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-rwxr-xr-x[-rw-r--r--]plugins/dfsound/spu.c2980
1 files changed, 1490 insertions, 1490 deletions
diff --git a/plugins/dfsound/spu.c b/plugins/dfsound/spu.c
index 8ee6a820..4dcb00b5 100644..100755
--- a/plugins/dfsound/spu.c
+++ b/plugins/dfsound/spu.c
@@ -1,1490 +1,1490 @@
-/***************************************************************************
- spu.c - description
- -------------------
- begin : Wed May 15 2002
- copyright : (C) 2002 by Pete Bernert
- email : BlackDove@addcom.de
- ***************************************************************************/
-/***************************************************************************
- * *
- * This program is free software; you can redistribute it and/or modify *
- * it under the terms of the GNU General Public License as published by *
- * the Free Software Foundation; either version 2 of the License, or *
- * (at your option) any later version. See also the license.txt file for *
- * additional informations. *
- * *
- ***************************************************************************/
-
-#include "stdafx.h"
-
-#define _IN_SPU
-
-#include "externals.h"
-#include "cfg.h"
-#include "dsoundoss.h"
-#include "regs.h"
-
-#ifdef _WINDOWS
-#include "debug.h"
-#include "record.h"
-#elif defined(_MACOSX)
-#include "maccfg.h"
-#endif
-
-#ifdef ENABLE_NLS
-#include <libintl.h>
-#include <locale.h>
-#define _(x) gettext(x)
-#define N_(x) (x)
-//If running under Mac OS X, use the Localizable.strings file instead.
-#elif defined(_MACOSX)
-#ifdef PCSXRCORE
-extern char* Pcsxr_locale_text(char* toloc);
-#define _(String) Pcsxr_locale_text(String)
-#define N_(String) String
-#else
-#ifndef PCSXRPLUG
-#warning please define the plug being built to use Mac OS X localization!
-#define _(msgid) msgid
-#define N_(msgid) msgid
-#else
-//Kludge to get the preprocessor to accept PCSXRPLUG as a variable.
-#define PLUGLOC_x(x,y) x ## y
-#define PLUGLOC_y(x,y) PLUGLOC_x(x,y)
-#define PLUGLOC PLUGLOC_y(PCSXRPLUG,_locale_text)
-extern char* PLUGLOC(char* toloc);
-#define _(String) PLUGLOC(String)
-#define N_(String) String
-#endif
-#endif
-#else
-#define _(x) (x)
-#define N_(x) (x)
-#endif
-
-#if defined (_WINDOWS)
-static char * libraryName = N_("DirectSound Driver");
-#elif 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 (USEOPENAL)
-static char * libraryName = N_("OpenAL Sound");
-#elif defined (USEPULSEAUDIO)
-static char * libraryName = N_("PulseAudio Sound");
-#else
-static char * libraryName = N_("NULL Sound");
-#endif
-
-static char * libraryInfo = N_("P.E.Op.S. Sound Driver V1.7\nCoded by Pete Bernert and the P.E.Op.S. team\n");
-
-// globals
-
-// psx buffer / addresses
-
-unsigned short regArea[10000];
-unsigned short spuMem[256*1024];
-unsigned char * spuMemC;
-unsigned char * pSpuIrq=0;
-unsigned char * pSpuBuffer;
-unsigned char * pMixIrq=0;
-
-// user settings
-
-int iVolume=3;
-int iXAPitch=1;
-int iUseTimer=2;
-int iSPUIRQWait=1;
-int iDebugMode=0;
-int iRecordMode=0;
-int iUseReverb=2;
-int iUseInterpolation=2;
-int iDisStereo=0;
-int iFreqResponse=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
-unsigned long dwNoiseCount; // global noise generator
-unsigned long dwNoiseClock; // global noise generator
-int iSpuAsyncWait=0;
-
-unsigned int decoded_ptr = 0;
-unsigned int bIrqHit = 0;
-
-unsigned short spuCtrl=0; // some vars to store psx reg infos
-unsigned short spuStat=0;
-unsigned short spuIrq=0;
-unsigned long spuAddr=0xffffffff; // address into spu mem
-int bEndThread=0; // thread handlers
-int bThreadEnded=0;
-int bSpuInit=0;
-int bSPUIsOpen=0;
-
-#ifdef _WINDOWS
-HWND hWMain=0; // window handle
-HWND hWDebug=0;
-HWND hWRecord=0;
-static HANDLE hMainThread;
-#else
-static pthread_t thread = (pthread_t)-1; // thread id (linux)
-#endif
-
-uint32_t dwNewChannel=0; // flags for faster testing, if new channel starts
-
-void (CALLBACK *irqCallback)(void)=0; // func of main emu, called on spu irq
-void (CALLBACK *cddavCallback)(unsigned short,unsigned short)=0;
-
-// certain globals (were local before, but with the new timeproc I need em global)
-
-static const int f[5][2] = { { 0, 0 },
- { 60, 0 },
- { 115, -52 },
- { 98, -55 },
- { 122, -60 } };
-int SSumR[NSSIZE];
-int SSumL[NSSIZE];
-int iFMod[NSSIZE];
-int iCycle = 0;
-short * pS;
-
-int lastch=-1; // last channel processed on spu irq in timer mode
-static int lastns=0; // last ns pos
-static int iSecureStart=0; // secure start counter
-
-////////////////////////////////////////////////////////////////////////
-// CODE AREA
-////////////////////////////////////////////////////////////////////////
-
-// dirty inline func includes
-
-#include "reverb.c"
-#include "adsr.c"
-
-////////////////////////////////////////////////////////////////////////
-// helpers for simple interpolation
-
-//
-// easy interpolation on upsampling, no special filter, just "Pete's common sense" tm
-//
-// instead of having n equal sample values in a row like:
-// ____
-// |____
-//
-// we compare the current delta change with the next delta change.
-//
-// if curr_delta is positive,
-//
-// - and next delta is smaller (or changing direction):
-// \.
-// -__
-//
-// - and next delta significant (at least twice) bigger:
-// --_
-// \.
-//
-// - and next delta is nearly same:
-// \.
-// \.
-//
-//
-// if curr_delta is negative,
-//
-// - and next delta is smaller (or changing direction):
-// _--
-// /
-//
-// - and next delta significant (at least twice) bigger:
-// /
-// __-
-//
-// - and next delta is nearly same:
-// /
-// /
-//
-
-
-static 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
-//
-
-static INLINE void InterpolateDown(int ch)
-{
- if(s_chan[ch].sinc>=0x20000L) // we would skip at least one val?
- {
- s_chan[ch].SB[29]+=(s_chan[ch].SB[30]-s_chan[ch].SB[29])/2; // add easy weight
- if(s_chan[ch].sinc>=0x30000L) // we would skip even more vals?
- s_chan[ch].SB[29]+=(s_chan[ch].SB[31]-s_chan[ch].SB[30])/2;// add additional next weight
- }
-}
-
-////////////////////////////////////////////////////////////////////////
-// helpers for gauss interpolation
-
-#define gval0 (((short*)(&s_chan[ch].SB[29]))[gpos])
-#define gval(x) (((short*)(&s_chan[ch].SB[29]))[(gpos+x)&3])
-
-#include "gauss_i.h"
-
-////////////////////////////////////////////////////////////////////////
-
-#include "xa.c"
-
-////////////////////////////////////////////////////////////////////////
-// START SOUND... called by main thread to setup a new sound on a channel
-////////////////////////////////////////////////////////////////////////
-
-static 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].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].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
-////////////////////////////////////////////////////////////////////////
-
-static INLINE void VoiceChangeFrequency(int ch)
-{
- s_chan[ch].iUsedFreq=s_chan[ch].iActFreq; // -> take it and calc steps
- s_chan[ch].sinc=s_chan[ch].iRawPitch<<4;
- if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
- if(iUseInterpolation==1) s_chan[ch].SB[32]=1; // -> freq change in simle imterpolation mode: set flag
-}
-
-////////////////////////////////////////////////////////////////////////
-
-static INLINE void FModChangeFrequency(int ch,int ns)
-{
- int NP=s_chan[ch].iRawPitch;
-
- NP=((32768L+iFMod[ns])*NP)/32768L;
-
- if(NP>0x3fff) NP=0x3fff;
- if(NP<0x1) NP=0x1;
-
- NP=(44100L*NP)/(4096L); // calc frequency
-
- s_chan[ch].iActFreq=NP;
- s_chan[ch].iUsedFreq=NP;
- s_chan[ch].sinc=(((NP/10)<<16)/4410);
- if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
- if(iUseInterpolation==1) // freq change in simple interpolation mode
- s_chan[ch].SB[32]=1;
- iFMod[ns]=0;
-}
-
-////////////////////////////////////////////////////////////////////////
-
-/*
-Noise Algorithm
-- Dr.Hell (Xebra PS1 emu)
-- 100% accurate (waveform + frequency)
-- http://drhell.web.fc2.com
-
-
-Level change cycle
-Freq = 0x8000 >> (NoiseClock >> 2);
-
-Frequency of half cycle
-Half = ((NoiseClock & 3) * 2) / (4 + (NoiseClock & 3));
-- 0 = (0*2)/(4+0) = 0/4
-- 1 = (1*2)/(4+1) = 2/5
-- 2 = (2*2)/(4+2) = 4/6
-- 3 = (3*2)/(4+3) = 6/7
-
--------------------------------
-
-5*6*7 = 210
-4 - 0*0 = 0
-5 - 42*2 = 84
-6 - 35*4 = 140
-7 - 30*6 = 180
-*/
-
-// Noise Waveform - Dr. Hell (Xebra)
-char NoiseWaveAdd [64] = {
- 1, 0, 0, 1, 0, 1, 1, 0,
- 1, 0, 0, 1, 0, 1, 1, 0,
- 1, 0, 0, 1, 0, 1, 1, 0,
- 1, 0, 0, 1, 0, 1, 1, 0,
- 0, 1, 1, 0, 1, 0, 0, 1,
- 0, 1, 1, 0, 1, 0, 0, 1,
- 0, 1, 1, 0, 1, 0, 0, 1,
- 0, 1, 1, 0, 1, 0, 0, 1
-};
-
-unsigned short NoiseFreqAdd[5] = {
- 0, 84, 140, 180, 210
-};
-
-static INLINE void NoiseClock()
-{
- unsigned int level;
-
- level = 0x8000 >> (dwNoiseClock >> 2);
- level <<= 16;
-
- dwNoiseCount += 0x10000;
-
- // Dr. Hell - fraction
- dwNoiseCount += NoiseFreqAdd[ dwNoiseClock & 3 ];
- if( (dwNoiseCount&0xffff) >= NoiseFreqAdd[4] ) {
- dwNoiseCount += 0x10000;
- dwNoiseCount -= NoiseFreqAdd[ dwNoiseClock & 3 ];
- }
-
- if( dwNoiseCount >= level )
- {
- while( dwNoiseCount >= level )
- dwNoiseCount -= level;
-
- // Dr. Hell - form
- dwNoiseVal = (dwNoiseVal<<1) | NoiseWaveAdd[ (dwNoiseVal>>10) & 63 ];
- }
-}
-
-static INLINE int iGetNoiseVal(int ch)
-{
- int fa;
-
- fa = (short) dwNoiseVal;
-
- // no clip need
- //if(fa>32767L) fa=32767L;
- //if(fa<-32767L) fa=-32767L;
-
- // don't upset VAG decoder
- //if(iUseInterpolation<2) // no gauss/cubic interpolation?
