--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/playercode/virtch2.c	Fri Jan 23 16:05:08 1998 +0000
@@ -0,0 +1,927 @@
+/*
+
+Name:  VIRTCH2.C
+
+Description:
+ All-c sample mixing routines, using a 32 bits mixing buffer, interpolation,
+ and sample smoothing [improves sound quality and removes clicks].
+
+ Future Additions:
+   Low-Pass filter to remove annoying staticy buzz.
+
+
+C Mixer Portability:
+ All Systems -- All compilers.
+
+Assembly Mixer Portability:
+
+ MSDOS:  BC(?)   Watcom(y)   DJGPP(y)
+ Win95:  ?
+ Os2:    ?
+ Linux:  y
+
+ (y) - yes
+ (n) - no (not possible or not useful)
+ (?) - may be possible, but not tested
+
+*/
+
+#include <stddef.h>
+#include <string.h>
+#include "mikmod.h"
+
+
+// REVERBERATION : Larger numbers result in shorter reverb duration.
+//      Longer reverb durations can cause unwanted static and make the
+//      reverb sound more like an echo.
+
+#define REVERBERATION 28000l
+
+// 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 soudn 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.
+
+#define SAMPLING_SHIFT  2
+#define SAMPLING_FACTOR (SLONG)(1<<SAMPLING_SHIFT)
+
+
+// FRACBITS : the number of bits per integer devoted to the fractional
+//      part of the number.  This value HAS to be changed if the compiler
+//      does not support 64 bit integers.  If 32 bit integers are used,
+//      FRACBITS _must_ be 9 or smaller.
+
+#define FRACBITS 28
+#define FRACMASK ((1l<<FRACBITS)-1)
+
+#define TICKLSIZE 4096
+#define TICKWSIZE (TICKLSIZE*2)
+#define TICKBSIZE (TICKWSIZE*2)
+
+#ifndef MIN
+#define MIN(a,b) (((a)<(b)) ? (a) : (b))
+#endif
+
+#ifndef MAX
+#define MAX(a,b) (((a)>(b))?(a):(b))
+#endif
+
+
+typedef struct
+{   UBYTE  kick;                  // =1 -> 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
+    UWORD  vol;                   // current volume
+    UWORD  pan;                   // current panning position
+    SDOUBLE  current;               // current index in the sample
+    SDOUBLE  increment;             // fixed-point increment value
+} VINFO;
+
+
+static SWORD   **Samples;
+
+static VINFO   *vinf = NULL;
+static VINFO   *vnf;
+static ULONG   samplesthatfit;
+static int     vc_memory, vc_softchn;
+static SDOUBLE idxsize,idxlpos,idxlend;
+static SLONG   TICKLEFT;
+static SLONG   *VC2_TICKBUF = NULL;
+static UWORD   vc_mode;
+static int     bitshift;           // Amplification shift (amount to decrease 32 bit mixing buffer by!)
+static SLONG   lvolsel, rvolsel;   // Volume factor .. range 0-255
+
+
+// Reverb control variables
+// ========================
+
+static int     RVc1, RVc2, RVc3, RVc4, RVc5, RVc6, RVc7, RVc8;
+static ULONG   RVRindex;
+
+// For Mono or Left Channel
+static SLONG  *RVbuf1 = NULL, *RVbuf2 = NULL, *RVbuf3 = NULL,
+              *RVbuf4 = NULL, *RVbuf5 = NULL, *RVbuf6 = NULL,
+              *RVbuf7 = NULL, *RVbuf8 = NULL;
+
+// For Stereo only (Right Channel)
+static SLONG  *RVbuf9 = NULL, *RVbuf10 = NULL, *RVbuf11 = NULL,
+              *RVbuf12 = NULL, *RVbuf13 = NULL, *RVbuf14 = NULL,
+              *RVbuf15 = NULL, *RVbuf16 = NULL;
+
+
+// ==============================================================
+//  16 bit sample mixers!
