/* MikMod sound library (c) 1998, 1999, 2000 Miodrag Vallat and others - see file AUTHORS for complete list. This library is free software; you can redistribute it and/or modify it under the terms of the GNU Library General Public License as published by the Free Software Foundation; either version 2 of the License, or (at your option) any later version. This program 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 Library General Public License for more details. You should have received a copy of the GNU Library 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. */ /*============================================================================== $Id$ High-quality sample mixing routines, using a 32 bits mixing buffer, interpolation, and sample smoothing to improve sound quality and remove clicks. ==============================================================================*/ /* Future Additions: Low-Pass filter to remove annoying staticy buzz. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #ifdef HAVE_MEMORY_H #include #endif #include #include "mikmod_internals.h" /* Constant Definitions ==================== MAXVOL_FACTOR (was BITSHIFT in virtch.c) Controls the maximum volume of the output data. All mixed data is divided by this number after mixing, so larger numbers result in quieter mixing. Smaller numbers will increase the likeliness of distortion on loud modules. REVERBERATION Larger numbers result in shorter reverb duration. Longer reverb durations can cause unwanted static and make the reverb sound more like a crappy echo. SAMPLING_SHIFT Specified the shift multiplier which controls by how much the mixing rate is multiplied while mixing. Higher values can improve quality by smoothing the sound and reducing pops and clicks. Note, this is a shift value, so a value of 2 becomes a mixing-rate multiplier of 4, and a value of 3 = 8, etc. FRACBITS The number of bits per integer devoted to the fractional part of the number. Generally, this number should not be changed for any reason. !!! IMPORTANT !!! All values below MUST ALWAYS be greater than 0 */ #define MAXVOL_FACTOR (1<<9) #define REVERBERATION 11000L #define SAMPLING_SHIFT 2 #define SAMPLING_FACTOR (1UL< sample has to be restarted */ UBYTE active; /* =1 -> sample is playing */ UWORD flags; /* 16/8 bits looping/one-shot */ SWORD handle; /* identifies the sample */ ULONG start; /* start index */ ULONG size; /* samplesize */ ULONG reppos; /* loop start */ ULONG repend; /* loop end */ ULONG frq; /* current frequency */ int vol; /* current volume */ int pan; /* current panning position */ int click; int rampvol; SLONG lastvalL,lastvalR; int lvolsel,rvolsel; /* Volume factor in range 0-255 */ int oldlvol,oldrvol; SLONGLONG current; /* current index in the sample */ SLONGLONG increment; /* increment value */ } VINFO; static SWORD **Samples; static VINFO *vinf=NULL,*vnf; static long tickleft,samplesthatfit,vc_memory=0; static int vc_softchn; static SLONGLONG idxsize,idxlpos,idxlend; static SLONG *vc_tickbuf=NULL; static UWORD vc_mode; /* Reverb control variables */ static int RVc1, RVc2, RVc3, RVc4, RVc5, RVc6, RVc7, RVc8; static ULONG RVRindex; /* For Mono or Left Channel */ static SLONG *RVbufL1=NULL,*RVbufL2=NULL,*RVbufL3=NULL,*RVbufL4=NULL, *RVbufL5=NULL,*RVbufL6=NULL,*RVbufL7=NULL,*RVbufL8=NULL; /* For Stereo only (Right Channel) */ static SLONG *RVbufR1=NULL,*RVbufR2=NULL,*RVbufR3=NULL,*RVbufR4=NULL, *RVbufR5=NULL,*RVbufR6=NULL,*RVbufR7=NULL,*RVbufR8=NULL; #ifdef NATIVE_64BIT_INT #define NATIVE SLONGLONG #else #define NATIVE SLONG #endif /*========== 32 bit sample mixers - only for 32 bit platforms */ #ifndef NATIVE_64BIT_INT static SLONG Mix32MonoNormal(const SWORD* srce,SLONG* dest,SLONG index,SLONG increment,SLONG todo) { SWORD sample=0; SLONG i,f; while(todo--) { i=index>>FRACBITS,f=index&FRACMASK; sample=(((SLONG)(srce[i]*(FRACMASK+1L-f)) + ((SLONG)srce[i+1]*f)) >> FRACBITS); index+=increment; if(vnf->rampvol) { *dest++ += (long)( ( ( (SLONG)(vnf->oldlvol*vnf->rampvol) + (vnf->lvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONG)sample ) >> CLICK_SHIFT ); vnf->rampvol--; } else if(vnf->click) { *dest++ += (long)( ( ( ((SLONG)vnf->lvolsel*(CLICK_BUFFER-vnf->click)) * (SLONG)sample ) + (vnf->lastvalL*vnf->click) ) >> CLICK_SHIFT ); vnf->click--; } else *dest++ +=vnf->lvolsel*sample; } vnf->lastvalL=vnf->lvolsel * sample; return index; } static SLONG Mix32StereoNormal(const SWORD* srce,SLONG* dest,SLONG index,SLONG increment,ULONG todo) { SWORD sample=0; SLONG i,f; while(todo--) { i=index>>FRACBITS,f=index&FRACMASK; sample=((((SLONG)srce[i]*(FRACMASK+1L-f)) + ((SLONG)srce[i+1] * f)) >> FRACBITS); index += increment; if(vnf->rampvol) { *dest++ += (long)( ( ( ((SLONG)vnf->oldlvol*vnf->rampvol) + (vnf->lvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONG)sample ) >> CLICK_SHIFT ); *dest++ += (long)( ( ( ((SLONG)vnf->oldrvol*vnf->rampvol) + (vnf->rvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONG)sample ) >> CLICK_SHIFT ); vnf->rampvol--; } else if(vnf->click) { *dest++ += (long)( ( ( (SLONG)(vnf->lvolsel*(CLICK_BUFFER-vnf->click)) * (SLONG)sample ) + (vnf->lastvalL * vnf->click) ) >> CLICK_SHIFT ); *dest++ += (long)( ( ( ((SLONG)vnf->rvolsel*(CLICK_BUFFER-vnf->click)) * (SLONG)sample ) + (vnf->lastvalR * vnf->click) ) >> CLICK_SHIFT ); vnf->click--; } else { *dest++ +=vnf->lvolsel*sample; *dest++ +=vnf->rvolsel*sample; } } vnf->lastvalL=vnf->lvolsel*sample; vnf->lastvalR=vnf->rvolsel*sample; return index; } static SLONG Mix32StereoSurround(const