa<-32767L) fa=-32767L;
s_chan[ch].iOldNoise=fa;
if(iUseInterpolation<2) // no gauss/cubic interpolation?
s_chan[ch].SB[29] = fa; // -> store noise val in "current sample" slot
} //----------------------------------------
else // NO NOISE (NORMAL SAMPLE DATA) HERE
{//------------------------------------------//
if(iUseInterpolation==3) // cubic interpolation
{
long xd;
xd = ((s_chan[ch].spos) >> 1)+1;
gpos = s_chan[ch].SB[28];
fa = gval(3) - 3*gval(2) + 3*gval(1) - gval0;
fa *= (xd - (2<<15)) / 6;
fa >>= 15;
fa += gval(2) - gval(1) - gval(1) + gval0;
fa *= (xd - (1<<15)) >> 1;
fa >>= 15;
fa += gval(1) - gval0;
fa *= xd;
fa >>= 15;
fa = fa + gval0;
}
//------------------------------------------//
else
if(iUseInterpolation==2) // gauss interpolation
{
int vl, vr;
vl = (s_chan[ch].spos >> 6) & ~3;
gpos = s_chan[ch].SB[28];
vr=(gauss[vl]*gval0)&~2047;
vr+=(gauss[vl+1]*gval(1))&~2047;
vr+=(gauss[vl+2]*gval(2))&~2047;
vr+=(gauss[vl+3]*gval(3))&~2047;
fa = vr>>11;
}
//------------------------------------------//
else
if(iUseInterpolation==1) // simple interpolation
{
if(s_chan[ch].sinc<0x10000L) // -> upsampling?
InterpolateUp(ch); // --> interpolate up
else InterpolateDown(ch); // --> else down
fa=s_chan[ch].SB[29];
}
//------------------------------------------//
else fa=s_chan[ch].SB[29]; // no interpolation
}
s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // add adsr
if(s_chan[ch].bFMod==2) // fmod freq channel
{
int NP=((32768L+s_chan[ch].sval)*s_chan[ch+1].iRawPitch)>>14;
if(NP>0x3fff) NP=0x3fff;
else if(NP<0x1) NP=0x1;
NP=(48000L*NP)>>12; // calc frequency ( 48hz )
s_chan[ch+1].iActFreq=NP;
s_chan[ch+1].iUsedFreq=NP;
s_chan[ch+1].sinc=(((NP/10)<<16)/48000); // check , was 4800
if(!s_chan[ch+1].sinc) s_chan[ch+1].sinc=1;
if(iUseInterpolation==1) // freq change in sipmle interpolation mode
s_chan[ch+1].SB[32]=1;
}
else
{
//////////////////////////////////////////////
// ok, left/right sound volume (ps2 volume goes from 0 ... 0x3fff)
if(s_chan[ch].iMute) s_chan[ch].sval=0; // debug mute
else
{
if(s_chan[ch].bVolumeL)
SSumL[ns]+=(s_chan[ch].sval*s_chan[ch].iLeftVolume)>>14;
if(s_chan[ch].bVolumeR)
SSumR[ns]+=(s_chan[ch].sval*s_chan[ch].iRightVolume)>>14;
}
//////////////////////////////////////////////
// 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 all channels (including reverb) into one buffer
for(ns=0;ns<NSSIZE;ns++)
{
DirectInputC0.Left = 0;
DirectInputC0.Right= 0;
DirectInputC1.Left = 0;
DirectInputC1.Right= 0;
if((regArea[PS2_C0_MMIX] & 0xC0) && regArea[PS2_C0_ADMAS] & 0x1 && !(spuCtrl2[0] & 0x30))
{
DirectInputC0.Left = ((short*)spuMem)[0x2000+Adma4.Index];
DirectInputC0.Right = ((short*)spuMem)[0x2200+Adma4.Index];
Adma4.Index +=1;
if(Adma4.Index == 128 || Adma4.Index == 384)
{
if(ADMAS4Write())
{
spuStat2[0]&=~0x80;
irqCallbackDMA4();
}
else MemAddr[0] += 1024;
}
if(Adma4.Index == 512){
Adma4.Index = 0;
}
}
if((regArea[PS2_C1_MMIX] & 0xC0) && regArea[PS2_C1_ADMAS] & 0x2 && !(spuCtrl2[1] & 0x30))
{
DirectInputC1.Left = ((short*)spuMem)[0x2400+Adma7.Index];
DirectInputC1.Right = ((short*)spuMem)[0x2600+Adma7.Index];
Adma7.Index +=1;
if(Adma7.Index == 128 || Adma7.Index == 384)
{
if(ADMAS7Write())
{
spuStat2[1]&=~0x80;
irqCallbackDMA7();
}else MemAddr[1] += 1024;
}
if(Adma7.Index == 512) Adma7.Index = 0;
}
SSumL[ns]+=MixREVERBLeft(ns,0);
SSumL[ns]+=MixREVERBLeft(ns,1);
if((regArea[PS2_C0_MMIX] & 0x80)) SSumL[ns] += (DirectInputC0.Left*(int)regArea[PS2_C0_BVOLL])>>16;
if((regArea[PS2_C1_MMIX] & 0x80)) SSumL[ns] += (DirectInputC1.Left*(int)regArea[PS2_C1_BVOLL])>>16;
UpdateMainVolL();
d=SSumL[ns]/iVolume;
SSumL[ns]=0;
*pS++=(d<-32767) ? -32767 : ((d>32767) ? 32767 : d);
SSumR[ns]+=MixREVERBRight(0);
SSumR[ns]+=MixREVERBRight(1);