- //pChannel->SB[29] = fa; // -> store noise val in "current sample" slot
-
- // boost volume - no more!
- //return fa * 3 / 2;
- return fa;
-}
-
-////////////////////////////////////////////////////////////////////////
-
-static INLINE void StoreInterpolationVal(int ch,int fa)
-{
- /*
- // fmod channel = sound output
- if(s_chan[ch].bFMod==2) // fmod freq channel
- s_chan[ch].SB[29]=fa;
- else
- */
- {
- if((spuCtrl&0x4000)==0) fa=0; // muted?
- else // else adjust
- {
- if(fa>32767L) fa=32767L;
- if(fa<-32767L) fa=-32767L;
- }
-
- if(iUseInterpolation>=2) // gauss/cubic interpolation
- {
- int gpos = s_chan[ch].SB[28];
- gval0 = fa;
- gpos = (gpos+1) & 3;
- s_chan[ch].SB[28] = gpos;
- }
- else
- if(iUseInterpolation==1) // simple interpolation
- {
- s_chan[ch].SB[28] = 0;
- s_chan[ch].SB[29] = s_chan[ch].SB[30]; // -> helpers for simple linear interpolation: delay real val for two slots, and calc the two deltas, for a 'look at the future behaviour'
- s_chan[ch].SB[30] = s_chan[ch].SB[31];
- s_chan[ch].SB[31] = fa;
- s_chan[ch].SB[32] = 1; // -> flag: calc new interolation
- }
- else s_chan[ch].SB[29]=fa; // no interpolation
- }
-}
-
-////////////////////////////////////////////////////////////////////////
-
-static INLINE int iGetInterpolationVal(int ch)
-{
- int fa;
-
- // fmod channel = sound output
- //if(s_chan[ch].bFMod==2) return s_chan[ch].SB[29];
-
- switch(iUseInterpolation)
- {
- //--------------------------------------------------//
- case 3: // cubic interpolation
- {
- long xd;int gpos;
- xd = ((s_chan[ch].spos) >> 1)+1;
- gpos = s_chan[ch].SB[28];
-
- fa = gval(3) - 3*gval(2) + 3*gval(1) - gval0;
- fa *= (xd - (2<<15)) / 6;
- fa >>= 15;
- fa += gval(2) - gval(1) - gval(1) + gval0;
- fa *= (xd - (1<<15)) >> 1;
- fa >>= 15;
- fa += gval(1) - gval0;
- fa *= xd;
- fa >>= 15;
- fa = fa + gval0;
-
- } break;
- //--------------------------------------------------//
- case 2: // gauss interpolation
- {
- int vl, vr;int gpos;
- vl = (s_chan[ch].spos >> 6) & ~3;
- gpos = s_chan[ch].SB[28];
- vr=(gauss[vl]*gval0)&~2047;
- vr+=(gauss[vl+1]*gval(1))&~2047;
- vr+=(gauss[vl+2]*gval(2))&~2047;
- vr+=(gauss[vl+3]*gval(3))&~2047;
- fa = vr>>11;
- } break;
- //--------------------------------------------------//
- case 1: // simple interpolation
- {
- if(s_chan[ch].sinc<0x10000L) // -> upsampling?
- InterpolateUp(ch); // --> interpolate up
- else InterpolateDown(ch); // --> else down
- fa=s_chan[ch].SB[29];
- } break;
- //--------------------------------------------------//
- default: // no interpolation
- {
- fa=s_chan[ch].SB[29];
- } break;
- //--------------------------------------------------//
- }
-
- return fa;
-}
-
-////////////////////////////////////////////////////////////////////////
-// MAIN SPU FUNCTION
-// here is the main job handler... thread, timer or direct func call
-// basically the whole sound processing is done in this fat func!
-////////////////////////////////////////////////////////////////////////
-
-// 5 ms waiting phase, if buffer is full and no new sound has to get started
-// .. can be made smaller (smallest val: 1 ms), but bigger waits give
-// better performance
-
-#define PAUSE_W 1
-#define PAUSE_L 1000
-
-////////////////////////////////////////////////////////////////////////
-
-#ifdef _WINDOWS
-static VOID CALLBACK MAINProc(UINT nTimerId, UINT msg, DWORD dwUser, DWORD dwParam1, DWORD dwParam2)
-#else
-static void *MAINThread(void *arg)
-#endif
-{
- int s_1,s_2,fa,ns;
- int voldiv = iVolume;
-
- unsigned char * start;unsigned int nSample;
- int ch,predict_nr,shift_factor,flags,d,s;
- int bIRQReturn=0;
- unsigned int decoded_voice=0;
-
- // mute output
- if( voldiv == 5 ) voldiv = 0x7fffffff;
-
- while(!bEndThread) // until we are shutting down
- {
- // ok, at the beginning we are looking if there is
- // enuff free place in the dsound/oss buffer to
- // fill in new data, or if there is a new channel to start.
- // if not, we wait (thread) or return (timer/spuasync)
- // until enuff free place is available/a new channel gets
- // started
-
- if(dwNewChannel) // new channel should start immedately?
- { // (at least one bit 0 ... MAXCHANNEL is set?)
- iSecureStart++; // -> set iSecure
- if(iSecureStart>1) 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
-
-#ifdef _WINDOWS
- if(iUseTimer) // no-thread mode?
- {
- if(iUseTimer==1) // -> ok, timer mode 1: setup a oneshot timer of x ms to wait
- timeSetEvent(PAUSE_W,1,MAINProc,0,TIME_ONESHOT);
- return; // -> and done this time (timer mode 1 or 2)
- }
- // win thread mode:
- Sleep(PAUSE_W); // sleep for x ms (win)
-#else
- if(iUseTimer) return 0; // linux no-thread mode? bye
- usleep(PAUSE_L); // else sleep for x ms (linux)
-#endif
-
- if(dwNewChannel) iSecureStart=1; // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop
- }
-
- //--------------------------------------------------// continue from irq handling in timer mode?
-
- if(lastch>=0) // will be -1 if no continue is pending
- {
- ch=lastch; ns=lastns; lastch=-1; // -> setup all kind of vars to continue
- goto GOON; // -> directly jump to the continue point
- }
-
- //--------------------------------------------------//
- //- main channel loop -//
- //--------------------------------------------------//
- {
- ns=0;
- decoded_voice = decoded_ptr;
-
- while(ns<NSSIZE) // loop until 1 ms of data is reached
- {
- SSumL[ns]=0;
- SSumR[ns]=0;
-
-
- // decoded buffer values - dummy
- spuMem[ (0x000 + decoded_voice) / 2 ] = (short) 0;
- spuMem[ (0x400 + decoded_voice) / 2 ] = (short) 0;
- spuMem[ (0x800 + decoded_voice) / 2 ] = (short) 0;
- spuMem[ (0xc00 + decoded_voice) / 2 ] = (short) 0;
-
-
- NoiseClock();
-
- 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) {
-#if 1
- StartSound(ch); // start new sound
- dwNewChannel&=~(1<<ch); // clear new channel bit
-#else
- if( s_chan[ch].ADSRX.StartDelay == 0 ) {
- StartSound(ch); // start new sound
- dwNewChannel&=~(1<<ch); // clear new channel bit
- } else {
- s_chan[ch].ADSRX.StartDelay--;
- }
-#endif
- }
- if(!s_chan[ch].bOn) continue; // channel not playing? next
-
- if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq) // new psx frequency?
- VoiceChangeFrequency(ch);
-
- if(s_chan[ch].bFMod==1 && iFMod[ns]) // fmod freq channel
- FModChangeFrequency(ch,ns);
-
- while(s_chan[ch].spos>=0x10000L)
- {
- if(s_chan[ch].iSBPos==28) // 28 reached?