SWORD* srce,SLONG* dest,SLONG index,SLONG increment,ULONG todo) { SWORD sample=0; long whoop; SLONG i, f; while(todo--) { i=index>>FRACBITS,f=index&FRACMASK; sample=((((SLONG)srce[i]*(FRACMASK+1L-f)) + ((SLONG)srce[i+1]*f)) >> FRACBITS); index+=increment; if(vnf->rampvol) { whoop=(long)( ( ( (SLONG)(vnf->oldlvol*vnf->rampvol) + (vnf->lvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONG)sample) >> CLICK_SHIFT ); *dest++ +=whoop; *dest++ -=whoop; vnf->rampvol--; } else if(vnf->click) { whoop = (long)( ( ( ((SLONG)vnf->lvolsel*(CLICK_BUFFER-vnf->click)) * (SLONG)sample) + (vnf->lastvalL * vnf->click) ) >> CLICK_SHIFT ); *dest++ +=whoop; *dest++ -=whoop; vnf->click--; } else { *dest++ +=vnf->lvolsel*sample; *dest++ -=vnf->lvolsel*sample; } } vnf->lastvalL=vnf->lvolsel*sample; vnf->lastvalR=vnf->lvolsel*sample; return index; } #endif /*========== 64 bit mixers */ static SLONGLONG MixMonoNormal(const SWORD* srce,SLONG* dest,SLONGLONG index,SLONGLONG increment,SLONG todo) { SWORD sample=0; SLONGLONG i,f; while(todo--) { i=index>>FRACBITS,f=index&FRACMASK; sample=(((SLONGLONG)(srce[i]*(FRACMASK+1L-f)) + ((SLONGLONG)srce[i+1]*f)) >> FRACBITS); index+=increment; if(vnf->rampvol) { *dest++ += (long)( ( ( (SLONGLONG)(vnf->oldlvol*vnf->rampvol) + (vnf->lvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONGLONG)sample ) >> CLICK_SHIFT ); vnf->rampvol--; } else if(vnf->click) { *dest++ += (long)( ( ( ((SLONGLONG)vnf->lvolsel*(CLICK_BUFFER-vnf->click)) * (SLONGLONG)sample ) + (vnf->lastvalL*vnf->click) ) >> CLICK_SHIFT ); vnf->click--; } else *dest++ +=vnf->lvolsel*sample; } vnf->lastvalL=vnf->lvolsel * sample; return index; } static SLONGLONG MixStereoNormal(const SWORD* srce,SLONG* dest,SLONGLONG index,SLONGLONG increment,ULONG todo) { SWORD sample=0; SLONGLONG i,f; while(todo--) { i=index>>FRACBITS,f=index&FRACMASK; sample=((((SLONGLONG)srce[i]*(FRACMASK+1L-f)) + ((SLONGLONG)srce[i+1] * f)) >> FRACBITS); index += increment; if(vnf->rampvol) { *dest++ += (long)( ( ( ((SLONGLONG)vnf->oldlvol*vnf->rampvol) + (vnf->lvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONGLONG)sample ) >> CLICK_SHIFT ); *dest++ += (long)( ( ( ((SLONGLONG)vnf->oldrvol*vnf->rampvol) + (vnf->rvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONGLONG)sample ) >> CLICK_SHIFT ); vnf->rampvol--; } else if(vnf->click) { *dest++ += (long)( ( ( (SLONGLONG)(vnf->lvolsel*(CLICK_BUFFER-vnf->click)) * (SLONGLONG)sample ) + (vnf->lastvalL * vnf->click) ) >> CLICK_SHIFT ); *dest++ += (long)( ( ( ((SLONGLONG)vnf->rvolsel*(CLICK_BUFFER-vnf->click)) * (SLONGLONG)sample ) + (vnf->lastvalR * vnf->click) ) >> CLICK_SHIFT ); vnf->click--; } else { *dest++ +=vnf->lvolsel*sample; *dest++ +=vnf->rvolsel*sample; } } vnf->lastvalL=vnf->lvolsel*sample; vnf->lastvalR=vnf->rvolsel*sample; return