+
+static SDOUBLE MixStereoNormal(SWORD *srce, SLONG *dest, SDOUBLE index, SDOUBLE increment, ULONG todo)
+{
+    SWORD  sample;
+
+    for(; todo; todo--)
+    {   sample = (index & FRACBITS) ?
+                    (((srce[index >> FRACBITS] * (FRACBITS - (index & FRACBITS))) +
+                    (srce[(index >> FRACBITS)+1] * (index & FRACBITS))) / FRACBITS)
+                 :  srce[index >> FRACBITS];
+        index += increment;
+
+        *dest++ += lvolsel * sample;
+        *dest++ += rvolsel * sample;
+    }
+
+    return index;
+}
+
+
+static SDOUBLE MixStereoSurround(SWORD *srce, SLONG *dest, SDOUBLE index, SDOUBLE increment, ULONG todo)
+{
+    SWORD  sample;
+
+    for(dest--; todo; todo--)
+    {   sample = (index & FRACBITS) ?
+                    (((srce[index >> FRACBITS] * (FRACBITS - (index & FRACBITS))) +
+                    (srce[(index >> FRACBITS)+1] * (index & FRACBITS))) / FRACBITS)
+                 :  srce[index >> FRACBITS];
+        index += increment;
+
+        *dest++ += lvolsel * sample;
+        *dest++ -= lvolsel * sample;
+    }
+    
+    return index;
+}
+
+
+static SDOUBLE MixMonoNormal(SWORD *srce, SLONG *dest, SDOUBLE index, SDOUBLE increment, SLONG todo)
+{
+    SWORD  sample;
+    
+    for(; todo; todo--)
+    {   sample = (index & FRACBITS) ?
+                    (((srce[index >> FRACBITS] * (FRACBITS - (index & FRACBITS))) +
+                    (srce[(index >> FRACBITS)+1] * (index & FRACBITS))) / FRACBITS)
+                 :  srce[index >> FRACBITS];
+        index += increment;
+
+        *dest++ += lvolsel * sample;
+    }
+
+    return index;
+}
+
+
+static void (*Mix32to16)(SWORD *dste, SLONG *srce, SLONG count);
+static void (*Mix32to8)(SBYTE *dste, SLONG *srce, SLONG count);
+static void (*MixReverb)(SLONG *srce, SLONG count);
+
+
+static void MixReverb_Normal(SLONG *srce, SLONG count)
+{
+    SLONG        speedup;
+    int          ReverbPct;
+    unsigned int loc1, loc2, loc3, loc4,
+                 loc5, loc6, loc7, loc8;
+
+    ReverbPct = 63 + (md_reverb*4);
+
+    loc1 = RVRindex % RVc1;
+    loc2 = RVRindex % RVc2;
+    loc3 = RVRindex % RVc3;
+    loc4 = RVRindex % RVc4;
+    loc5 = RVRindex % RVc5;
+    loc6 = RVRindex % RVc6;
+    loc7 = RVRindex % RVc7;
+    loc8 = RVRindex % RVc8;
+
+    for(; count; count--)
+    {
+        // Compute the LEFT CHANNEL echo buffers!
+
+        speedup = *srce >> 3;
+
+        RVbuf1[loc1] = speedup + ((ReverbPct * RVbuf1[loc1]) / 128);
+        RVbuf2[loc2] = speedup + ((ReverbPct * RVbuf2[loc2]) / 128);
+        RVbuf3[loc3] = speedup + ((ReverbPct * RVbuf3[loc3]) / 128);
+        RVbuf4[loc4] = speedup + ((ReverbPct * RVbuf4[loc4]) / 128);
+        RVbuf5[loc5] = speedup + ((ReverbPct * RVbuf5[loc5]) / 128);
+        RVbuf6[loc6] = speedup + ((ReverbPct * RVbuf6[loc6]) / 128);
+        RVbuf7[loc7] = speedup + ((ReverbPct * RVbuf7[loc7]) / 128);
+        RVbuf8[loc8] = speedup + ((ReverbPct * RVbuf8[loc8]) / 128);
+
+        // Prepare to compute actual finalized data!