- {
- // Xenogears - Anima Relic dungeon (exp gain)
- if( s_chan[ch].bLoopJump == 1 )
- s_chan[ch].pCurr = s_chan[ch].pLoop;
-
- s_chan[ch].bLoopJump = 0;
-
-
- start=s_chan[ch].pCurr; // set up the current pos
-
- if (s_chan[ch].iSilent==1 || start == (unsigned char*)-1) // special "stop" sign
- {
- // silence = let channel keep running (IRQs)
- //s_chan[ch].bOn=0; // -> turn everything off
- s_chan[ch].iSilent=2;
-
- s_chan[ch].ADSRX.lVolume=0;
- s_chan[ch].ADSRX.EnvelopeVol=0;
- }
-
- 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++;
-
- // Silhouette Mirage - Serah fight
- if( predict_nr > 4 ) predict_nr = 0;
-
- // -------------------------------------- //
-
- 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);
-
- // snes brr clamps
- fa = CLAMP16(fa);
-
- 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);
-
- // snes brr clamps
- fa = CLAMP16(fa);
-
- s_2=s_1;s_1=fa;
-
- s_chan[ch].SB[nSample++]=fa;
- }
-
- //////////////////////////////////////////// irq check
-
-#if 1
- // ?? (-8)
- if( Check_IRQ( (start-spuMemC)-8, 0 ) ||
- Check_IRQ( (start-spuMemC)-0, 0 ) )
- {
-#else
- 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)))
-#endif
- {
- 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
-
- /*
- SPU2-X:
- $4 = set loop to current block
- $2 = keep envelope on (no mute)
- $1 = jump to loop address
-
- silence means no volume (ADSR keeps playing!!)
- */
-
- if(flags&4)
- s_chan[ch].pLoop=start-16;
-
-
- // Jungle Book - Rhythm 'n Groove - don't reset ignore status
- // - fixes gameplay speed (IRQ hits)
- //s_chan[ch].bIgnoreLoop = 0;
-
-
- if(flags&1)
- {
- // ...?
- //s_chan[ch].bIgnoreLoop = 0;
-
- // Xenogears - 7 = play missing sounds
- // set jump flag
- s_chan[ch].bLoopJump = 1;
-
-
- // silence = keep playing..?
- if( (flags&2) == 0 ) {
- s_chan[ch].iSilent = 1;
-
- // silence = don't start release phase
- //s_chan[ch].bStop = 1;
-
- //start = (unsigned char *) -1;
- }
- }
-
-#if 0
- // crash check
- if( start == 0 )
- start = (unsigned char *) -1;
- if( start >= spuMemC + 0x80000 )
- start = spuMemC - 0x80000;
-#endif
-
-
- // Silhouette Mirage - ending mini-game
-
- // ??
- if( start - spuMemC >= 0x80000 ) {
- start -= 16;
-
- s_chan[ch].iSilent = 1;
- s_chan[ch].bStop = 1;
- }
-
-
- 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)
-#ifdef _WINDOWS
- Sleep(1);
-#else
- usleep(1000L);
-#endif
- }
- else
- {
- lastch=ch;
- lastns=ns;
-
-#ifdef _WINDOWS
- return;
-#else
- return 0;
-#endif
- }
- }
-
-GOON: ;
- }
-
- fa=s_chan[ch].SB[s_chan[ch].iSBPos++]; // get sample data
-
- StoreInterpolationVal(ch,fa); // store val for later interpolation
-
- s_chan[ch].spos -= 0x10000L;
- }
-
- if(s_chan[ch].bNoise)
- fa=iGetNoiseVal(ch); // get noise val
- else fa=iGetInterpolationVal(ch); // get sample val
-
-
- // Voice 1/3 decoded buffer
- if( ch == 0 ) {
- spuMem[ (0x800 + decoded_voice) / 2 ] = (short) fa;
- } else if( ch == 2 ) {
- spuMem[ (0xc00 + decoded_voice) / 2 ] = (short) fa;
- }
-
-
- s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // mix adsr
-
- if(s_chan[ch].bFMod==2) // fmod freq channel
- iFMod[ns]=s_chan[ch].sval; // -> store 1T sample data, use that to do fmod on next channel
-
- // mix fmod channel into output
- // - Xenogears save icon (high pitch)
- {
- //////////////////////////////////////////////
- // 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);
- }
-
- s_chan[ch].spos += s_chan[ch].sinc;
- }
-
- ////////////////////////////////////////////////
- // ok, go on until 1 ms data of this channel is collected
-
- // decoded buffer - voice
- decoded_voice += 2;
- decoded_voice &= 0x400;
-
-
- // status flag
- if( decoded_voice >= 0x200 ) {
- spuStat |= STAT_DECODED;
- } else {
- spuStat &= ~STAT_DECODED;
- }
-
-
- // IRQ work
- {
- unsigned char *old_irq;
- unsigned int old_ptr;
-
- old_irq = pSpuIrq;
- old_ptr = decoded_voice;
-
-#if 0
- // align to boundaries ($0, $200, $400, $600)
- pSpuIrq = ((pSpuIrq - spuMemC) & (~0x1ff)) + spuMemC;
- decoded_voice = decoded_voice & (~0x1ff);
-#endif
-
- // check all decoded buffer IRQs - timing issue
- Check_IRQ( decoded_voice + 0x000, 0 );
- Check_IRQ( decoded_voice + 0x400, 0 );
- Check_IRQ( decoded_voice + 0x800, 0 );
- Check_IRQ( decoded_voice + 0xc00, 0 );
-
- pSpuIrq = old_irq;
- decoded_voice = old_ptr;
- }
-
-
- ns++;
- } // end ns
- }
-
-
- //---------------------------------------------------//
- //- here we have another 1 ms of sound data
- //---------------------------------------------------//
- // mix XA infos (if any)
-
- MixXA();
-
-
- // now safe to update deocded buffer ptr
- decoded_ptr += ns * 2;
- decoded_ptr &= 0x400;
-
-
- ///////////////////////////////////////////////////////
- // mix all channels (including reverb) into one buffer
-
- if(iDisStereo) // no stereo?
- {
- int dl, dr;
- for (ns = 0; ns < NSSIZE; ns++)
- {
- SSumL[ns] += MixREVERBLeft(ns);
-
- dl = SSumL[ns] / voldiv; SSumL[ns] = 0;
- if (dl < -32767) dl = -32767; if (dl > 32767) dl = 32767;
-
- SSumR[ns] += MixREVERBRight();
-
- dr = SSumR[ns] / voldiv; SSumR[ns] = 0;
- if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767;
- *pS++ = (dl + dr) / 2;
- }
- }
- else // stereo:
- for (ns = 0; ns < NSSIZE; ns++)
- {
- static double _interpolation_coefficient = 3.759285613;
-
- if(iFreqResponse) {
- int sl,sr;
- double ldiff, rdiff, avg, tmp;
-
- SSumL[ns]+=MixREVERBLeft(ns);
- SSumR[ns]+=MixREVERBRight();
-
- sl = SSumL[ns]; SSumL[ns]=0;
- sr = SSumR[ns]; SSumR[ns]=0;
-
-
- /*
- Frequency Response
- - William Pitcock (nenolod) (UPSE PSF player)
- - accurate (!)
- - http://nenolod.net
- */
-
- avg = ((sl + sr) / 2);
- ldiff = sl - avg;
- rdiff = sr - avg;
-
- tmp = avg + ldiff * _interpolation_coefficient;
- if (tmp < -32768)
- tmp = -32768;
- if (tmp > 32767)
- tmp = 32767;
- sl = (int)tmp;
-
- tmp = avg + rdiff * _interpolation_coefficient;
- if (tmp < -32768)
- tmp = -32768;
- if (tmp > 32767)
- tmp = 32767;
- sr = (int)tmp;
-
-
- *pS++=sl/voldiv;
- *pS++=sr/voldiv;
- } else {
- SSumL[ns]+=MixREVERBLeft(ns);
-
- d=SSumL[ns]/voldiv;SSumL[ns]=0;
- if(d<-32767) d=-32767;if(d>32767) d=32767;
- *pS++=d;
-
- SSumR[ns]+=MixREVERBRight();
-
- d=SSumR[ns]/voldiv;SSumR[ns]=0;
- if(d<-32767) d=-32767;if(d>32767) d=32767;
- *pS++=d;
- }
- }
-
- //////////////////////////////////////////////////////
- // special irq handling in the decode buffers (0x0000-0x1000)
- // we know:
- // the decode buffers are located in spu memory in the following way:
- // 0x0000-0x03ff CD audio left
- // 0x0400-0x07ff CD audio right
- // 0x0800-0x0bff Voice 1
- // 0x0c00-0x0fff Voice 3
- // and decoded data is 16 bit for one sample
- // we assume:
- // even if voices 1/3 are off or no cd audio is playing, the internal
- // play positions will move on and wrap after 0x400 bytes.
- // Therefore: we just need a pointer from spumem+0 to spumem+3ff, and
- // increase this pointer on each sample by 2 bytes. If this pointer
- // (or 0x400 offsets of this pointer) hits the spuirq address, we generate
- // an IRQ. Only problem: the "wait for cpu" option is kinda hard to do here
- // in some of Peops timer modes. So: we ignore this option here (for now).
-
-#if 0
- if(pMixIrq && irqCallback)
- {
- for(ns=0;ns<NSSIZE;ns++)
- {
- if((spuCtrl&0x40) && pSpuIrq && pSpuIrq<spuMemC+0x1000)
- {
- for(ch=0;ch<4;ch++)
- {
- if(pSpuIrq>=pMixIrq+(ch*0x400) && pSpuIrq<pMixIrq+(ch*0x400)+2)
- {irqCallback();s_chan[ch].iIrqDone=1;}
- }
- }
- pMixIrq+=2;if(pMixIrq>spuMemC+0x3ff) pMixIrq=spuMemC;
- }
- }
-#endif
-
- InitREVERB();
-
- //////////////////////////////////////////////////////
- // feed the sound
- // latency = 25 ms (less pops, crackles, smoother)
-
- //if(iCycle++>=20)
- iCycle += APU_CYCLES_UPDATE;
- if(iCycle > 44000/1000*LATENCY + 100*LATENCY/1000)
- {
- SoundFeedStreamData((unsigned char *)pSpuBuffer,
- ((unsigned char *)pS) - ((unsigned char *)pSpuBuffer));
- pS = (short *)pSpuBuffer;
- iCycle = 0;
- }
-
-
- if( iUseTimer == 2 )
- break;
- }
-
- // end of big main loop...