if((regArea[PS2_C0_MMIX] & 0x40)) SSumR[ns] += (DirectInputC0.Right*(int)regArea[PS2_C0_BVOLR])>>16;
if((regArea[PS2_C1_MMIX] & 0x40)) SSumR[ns] += (DirectInputC1.Right*(int)regArea[PS2_C1_BVOLR])>>16;
UpdateMainVolR();
d=SSumR[ns]/iVolume;
SSumR[ns]=0;
*pS++=(d<-32767) ? -32767 : ((d>32767) ? 32767 : d);
}
InitREVERB();
//////////////////////////////////////////////////////
// feed the sound
// wanna have around 1/60 sec (16.666 ms) updates
if(iCycle++>16)
{
SoundFeedVoiceData((unsigned char*)pSpuBuffer,
((unsigned char *)pS)-
((unsigned char *)pSpuBuffer));
pS=(short *)pSpuBuffer;
iCycle=0;
}
}
// end of big main loop...
bThreadEnded=1;
return;
}
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
DWORD WINAPI MAINThreadEx(LPVOID lpParameter)
{
MAINProc(0,0,0,0,0);
return 0;
}
////////////////////////////////////////////////////////////////////////
// SPU ASYNC... even newer epsxe func
// 1 time every 'cycle' cycles... harhar
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2async(unsigned long cycle)
{
SPUCycles += cycle;
if(interrupt & (1<<2)){
if(SPUCycles - SPUStartCycle[1] >= SPUTargetCycle[1]){
interrupt &= ~(1<<2);
irqCallbackDMA7();
}
}
if(interrupt & (1<<1)){
if(SPUCycles - SPUStartCycle[0] >= SPUTargetCycle[0]){
interrupt &= ~(1<<1);
irqCallbackDMA4();
}
}
if(iSpuAsyncWait)
{
iSpuAsyncWait++;
if(iSpuAsyncWait<=64) return;
iSpuAsyncWait=0;
}
if(iDebugMode==2)
{
if(IsWindow(hWDebug)) DestroyWindow(hWDebug);
hWDebug=0;iDebugMode=0;
}
if(iRecordMode==2)
{
if(IsWindow(hWRecord)) DestroyWindow(hWRecord);
hWRecord=0;iRecordMode=0;
}
if(iUseTimer==0) // does the emu support SPUAsync, is it in thread mode, and in Thread Sync ON?
{
aSyncMode=1; // Ten, activate main function Sync system flag
aSyncTimerOld = aSyncTimerNew; // Recalculate, AsyncWait (ms)
aSyncTimerNew = timeGetTime();
aSyncWait =(unsigned int)((aSyncTimerNew - aSyncTimerOld)/2);
aSyncCounter += cycle ;
return;
}
if(iUseTimer==2) // special mode, only used in Linux by this spu (or if you enable the experimental Windows mode)
{
aSyncMode = 0;
if(!bSpuInit) return; // -> no init, no call
MAINProc(0,0,0,0,0); // -> experimental win mode... not really tested... don't like the drawbacks
}
}
////////////////////////////////////////////////////////////////////////
// INIT/EXIT STUFF
////////////////////////////////////////////////////////////////////////
////////////////////////////////////////////////////////////////////////
// SPUINIT: this func will be called first by the main emu
////////////////////////////////////////////////////////////////////////
#ifdef _WINDOWS
static HINSTANCE hIRE = NULL;
#endif
EXPORT_GCC long CALLBACK SPU2init(void)
{
spuMemC=(unsigned char *)spuMem; // just small setup
memset((void *)s_chan,0,MAXCHAN*sizeof(SPUCHAN));
memset(rvb,0,2*sizeof(REVERBInfo));
InitADSR();
if(hIRE==NULL) hIRE=LoadLibrary("Riched32.dll "); // needed for debug output
return 0;
}
////////////////////////////////////////////////////////////////////////
// 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));
pS=(short *)pSpuBuffer; // setup soundbuffer pointer
pS1=(short *)pSpuStreamBuffer[0]; // setup soundbuffer pointer
bEndThread=0; // init thread vars
bSpuInit=1; // flag: we are inited
bSpuInit=1; // flag: we are inited
#ifdef _WINDOWS
if(iUseTimer==0) // windows: use thread
{
//_beginthread(MAINThread,0,NULL);
DWORD dw;
hMainThread=CreateThread(NULL,0,MAINThreadEx,0,0,&dw);
SetThreadPriority(hMainThread,
//THREAD_PRIORITY_TIME_CRITICAL);
THREAD_PRIORITY_HIGHEST);
}
#endif
}
////////////////////////////////////////////////////////////////////////
// REMOVETIMER: kill threads/timers
////////////////////////////////////////////////////////////////////////
void RemoveTimer(void)
{
bEndThread=1; // raise flag to end thread
if(iUseTimer!=2) // windows thread?