index; } static SLONGLONG MixStereoSurround(const SWORD* srce,SLONG* dest,SLONGLONG index,SLONGLONG increment,ULONG todo) { SWORD sample=0; long whoop; SLONGLONG i, f; while(todo--) { i=index>>FRACBITS,f=index&FRACMASK; sample=((((SLONGLONG)srce[i]*(FRACMASK+1L-f)) + ((SLONGLONG)srce[i+1]*f)) >> FRACBITS); index+=increment; if(vnf->rampvol) { whoop=(long)( ( ( (SLONGLONG)(vnf->oldlvol*vnf->rampvol) + (vnf->lvolsel*(CLICK_BUFFER-vnf->rampvol)) ) * (SLONGLONG)sample) >> CLICK_SHIFT ); *dest++ +=whoop; *dest++ -=whoop; vnf->rampvol--; } else if(vnf->click) { whoop = (long)( ( ( ((SLONGLONG)vnf->lvolsel*(CLICK_BUFFER-vnf->click)) * (SLONGLONG)sample) + (vnf->lastvalL * vnf->click) ) >> CLICK_SHIFT ); *dest++ +=whoop; *dest++ -=whoop; vnf->click--; } else { *dest++ +=vnf->lvolsel*sample; *dest++ -=vnf->lvolsel*sample; } } vnf->lastvalL=vnf->lvolsel*sample; vnf->lastvalR=vnf->lvolsel*sample; return index; } static void(*Mix32to16)(SWORD* dste,const SLONG* srce,NATIVE count); static void(*Mix32to8)(SBYTE* dste,const SLONG* srce,NATIVE count); static void(*MixReverb)(SLONG* srce,NATIVE count); /* Reverb macros */ #define COMPUTE_LOC(n) loc##n = RVRindex % RVc##n #define COMPUTE_LECHO(n) RVbufL##n [loc##n ]=speedup+((ReverbPct*RVbufL##n [loc##n ])>>7) #define COMPUTE_RECHO(n) RVbufR##n [loc##n ]=speedup+((ReverbPct*RVbufR##n [loc##n ])>>7) static void MixReverb_Normal(SLONG* srce,NATIVE count) { NATIVE speedup; int ReverbPct; unsigned int loc1,loc2,loc3,loc4,loc5,loc6,loc7,loc8; ReverbPct=58+(md_reverb*4); COMPUTE_LOC(1); COMPUTE_LOC(2); COMPUTE_LOC(3); COMPUTE_LOC(4); COMPUTE_LOC(5); COMPUTE_LOC(6); COMPUTE_LOC(7); COMPUTE_LOC(8); while(count--) { /* Compute the left channel echo buffers */ speedup = *srce >> 3; COMPUTE_LECHO(1); COMPUTE_LECHO(2); COMPUTE_LECHO(3); COMPUTE_LECHO(4); COMPUTE_LECHO(5); COMPUTE_LECHO(6); COMPUTE_LECHO(7); COMPUTE_LECHO(8); /* Prepare to compute actual finalized data */ RVRindex++; COMPUTE_LOC(1); COMPUTE_LOC(2); COMPUTE_LOC(3); COMPUTE_LOC(4); COMPUTE_LOC(5); COMPUTE_LOC(6); COMPUTE_LOC(7); COMPUTE_LOC(8); /* left channel */ *srce++ +=RVbufL1[loc1]-RVbufL2[loc2]+RVbufL3[loc3]-RVbufL4[loc4]+ RVbufL5[loc5]-RVbufL6[loc6]+RVbufL7[loc7]-RVbufL8[loc8]; } } static void MixReverb_Stereo(SLONG *srce,NATIVE count) { NATIVE speedup; int ReverbPct; unsigned int loc1,loc2,loc3,loc4,loc5,loc6,loc7,loc8; ReverbPct=58+(md_reverb*4); COMPUTE_LOC(1); COMPUTE_LOC(2); COMPUTE_LOC(3); COMPUTE_LOC(4); COMPUTE_LOC(5); COMPUTE_LOC(6); COMPUTE_LOC(7); COMPUTE_LOC(8); while(count--) { /* Compute the left channel echo buffers */ speedup = *srce >> 3; COMPUTE_LECHO(1); COMPUTE_LECHO(2); COMPUTE_LECHO(3); COMPUTE_LECHO(4); COMPUTE_LECHO(5); COMPUTE_LECHO(6); COMPUTE_LECHO(7); COMPUTE_LECHO(8); /* Compute the right