-
- bThreadEnded = 1;
-
-#ifndef _WINDOWS
- return 0;
-#endif
-}
-
-////////////////////////////////////////////////////////////////////////
-// WINDOWS THREAD... simply calls the timer func and stays forever :)
-////////////////////////////////////////////////////////////////////////
-
-#ifdef _WINDOWS
-
-DWORD WINAPI MAINThreadEx(LPVOID lpParameter)
-{
- MAINProc(0,0,0,0,0);
- return 0;
-}
-
-#endif
-
-// SPU ASYNC... even newer epsxe func
-// 1 time every 'cycle' cycles... harhar
-
-long cpu_cycles;
-void CALLBACK SPUasync(unsigned long cycle)
-{
- cpu_cycles += cycle;
-
- if(iSpuAsyncWait)
- {
- iSpuAsyncWait++;
- if(iSpuAsyncWait<=64) return;
- iSpuAsyncWait=0;
-
- cpu_cycles = cycle;
- }
-
-#ifdef _WINDOWS
- if(iDebugMode==2)
- {
- if(IsWindow(hWDebug)) DestroyWindow(hWDebug);
- hWDebug=0;iDebugMode=0;
- }
- if(iRecordMode==2)
- {
- if(IsWindow(hWRecord)) DestroyWindow(hWRecord);
- hWRecord=0;iRecordMode=0;
- }
-#endif
-
- 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
-
- // note: usable precision difference (not using interval_time)
- while( cpu_cycles >= CPU_CLOCK / 44100 * NSSIZE )
- {
- #ifdef _WINDOWS
- MAINProc(0,0,0,0,0); // -> experimental win mode... not really tested... don't like the drawbacks
- #else
- MAINThread(0); // -> linux high-compat mode
- #endif
-
- cpu_cycles -= CPU_CLOCK / 44100 * NSSIZE;
- }
- }
-}
-
-// SPU UPDATE... new epsxe func
-// 1 time every 32 hsync lines
-// (312/32)x50 in pal
-// (262/32)x60 in ntsc
-
-// since epsxe 1.5.2 (linux) uses SPUupdate, not SPUasync, I will
-// leave that func in the linux port, until epsxe linux is using
-// the async function as well
-
-void CALLBACK SPUupdate(void)
-{
- SPUasync(0);
-}
-
-// XA AUDIO
-
-void CALLBACK SPUplayADPCMchannel(xa_decode_t *xap)
-{
- if(!xap) return;
- if(!xap->freq) return; // no xa freq ? bye
-
- FeedXA(xap); // call main XA feeder
-}
-
-// CDDA AUDIO
-void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
-{
- if (!pcm) return;
- if (nbytes<=0) return;
-
- FeedCDDA((unsigned char *)pcm, nbytes);
-}
-
-// SETUPTIMER: init of certain buffers and threads/timers
-void SetupTimer(void)
-{
- memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers
- memset(SSumL,0,NSSIZE*sizeof(int));
- memset(iFMod,0,NSSIZE*sizeof(int));
- pS=(short *)pSpuBuffer; // setup soundbuffer pointer
-
- bEndThread=0; // init thread vars
- bThreadEnded=0;
- bSpuInit=1; // flag: we are inited
-
-#ifdef _WINDOWS
-
- if(iUseTimer==1) // windows: use timer
- {
- timeBeginPeriod(1);
- timeSetEvent(1,1,MAINProc,0,TIME_ONESHOT);
- }
- else
- 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);
- }
-
-#else
-
- if(!iUseTimer) // linux: use thread
- {
- pthread_create(&thread, NULL, MAINThread, NULL);
- }
-
-#endif
-}
-
-// REMOVETIMER: kill threads/timers
-void RemoveTimer(void)
-{
- bEndThread=1; // raise flag to end thread
-
-#ifdef _WINDOWS
-
- if(iUseTimer!=2) // windows thread?
- {
- while(!bThreadEnded) {Sleep(5L);} // -> wait till thread has ended
- Sleep(5L);
- }
- if(iUseTimer==1) timeEndPeriod(1); // windows timer? stop it
-
-#else
- 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
- }
-
-#endif
-
- bThreadEnded=0; // no more spu is running
- bSpuInit=0;
-}
-
-// SETUPSTREAMS: init most of the spu buffers
-void SetupStreams(void)
-{
- int i;
-
- pSpuBuffer=(unsigned char *)malloc(32768); // alloc mixing buffer
-
- if(iUseReverb==1) i=88200*2;
- else i=NSSIZE*2;
-
- sRVBStart = (int *)malloc(i*4); // alloc reverb buffer
- memset(sRVBStart,0,i*4);
- sRVBEnd = sRVBStart + i;
- sRVBPlay = sRVBStart;
-
- XAStart = // alloc xa buffer
- (uint32_t *)malloc(44100 * sizeof(uint32_t));
- XAEnd = XAStart + 44100;
- XAPlay = XAStart;
- XAFeed = XAStart;
-
- CDDAStart = // alloc cdda buffer
- (uint32_t *)malloc(44100 * sizeof(uint32_t));
- CDDAEnd = CDDAStart + 44100;
- 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 = 1024; // -> init sustain
- s_chan[i].iMute=0;
- s_chan[i].iIrqDone=0;
- s_chan[i].pLoop=spuMemC;
- s_chan[i].pStart=spuMemC;
- s_chan[i].pCurr=spuMemC;
- }
-
- pMixIrq=spuMemC; // enable decoded buffer irqs by setting the address
-}
-
-// REMOVESTREAMS: free most buffer
-void RemoveStreams(void)
-{
- free(pSpuBuffer); // free mixing buffer
- pSpuBuffer = NULL;
- free(sRVBStart); // free reverb buffer
- sRVBStart = NULL;
- free(XAStart); // free XA buffer
- XAStart = NULL;
- free(CDDAStart); // free CDDA buffer
- CDDAStart = NULL;
-}
-
-// INIT/EXIT STUFF
-
-// SPUINIT: this func will be called first by the main emu
-long CALLBACK SPUinit(void)
-{
- spuMemC = (unsigned char *)spuMem; // just small setup
- memset((void *)&rvb, 0, sizeof(REVERBInfo));
- InitADSR();
-
- iVolume = 3;
- iReverbOff = -1;
- spuIrq = 0;
- spuAddr = 0xffffffff;
- bEndThread = 0;
- bThreadEnded = 0;
- spuMemC = (unsigned char *)spuMem;
- pMixIrq = 0;
- memset((void *)s_chan, 0, (MAXCHAN + 1) * sizeof(SPUCHAN));
- pSpuIrq = 0;
- iSPUIRQWait = 1;
- lastch = -1;
-
- ReadConfig(); // read user stuff
- SetupStreams(); // prepare streaming
-
- return 0;
-}
-
-// SPUOPEN: called by main emu after init
-#ifdef _WINDOWS
-long CALLBACK SPUopen(HWND hW)
-#else
-long SPUopen(void)
-#endif
-{
- if (bSPUIsOpen) return 0; // security for some stupid main emus
-
-#ifdef _WINDOWS
- LastWrite=0xffffffff;LastPlay=0; // init some play vars
- if(!IsWindow(hW)) hW=GetActiveWindow();
- hWMain = hW; // store hwnd
-#endif
-
- SetupSound(); // setup sound (before init!)
- SetupTimer(); // timer for feeding data
-
- bSPUIsOpen = 1;
-
-#ifdef _WINDOWS
- 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);
- }
-#endif
-
- return PSE_SPU_ERR_SUCCESS;
-}
-
-// SPUCLOSE: called before shutdown
-long CALLBACK SPUclose(void)
-{
- if (!bSPUIsOpen) return 0; // some security
-
- bSPUIsOpen = 0; // no more open
-
-#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
-
- return 0;
-}
-
-// SPUSHUTDOWN: called by main emu on final exit
-long CALLBACK SPUshutdown(void)
-{
- SPUclose();
- RemoveStreams(); // no more streaming
-
- return 0;
-}
-
-// SPUTEST: we don't test, we are always fine ;)
-long CALLBACK SPUtest(void)
-{
- return 0;
-}
-
-// SPUCONFIGURE: call config dialog
-long CALLBACK SPUconfigure(void)
-{
-#if defined (_WINDOWS)
- DialogBox(hInst,MAKEINTRESOURCE(IDD_CFGDLG),
- GetActiveWindow(),(DLGPROC)DSoundDlgProc);
-#elif defined (_MACOSX)
- DoConfiguration();
-#else
- StartCfgTool("CFG");
-#endif
-
- return 0;
-}
-
-// SPUABOUT: show about window
-void CALLBACK SPUabout(void)
-{
-#if defined (_WINDOWS)
- DialogBox(hInst,MAKEINTRESOURCE(IDD_ABOUT),
- GetActiveWindow(),(DLGPROC)AboutDlgProc);
-#elif defined (_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) | (1 << 8);
-}
-
-char * SPUgetLibInfos(void)
-{
- return _(libraryInfo);
-}
+/***************************************************************************
+ spu.c - description
+ -------------------
+ begin : Wed May 15 2002
+ copyright : (C) 2002 by Pete Bernert
+ email : BlackDove@addcom.de
+ ***************************************************************************/
+/***************************************************************************
+ * *
+ * This program is free software; you can redistribute it and/or modify *
+ * it under the terms of the GNU General Public License as published by *
+ * the Free Software Foundation; either version 2 of the License, or *
+ * (at your option) any later version. See also the license.txt file for *
+ * additional informations. *
+ * *
+ ***************************************************************************/
+
+#include "stdafx.h"
+
+#define _IN_SPU
+
+#include "externals.h"
+#include "cfg.h"
+#include "dsoundoss.h"
+#include "regs.h"
+
+#ifdef _WINDOWS
+#include "debug.h"
+#include "record.h"
+#elif defined(_MACOSX)
+#include "maccfg.h"
+#endif
+
+#ifdef ENABLE_NLS
+#include <libintl.h>
+#include <locale.h>
+#define _(x) gettext(x)
+#define N_(x) (x)
+//If running under Mac OS X, use the Localizable.strings file instead.
