diff --git a/plugins/opl2/CMakeLists.txt b/plugins/opl2/CMakeLists.txt index 0b9bd32cd..981ec53a3 100644 --- a/plugins/opl2/CMakeLists.txt +++ b/plugins/opl2/CMakeLists.txt @@ -1,3 +1,3 @@ INCLUDE(BuildPlugin) -BUILD_PLUGIN(OPL2 opl2instrument.cpp opl2instrument.h opl.h kemuopl.h adlibemu.c adlibemu.h fmopl.c fmopl.h temuopl.cpp temuopl.h MOCFILES opl2instrument.h EMBEDDED_RESOURCES ${CMAKE_CURRENT_SOURCE_DIR}/*.png) \ No newline at end of file +BUILD_PLUGIN(OPL2 opl2instrument.cpp opl2instrument.h opl.h fmopl.c fmopl.h temuopl.cpp temuopl.h MOCFILES opl2instrument.h EMBEDDED_RESOURCES ${CMAKE_CURRENT_SOURCE_DIR}/*.png) \ No newline at end of file diff --git a/plugins/opl2/adlibemu.c b/plugins/opl2/adlibemu.c deleted file mode 100644 index 9aac3a6fc..000000000 --- a/plugins/opl2/adlibemu.c +++ /dev/null @@ -1,608 +0,0 @@ -/* - * ADLIBEMU.C - * Copyright (C) 1998-2001 Ken Silverman - * Ken Silverman's official web site: "http://www.advsys.net/ken" - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -/* -This file is a digital Adlib emulator for OPL2 and possibly OPL3 - -Features that could be added in a future version: -- Amplitude and Frequency Vibrato Bits (not hard, but a big speed hit) -- Global Keyboard Split Number Bit (need to research this one some more) -- 2nd Adlib chip for OPL3 (simply need to make my cell array bigger) -- Advanced connection modes of OPL3 (Just need to add more "docell" cases) -- L/R Stereo bits of OPL3 (Need adlibgetsample to return stereo) - -Features that aren't worth supporting: -- Anything related to adlib timers&interrupts (Sorry - I always used IRQ0) -- Composite sine wave mode (CSM) (Supported only on ancient cards) - -I'm not sure about a few things in my code: -- Attack curve. What function is this anyway? I chose to use an order-3 - polynomial to approximate but this doesn't seem right. -- Attack/Decay/Release constants - my constants may not be exact -- What should ADJUSTSPEED be? -- Haven't verified that Global Keyboard Split Number Bit works yet -- Some of the drums don't always sound right. It's pretty hard to guess - the exact waveform of drums when you look at random data which is - slightly randomized due to digital ADC recording. -- Adlib seems to have a lot more treble than my emulator does. I'm not - sure if this is simply unfixable due to the sound blaster's different - filtering on FM and digital playback or if it's a serious bug in my - code. -*/ - -#include -#include - -#if !defined(max) && !defined(__cplusplus) -#define max(a,b) (((a) > (b)) ? (a) : (b)) -#endif -#if !defined(min) && !defined(__cplusplus) -#define min(a,b) (((a) < (b)) ? (a) : (b)) -#endif - -#define PI 3.141592653589793 -#define MAXCELLS 18 -#define WAVPREC 2048 - -static float AMPSCALE=(8192.0); -#define FRQSCALE (49716/512.0) - -//Constants for Ken's Awe32, on a PII-266 (Ken says: Use these for KSM's!) -#define MODFACTOR 4.0 //How much of modulator cell goes into carrier -#define MFBFACTOR 1.0 //How much feedback goes back into modulator -#define ADJUSTSPEED 0.75 //0<=x<=1 Simulate finite rate of change of state - -//Constants for Ken's Awe64G, on a P-133 -//#define MODFACTOR 4.25 //How much of modulator cell goes into carrier -//#define MFBFACTOR 0.5 //How much feedback goes back into modulator -//#define ADJUSTSPEED 0.85 //0<=x<=1 Simulate finite rate of change of state - -typedef struct -{ - float val, t, tinc, vol, sustain, amp, mfb; - float a0, a1, a2, a3, decaymul, releasemul; - short *waveform; - long wavemask; - void (*cellfunc)(void *, float); - unsigned char flags, dum0, dum1, dum2; -} celltype; - -static long numspeakers, bytespersample; -static float recipsamp; -static celltype cell[MAXCELLS]; -static signed short wavtable[WAVPREC*3]; -static float kslmul[4] = {0.0,0.5,0.25,1.0}; -static float frqmul[16] = {.5,1,2,3,4,5,6,7,8,9,10,10,12,12,15,15}, nfrqmul[16]; -static