+
+        RVRindex++;
+        loc1 = RVRindex % RVc1;
+        loc2 = RVRindex % RVc2;
+        loc3 = RVRindex % RVc3;
+        loc4 = RVRindex % RVc4;
+        loc5 = RVRindex % RVc5;
+        loc6 = RVRindex % RVc6;
+        loc7 = RVRindex % RVc7;
+        loc8 = RVRindex % RVc8;
+
+        // Left Channel!
+        
+        *srce++ += (RVbuf1[loc1] - RVbuf2[loc2] + RVbuf3[loc3] - RVbuf4[loc4] +
+                    RVbuf5[loc5] - RVbuf6[loc6] + RVbuf7[loc7] - RVbuf8[loc8]);
+    }
+}
+
+
+static void MixReverb_Stereo(SLONG *srce, SLONG count)
+{
+    SLONG        speedup;
+    int          ReverbPct;
+    unsigned int loc1, loc2, loc3, loc4,
+                 loc5, loc6, loc7, loc8;
+
+    ReverbPct = 63 + (md_reverb*4);
+
+    loc1 = RVRindex % RVc1;
+    loc2 = RVRindex % RVc2;
+    loc3 = RVRindex % RVc3;
+    loc4 = RVRindex % RVc4;
+    loc5 = RVRindex % RVc5;
+    loc6 = RVRindex % RVc6;
+    loc7 = RVRindex % RVc7;
+    loc8 = RVRindex % RVc8;
+
+    for(; count; count--)
+    {
+        // Compute the LEFT CHANNEL echo buffers!
+
+        speedup = *srce >> 3;
+
+        RVbuf1[loc1] = speedup + ((ReverbPct * RVbuf1[loc1]) / 128);
+        RVbuf2[loc2] = speedup + ((ReverbPct * RVbuf2[loc2]) / 128);
+        RVbuf3[loc3] = speedup + ((ReverbPct * RVbuf3[loc3]) / 128);
+        RVbuf4[loc4] = speedup + ((ReverbPct * RVbuf4[loc4]) / 128);
+        RVbuf5[loc5] = speedup + ((ReverbPct * RVbuf5[loc5]) / 128);
+        RVbuf6[loc6] = speedup + ((ReverbPct * RVbuf6[loc6]) / 128);
+        RVbuf7[loc7] = speedup + ((ReverbPct * RVbuf7[loc7]) / 128);
+        RVbuf8[loc8] = speedup + ((ReverbPct * RVbuf8[loc8]) / 128);
+
+        // Compute the RIGHT CHANNEL echo buffers!
+        
+        speedup = srce[1] >> 3;
+
+        RVbuf9[loc1] = speedup + ((ReverbPct * RVbuf9[loc1]) / 128);
+        RVbuf10[loc2] = speedup + ((ReverbPct * RVbuf11[loc2]) / 128);
+        RVbuf11[loc3] = speedup + ((ReverbPct * RVbuf12[loc3]) / 128);
+        RVbuf12[loc4] = speedup + ((ReverbPct * RVbuf12[loc4]) / 128);
+        RVbuf13[loc5] = speedup + ((ReverbPct * RVbuf13[loc5]) / 128);
+        RVbuf14[loc6] = speedup + ((ReverbPct * RVbuf14[loc6]) / 128);
+        RVbuf15[loc7] = speedup + ((ReverbPct * RVbuf15[loc7]) / 128);
+        RVbuf16[loc8] = speedup + ((ReverbPct * RVbuf16[loc8]) / 128);
+
+        // Prepare to compute actual finalized data!
+
+        RVRindex++;
+        loc1 = RVRindex % RVc1;
+        loc2 = RVRindex % RVc2;
+        loc3 = RVRindex % RVc3;
+        loc4 = RVRindex % RVc4;
+        loc5 = RVRindex % RVc5;
+        loc6 = RVRindex % RVc6;
+        loc7 = RVRindex % RVc7;
+        loc8 = RVRindex % RVc8;
+
+        // Left Channel!