{
while(!bThreadEnded) {Sleep(5L);} // -> wait till thread has ended
Sleep(5L);
}
bSpuInit=0;
bThreadEnded=0; // no more spu is running
}
////////////////////////////////////////////////////////////////////////
// SETUPSTREAMS: init most of the spu buffers
////////////////////////////////////////////////////////////////////////
void SetupStreams(void)
{
int i;
pSpuBuffer=(unsigned char *)malloc(38400); // alloc mixing buffer
i=NSSIZE*2;
sRVBStart[0] = (int *)malloc(i*4); // alloc reverb buffer
memset(sRVBStart[0],0,i*4);
sRVBEnd[0] = sRVBStart[0] + i;
sRVBPlay[0] = sRVBStart[0];
sRVBStart[1] = (int *)malloc(i*4); // alloc reverb buffer
memset(sRVBStart[1],0,i*4);
sRVBEnd[1] = sRVBStart[1] + i;
sRVBPlay[1] = sRVBStart[1];
for(i=0;i<MAXCHAN;i++) // loop sound channels
{
// we don't use mutex sync... not needed, would only
// slow us down:
// s_chan[i].hMutex=CreateMutex(NULL,FALSE,NULL);
s_chan[i].ADSRX.SustainLevel = 1024; // -> init sustain
s_chan[i].iMute=0;
s_chan[i].iIrqDone=0;
s_chan[i].pLoop=spuMemC+(s_chan[i].iStartAdr<<1);
s_chan[i].pStart=spuMemC+(s_chan[i].iStartAdr<<1);
s_chan[i].pCurr=spuMemC+(s_chan[i].iStartAdr<<1);
}
}
////////////////////////////////////////////////////////////////////////
// REMOVESTREAMS: free most buffer
////////////////////////////////////////////////////////////////////////
void RemoveStreams(void)
{
free(pSpuBuffer); // free mixing buffer
pSpuBuffer=NULL;
free(sRVBStart[0]); // free reverb buffer
sRVBStart[0]=0;
free(sRVBStart[1]); // free reverb buffer
sRVBStart[1]=0;
}
////////////////////////////////////////////////////////////////////////
// SPUOPEN: called by main emu after init
////////////////////////////////////////////////////////////////////////
#include <stdio.h>
FILE * LogFile;
EXPORT_GCC long CALLBACK SPU2open(void* pWindow)
{
#ifdef _WINDOWS
HWND hW= pWindow == NULL ? NULL : *(HWND*)pWindow;
#endif
if(bSPUIsOpen) return 0; // security for some stupid main emus
LogFile = fopen("Logs/spu2.txt","wb");
iVolume=3;
bEndThread=0;
bThreadEnded=0;
spuMemC=(unsigned char *)spuMem;
memset((void *)s_chan,0,(MAXCHAN+1)*sizeof(SPUCHAN));
pSpuIrq[0]=spuMemC;
pSpuIrq[1]=spuMemC;
dwNewChannel2[0]=0;
dwNewChannel2[1]=0;
dwEndChannel2[0]=0;
dwEndChannel2[1]=0;
spuCtrl2[0]=0;
spuCtrl2[1]=0;
spuStat2[0]=0;
spuStat2[1]=0;
spuIrq2[0]=0;
spuIrq2[1]=0;
spuAddr2[0]=0x0;
spuAddr2[1]=0x0;
spuRvbAddr2[0]=0;
spuRvbAddr2[1]=0;
spuRvbAEnd2[0]=0;
spuRvbAEnd2[1]=0;
memset(&Adma4,0,sizeof(ADMA));
memset(&Adma7,0,sizeof(ADMA));
memset(&DirectInputC0,0,sizeof(DINPUT));
memset(&DirectInputC1,0,sizeof(DINPUT));
LastWrite=0x00000000;LastPlay=0; // init some play vars
if(!IsWindow(hW)) hW=GetActiveWindow();
hWMain = hW; // store hwnd
ReadConfig(); // read user stuff
SetupSound(); // setup midas (before init!)