unsigned char adlibreg[256], ksl[8][16]; -static unsigned char modulatorbase[9] = {0,1,2,8,9,10,16,17,18}; -static unsigned char odrumstat = 0; -static unsigned char base2cell[22] = {0,1,2,0,1,2,0,0,3,4,5,3,4,5,0,0,6,7,8,6,7,8}; - -float lvol[9] = {1,1,1,1,1,1,1,1,1}; //Volume multiplier on left speaker -float rvol[9] = {1,1,1,1,1,1,1,1,1}; //Volume multiplier on right speaker -long lplc[9] = {0,0,0,0,0,0,0,0,0}; //Samples to delay on left speaker -long rplc[9] = {0,0,0,0,0,0,0,0,0}; //Samples to delay on right speaker - -long nlvol[9], nrvol[9]; -long nlplc[9], nrplc[9]; -long rend = 0; -#define FIFOSIZ 256 -static float *rptr[9], *nrptr[9]; -static float rbuf[9][FIFOSIZ*2]; -static float snd[FIFOSIZ*2]; - -#ifndef USING_ASM -#define _inline -#endif - -#ifdef USING_ASM -static _inline void ftol (float f, long *a) -{ - _asm - { - mov eax, a - fld f - fistp dword ptr [eax] - } -} -#else -static void ftol(float f, long *a) { - *a=f; -} -#endif - -#define ctc ((celltype *)c) //A rare attempt to make code easier to read! -void docell4 (void *c, float modulator) { } -void docell3 (void *c, float modulator) -{ - long i; - - ftol(ctc->t+modulator,&i); - ctc->t += ctc->tinc; - ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED; -} -void docell2 (void *c, float modulator) -{ - long i; - - ftol(ctc->t+modulator,&i); - - void *amp_void = &ctc->amp; - long *amp_long = (long *)amp_void; - if (*amp_long <= 0x37800000) - { - ctc->amp = 0; - ctc->cellfunc = docell4; - } - ctc->amp *= ctc->releasemul; - - ctc->t += ctc->tinc; - ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED; -} -void docell1 (void *c, float modulator) -{ - long i; - - ftol(ctc->t+modulator,&i); - - void *amp_void = &ctc->amp; - long *amp_long = (long *)amp_void; - void *sustain_void = &ctc->sustain; - long *sustain_long = (long *)sustain_void; - if (*amp_long <= *sustain_long) - { - if (ctc->flags&32) - { - ctc->amp = ctc->sustain; - ctc->cellfunc = docell3; - } - else - ctc->cellfunc = docell2; - } - else - ctc->amp *= ctc->decaymul; - - ctc->t += ctc->tinc; - ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED; -} -void docell0 (void *c, float modulator) -{ - long i; - - ftol(ctc->t+modulator,&i); - - ctc->amp = ((ctc->a3*ctc->amp + ctc->a2)*ctc->amp + ctc->a1)*ctc->amp + ctc->a0; - void *amp_void = &ctc->amp; - long *amp_long = (long *)amp_void; - if (*amp_long > 0x3f800000) - { - ctc->amp = 1; - ctc->cellfunc = docell1; - } - - ctc->t += ctc->tinc; - ctc->val += (ctc->amp*ctc->vol*((float)ctc->waveform[i&ctc->wavemask])-ctc->val)*ADJUSTSPEED; -} - - -static long waveform[8] = {WAVPREC,WAVPREC>>1,WAVPREC,(WAVPREC*3)>>2,0,0,(WAVPREC*5)>>2,WAVPREC<<1}; -static long wavemask[8] = {WAVPREC-1,WAVPREC-1,(WAVPREC>>1)-1,(WAVPREC>>1)-1,WAVPREC-1,((WAVPREC*3)>>2)-1,WAVPREC>>1,WAVPREC-1}; -static long wavestart[8] = {0,WAVPREC>>1,0,WAVPREC>>2,0,0,0,WAVPREC>>3}; -static float attackconst[4] = {1/2.82624,1/2.25280,1/1.88416,1/1.59744}; -static float decrelconst[4] = {1/39.28064,1/31.41608,1/26.17344,1/22.44608}; -void cellon (long i, long j, celltype *c, unsigned char iscarrier) -{ - long frn, oct, toff; - float f; - - frn = ((((long)adlibreg[i+0xb0])&3)<<8) + (long)adlibreg[i+0xa0]; - oct = ((((long)adlibreg[i+0xb0])>>2)&7); - toff = (oct<<1) + ((frn>>9)&((frn>>8)|(((adlibreg[8]>>6)&1)^1))); - if (!(adlibreg[j+0x20]&16)) toff >>= 2; - - f = pow(2.0,(adlibreg[j+0x60]>>4)+(toff>>2)-1)*attackconst[toff&3]*recipsamp; - c->a0 = .0377*f; c->a1 = 10.73*f+1; c->a2 = -17.57*f; c->a3 = 7.42*f; - f = -7.4493*decrelconst[toff&3]*recipsamp; - c->decaymul = pow(2.0,f*pow(2.0,(adlibreg[j+0x60]&15)+(toff>>2))); - c->releasemul = pow(2.0,f*pow(2.0,(adlibreg[j+0x80]&15)+(toff>>2))); - c->wavemask = wavemask[adlibreg[j+0xe0]&7]; - c->waveform = &wavtable[waveform[adlibreg[j+0xe0]&7]]; - if (!