channel echo buffers */ speedup = srce[1] >> 3; COMPUTE_RECHO(1); COMPUTE_RECHO(2); COMPUTE_RECHO(3); COMPUTE_RECHO(4); COMPUTE_RECHO(5); COMPUTE_RECHO(6); COMPUTE_RECHO(7); COMPUTE_RECHO(8); /* Prepare to compute actual finalized data */ RVRindex++; COMPUTE_LOC(1); COMPUTE_LOC(2); COMPUTE_LOC(3); COMPUTE_LOC(4); COMPUTE_LOC(5); COMPUTE_LOC(6); COMPUTE_LOC(7); COMPUTE_LOC(8); /* left channel */ *srce++ +=RVbufL1[loc1]-RVbufL2[loc2]+RVbufL3[loc3]-RVbufL4[loc4]+ RVbufL5[loc5]-RVbufL6[loc6]+RVbufL7[loc7]-RVbufL8[loc8]; /* right channel */ *srce++ +=RVbufR1[loc1]-RVbufR2[loc2]+RVbufR3[loc3]-RVbufR4[loc4]+ RVbufR5[loc5]-RVbufR6[loc6]+RVbufR7[loc7]-RVbufR8[loc8]; } } /* Mixing macros */ #define EXTRACT_SAMPLE(var,attenuation) var=*srce++/(MAXVOL_FACTOR*attenuation) #define CHECK_SAMPLE(var,bound) var=(var>=bound)?bound-1:(var<-bound)?-bound:var static void Mix32To16_Normal(SWORD* dste,const SLONG* srce,NATIVE count) { NATIVE x1,x2,tmpx; int i; for(count/=SAMPLING_FACTOR;count;count--) { tmpx=0; for(i=SAMPLING_FACTOR/2;i;i--) { EXTRACT_SAMPLE(x1,1); EXTRACT_SAMPLE(x2,1); CHECK_SAMPLE(x1,32768); CHECK_SAMPLE(x2,32768); tmpx+=x1+x2; } *dste++ =tmpx/SAMPLING_FACTOR; } } static void Mix32To16_Stereo(SWORD* dste,const SLONG* srce,NATIVE count) { NATIVE x1,x2,x3,x4,tmpx,tmpy; int i; for(count/=SAMPLING_FACTOR;count;count--) { tmpx=tmpy=0; for(i=SAMPLING_FACTOR/2;i;i--) { EXTRACT_SAMPLE(x1,1); EXTRACT_SAMPLE(x2,1); EXTRACT_SAMPLE(x3,1); EXTRACT_SAMPLE(x4,1); CHECK_SAMPLE(x1,32768); CHECK_SAMPLE(x2,32768); CHECK_SAMPLE(x3,32768); CHECK_SAMPLE(x4,32768); tmpx+=x1+x3; tmpy+=x2+x4; } *dste++ =tmpx/SAMPLING_FACTOR; *dste++ =tmpy/SAMPLING_FACTOR; } } static void Mix32To8_Normal(SBYTE* dste,const SLONG* srce,NATIVE count) { NATIVE x1,x2,tmpx; int i; for(count/=SAMPLING_FACTOR;count;count--) { tmpx = 0; for(i=SAMPLING_FACTOR/2;i;i--) { EXTRACT_SAMPLE(x1,256); EXTRACT_SAMPLE(x2,256); CHECK_SAMPLE(x1,128); CHECK_SAMPLE(x2,128); tmpx+=x1+x2; } *dste++ =(tmpx/SAMPLING_FACTOR)+128; } } static void Mix32To8_Stereo(SBYTE* dste,const SLONG* srce,NATIVE count) { NATIVE x1,x2,x3,x4,tmpx,tmpy; int i; for(count/=SAMPLING_FACTOR;count;count--) { tmpx=tmpy=0; for(i=SAMPLING_FACTOR/2;i;i--) { EXTRACT_SAMPLE(x1,256); EXTRACT_SAMPLE(x2,256); EXTRACT_SAMPLE(x3,256); EXTRACT_SAMPLE(x4,256); CHECK_SAMPLE(x1,128); CHECK_SAMPLE(x2,128); CHECK_SAMPLE(x3,128); CHECK_SAMPLE(x4,128); tmpx+=x1+x3; tmpy+=x2+x4; } *dste++ =(tmpx/SAMPLING_FACTOR)+128; *dste++ =(tmpy/SAMPLING_FACTOR)+128; } } static void AddChannel(SLONG* ptr,NATIVE todo) { SLONGLONG end,done; SWORD *s; if(!