+        
+        *srce++ += (RVbuf1[loc1] - RVbuf2[loc2] + RVbuf3[loc3] - RVbuf4[loc4] +
+                    RVbuf5[loc5] - RVbuf6[loc6] + RVbuf7[loc7] - RVbuf8[loc8]);
+
+        // Right Channel!
+        
+        *srce++ += (RVbuf9[loc1] - RVbuf10[loc2] + RVbuf11[loc3] - RVbuf12[loc4] +
+                    RVbuf13[loc5] - RVbuf14[loc6] + RVbuf15[loc7] - RVbuf16[loc8]);
+    }
+}
+
+
+static void Mix32To16_Normal(SWORD *dste, SLONG *srce, SLONG count)
+{
+    SLONG    x1, x2, tmpx;
+    int      i;
+    
+    for(count/=SAMPLING_FACTOR; count; count--)
+    {   tmpx = 0;
+
+        for(i=SAMPLING_FACTOR/2; i; i--)
+        {  x1 = *srce++ / bitshift;
+           x2 = *srce++ / bitshift;
+
+           x1 = (x1 > 32767) ? 32767 : (x1 < -32768) ? -32768 : x1;
+           x2 = (x2 > 32767) ? 32767 : (x2 < -32768) ? -32768 : x2;
+
+           tmpx += x1 + x2;
+        }
+
+        *dste++ = tmpx / SAMPLING_FACTOR;
+    }
+}
+
+
+static void Mix32To16_Stereo(SWORD *dste, SLONG *srce, SLONG count)
+{
+    SLONG   x1, x2, x3, x4, tmpx, tmpy;
+    int     i;
+
+    for(count/=SAMPLING_FACTOR; count; count--)
+    {   tmpx = tmpy = 0;
+
+        for(i=SAMPLING_FACTOR/2; i; i--)
+        {   x1 = *srce++ / bitshift;
+            x2 = *srce++ / bitshift;
+            x3 = *srce++ / bitshift;
+            x4 = *srce++ / bitshift;
+
+            x1 = (x1 > 32767) ? 32767 : (x1 < -32768) ? -32768 : x1;
+            x2 = (x2 > 32767) ? 32767 : (x2 < -32768) ? -32768 : x2;
+            x3 = (x3 > 32767) ? 32767 : (x3 < -32768) ? -32768 : x3;
+            x4 = (x4 > 32767) ? 32767 : (x4 < -32768) ? -32768 : x4;
+
+            tmpx += x1 + x3;
+            tmpy += x2 + x4;
+        }
+
+        *dste++ = tmpx / SAMPLING_FACTOR;
+        *dste++ = tmpy / SAMPLING_FACTOR;
+    }
+}
+
+
+static void Mix32To8_Normal(SBYTE *dste, SLONG *srce, SLONG count)
+{
+    int    x1, x2;
+    int    i, tmpx;
+    
+    for(count/=SAMPLING_FACTOR; count; count--)
+    {   tmpx = 0;
+
+        for(i=SAMPLING_FACTOR/2; i; i--)
+        {   x1 = *srce++ / bitshift;
+            x2 = *srce++ / bitshift;
+
+            x1 = (x1 > 127) ? 127 : (x1 < -128) ? -128 : x1;
+            x2 = (x2 > 127) ? 127 : (x2 < -128) ? -128 : x2;
+
+            tmpx += x1 + x2;
+        }
+
+        *dste++ = (tmpx / SAMPLING_FACTOR) + 128;
+    }
+}
+
+
+static void Mix32To8_Stereo(SBYTE *dste, SLONG *srce, SLONG count)
+{
+    int    x1, x2, x3, x4;
+    int    i, tmpx, tmpy;
+
+    for(count/=SAMPLING_FACTOR; count; count--)
+    {   tmpx = tmpy = 0;
+
+        for(i=SAMPLING_FACTOR/2; i; i--)
+        {   x1 = *srce++ / bitshift;
+            x2 = *srce++ / bitshift;
+            x3 = *srce++ / bitshift;
+            x4 = *srce++ / bitshift;
+    
+            x1 = (x1 > 127) ? 127 : (x1 < -128) ? -128 : x1;
+            x2 = (x2 > 127) ? 127 : (x2 < -128) ? -128 : x2;
+            x3 = (x3 > 127) ? 127 : (x3 < -128) ? -128 : x3;
+            x4 = (x4 > 127) ? 127 : (x4 < -128) ? -128 : x4;