(adlibreg[1]&0x20)) c->waveform = &wavtable[WAVPREC]; - c->t = wavestart[adlibreg[j+0xe0]&7]; - c->flags = adlibreg[j+0x20]; - c->cellfunc = docell0; - c->tinc = (float)(frn<vol = pow(2.0,((float)(adlibreg[j+0x40]&63) + - (float)kslmul[adlibreg[j+0x40]>>6]*ksl[oct][frn>>6]) * -.125 - 14); - c->sustain = pow(2.0,(float)(adlibreg[j+0x80]>>4) * -.5); - if (!iscarrier) c->amp = 0; - c->mfb = pow(2.0,((adlibreg[i+0xc0]>>1)&7)+5)*(WAVPREC/2048.0)*MFBFACTOR; - if (!(adlibreg[i+0xc0]&14)) c->mfb = 0; - c->val = 0; -} - -//This function (and bug fix) written by Chris Moeller -void cellfreq (signed long i, signed long j, celltype *c) -{ - long frn, oct; - - frn = ((((long)adlibreg[i+0xb0])&3)<<8) + (long)adlibreg[i+0xa0]; - oct = ((((long)adlibreg[i+0xb0])>>2)&7); - - c->tinc = (float)(frn<vol = pow(2.0,((float)(adlibreg[j+0x40]&63) + - (float)kslmul[adlibreg[j+0x40]>>6]*ksl[oct][frn>>6]) * -.125 - 14); -} - -static long initfirstime = 0; -void adlibinit (long dasamplerate, long danumspeakers, long dabytespersample) -{ - long i, j, frn, oct; - - memset((void *)adlibreg,0,sizeof(adlibreg)); - memset((void *)cell,0,sizeof(celltype)*MAXCELLS); - memset((void *)rbuf,0,sizeof(rbuf)); - rend = 0; odrumstat = 0; - - for(i=0;i=0;i--) nfrqmul[i] = frqmul[i]*recipsamp*FRQSCALE*(WAVPREC/2048.0); - - if (!initfirstime) - { - initfirstime = 1; - - for(i=0;i<(WAVPREC>>1);i++) - { - wavtable[i] = - wavtable[(i<<1) +WAVPREC] = (signed short)(16384*sin((float)((i<<1) )*PI*2/WAVPREC)); - wavtable[(i<<1)+1+WAVPREC] = (signed short)(16384*sin((float)((i<<1)+1)*PI*2/WAVPREC)); - } - for(i=0;i<(WAVPREC>>3);i++) - { - wavtable[i+(WAVPREC<<1)] = wavtable[i+(WAVPREC>>3)]-16384; - wavtable[i+((WAVPREC*17)>>3)] = wavtable[i+(WAVPREC>>2)]+16384; - } - - //[table in book]*8/3 - ksl[7][0] = 0; ksl[7][1] = 24; ksl[7][2] = 32; ksl[7][3] = 37; - ksl[7][4] = 40; ksl[7][5] = 43; ksl[7][6] = 45; ksl[7][7] = 47; - ksl[7][8] = 48; for(i=9;i<16;i++) ksl[7][i] = i+41; - for(j=6;j>=0;j--) - for(i=0;i<16;i++) - { - oct = (long)ksl[j+1][i]-8; if (oct < 0) oct = 0; - ksl[j][i] = (unsigned char)oct; - } - } - else - { - for(i=0;i<9;i++) - { - frn = ((((long)adlibreg[i+0xb0])&3)<<8) + (long)adlibreg[i+0xa0]; - oct = ((((long)adlibreg[i+0xb0])>>2)&7); - cell[i].tinc = (float)(frn< (odrumstat&16)) //BassDrum - { - cellon(6,16,&cell[6],0); - cellon(6,19,&cell[15],1); - cell[15].vol *= 2; - } - if ((v&8) > (odrumstat&8)) //Snare - { - cellon(16,20,&cell[16],0); - cell[16].tinc *= 2*(nfrqmul[adlibreg[17+0x20]&15] / nfrqmul[adlibreg[20+0x20]&15]); - if (((adlibreg[20+0xe0]&7) >= 3) && ((adlibreg[20+0xe0]&7) <= 5)) cell[16].vol = 0; - cell[16].vol *= 2; - } - if ((v&4) > (odrumstat&4)) //TomTom - { - cellon(8,18,&cell[8],0); - cell[8].vol *= 2; - } - if ((v&2) > (odrumstat&2)) //Cymbal - { - cellon(17,21,&cell[17],0); - - cell[17].wavemask = wavemask[5]; - cell[17].waveform = &wavtable[waveform[5]]; - cell[17].tinc *= 16; cell[17].vol *= 2; - - //cell[17].waveform = &wavtable[WAVPREC]; cell[17].wavemask = 0; - //if (((adlibreg[21+0xe0]&7) == 0) || ((adlibreg[21+0xe0]&7) == 6)) - // cell[17].waveform = &wavtable[(WAVPREC*7)>>2]; - //if (((adlibreg[21+0xe0]&7) == 2) || ((adlibreg[21+0xe0]&7) == 3)) - // cell[17].waveform = &wavtable[(WAVPREC*5)>>2]; - } - if ((v&1) > (odrumstat&1)) //Hihat - { - cellon(7,17,&cell[7],0); - if (((adlibreg[17+0xe0]&7) == 1) || ((adlibreg[17+0xe0]&7) == 4) || - ((adlibreg[17+0xe0]&7) == 5) || ((adlibreg[17+0xe0]&7) == 7)) cell[7].vol = 0; - if ((adlibreg[17+0xe0]&7) == 6) { cell[7].wavemask = 0; cell[7].waveform = &wavtable[(WAVPREC*7)>>2]; } - } - - odrumstat = v; - } - else if (((unsigned)(i-0x40) < (unsigned)22) && ((i&7) < 6)) - { - if ((i&7) < 3) // Modulator - cellfreq(base2cell[i-0x40],i-0x40,&cell[base2cell[i-0x40]]); - else // Carrier - cellfreq(base2cell[i-0x40],i-0x40,&cell[base2cell[i-0x40]+9]); - } - else if ((unsigned)(i-0xa0) < (unsigned)9) - { - cellfreq(i-0xa0,modulatorbase[i-0xa0],&cell[i-0xa0]); - cellfreq(i-0xa0,modulatorbase[i-0xa0]+3,&cell[i-0xa0+9]); - } - else if ((unsigned)(i-0xb0) < (unsigned)9) - { - if ((v&32) > (tmp&32)) - { - cellon(i-0xb0,modulatorbase[i-0xb0],&cell[i-0xb0],0); - cellon(i-0xb0,modulatorbase[i-0xb0]+3,&cell[i-0xb0+9],1); - } - else if ((v&32) < (tmp&32)) - cell[i-0xb0].cellfunc = cell[i-0xb0+9].cellfunc = docell2; - cellfreq(i-0xb0,modulatorbase[i-0xb0],&cell[i-0xb0]); - cellfreq(i-0xb0,modulatorbase[i-0xb0]+3,&cell[i-0xb0+9]); - } - - //outdata(i,v); -} - -#ifdef USING_ASM -static long fpuasm; -static float fakeadd = 8388608.0+128.0; -static _inline void clipit8 (float f, long a) -{ - _asm - { - mov edi, a - fld dword ptr f - fadd dword ptr fakeadd - fstp dword ptr fpuasm - mov eax, fpuasm - test eax, 0x007fff00 - jz short skipit - shr eax, 16 - xor eax, -1 - skipit: mov byte ptr [edi], al - } -} - -static _inline void clipit16 (float f, long a) -{ - _asm - { - mov eax, a - fld dword ptr f - fist word ptr [eax] - cmp word ptr [eax], 0x8000 - jne short skipit2 - fst dword ptr [fpuasm] - cmp fpuasm, 0x80000000 - sbb word ptr [eax], 0 - skipit2: fstp st - } -} -#else -static void clipit8(float f,unsigned char *a) { - f/=256.0; - f+=128.0; - if (f>254.5) *a=255; - else if (f<0.5) *a=0; - else *a=f; -} - -static void clipit16(float f,short *a) { - if (f>32766.5) *a=32767; - else if (f<-32767.5) *a=-32768; - else *a=f; -} -#endif - -void adlibsetvolume(int i) { - AMPSCALE=i; -} - -void adlibgetsample (unsigned char *sndptr, long numbytes) -{ - long i, j, k=0, ns, endsamples, rptrs, numsamples; - celltype *cptr; - float f; - short *sndptr2=(short *)sndptr; - - numsamples = (numbytes>>(numspeakers+bytespersample-2)); - - if (bytespersample == 1) f = AMPSCALE/256.0; else f = AMPSCALE; - if (numspeakers == 1) - { - nlvol[0] = lvol[0]*f; - for(i=0;i<9;i++) rptr[i] = &rbuf[0][0]; - rptrs = 1; - } - else - { - rptrs = 0; - for(i=0;i<9;i++) - { - if ((!i) || (lvol[i] != lvol[i-1]) || (rvol[i] != rvol[i-1]) || - (lplc[i] != lplc[i-1]) || (rplc[i] != rplc[i-1])) - { - nlvol[rptrs] = lvol[i]*f; - nrvol[rptrs] = rvol[i]*f; - nlplc[rptrs] = rend-min(max(lplc[i],0),FIFOSIZ); - nrplc[rptrs] = rend-min(max(rplc[i],0),FIFOSIZ); - rptrs++; - } - rptr[i] = &rbuf[rptrs-1][0]; - } - } - - - //CPU time used to be somewhat less when emulator was only mono! - // Because of no delay fifos! - - for(ns=0;ns>1)-1)); //Snare - (cell[7].cellfunc)((void *)&cell[7],k&(WAVPREC-1)); //Hihat - (cell[17].cellfunc)((void *)&cell[17],k&((WAVPREC>>3)-1)); //Cymbal - (cell[8].cellfunc)((void *)&cell[8],0.0); //TomTom - nrptr[7][i] += cell[7].val + cell[16].val; - nrptr[8][i] += cell[8].val + cell[17].val; - } - } - } - for(j=9-1;j>=0;j--) - { - if ((adlibreg[0xbd]&0x20) && (j >= 6) && (j < 9)) continue; - - cptr = &cell[j]; k = j; - if (adlibreg[0xc0+k]&1) - { - if ((cptr[9].cellfunc == docell4) && (cptr->cellfunc == docell4)) continue; - for(i=0;icellfunc)((void *)cptr,cptr->val*cptr->mfb); - (cptr->cellfunc)((void *)&cptr[9],0); - nrptr[j][i] += cptr[9].val + cptr->val; - } - } - else - { - if (cptr[9].cellfunc == docell4) continue; - for(i=0;icellfunc)((void *)cptr,cptr->val*cptr->mfb); - (cptr[9].cellfunc)((void *)&cptr[9],cptr->val*WAVPREC*MODFACTOR); - nrptr[j][i] += cptr[9].val; - } - } - } - - if (numspeakers == 1) - { - if (bytespersample == 1) - { - for(i=endsamples-1;i>=0;i--) - clipit8(nrptr[0][i]*nlvol[0],sndptr+1); - } - else - { - for(i=endsamples-1;i>=0;i--) - clipit16(nrptr[0][i]*nlvol[0],sndptr2+i); - } - } - else - { - memset((void *)snd,0,endsamples*sizeof(float)*2); - for(j=0;j=0;i--) - clipit8(snd[i],sndptr+i); - } - else - { - for(i=(endsamples<<1)-1;i>=0;i--) - clipit16(snd[i],sndptr2+i); - } - } - - sndptr = sndptr+(numspeakers*endsamples); - sndptr2 = sndptr2+(numspeakers*endsamples); - rend = ((rend+endsamples)&(FIFOSIZ*2-1)); - } -} diff --git a/plugins/opl2/adlibemu.h b/plugins/opl2/adlibemu.h deleted file mode 100644 index 8600d787d..000000000 --- a/plugins/opl2/adlibemu.h +++ /dev/null @@ -1,26 +0,0 @@ -/* - * ADLIBEMU.H - * Copyright (C) 1998-2001 Ken Silverman - * Ken Silverman's official web site: "http://www.advsys.net/ken" - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - */ - -void adlibinit(long dasamplerate,long danumspeakers,long dabytespersample); -void adlib0(long i,long v); -void adlibgetsample(void *sndptr,long numbytes); -void adlibsetvolume(int i); -void randoinsts(); -extern float lvol[9],rvol[9],lplc[9],rplc[9]; diff --git a/plugins/opl2/fmopl.c b/plugins/opl2/fmopl.c index 2b0e82b0c..db5180189 100644 --- a/plugins/opl2/fmopl.c +++ b/plugins/opl2/fmopl.c @@ -596,7 +596,7 @@ static void init_timetables( FM_OPL *OPL , int ARRATE , int DRRATE ) OPL->AR_TABLE[i] = rate / ARRATE; OPL->DR_TABLE[i] = rate / DRRATE; } - for (i = 60;i < 76;i++) + for (i = 60;i < 75;i++) { OPL->AR_TABLE[i] = EG_AED-1; OPL->DR_TABLE[i] = OPL->DR_TABLE[60]; diff --git a/plugins/opl2/kemuopl.h b/plugins/opl2/kemuopl.h deleted file mode 100644 index d2ca6e288..000000000 --- a/plugins/opl2/kemuopl.h +++ /dev/null @@ -1,61 +0,0 @@ -/* - * Adplug - Replayer for many OPL2/OPL3 audio file formats. - * Copyright (C) 1999 - 2005 Simon Peter, , et al. - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public - * License as published by the Free Software Foundation; either - * version 2.1 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA - * - * kemuopl.h - Emulated OPL using Ken Silverman's emulator, by Simon Peter - * - */ - -#ifndef H_ADPLUG_KEMUOPL -#define H_ADPLUG_KEMUOPL - -#include "opl.h" -extern "C" { -#include "adlibemu.h" -} - -class CKemuopl: public Copl -{ -public: - CKemuopl(int rate, bool bit16, bool usestereo) - : use16bit(bit16), stereo(usestereo) - { - adlibinit(rate, usestereo ? 2 : 1, bit16 ? 2 : 1); - currType = TYPE_OPL2; - }; - - void update(short *buf, int samples) - { - if(use16bit) samples *= 2; - if(stereo) samples *= 2; - adlibgetsample(buf, samples); - } - - // template methods - void write(int reg, int val) - { - if(currChip == 0) - adlib0(reg, val); - }; - - void init() {}; - -private: - bool use16bit,stereo; -}; - -#endif diff --git a/plugins/opl2/logo.png b/plugins/opl2/logo.png index a5b8256e4..fab030160 100644 Binary files a/plugins/opl2/logo.png and b/plugins/opl2/logo.png differ diff --git a/plugins/opl2/opl2instrument.cpp b/plugins/opl2/opl2instrument.cpp index 34fd186df..0718a3796 100644 --- a/plugins/opl2/opl2instrument.cpp +++ b/plugins/opl2/opl2instrument.cpp @@ -1,7 +1,7 @@ /* * OPL2 FM synth * - * Copyright (c) 2013 Raine M. Ekman + * Copyright (c) 2014 Raine M. Ekman * * This file is part of Linux MultiMedia Studio - http://lmms.sourceforge.net * @@ -23,10 +23,10 @@ */ // TODO: -// - Pitch bend -// - Velocity (and aftertouch) sensitivity -// - in FM mode: OP2 level, add mode: OP1 and OP2 levels +// - Better voice allocation: long releases get cut short :( // - .sbi (or similar) file loading into models +// - RT safety = get rid of mutex = make emulator code thread-safe + // - Extras: // - double release: first release is in effect until noteoff (heard if percussive sound), // second is switched in just before key bit cleared (is this useful???) @@ -53,7 +53,6 @@ #include "opl.h" #include "temuopl.h" -#include "kemuopl.h" #include "embed.cpp" #include "math.h" @@ -92,6 +91,9 @@ Plugin * PLUGIN_EXPORT lmms_plugin_main( Model *, void * _data ) // the emulator code isn't really ready for threads QMutex opl2instrument::emulatorMutex; +// Weird ordering of voice parameters +const unsigned int adlib_opadd[9] = {0x00, 0x01, 0x02, 0x08, 0x09, 0x0A, 0x10, 0x11, 