+#elif defined(_MACOSX)
+#ifdef PCSXRCORE
+extern char* Pcsxr_locale_text(char* toloc);
+#define _(String) Pcsxr_locale_text(String)
+#define N_(String) String
+#else
+#ifndef PCSXRPLUG
+#warning please define the plug being built to use Mac OS X localization!
+#define _(msgid) msgid
+#define N_(msgid) msgid
+#else
+//Kludge to get the preprocessor to accept PCSXRPLUG as a variable.
+#define PLUGLOC_x(x,y) x ## y
+#define PLUGLOC_y(x,y) PLUGLOC_x(x,y)
+#define PLUGLOC PLUGLOC_y(PCSXRPLUG,_locale_text)
+extern char* PLUGLOC(char* toloc);
+#define _(String) PLUGLOC(String)
+#define N_(String) String
+#endif
+#endif
+#else
+#define _(x) (x)
+#define N_(x) (x)
+#endif
+
+#if defined (_WINDOWS)
+static char * libraryName = N_("DirectSound Driver");
+#elif 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 (USEOPENAL)
+static char * libraryName = N_("OpenAL Sound");
+#elif defined (USEPULSEAUDIO)
+static char * libraryName = N_("PulseAudio Sound");
+#else
+static char * libraryName = N_("NULL Sound");
+#endif
+
+static char * libraryInfo = N_("P.E.Op.S. Sound Driver V1.7\nCoded by Pete Bernert and the P.E.Op.S. team\n");
+
+// globals
+
+// psx buffer / addresses
+
+unsigned short regArea[10000];
+unsigned short spuMem[256*1024];
+unsigned char * spuMemC;
+unsigned char * pSpuIrq=0;
+unsigned char * pSpuBuffer;
+unsigned char * pMixIrq=0;
+
+// user settings
+
+int iVolume=3;
+int iXAPitch=1;
+int iUseTimer=2;
+int iSPUIRQWait=1;
+int iDebugMode=0;
+int iRecordMode=0;
+int iUseReverb=2;
+int iUseInterpolation=2;
+int iDisStereo=0;
+int iFreqResponse=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
+unsigned long dwNoiseCount; // global noise generator
+unsigned long dwNoiseClock; // global noise generator
+int iSpuAsyncWait=0;
+
+unsigned int decoded_ptr = 0;
+unsigned int bIrqHit = 0;
+
+unsigned short spuCtrl=0; // some vars to store psx reg infos
+unsigned short spuStat=0;
+unsigned short spuIrq=0;
+unsigned long spuAddr=0xffffffff; // address into spu mem
+int bEndThread=0; // thread handlers
+int bThreadEnded=0;
+int bSpuInit=0;
+int bSPUIsOpen=0;
+
+#ifdef _WINDOWS
+HWND hWMain=0; // window handle
+HWND hWDebug=0;
+HWND hWRecord=0;
+static HANDLE hMainThread;
+#else
+static pthread_t thread = (pthread_t)-1; // thread id (linux)
+#endif
+
+uint32_t dwNewChannel=0; // flags for faster testing, if new channel starts
+
+void (CALLBACK *irqCallback)(void)=0; // func of main emu, called on spu irq
+void (CALLBACK *cddavCallback)(unsigned short,unsigned short)=0;
+
+// certain globals (were local before, but with the new timeproc I need em global)
+
+static const int f[5][2] = { { 0, 0 },
+ { 60, 0 },
+ { 115, -52 },
+ { 98, -55 },
+ { 122, -60 } };
+int SSumR[NSSIZE];
+int SSumL[NSSIZE];
+int iFMod[NSSIZE];
+int iCycle = 0;
+short * pS;
+
+int lastch=-1; // last channel processed on spu irq in timer mode
+static int lastns=0; // last ns pos
+static int iSecureStart=0; // secure start counter
+
+////////////////////////////////////////////////////////////////////////
+// CODE AREA
+////////////////////////////////////////////////////////////////////////
+
+// dirty inline func includes
+
+#include "reverb.c"
+#include "adsr.c"
+
+////////////////////////////////////////////////////////////////////////
+// helpers for simple interpolation
+
+//
+// easy interpolation on upsampling, no special filter, just "Pete's common sense" tm
+//
+// instead of having n equal sample values in a row like:
+// ____
+// |____
+//
+// we compare the current delta change with the next delta change.
+//
+// if curr_delta is positive,
+//
+// - and next delta is smaller (or changing direction):
+// \.
+// -__
+//
+// - and next delta significant (at least twice) bigger:
+// --_
+// \.
+//
+// - and next delta is nearly same:
+// \.
+// \.
+//
+//
+// if curr_delta is negative,
+//
+// - and next delta is smaller (or changing direction):
+// _--
+// /
+//
+// - and next delta significant (at least twice) bigger:
+// /
+// __-
+//
+// - and next delta is nearly same:
+// /
+// /
+//
+
+
+static 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
+//
+
+static INLINE void InterpolateDown(int ch)
+{
+ if(s_chan[ch].sinc>=0x20000L) // we would skip at least one val?
+ {
+ s_chan[ch].SB[29]+=(s_chan[ch].SB[30]-s_chan[ch].SB[29])/2; // add easy weight
+ if(s_chan[ch].sinc>=0x30000L) // we would skip even more vals?
+ s_chan[ch].SB[29]+=(s_chan[ch].SB[31]-s_chan[ch].SB[30])/2;// add additional next weight
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// helpers for gauss interpolation
+
+#define gval0 (((short*)(&s_chan[ch].SB[29]))[gpos])
+#define gval(x) (((short*)(&s_chan[ch].SB[29]))[(gpos+x)&3])
+
+#include "gauss_i.h"
+
+////////////////////////////////////////////////////////////////////////
+
+#include "xa.c"
+
+////////////////////////////////////////////////////////////////////////
+// START SOUND... called by main thread to setup a new sound on a channel
+////////////////////////////////////////////////////////////////////////
+
+static 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].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].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
+////////////////////////////////////////////////////////////////////////
+
+static INLINE void VoiceChangeFrequency(int ch)
+{
+ s_chan[ch].iUsedFreq=s_chan[ch].iActFreq; // -> take it and calc steps
+ s_chan[ch].sinc=s_chan[ch].iRawPitch<<4;
+ if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
+ if(iUseInterpolation==1) s_chan[ch].SB[32]=1; // -> freq change in simle imterpolation mode: set flag
+}
+
+////////////////////////////////////////////////////////////////////////
+
+static INLINE void FModChangeFrequency(int ch,int ns)
+{
+ int NP=s_chan[ch].iRawPitch;
+
+ NP=((32768L+iFMod[ns])*NP)/32768L;
+
+ if(NP>0x3fff) NP=0x3fff;
+ if(NP<0x1) NP=0x1;
+
+ NP=(44100L*NP)/(4096L); // calc frequency
+
+ s_chan[ch].iActFreq=NP;
+ s_chan[ch].iUsedFreq=NP;
+ s_chan[ch].sinc=(((NP/10)<<16)/4410);
+ if(!s_chan[ch].sinc) s_chan[ch].sinc=1;
+ if(iUseInterpolation==1) // freq change in simple interpolation mode
+ s_chan[ch].SB[32]=1;
+ iFMod[ns]=0;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+/*
+Noise Algorithm
+- Dr.Hell (Xebra PS1 emu)
+- 100% accurate (waveform + frequency)
+- http://drhell.web.fc2.com
+
+
+Level change cycle
+Freq = 0x8000 >> (NoiseClock >> 2);
+
+Frequency of half cycle
+Half = ((NoiseClock & 3) * 2) / (4 + (NoiseClock & 3));
+- 0 = (0*2)/(4+0) = 0/4
+- 1 = (1*2)/(4+1) = 2/5
+- 2 = (2*2)/(4+2) = 4/6
+- 3 = (3*2)/(4+3) = 6/7
+
+-------------------------------
+
+5*6*7 = 210
+4 - 0*0 = 0
+5 - 42*2 = 84
+6 - 35*4 = 140
+7 - 30*6 = 180
+*/
+
+// Noise Waveform - Dr. Hell (Xebra)
+char NoiseWaveAdd [64] = {
+ 1, 0, 0, 1, 0, 1, 1, 0,
+ 1, 0, 0, 1, 0, 1, 1, 0,
+ 1, 0, 0, 1, 0, 1, 1, 0,
+ 1, 0, 0, 1, 0, 1, 1, 0,
+ 0, 1, 1, 0, 1, 0, 0, 1,
+ 0, 1, 1, 0, 1, 0, 0, 1,
+ 0, 1, 1, 0, 1, 0, 0, 1,
+ 0, 1, 1, 0, 1, 0, 0, 1
+};
+
+unsigned short NoiseFreqAdd[5] = {
+ 0, 84, 140, 180, 210
+};
+
+static INLINE void NoiseClock()
+{
+ unsigned int level;
+
+ level = 0x8000 >> (dwNoiseClock >> 2);
+ level <<= 16;
+
+ dwNoiseCount += 0x10000;
+
+ // Dr. Hell - fraction
+ dwNoiseCount += NoiseFreqAdd[ dwNoiseClock & 3 ];
+ if( (dwNoiseCount&0xffff) >= NoiseFreqAdd[4] ) {
+ dwNoiseCount += 0x10000;
+ dwNoiseCount -= NoiseFreqAdd[ dwNoiseClock & 3 ];
+ }
+
+ if( dwNoiseCount >= level )
+ {
+ while( dwNoiseCount >= level )
+ dwNoiseCount -= level;
+
+ // Dr. Hell - form
+ dwNoiseVal = (dwNoiseVal<<1) | NoiseWaveAdd[ (dwNoiseVal>>10) & 63 ];
+ }
+}
+
+static INLINE int iGetNoiseVal(int ch)
+{
+ int fa;
+
+ fa = (short) dwNoiseVal;
+
+ // no clip need
+ //if(fa>32767L) fa=32767L;
+ //if(fa<-32767L) fa=-32767L;
+
+ // don't upset VAG decoder
+ //if(iUseInterpolation<2) // no gauss/cubic interpolation?