(s=Samples[vnf->handle])) { vnf->current = vnf->active = 0; vnf->lastvalL = vnf->lastvalR = 0; return; } /* update the 'current' index so the sample loops, or stops playing if it reached the end of the sample */ while(todo>0) { SLONGLONG endpos; if(vnf->flags & SF_REVERSE) { /* The sample is playing in reverse */ if((vnf->flags&SF_LOOP)&&(vnf->currentflags & SF_BIDI) { /* sample is doing bidirectional loops, so 'bounce' the current index against the idxlpos */ vnf->current = idxlpos+(idxlpos-vnf->current); vnf->flags &= ~SF_REVERSE; vnf->increment = -vnf->increment; } else /* normal backwards looping, so set the current position to loopend index */ vnf->current=idxlend-(idxlpos-vnf->current); } else { /* the sample is not looping, so check if it reached index 0 */ if(vnf->current < 0) { /* playing index reached 0, so stop playing this sample */ vnf->current = vnf->active = 0; break; } } } else { /* The sample is playing forward */ if((vnf->flags & SF_LOOP) && (vnf->current >= idxlend)) { /* the sample is looping, check the loopend index */ if(vnf->flags & SF_BIDI) { /* sample is doing bidirectional loops, so 'bounce' the current index against the idxlend */ vnf->flags |= SF_REVERSE; vnf->increment = -vnf->increment; vnf->current = idxlend-(vnf->current-idxlend); } else /* normal backwards looping, so set the current position to loopend index */ vnf->current=idxlpos+(vnf->current-idxlend); } else { /* sample is not looping, so check if it reached the last position */ if(vnf->current >= idxsize) { /* yes, so stop playing this sample */ vnf->current = vnf->active = 0; break; } } } end=(vnf->flags&SF_REVERSE)?(vnf->flags&SF_LOOP)?idxlpos:0: (vnf->flags&SF_LOOP)?idxlend:idxsize; /* if the sample is not blocked... */ if((end==vnf->current)||(!vnf->increment)) done=0; else { done=MIN((end-vnf->current)/vnf->increment+1,todo); if(done<0) done=0; } if(!done) { vnf->active = 0; break; } endpos=vnf->current+done*vnf->increment; if(vnf->vol || vnf->rampvol) { #ifndef NATIVE_64BIT_INT /* use the 32 bit mixers as often as we can (they're much faster) */ if((vnf->current<0x7fffffff)&&(endpos<0x7fffffff)) { if(vc_mode & DMODE_STEREO) { if((vnf->pan==PAN_SURROUND)&&(vc_mode&DMODE_SURROUND)) vnf->current=Mix32StereoSurround (s,ptr,vnf->current,vnf->increment,done); else vnf->current=Mix32StereoNormal (s,ptr,vnf->current,vnf->increment,done); } else vnf->current=Mix32MonoNormal (s,ptr,vnf->current,vnf->increment,done); } else #endif { if(vc_mode & DMODE_STEREO) { if((vnf->pan==PAN_SURROUND)&&(vc_mode&DMODE_SURROUND)) vnf->current=MixStereoSurround (s,ptr,vnf->current,vnf->increment,done); else vnf->current=MixStereoNormal (s,ptr,vnf->current,vnf->increment,done); } else vnf->current=MixMonoNormal (s,ptr,vnf->current,vnf->increment,done); } } else { vnf->lastvalL = vnf->lastvalR = 0; /* update sample position */ vnf->current=endpos; } todo -= done; ptr +=(vc_mode & DMODE_STEREO)?