0x12}; + opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) : Instrument( _instrument_track, &OPL2_plugin_descriptor ), m_patchModel( 0, 0, 127, this, tr( "Patch" ) ), @@ -139,10 +141,11 @@ opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) : InstrumentPlayHandle * iph = new InstrumentPlayHandle( this ); engine::mixer()->addPlayHandle( iph ); + // Voices are laid out in a funny way... + // adlib_opadd = {0x00, 0x01, 0x02, 0x08, 0x09, 0x0A, 0x10, 0x11, 0x12}; + // Create an emulator - samplerate, 16 bit, mono - // CTemuopl is the better one, CKemuopl kinda sucks (some sounds silent, pitch goes flat after a while) emulatorMutex.lock(); - // theEmulator = new CKemuopl(engine::mixer()->processingSampleRate(), true, false); theEmulator = new CTemuopl(engine::mixer()->processingSampleRate(), true, false); theEmulator->init(); // Enable waveform selection @@ -155,12 +158,15 @@ opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) : frameCount = engine::mixer()->framesPerPeriod(); renderbuffer = new short[frameCount]; - // Some kind of sane default + // Some kind of sane defaults + pitchbend = 0; tuneEqual(69, 440); for(int i=1; i<9; ++i) { voiceNote[i] = OPL2_VOICE_FREE; + voiceLRU[i] = i; } + connect( engine::mixer(), SIGNAL( sampleRateChanged() ), this, SLOT( reloadEmulator() ) ); // Connect knobs @@ -207,8 +213,15 @@ opl2instrument::opl2instrument( InstrumentTrack * _instrument_track ) : MOD_CON( trem_depth_mdl ); } +opl2instrument::~opl2instrument() { + delete theEmulator; + engine::mixer()->removePlayHandles( instrumentTrack() ); + delete [] renderbuffer; +} + // Samplerate changes when choosing oversampling, so this is more or less mandatory void opl2instrument::reloadEmulator() { + delete theEmulator; emulatorMutex.lock(); theEmulator = new CTemuopl(engine::mixer()->processingSampleRate(), true, false); theEmulator->init(); @@ -216,50 +229,135 @@ void opl2instrument::reloadEmulator() { emulatorMutex.unlock(); for(int i=1; i<9; ++i) { voiceNote[i] = OPL2_VOICE_FREE; + voiceLRU[i] = i; } updatePatch(); } +// This shall only be called from code protected by the holy Mutex! +void opl2instrument::setVoiceVelocity(int voice, int vel) { + int vel_adjusted; + // Velocity calculation, some kind of approximation + // Only calculate for operator 1 if in adding mode, don't want to change timbre + if( fm_mdl.value() == false ) { + vel_adjusted = 63 - ( op1_lvl_mdl.value() * vel/127.0) ; + } else { + vel_adjusted = 63 - op1_lvl_mdl.value(); + } + theEmulator->write(0x40+adlib_opadd[voice], + ( (int)op1_scale_mdl.value() & 0x03 << 6) + + ( vel_adjusted & 0x3f ) ); + + + vel_adjusted = 63 - ( op2_lvl_mdl.value() * vel/127.0 ); + // vel_adjusted = 63 - op2_lvl_mdl.value(); + theEmulator->write(0x43+adlib_opadd[voice], + ( (int)op2_scale_mdl.value() & 0x03 << 6) + + ( vel_adjusted & 0x3f ) ); + // printf("vel %d for voice %d (%f)\n",vel_adjusted,voice,op2_lvl_mdl.value() ); +} + +// Pop least recently used voice - why does it sometimes lose a voice (mostly 0)? +int opl2instrument::popVoice() { + int tmp = voiceLRU[0]; + for( int i=0; i<8; ++i) { + voiceLRU[i] = voiceLRU[i+1]; + } + voiceLRU[8] = OPL2_NO_VOICE; + /* printf("pop: %d %d %d %d %d %d %d %d %d \n", + voiceLRU[0],voiceLRU[1],voiceLRU[2], + voiceLRU[3],voiceLRU[4],voiceLRU[5], + voiceLRU[6],voiceLRU[7],voiceLRU[8]); */ + return tmp; +} + +int opl2instrument::pushVoice(int v) { + int i; + for(i=8; i>0; --i) { + if( voiceLRU[i-1] != OPL2_NO_VOICE ) { + break; + } + } + voiceLRU[i] = v; + /*printf("%d %d %d %d %d %d %d %d %d \n", + voiceLRU[0],voiceLRU[1],voiceLRU[2], + voiceLRU[3],voiceLRU[4],voiceLRU[5], + voiceLRU[6],voiceLRU[7],voiceLRU[8]); */ + return i; +} + bool opl2instrument::handleMidiEvent( const midiEvent & _me, const midiTime & _time ) { emulatorMutex.lock(); - // Real dummy version... Should at least add: - // - smarter voice allocation: - // - reuse same note, now we have round robin-ish - // - what to do when voices run out and so on... - // - mono mode - // - int key; - static int lastvoice=0; - if( _me.m_type == MidiNoteOn ) { - // to get us in line with MIDI + int key, vel, voice, tmp_pb; + + switch(_me.m_type) { + case MidiNoteOn: + // to get us in line with MIDI(?) key = _me.key() +12; - for(int i=lastvoice+1; i!