+ //pChannel->SB[29] = fa; // -> store noise val in "current sample" slot
+
+ // boost volume - no more!
+ //return fa * 3 / 2;
+ return fa;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+static INLINE void StoreInterpolationVal(int ch,int fa)
+{
+ /*
+ // fmod channel = sound output
+ if(s_chan[ch].bFMod==2) // fmod freq channel
+ s_chan[ch].SB[29]=fa;
+ else
+ */
+ {
+ if((spuCtrl&0x4000)==0) fa=0; // muted?
+ else // else adjust
+ {
+ if(fa>32767L) fa=32767L;
+ if(fa<-32767L) fa=-32767L;
+ }
+
+ if(iUseInterpolation>=2) // gauss/cubic interpolation
+ {
+ int gpos = s_chan[ch].SB[28];
+ gval0 = fa;
+ gpos = (gpos+1) & 3;
+ s_chan[ch].SB[28] = gpos;
+ }
+ else
+ if(iUseInterpolation==1) // simple interpolation
+ {
+ s_chan[ch].SB[28] = 0;
+ s_chan[ch].SB[29] = s_chan[ch].SB[30]; // -> helpers for simple linear interpolation: delay real val for two slots, and calc the two deltas, for a 'look at the future behaviour'
+ s_chan[ch].SB[30] = s_chan[ch].SB[31];
+ s_chan[ch].SB[31] = fa;
+ s_chan[ch].SB[32] = 1; // -> flag: calc new interolation
+ }
+ else s_chan[ch].SB[29]=fa; // no interpolation
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+static INLINE int iGetInterpolationVal(int ch)
+{
+ int fa;
+
+ // fmod channel = sound output
+ //if(s_chan[ch].bFMod==2) return s_chan[ch].SB[29];
+
+ switch(iUseInterpolation)
+ {
+ //--------------------------------------------------//
+ case 3: // cubic interpolation
+ {
+ long xd;int gpos;
+ xd = ((s_chan[ch].spos) >> 1)+1;
+ gpos = s_chan[ch].SB[28];
+
+ fa = gval(3) - 3*gval(2) + 3*gval(1) - gval0;
+ fa *= (xd - (2<<15)) / 6;
+ fa >>= 15;
+ fa += gval(2) - gval(1) - gval(1) + gval0;
+ fa *= (xd - (1<<15)) >> 1;
+ fa >>= 15;
+ fa += gval(1) - gval0;
+ fa *= xd;
+ fa >>= 15;
+ fa = fa + gval0;
+
+ } break;
+ //--------------------------------------------------//
+ case 2: // gauss interpolation
+ {
+ int vl, vr;int gpos;
+ vl = (s_chan[ch].spos >> 6) & ~3;
+ gpos = s_chan[ch].SB[28];
+ vr=(gauss[vl]*gval0)&~2047;
+ vr+=(gauss[vl+1]*gval(1))&~2047;
+ vr+=(gauss[vl+2]*gval(2))&~2047;
+ vr+=(gauss[vl+3]*gval(3))&~2047;
+ fa = vr>>11;
+ } break;
+ //--------------------------------------------------//
+ case 1: // simple interpolation
+ {
+ if(s_chan[ch].sinc<0x10000L) // -> upsampling?
+ InterpolateUp(ch); // --> interpolate up
+ else InterpolateDown(ch); // --> else down
+ fa=s_chan[ch].SB[29];
+ } break;
+ //--------------------------------------------------//
+ default: // no interpolation
+ {
+ fa=s_chan[ch].SB[29];
+ } break;
+ //--------------------------------------------------//
+ }
+
+ return fa;
+}
+
+////////////////////////////////////////////////////////////////////////
+// MAIN SPU FUNCTION
+// here is the main job handler... thread, timer or direct func call
+// basically the whole sound processing is done in this fat func!
+////////////////////////////////////////////////////////////////////////
+
+// 5 ms waiting phase, if buffer is full and no new sound has to get started
+// .. can be made smaller (smallest val: 1 ms), but bigger waits give
+// better performance
+
+#define PAUSE_W 1
+#define PAUSE_L 1000
+
+////////////////////////////////////////////////////////////////////////
+
+#ifdef _WINDOWS
+static VOID CALLBACK MAINProc(UINT nTimerId, UINT msg, DWORD dwUser, DWORD dwParam1, DWORD dwParam2)
+#else
+static void *MAINThread(void *arg)
+#endif
+{
+ int s_1,s_2,fa,ns;
+ int voldiv = iVolume;
+
+ unsigned char * start;unsigned int nSample;
+ int ch,predict_nr,shift_factor,flags,d,s;
+ int bIRQReturn=0;
+ unsigned int decoded_voice=0;
+
+ // mute output
+ if( voldiv == 5 ) voldiv = 0x7fffffff;
+
+ while(!bEndThread) // until we are shutting down
+ {
+ // ok, at the beginning we are looking if there is
+ // enuff free place in the dsound/oss buffer to
+ // fill in new data, or if there is a new channel to start.
+ // if not, we wait (thread) or return (timer/spuasync)
+ // until enuff free place is available/a new channel gets
+ // started
+
+ if(dwNewChannel) // new channel should start immedately?
+ { // (at least one bit 0 ... MAXCHANNEL is set?)
+ iSecureStart++; // -> set iSecure
+ if(iSecureStart>1) 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
+
+#ifdef _WINDOWS
+ if(iUseTimer) // no-thread mode?
+ {
+ if(iUseTimer==1) // -> ok, timer mode 1: setup a oneshot timer of x ms to wait
+ timeSetEvent(PAUSE_W,1,MAINProc,0,TIME_ONESHOT);
+ return; // -> and done this time (timer mode 1 or 2)
+ }
+ // win thread mode:
+ Sleep(PAUSE_W); // sleep for x ms (win)
+#else
+ if(iUseTimer) return 0; // linux no-thread mode? bye
+ usleep(PAUSE_L); // else sleep for x ms (linux)
+#endif
+
+ if(dwNewChannel) iSecureStart=1; // if a new channel kicks in (or, of course, sound buffer runs low), we will leave the loop
+ }
+
+ //--------------------------------------------------// continue from irq handling in timer mode?
+
+ if(lastch>=0) // will be -1 if no continue is pending
+ {
+ ch=lastch; ns=lastns; lastch=-1; // -> setup all kind of vars to continue
+ goto GOON; // -> directly jump to the continue point
+ }
+
+ //--------------------------------------------------//
+ //- main channel loop -//
+ //--------------------------------------------------//
+ {
+ ns=0;
+ decoded_voice = decoded_ptr;
+
+ while(ns<NSSIZE) // loop until 1 ms of data is reached
+ {
+ SSumL[ns]=0;
+ SSumR[ns]=0;
+
+
+ // decoded buffer values - dummy
+ spuMem[ (0x000 + decoded_voice) / 2 ] = (short) 0;
+ spuMem[ (0x400 + decoded_voice) / 2 ] = (short) 0;
+ spuMem[ (0x800 + decoded_voice) / 2 ] = (short) 0;
+ spuMem[ (0xc00 + decoded_voice) / 2 ] = (short) 0;
+
+
+ NoiseClock();
+
+ 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) {
+#if 1
+ StartSound(ch); // start new sound
+ dwNewChannel&=~(1<<ch); // clear new channel bit
+#else
+ if( s_chan[ch].ADSRX.StartDelay == 0 ) {
+ StartSound(ch); // start new sound
+ dwNewChannel&=~(1<<ch); // clear new channel bit
+ } else {
+ s_chan[ch].ADSRX.StartDelay--;
+ }
+#endif
+ }
+ if(!s_chan[ch].bOn) continue; // channel not playing? next
+
+ if(s_chan[ch].iActFreq!=s_chan[ch].iUsedFreq) // new psx frequency?
+ VoiceChangeFrequency(ch);
+
+ if(s_chan[ch].bFMod==1 && iFMod[ns]) // fmod freq channel
+ FModChangeFrequency(ch,ns);
+
+ while(s_chan[ch].spos>=0x10000L)
+ {
+ if(s_chan[ch].iSBPos==28) // 28 reached?
+ {
+ // Xenogears - Anima Relic dungeon (exp gain)
+ if( s_chan[ch].bLoopJump == 1 )
+ s_chan[ch].pCurr = s_chan[ch].pLoop;
+
+ s_chan[ch].bLoopJump = 0;
+
+
+ start=s_chan[ch].pCurr; // set up the current pos
+
+ if (s_chan[ch].iSilent==1 || start == (unsigned char*)-1) // special "stop" sign
+ {
+ // silence = let channel keep running (IRQs)
+ //s_chan[ch].bOn=0; // -> turn everything off
+ s_chan[ch].iSilent=2;
+
+ s_chan[ch].ADSRX.lVolume=0;
+ s_chan[ch].ADSRX.EnvelopeVol=0;
+ }
+
+ 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++;
+
+ // Silhouette Mirage - Serah fight
+ if( predict_nr > 4 ) predict_nr = 0;
+
+ // -------------------------------------- //
+
+ 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);
+
+ // snes brr clamps
+ fa = CLAMP16(fa);
+
+ 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);
+
+ // snes brr clamps
+ fa = CLAMP16(fa);
+
+ s_2=s_1;s_1=fa;
+
+ s_chan[ch].SB[nSample++]=fa;
+ }
+
+ //////////////////////////////////////////// irq check
+
+#if 1
+ // ?? (-8)
+ if( Check_IRQ( (start-spuMemC)-8, 0 ) ||
+ Check_IRQ( (start-spuMemC)-0, 0 ) )
+ {
+#else
+ 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)))
+#endif
+ {
+ 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
+
+ /*
+ SPU2-X:
+ $4 = set loop to current block
+ $2 = keep envelope on (no mute)
+ $1 = jump to loop address
+
+ silence means no volume (ADSR keeps playing!!)