(done<<1):done; } } #define _IN_VIRTCH_ #define VC1_SilenceBytes VC2_SilenceBytes #define VC1_WriteSamples VC2_WriteSamples #define VC1_WriteBytes VC2_WriteBytes #define VC1_Exit VC2_Exit #define VC1_VoiceSetVolume VC2_VoiceSetVolume #define VC1_VoiceGetVolume VC2_VoiceGetVolume #define VC1_VoiceSetPanning VC2_VoiceSetPanning #define VC1_VoiceGetPanning VC2_VoiceGetPanning #define VC1_VoiceSetFrequency VC2_VoiceSetFrequency #define VC1_VoiceGetFrequency VC2_VoiceGetFrequency #define VC1_VoicePlay VC2_VoicePlay #define VC1_VoiceStop VC2_VoiceStop #define VC1_VoiceStopped VC2_VoiceStopped #define VC1_VoiceGetPosition VC2_VoiceGetPosition #define VC1_SampleUnload VC2_SampleUnload #define VC1_SampleLoad VC2_SampleLoad #define VC1_SampleSpace VC2_SampleSpace #define VC1_SampleLength VC2_SampleLength #define VC1_VoiceRealVolume VC2_VoiceRealVolume #include "virtch_common.c" #undef _IN_VIRTCH_ void VC2_WriteSamples(SBYTE* buf,ULONG todo) { int left,portion=0; SBYTE *buffer; int t,pan,vol; todo*=SAMPLING_FACTOR; while(todo) { if(!tickleft) { if(vc_mode & DMODE_SOFT_MUSIC) md_player(); tickleft=(md_mixfreq*125L*SAMPLING_FACTOR)/(md_bpm*50L); tickleft&=~(SAMPLING_FACTOR-1); } left = MIN(tickleft, (long)todo); buffer = buf; tickleft -= left; todo -= left; buf += samples2bytes(left)/SAMPLING_FACTOR; while(left) { portion = MIN(left, samplesthatfit); memset(vc_tickbuf,0,portion<<((vc_mode&DMODE_STEREO)?3:2)); for(t=0;tkick) { vnf->current=((SLONGLONG)(vnf->start))<kick = 0; vnf->active = 1; vnf->click = CLICK_BUFFER; vnf->rampvol = 0; } if(!vnf->frq) vnf->active = 0; if(vnf->active) { vnf->increment=((SLONGLONG)(vnf->frq)<<(FRACBITS-SAMPLING_SHIFT)) /md_mixfreq; if(vnf->flags&SF_REVERSE) vnf->increment=-vnf->increment; vol = vnf->vol; pan = vnf->pan; vnf->oldlvol=vnf->lvolsel;vnf->oldrvol=vnf->rvolsel; if(vc_mode & DMODE_STEREO) { if(pan!=PAN_SURROUND) { vnf->lvolsel=(vol*(PAN_RIGHT-pan))>>8; vnf->rvolsel=(vol*pan)>>8; } else { vnf->lvolsel=vnf->rvolsel=(vol * 256L) / 480; } } else vnf->lvolsel=vol; idxsize=(vnf->size)?((SLONGLONG)(vnf->size)<repend)?((SLONGLONG)(vnf->repend)<reppos)<15) md_reverb=15; MixReverb(vc_tickbuf,portion); } if(vc_mode & DMODE_16BITS) Mix32to16((SWORD*)buffer,vc_tickbuf,portion); else Mix32to8((SBYTE*)buffer,vc_tickbuf,portion); buffer += samples2bytes(portion) / SAMPLING_FACTOR; left -= portion; } } } BOOL VC2_Init(void) { VC_SetupPointers(); if (!(md_mode&DMODE_HQMIXER)) return VC1_Init(); if(!(Samples=(SWORD**)MikMod_calloc(MAXSAMPLEHANDLES,sizeof(SWORD*)))) { _mm_errno = MMERR_INITIALIZING_MIXER; return 1; } if(!vc_tickbuf) if(!