=lastvoice; ++i,i%=9) { - if( voiceNote[i] == OPL2_VOICE_FREE ) { - theEmulator->write(0xA0+i, fnums[key] & 0xff); - theEmulator->write(0xB0+i, 32 + ((fnums[key] & 0x1f00) >> 8) ); - // printf("%d: %d %d\n", key, (fnums[key] & 0x1c00) >> 10, fnums[key] & 0x3ff); - voiceNote[i] = key; - // printf("Voice %d on\n",i); - lastvoice=i; - break; + vel = _me.velocity(); + + voice = popVoice(); + if( voice != OPL2_NO_VOICE ) { + // Turn voice on, NB! the frequencies are straight by voice number, + // not by the adlib_opadd table! + theEmulator->write(0xA0+voice, fnums[key] & 0xff); + theEmulator->write(0xB0+voice, 32 + ((fnums[key] & 0x1f00) >> 8) ); + setVoiceVelocity(voice, vel); + voiceNote[voice] = key; + velocities[key] = vel; + // printf("%d %d\n",voice,vel); + } + break; + case MidiNoteOff: + key = _me.key() +12; + for(voice=0; voice<9; ++voice) { + if( voiceNote[voice] == key ) { + theEmulator->write(0xA0+voice, fnums[key] & 0xff); + theEmulator->write(0xB0+voice, (fnums[key] & 0x1f00) >> 8 ); + voiceNote[voice] = OPL2_VOICE_FREE; + pushVoice(voice); + } + } + velocities[key] = 0; + break; + case MidiKeyPressure: + key = _me.key() +12; + vel = _me.velocity(); + if( velocities[key] != 0) { + velocities[key] = vel; + } + for(voice=0; voice<9; ++voice) { + if(voiceNote[voice] == key) { + setVoiceVelocity(voice, vel); } } - } else if( _me.m_type == MidiNoteOff ) { - key = _me.key() +12; - for(int i=0; i<9; ++i) { - if( voiceNote[i] == key ) { - theEmulator->write(0xA0+i, fnums[key] & 0xff); - theEmulator->write(0xB0+i, (fnums[key] & 0x1f00) >> 8 ); - voiceNote[i] = OPL2_VOICE_FREE; - } + break; + case MidiPitchBend: + // Update fnumber table + // Pitchbend should be in the range 0...16383 but the new range knob gets it wrong. + // tmp_pb = (2*BEND_CENTS)*((float)_me.m_data.m_param[0]/16383)-BEND_CENTS; + + // Something like 100 cents = 8192, but offset by 8192 so the +/-100 cents range goes from 0...16383? + tmp_pb = ( _me.m_data.m_param[0]-8192 ) * BEND_CENTS / 8192; + + printf("Pitch bend: %d -> %d cents\n",_me.m_data.m_param[0],tmp_pb); + if( tmp_pb != pitchbend ) { + pitchbend = tmp_pb; + tuneEqual(69, 440.0); } - } else { - printf("Midi event type %d\n",_me.m_type); - // 224 - pitch wheel - // 160 - aftertouch? - } + // Update pitch of sounding notes + for( int v=0; v<9; ++v ) { + if( voiceNote[v] != OPL2_VOICE_FREE ) { + theEmulator->write(0xA0+v, fnums[voiceNote[v] ] & 0xff); + theEmulator->write(0xB0+v, 32 + ((fnums[voiceNote[v]] & 0x1f00) >> 8) ); + } + } + break; + default: + printf("Midi event type %d\n",_me.m_type); + } emulatorMutex.unlock(); return true; } @@ -332,7 +430,6 @@ void opl2instrument::saveSettings( QDomDocument & _doc, QDomElement & _this ) void opl2instrument::loadSettings( const QDomElement & _this ) { - printf("loadSettings!\n"); op1_a_mdl.loadSettings( _this, "op1_a" ); op1_d_mdl.loadSettings( _this, "op1_d" ); op1_s_mdl.loadSettings( _this, "op1_s" ); @@ -366,19 +463,14 @@ void opl2instrument::loadSettings( const QDomElement & _this ) } -// Load a preset in binary form +// Load a patch into the emulator void opl2instrument::loadPatch(unsigned char inst[14]) { - const unsigned int adlib_opadd[] = {0x00, 0x01, 0x02, 0x08, 0x09, 0x0A, 0x10, 0x11, 0x12}; - // Set all voices - printf("%02x %02x %02x %02x %02x ",inst[0],inst[1],inst[2],inst[3],inst[4]); - printf("%02x %02x %02x %02x %02x %02x\n",inst[5],inst[6],inst[7],inst[8],inst[9],inst[10]); - emulatorMutex.lock(); for(int