+ */
+
+ if(flags&4)
+ s_chan[ch].pLoop=start-16;
+
+
+ // Jungle Book - Rhythm 'n Groove - don't reset ignore status
+ // - fixes gameplay speed (IRQ hits)
+ //s_chan[ch].bIgnoreLoop = 0;
+
+
+ if(flags&1)
+ {
+ // ...?
+ //s_chan[ch].bIgnoreLoop = 0;
+
+ // Xenogears - 7 = play missing sounds
+ // set jump flag
+ s_chan[ch].bLoopJump = 1;
+
+
+ // silence = keep playing..?
+ if( (flags&2) == 0 ) {
+ s_chan[ch].iSilent = 1;
+
+ // silence = don't start release phase
+ //s_chan[ch].bStop = 1;
+
+ //start = (unsigned char *) -1;
+ }
+ }
+
+#if 0
+ // crash check
+ if( start == 0 )
+ start = (unsigned char *) -1;
+ if( start >= spuMemC + 0x80000 )
+ start = spuMemC - 0x80000;
+#endif
+
+
+ // Silhouette Mirage - ending mini-game
+
+ // ??
+ if( start - spuMemC >= 0x80000 ) {
+ start -= 16;
+
+ s_chan[ch].iSilent = 1;
+ s_chan[ch].bStop = 1;
+ }
+
+
+ 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)
+#ifdef _WINDOWS
+ Sleep(1);
+#else
+ usleep(1000L);
+#endif
+ }
+ else
+ {
+ lastch=ch;
+ lastns=ns;
+
+#ifdef _WINDOWS
+ return;
+#else
+ return 0;
+#endif
+ }
+ }
+
+GOON: ;
+ }
+
+ fa=s_chan[ch].SB[s_chan[ch].iSBPos++]; // get sample data
+
+ StoreInterpolationVal(ch,fa); // store val for later interpolation
+
+ s_chan[ch].spos -= 0x10000L;
+ }
+
+ if(s_chan[ch].bNoise)
+ fa=iGetNoiseVal(ch); // get noise val
+ else fa=iGetInterpolationVal(ch); // get sample val
+
+
+ // Voice 1/3 decoded buffer
+ if( ch == 0 ) {
+ spuMem[ (0x800 + decoded_voice) / 2 ] = (short) fa;
+ } else if( ch == 2 ) {
+ spuMem[ (0xc00 + decoded_voice) / 2 ] = (short) fa;
+ }
+
+
+ s_chan[ch].sval = (MixADSR(ch) * fa) / 1023; // mix adsr
+
+ if(s_chan[ch].bFMod==2) // fmod freq channel
+ iFMod[ns]=s_chan[ch].sval; // -> store 1T sample data, use that to do fmod on next channel
+
+ // mix fmod channel into output
+ // - Xenogears save icon (high pitch)
+ {
+ //////////////////////////////////////////////
+ // 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);
+ }
+
+ s_chan[ch].spos += s_chan[ch].sinc;
+ }
+
+ ////////////////////////////////////////////////
+ // ok, go on until 1 ms data of this channel is collected
+
+ // decoded buffer - voice
+ decoded_voice += 2;
+ decoded_voice &= 0x400;
+
+
+ // status flag
+ if( decoded_voice >= 0x200 ) {
+ spuStat |= STAT_DECODED;
+ } else {
+ spuStat &= ~STAT_DECODED;
+ }
+
+
+ // IRQ work
+ {
+ unsigned char *old_irq;
+ unsigned int old_ptr;
+
+ old_irq = pSpuIrq;
+ old_ptr = decoded_voice;
+
+#if 0
+ // align to boundaries ($0, $200, $400, $600)
+ pSpuIrq = ((pSpuIrq - spuMemC) & (~0x1ff)) + spuMemC;
+ decoded_voice = decoded_voice & (~0x1ff);
+#endif
+
+ // check all decoded buffer IRQs - timing issue
+ Check_IRQ( decoded_voice + 0x000, 0 );
+ Check_IRQ( decoded_voice + 0x400, 0 );
+ Check_IRQ( decoded_voice + 0x800, 0 );
+ Check_IRQ( decoded_voice + 0xc00, 0 );
+
+ pSpuIrq = old_irq;
+ decoded_voice = old_ptr;
+ }
+
+
+ ns++;
+ } // end ns
+ }
+
+
+ //---------------------------------------------------//
+ //- here we have another 1 ms of sound data
+ //---------------------------------------------------//
+ // mix XA infos (if any)
+
+ MixXA();
+
+
+ // now safe to update deocded buffer ptr
+ decoded_ptr += ns * 2;
+ decoded_ptr &= 0x400;
+
+
+ ///////////////////////////////////////////////////////
+ // mix all channels (including reverb) into one buffer
+
+ if(iDisStereo) // no stereo?
+ {
+ int dl, dr;
+ for (ns = 0; ns < NSSIZE; ns++)
+ {
+ SSumL[ns] += MixREVERBLeft(ns);
+
+ dl = SSumL[ns] / voldiv; SSumL[ns] = 0;
+ if (dl < -32767) dl = -32767; if (dl > 32767) dl = 32767;
+
+ SSumR[ns] += MixREVERBRight();
+
+ dr = SSumR[ns] / voldiv; SSumR[ns] = 0;
+ if (dr < -32767) dr = -32767; if (dr > 32767) dr = 32767;
+ *pS++ = (dl + dr) / 2;
+ }
+ }
+ else // stereo:
+ for (ns = 0; ns < NSSIZE; ns++)
+ {
+ static double _interpolation_coefficient = 3.759285613;
+
+ if(iFreqResponse) {
+ int sl,sr;
+ double ldiff, rdiff, avg, tmp;
+
+ SSumL[ns]+=MixREVERBLeft(ns);
+ SSumR[ns]+=MixREVERBRight();
+
+ sl = SSumL[ns]; SSumL[ns]=0;
+ sr = SSumR[ns]; SSumR[ns]=0;
+
+
+ /*
+ Frequency Response
+ - William Pitcock (nenolod) (UPSE PSF player)
+ - accurate (!)
+ - http://nenolod.net
+ */
+
+ avg = ((sl + sr) / 2);
+ ldiff = sl - avg;
+ rdiff = sr - avg;
+
+ tmp = avg + ldiff * _interpolation_coefficient;
+ if (tmp < -32768)
+ tmp = -32768;
+ if (tmp > 32767)
+ tmp = 32767;
+ sl = (int)tmp;
+
+ tmp = avg + rdiff * _interpolation_coefficient;
+ if (tmp < -32768)
+ tmp = -32768;
+ if (tmp > 32767)
+ tmp = 32767;
+ sr = (int)tmp;
+
+
+ *pS++=sl/voldiv;
+ *pS++=sr/voldiv;
+ } else {
+ SSumL[ns]+=MixREVERBLeft(ns);
+
+ d=SSumL[ns]/voldiv;SSumL[ns]=0;
+ if(d<-32767) d=-32767;if(d>32767) d=32767;
+ *pS++=d;
+
+ SSumR[ns]+=MixREVERBRight();
+
+ d=SSumR[ns]/voldiv;SSumR[ns]=0;
+ if(d<-32767) d=-32767;if(d>32767) d=32767;
+ *pS++=d;
+ }
+ }
+
+ //////////////////////////////////////////////////////
+ // special irq handling in the decode buffers (0x0000-0x1000)
+ // we know:
+ // the decode buffers are located in spu memory in the following way:
+ // 0x0000-0x03ff CD audio left
+ // 0x0400-0x07ff CD audio right
+ // 0x0800-0x0bff Voice 1
+ // 0x0c00-0x0fff Voice 3
+ // and decoded data is 16 bit for one sample
+ // we assume:
+ // even if voices 1/3 are off or no cd audio is playing, the internal
+ // play positions will move on and wrap after 0x400 bytes.
+ // Therefore: we just need a pointer from spumem+0 to spumem+3ff, and
+ // increase this pointer on each sample by 2 bytes. If this pointer
+ // (or 0x400 offsets of this pointer) hits the spuirq address, we generate
+ // an IRQ. Only problem: the "wait for cpu" option is kinda hard to do here
+ // in some of Peops timer modes. So: we ignore this option here (for now).
+
+#if 0
+ if(pMixIrq && irqCallback)
+ {
+ for(ns=0;ns<NSSIZE;ns++)
+ {
+ if((spuCtrl&0x40) && pSpuIrq && pSpuIrq<spuMemC+0x1000)
+ {
+ for(ch=0;ch<4;ch++)
+ {
+ if(pSpuIrq>=pMixIrq+(ch*0x400) && pSpuIrq<pMixIrq+(ch*0x400)+2)
+ {irqCallback();s_chan[ch].iIrqDone=1;}
+ }
+ }
+ pMixIrq+=2;if(pMixIrq>spuMemC+0x3ff) pMixIrq=spuMemC;
+ }
+ }
+#endif
+
+ InitREVERB();
+
+ //////////////////////////////////////////////////////
+ // feed the sound
+ // latency = 25 ms (less pops, crackles, smoother)
+
+ //if(iCycle++>=20)
+ iCycle += APU_CYCLES_UPDATE;
+ if(iCycle > 44000/1000*LATENCY + 100*LATENCY/1000)
+ {
+ SoundFeedStreamData((unsigned char *)pSpuBuffer,
+ ((unsigned char *)pS) - ((unsigned char *)pSpuBuffer));
+ pS = (short *)pSpuBuffer;
+ iCycle = 0;
+ }
+
+
+ if( iUseTimer == 2 )
+ break;
+ }
+
+ // end of big main loop...