SetupStreams(); // prepare streaming
SetupTimer(); // timer for feeding data
bSPUIsOpen=1;
if(iDebugMode) // windows debug dialog
{
hWDebug=CreateDialog(hInst,MAKEINTRESOURCE(IDD_DEBUG),
NULL,(DLGPROC)DebugDlgProc);
SetWindowPos(hWDebug,HWND_TOPMOST,0,0,0,0,SWP_NOMOVE|SWP_NOSIZE|SWP_SHOWWINDOW|SWP_NOACTIVATE);
UpdateWindow(hWDebug);
SetFocus(hWMain);
}
if(iRecordMode) // windows recording dialog
{
hWRecord=CreateDialog(hInst,MAKEINTRESOURCE(IDD_RECORD),
NULL,(DLGPROC)RecordDlgProc);
SetWindowPos(hWRecord,HWND_TOPMOST,0,0,0,0,SWP_NOMOVE|SWP_NOSIZE|SWP_SHOWWINDOW|SWP_NOACTIVATE);
UpdateWindow(hWRecord);
SetFocus(hWMain);
}
return 0;
}
////////////////////////////////////////////////////////////////////////
// not used yet
#ifndef _WINDOWS
void SPU2setConfigFile(char * pCfg)
{
pConfigFile=pCfg;
}
#endif
////////////////////////////////////////////////////////////////////////
// SPUCLOSE: called before shutdown
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2close(void)
{
if(!bSPUIsOpen) return; // some security
bSPUIsOpen=0; // no more open
//fclose(LogFile);
#ifdef _WINDOWS
if(IsWindow(hWDebug)) DestroyWindow(hWDebug);
hWDebug=0;
if(IsWindow(hWRecord)) DestroyWindow(hWRecord);
hWRecord=0;
#endif
RemoveTimer(); // no more feeding
RemoveSound(); // no more sound handling
RemoveStreams(); // no more streaming
}
////////////////////////////////////////////////////////////////////////
// SPUSHUTDOWN: called by main emu on final exit
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2shutdown(void)
{
#ifdef _WINDOWS
if(hIRE!=NULL) {FreeLibrary(hIRE);hIRE=NULL;}
#endif
return;
}
////////////////////////////////////////////////////////////////////////
// SPUTEST: we don't test, we are always fine ;)
////////////////////////////////////////////////////////////////////////
EXPORT_GCC long CALLBACK SPU2test(void)
{
return 0;
}
////////////////////////////////////////////////////////////////////////
// SPUCONFIGURE: call config dialog
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2configure(void)
{
#ifdef _WINDOWS
DialogBox(hInst,MAKEINTRESOURCE(IDD_CFGDLG),
GetActiveWindow(),(DLGPROC)DSoundDlgProc);
#else
StartCfgTool("CFG");
#endif
}
////////////////////////////////////////////////////////////////////////
// SPUABOUT: show about window
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2about(void)
{
#ifdef _WINDOWS
DialogBox(hInst,MAKEINTRESOURCE(IDD_ABOUT),
GetActiveWindow(),(DLGPROC)AboutDlgProc);
#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
////////////////////////////////////////////////////////////////////////
EXPORT_GCC void CALLBACK SPU2irqCallback(void (CALLBACK *SPU2callback)(int),
void (CALLBACK *DMA4callback)(int),
void (CALLBACK *DMA7callback)(int))
{
irqCallbackSPU2 = SPU2callback;
irqCallbackDMA4 = DMA4callback;
irqCallbackDMA7 = DMA7callback;
}
////////////////////////////////////////////////////////////////////////
// COMMON PLUGIN INFO FUNCS
////////////////////////////////////////////////////////////////////////
EXPORT_GCC char * CALLBACK PS2EgetLibName(void)
{
return libraryName;
}
#define PS2E_LT_SPU2 0x4
EXPORT_GCC unsigned long CALLBACK PS2EgetLibType(void)
{
return PS2E_LT_SPU2;
}
EXPORT_GCC unsigned long CALLBACK PS2EgetLibVersion2(unsigned long type)
{
unsigned char v=version;
// key hack to fake a lower version:
//if(GetAsyncKeyState(VK_SHIFT)&0x8000) v--;
// compile hack to set lib version to PCSX2 0.6 standards
//v=2;
return v<<16|revision<<8|build;
}
////////////////////////////////////////////////////////////////////////