(vc_tickbuf=(SLONG*)MikMod_malloc((TICKLSIZE+32)*sizeof(SLONG)))) { _mm_errno = MMERR_INITIALIZING_MIXER; return 1; } if(md_mode & DMODE_STEREO) { Mix32to16 = Mix32To16_Stereo; Mix32to8 = Mix32To8_Stereo; MixReverb = MixReverb_Stereo; } else { Mix32to16 = Mix32To16_Normal; Mix32to8 = Mix32To8_Normal; MixReverb = MixReverb_Normal; } md_mode |= DMODE_INTERP; vc_mode = md_mode; return 0; } BOOL VC2_PlayStart(void) { md_mode|=DMODE_INTERP; samplesthatfit = TICKLSIZE; if(vc_mode & DMODE_STEREO) samplesthatfit >>= 1; tickleft = 0; RVc1 = (5000L * md_mixfreq) / (REVERBERATION * 10); RVc2 = (5078L * md_mixfreq) / (REVERBERATION * 10); RVc3 = (5313L * md_mixfreq) / (REVERBERATION * 10); RVc4 = (5703L * md_mixfreq) / (REVERBERATION * 10); RVc5 = (6250L * md_mixfreq) / (REVERBERATION * 10); RVc6 = (6953L * md_mixfreq) / (REVERBERATION * 10); RVc7 = (7813L * md_mixfreq) / (REVERBERATION * 10); RVc8 = (8828L * md_mixfreq) / (REVERBERATION * 10); if(!(RVbufL1=(SLONG*)MikMod_calloc((RVc1+1),sizeof(SLONG)))) return 1; if(!(RVbufL2=(SLONG*)MikMod_calloc((RVc2+1),sizeof(SLONG)))) return 1; if(!(RVbufL3=(SLONG*)MikMod_calloc((RVc3+1),sizeof(SLONG)))) return 1; if(!(RVbufL4=(SLONG*)MikMod_calloc((RVc4+1),sizeof(SLONG)))) return 1; if(!(RVbufL5=(SLONG*)MikMod_calloc((RVc5+1),sizeof(SLONG)))) return 1; if(!(RVbufL6=(SLONG*)MikMod_calloc((RVc6+1),sizeof(SLONG)))) return 1; if(!(RVbufL7=(SLONG*)MikMod_calloc((RVc7+1),sizeof(SLONG)))) return 1; if(!(RVbufL8=(SLONG*)MikMod_calloc((RVc8+1),sizeof(SLONG)))) return 1; if(!(RVbufR1=(SLONG*)MikMod_calloc((RVc1+1),sizeof(SLONG)))) return 1; if(!(RVbufR2=(SLONG*)MikMod_calloc((RVc2+1),sizeof(SLONG)))) return 1; if(!(RVbufR3=(SLONG*)MikMod_calloc((RVc3+1),sizeof(SLONG)))) return 1; if(!(RVbufR4=(SLONG*)MikMod_calloc((RVc4+1),sizeof(SLONG)))) return 1; if(!(RVbufR5=(SLONG*)MikMod_calloc((RVc5+1),sizeof(SLONG)))) return 1; if(!(RVbufR6=(SLONG*)MikMod_calloc((RVc6+1),sizeof(SLONG)))) return 1; if(!(RVbufR7=(SLONG*)MikMod_calloc((RVc7+1),sizeof(SLONG)))) return 1; if(!(RVbufR8=(SLONG*)MikMod_calloc((RVc8+1),sizeof(SLONG)))) return 1; RVRindex = 0; return 0; } void VC2_PlayStop(void) { if(RVbufL1) MikMod_free(RVbufL1); if(RVbufL2) MikMod_free(RVbufL2); if(RVbufL3) MikMod_free(RVbufL3); if(RVbufL4) MikMod_free(RVbufL4); if(RVbufL5) MikMod_free(RVbufL5); if(RVbufL6) MikMod_free(RVbufL6); if(RVbufL7) MikMod_free(RVbufL7); if(RVbufL8) MikMod_free(RVbufL8); if(RVbufR1) MikMod_free(RVbufR1); if(RVbufR2) MikMod_free(RVbufR2); if(RVbufR3) MikMod_free(RVbufR3); if(RVbufR4) MikMod_free(RVbufR4); if(RVbufR5) MikMod_free(RVbufR5); if(RVbufR6) MikMod_free(RVbufR6); if(RVbufR7) MikMod_free(RVbufR7); if(RVbufR8) MikMod_free(RVbufR8); RVbufL1=RVbufL2=RVbufL3=RVbufL4=RVbufL5=RVbufL6=RVbufL7=RVbufL8=NULL; RVbufR1=RVbufR2=RVbufR3=RVbufR4=RVbufR5=RVbufR6=RVbufR7=RVbufR8=NULL; } BOOL VC2_SetNumVoices(void) { int t; md_mode|=DMODE_INTERP; if(!(vc_softchn=md_softchn)) return 0; if(vinf) MikMod_free(vinf); if(!(vinf=MikMod_calloc(sizeof(VINFO),vc_softchn))) return 1; for(t=0;t