+
+ bThreadEnded = 1;
+
+#ifndef _WINDOWS
+ return 0;
+#endif
+}
+
+////////////////////////////////////////////////////////////////////////
+// WINDOWS THREAD... simply calls the timer func and stays forever :)
+////////////////////////////////////////////////////////////////////////
+
+#ifdef _WINDOWS
+
+DWORD WINAPI MAINThreadEx(LPVOID lpParameter)
+{
+ MAINProc(0,0,0,0,0);
+ return 0;
+}
+
+#endif
+
+// SPU ASYNC... even newer epsxe func
+// 1 time every 'cycle' cycles... harhar
+
+long cpu_cycles;
+void CALLBACK SPUasync(unsigned long cycle)
+{
+ cpu_cycles += cycle;
+
+ if(iSpuAsyncWait)
+ {
+ iSpuAsyncWait++;
+ if(iSpuAsyncWait<=64) return;
+ iSpuAsyncWait=0;
+
+ cpu_cycles = cycle;
+ }
+
+#ifdef _WINDOWS
+ if(iDebugMode==2)
+ {
+ if(IsWindow(hWDebug)) DestroyWindow(hWDebug);
+ hWDebug=0;iDebugMode=0;
+ }
+ if(iRecordMode==2)
+ {
+ if(IsWindow(hWRecord)) DestroyWindow(hWRecord);
+ hWRecord=0;iRecordMode=0;
+ }
+#endif
+
+ 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
+
+ // note: usable precision difference (not using interval_time)
+ while( cpu_cycles >= CPU_CLOCK / 44100 * NSSIZE )
+ {
+ #ifdef _WINDOWS
+ MAINProc(0,0,0,0,0); // -> experimental win mode... not really tested... don't like the drawbacks
+ #else
+ MAINThread(0); // -> linux high-compat mode
+ #endif
+
+ cpu_cycles -= CPU_CLOCK / 44100 * NSSIZE;
+ }
+ }
+}
+
+// SPU UPDATE... new epsxe func
+// 1 time every 32 hsync lines
+// (312/32)x50 in pal
+// (262/32)x60 in ntsc
+
+// since epsxe 1.5.2 (linux) uses SPUupdate, not SPUasync, I will
+// leave that func in the linux port, until epsxe linux is using
+// the async function as well
+
+void CALLBACK SPUupdate(void)
+{
+ SPUasync(0);
+}
+
+// XA AUDIO
+
+void CALLBACK SPUplayADPCMchannel(xa_decode_t *xap)
+{
+ if(!xap) return;
+ if(!xap->freq) return; // no xa freq ? bye
+
+ FeedXA(xap); // call main XA feeder
+}
+
+// CDDA AUDIO
+void CALLBACK SPUplayCDDAchannel(short *pcm, int nbytes)
+{
+ if (!pcm) return;
+ if (nbytes<=0) return;
+
+ FeedCDDA((unsigned char *)pcm, nbytes);
+}
+
+// SETUPTIMER: init of certain buffers and threads/timers
+void SetupTimer(void)
+{
+ memset(SSumR,0,NSSIZE*sizeof(int)); // init some mixing buffers
+ memset(SSumL,0,NSSIZE*sizeof(int));
+ memset(iFMod,0,NSSIZE*sizeof(int));
+ pS=(short *)pSpuBuffer; // setup soundbuffer pointer
+
+ bEndThread=0; // init thread vars
+ bThreadEnded=0;
+ bSpuInit=1; // flag: we are inited
+
+#ifdef _WINDOWS
+
+ if(iUseTimer==1) // windows: use timer
+ {
+ timeBeginPeriod(1);
+ timeSetEvent(1,1,MAINProc,0,TIME_ONESHOT);
+ }
+ else
+ 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);
+ }
+
+#else
+
+ if(!iUseTimer) // linux: use thread
+ {
+ pthread_create(&thread, NULL, MAINThread, NULL);
+ }
+
+#endif
+}
+
+// REMOVETIMER: kill threads/timers
+void RemoveTimer(void)
+{
+ bEndThread=1; // raise flag to end thread
+
+#ifdef _WINDOWS
+
+ if(iUseTimer!=2) // windows thread?
+ {
+ while(!bThreadEnded) {Sleep(5L);} // -> wait till thread has ended
+ Sleep(5L);
+ }
+ if(iUseTimer==1) timeEndPeriod(1); // windows timer? stop it
+
+#else
+ 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
+ }
+
+#endif
+
+ bThreadEnded=0; // no more spu is running
+ bSpuInit=0;
+}
+
+// SETUPSTREAMS: init most of the spu buffers
+void SetupStreams(void)
+{
+ int i;
+
+ pSpuBuffer=(unsigned char *)malloc(32768); // alloc mixing buffer
+
+ if(iUseReverb==1) i=88200*2;
+ else i=NSSIZE*2;
+
+ sRVBStart = (int *)malloc(i*4); // alloc reverb buffer
+ memset(sRVBStart,0,i*4);
+ sRVBEnd = sRVBStart + i;
+ sRVBPlay = sRVBStart;
+
+ XAStart = // alloc xa buffer
+ (uint32_t *)malloc(44100 * sizeof(uint32_t));
+ XAEnd = XAStart + 44100;
+ XAPlay = XAStart;
+ XAFeed = XAStart;
+
+ CDDAStart = // alloc cdda buffer
+ (uint32_t *)malloc(44100 * sizeof(uint32_t));
+ CDDAEnd = CDDAStart + 44100;
+ 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 = 1024; // -> init sustain
+ s_chan[i].iMute=0;
+ s_chan[i].iIrqDone=0;
+ s_chan[i].pLoop=spuMemC;
+ s_chan[i].pStart=spuMemC;
+ s_chan[i].pCurr=spuMemC;
+ }
+
+ pMixIrq=spuMemC; // enable decoded buffer irqs by setting the address
+}
+
+// REMOVESTREAMS: free most buffer
+void RemoveStreams(void)
+{
+ free(pSpuBuffer); // free mixing buffer
+ pSpuBuffer = NULL;
+ free(sRVBStart); // free reverb buffer
+ sRVBStart = NULL;
+ free(XAStart); // free XA buffer
+ XAStart = NULL;
+ free(CDDAStart); // free CDDA buffer
+ CDDAStart = NULL;
+}
+
+// INIT/EXIT STUFF
+
+// SPUINIT: this func will be called first by the main emu
+long CALLBACK SPUinit(void)
+{
+ spuMemC = (unsigned char *)spuMem; // just small setup
+ memset((void *)&rvb, 0, sizeof(REVERBInfo));
+ InitADSR();
+
+ iVolume = 3;
+ iReverbOff = -1;
+ spuIrq = 0;
+ spuAddr = 0xffffffff;
+ bEndThread = 0;
+ bThreadEnded = 0;
+ spuMemC = (unsigned char *)spuMem;
+ pMixIrq = 0;
+ memset((void *)s_chan, 0, (MAXCHAN + 1) * sizeof(SPUCHAN));
+ pSpuIrq = 0;
+ iSPUIRQWait = 1;
+ lastch = -1;
+
+ ReadConfig(); // read user stuff
+ SetupStreams(); // prepare streaming
+
+ return 0;
+}
+
+// SPUOPEN: called by main emu after init
+#ifdef _WINDOWS
+long CALLBACK SPUopen(HWND hW)
+#else
+long SPUopen(void)
+#endif
+{
+ if (bSPUIsOpen) return 0; // security for some stupid main emus
+
+#ifdef _WINDOWS
+ LastWrite=0xffffffff;LastPlay=0; // init some play vars
+ if(!IsWindow(hW)) hW=GetActiveWindow();
+ hWMain = hW; // store hwnd
+#endif
+
+ SetupSound(); // setup sound (before init!)
+ SetupTimer(); // timer for feeding data
+
+ bSPUIsOpen = 1;
+
+#ifdef _WINDOWS
+ 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);
+ }
+#endif
+
+ return PSE_SPU_ERR_SUCCESS;
+}
+
+// SPUCLOSE: called before shutdown
+long CALLBACK SPUclose(void)
+{
+ if (!bSPUIsOpen) return 0; // some security
+
+ bSPUIsOpen = 0; // no more open
+
+#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
+
+ return 0;
+}
+
+// SPUSHUTDOWN: called by main emu on final exit
+long CALLBACK SPUshutdown(void)
+{
+ SPUclose();
+ RemoveStreams(); // no more streaming
+
+ return 0;
+}
+
+// SPUTEST: we don't test, we are always fine ;)
+long CALLBACK SPUtest(void)
+{
+ return 0;
+}
+
+// SPUCONFIGURE: call config dialog
+long CALLBACK SPUconfigure(void)
+{
+#if defined (_WINDOWS)
+ DialogBox(hInst,MAKEINTRESOURCE(IDD_CFGDLG),
+ GetActiveWindow(),(DLGPROC)DSoundDlgProc);
+#elif defined (_MACOSX)
+ DoConfiguration();
+#else
+ StartCfgTool("CFG");
+#endif
+
+ return 0;
+}
+
+// SPUABOUT: show about window
+void CALLBACK SPUabout(void)
+{
+#if defined (_WINDOWS)
+ DialogBox(hInst,MAKEINTRESOURCE(IDD_ABOUT),
+ GetActiveWindow(),(DLGPROC)AboutDlgProc);
+#elif defined (_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) | (1 << 8);
+}
+
+char * SPUgetLibInfos(void)
+{
+ return _(libraryInfo);
+}
generated by cgit v1.2.3 (git 2.39.1) at 2026-08-21 02:07:34 +0000