+    
+            tmpx += x1 + x3;
+            tmpy += x2 + x4;
+        }
+
+        *dste++ = (tmpx / SAMPLING_FACTOR) + 128;        
+        *dste++ = (tmpy / SAMPLING_FACTOR) + 128;        
+    }
+}
+
+
+static ULONG samples2bytes(ULONG samples)
+{
+    if(vc_mode & DMODE_16BITS) samples <<= 1;
+    if(vc_mode & DMODE_STEREO) samples <<= 1;
+
+    return samples;
+}
+
+
+static ULONG bytes2samples(ULONG bytes)
+{
+    if(vc_mode & DMODE_16BITS) bytes >>= 1;
+    if(vc_mode & DMODE_STEREO) bytes >>= 1;
+
+    return bytes;
+}
+
+
+static void AddChannel(SLONG *ptr, SLONG todo)
+{
+    SDOUBLE  end;
+    SLONG  done;
+    SWORD  *s;
+
+    while(todo > 0)
+    {   // update the 'current' index so the sample loops, or
+        // stops playing if it reached the end of the sample
+
+        if(vnf->flags & SF_REVERSE)
+        {
+            // The sample is playing in reverse
+
+            if((vnf->flags & SF_LOOP) && (vnf->current < idxlpos))
+            {
+                // the sample is looping, and it has
+                // reached the loopstart index
+
+                if(vnf->flags & 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 = 0;
+                    vnf->active  = 0;
+                    break;
+                }
+            }
+        } else
+        {
+            // The sample is playing forward
+
+            if((vnf->flags & SF_LOOP) && (vnf->current > idxlend))
+            {
+                 // the sample is looping, so check if
+                 // it reached 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 = 0;
+                    vnf->active  = 0;
+                    break;
+                }
+            }
+        }
+
+        if(!(s=Samples[vnf->handle]))
+        {   vnf->current = 0;
+            vnf->active  = 0;
+            break;
+        }
+
+        end = (vnf->flags & SF_REVERSE) ? 
+                (vnf->flags & SF_LOOP) ? idxlpos : 0 :
+                (vnf->flags & SF_LOOP) ? idxlend : idxsize;
+
+        done = MIN((end - vnf->current) / vnf->increment + 1, todo);
+
+        if(!done)
+        {   vnf->active = 0;
+            break;
+        }
+
+        if(vnf->vol)
+        {   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);
+        }
+        todo -= done;
+        ptr  += (vc_mode & DMODE_STEREO) ? (done<<1) : done;
+    }
+}
+
+
+void VC2_WriteSamples(SBYTE *buf, ULONG todo)
+{
+    int    left, portion = 0;
+    SBYTE  *buffer;
+    int    t;
+    int    pan, vol;
+
+    todo *= SAMPLING_FACTOR;
+
+    while(todo)
+    {   if(TICKLEFT==0)
+        {   if(vc_mode & DMODE_SOFT_MUSIC) md_player();
+            TICKLEFT = (md_mixfreq*125l*SAMPLING_FACTOR) / (md_bpm*50l);
+            TICKLEFT &= ~(SAMPLING_FACTOR-1);
+        }
+
+        left = MIN(TICKLEFT, todo);
+        
+        buffer    = buf;
+        TICKLEFT -= left;
+        todo     -= left;
+
+        buf += samples2bytes(left) / SAMPLING_FACTOR;
+
+        while(left)