v=0; v<9; ++v) { theEmulator->write(0x20+adlib_opadd[v],inst[0]); // op1 AM/VIB/EG/KSR/Multiplier theEmulator->write(0x23+adlib_opadd[v],inst[1]); // op2 - theEmulator->write(0x40+adlib_opadd[v],inst[2]); // op1 KSL/Output Level - theEmulator->write(0x43+adlib_opadd[v],inst[3]); // op2 + // theEmulator->write(0x40+adlib_opadd[v],inst[2]); // op1 KSL/Output Level - these are handled by noteon/aftertouch code + // theEmulator->write(0x43+adlib_opadd[v],inst[3]); // op2 theEmulator->write(0x60+adlib_opadd[v],inst[4]); // op1 A/D theEmulator->write(0x63+adlib_opadd[v],inst[5]); // op2 theEmulator->write(0x80+adlib_opadd[v],inst[6]); // op1 S/R @@ -391,10 +483,10 @@ void opl2instrument::loadPatch(unsigned char inst[14]) { } void opl2instrument::tuneEqual(int center, float Hz) { + float tmp; for(int n=0; n<128; ++n) { - float tmp = Hz*pow(2, (n-center)/12.0); + tmp = Hz*pow( 2, ( n - center ) / 12.0 + pitchbend / 1200.0 ); fnums[n] = Hz2fnum( tmp ); - //printf("%d: %d %d %f\n", n, (fnums[n] & 0x1c00) >> 10, fnums[n] & 0x3ff,tmp); } } @@ -412,7 +504,6 @@ int opl2instrument::Hz2fnum(float Hz) { // Load one of the default patches void opl2instrument::loadGMPatch() { unsigned char *inst = midi_fm_instruments[m_patchModel.value()]; - // printf("loadGMPatch: %d ", m_patchModel.value()); loadPatch(inst); } @@ -423,7 +514,6 @@ void opl2instrument::loadGMPatch() { // Update patch from the models to the chip emulation void opl2instrument::updatePatch() { - printf("updatePatch()\n"); unsigned char *inst = midi_fm_instruments[0]; inst[0] = ( op1_trem_mdl.value() ? 128 : 0 ) + ( op1_vib_mdl.value() ? 64 : 0 ) + @@ -456,10 +546,16 @@ void opl2instrument::updatePatch() { inst[12] = 0; inst[13] = 0; - // Not part of the patch per se + // Not part of the per-voice patch info theEmulator->write(0xBD, (trem_depth_mdl.value() ? 128 : 0 ) + (vib_depth_mdl.value() ? 64 : 0 )); + // have to do this, as the level knobs might've changed + for( int voice = 0; voice < 9 ; ++voice) { + if(voiceNote[voice]!=OPL2_VOICE_FREE) { + setVoiceVelocity(voice, velocities[voiceNote[voice]] ); + } + } loadPatch(inst); } @@ -552,9 +648,9 @@ opl2instrumentView::opl2instrumentView( Instrument * _instrument, pal.setBrush( backgroundRole(), PLUGIN_NAME::getIconPixmap( "artwork" ) ); setPalette( pal ); - - - +} +opl2instrumentView::~opl2instrumentView() { + // Nobody else seems to delete their knobs and buttons? } void opl2instrumentView::modelChanged() @@ -604,4 +700,5 @@ void opl2instrumentView::modelChanged() } + #include "moc_opl2instrument.cxx" diff --git a/plugins/opl2/opl2instrument.h b/plugins/opl2/opl2instrument.h index 6933d7479..af2493f95 100644 --- a/plugins/opl2/opl2instrument.h +++ b/plugins/opl2/opl2instrument.h @@ -34,12 +34,17 @@ #include "pixmap_button.h" #define OPL2_VOICE_FREE 255 +#define OPL2_NO_VOICE 255 +// The "normal" range for LMMS pitchbends +#define BEND_CENTS 100 class opl2instrument : public Instrument { Q_OBJECT public: opl2instrument( InstrumentTrack * _instrument_track ); + virtual ~opl2instrument(); + virtual QString nodeName() const; virtual PluginView * instantiateView( QWidget * _parent ); @@ -107,20 +112,33 @@ private: fpp_t frameCount; short *renderbuffer; int voiceNote[9]; - int heldNotes[128]; + // Least recently used voices + int voiceLRU[9]; + // 0 - no note, >0 - note on velocity + int velocities[128]; // These include both octave and Fnumber int fnums[128]; + // in cents, range defaults to +/-100 cents (should this be changeable?) + int pitchbend; + + + + int popVoice(); + int pushVoice(int v); int Hz2fnum(float Hz); static QMutex emulatorMutex; + void setVoiceVelocity(int voice, int vel); }; + class opl2instrumentView : public InstrumentView { Q_OBJECT public: opl2instrumentView( Instrument * _instrument, QWidget * _parent ); + virtual ~opl2instrumentView(); lcdSpinBox *m_patch; void modelChanged();