+        {   portion = MIN(left, samplesthatfit);
+            memset(VC2_TICKBUF, 0, portion << ((vc_mode & DMODE_STEREO) ? 3 : 2));
+
+            for(t=0; t<vc_softchn; t++)
+            {   vnf = &vinf[t];
+
+                if(vnf->kick)
+                {   vnf->current = (SDOUBLE)(vnf->start) << FRACBITS;
+                    vnf->kick    = 0;
+                    vnf->active  = 1;
+                }
+                
+                if(vnf->frq == 0) vnf->active = 0;
+                
+                if(vnf->active)
+                {   vnf->increment = ((SDOUBLE)(vnf->frq) << (FRACBITS-SAMPLING_SHIFT)) / (SDOUBLE)md_mixfreq;
+                    if(vnf->flags & SF_REVERSE) vnf->increment=-vnf->increment;
+
+                    vol = vnf->vol;  pan = vnf->pan;
+
+                    if((vc_mode & DMODE_STEREO) && (pan!=PAN_SURROUND))
+                    {   lvolsel = (vol * (255-pan)) >> 8;
+                        rvolsel = (vol * pan) >> 8;
+                    } else
+                        lvolsel = (vol*256l) / 480;
+
+                    idxsize = (vnf->size)   ? ((SDOUBLE)(vnf->size) << FRACBITS)-1 : 0;
+                    idxlend = (vnf->repend) ? ((SDOUBLE)(vnf->repend) << FRACBITS)-1 : 0;
+                    idxlpos = (SDOUBLE)(vnf->reppos) << FRACBITS;
+                    AddChannel(VC2_TICKBUF, portion);
+                }
+            }
+
+            if(md_reverb) MixReverb(VC2_TICKBUF, portion);
+
+            if(vc_mode & DMODE_16BITS)
+                Mix32to16((SWORD *) buffer, VC2_TICKBUF, portion);
+            else
+                Mix32to8((SBYTE *) buffer, VC2_TICKBUF, portion);
+
+            buffer += samples2bytes(portion) / SAMPLING_FACTOR;
+            left   -= portion;
+        }
+    }
+}
+
+
+void VC2_SilenceBytes(SBYTE *buf, ULONG todo)
+
+//  Fill the buffer with 'todo' bytes of silence (it depends on the mixing
+//  mode how the buffer is filled)
+
+{
+    // clear the buffer to zero (16 bits
+    // signed ) or 0x80 (8 bits unsigned)
+
+    if(vc_mode & DMODE_16BITS)
+        memset(buf,0,todo);
+    else
+        memset(buf,0x80,todo);
+}
+
+
+ULONG VC2_WriteBytes(SBYTE *buf, ULONG todo)
+
+//  Writes 'todo' mixed SBYTES (!!) to 'buf'. It returns the number of
+//  SBYTES actually written to 'buf' (which is rounded to number of samples
+//  that fit into 'todo' bytes).
+
+{
+    if(vc_softchn == 0)
+    {   VC2_SilenceBytes(buf,todo);
+        return todo;
+    }
+
+    todo = bytes2samples(todo);
+    VC2_WriteSamples(buf,todo);
+
+    return samples2bytes(todo);
+}
+
+
+BOOL VC2_PlayStart(void)
+{
+    if(vc_softchn > 0)
+    {   bitshift = vc_softchn + 257;
+        if(!(vc_mode & DMODE_16BITS))
+            bitshift *= 256;
+        if(md_reverb) bitshift++;
+    }
+
+    samplesthatfit = TICKLSIZE;
+    if(vc_mode & DMODE_STEREO) samplesthatfit >>= 1;
+    TICKLEFT = 0;
+
+    /*  Original Reverb Code!
+        The stuff I use avoids floating point (below).
+
+    RVc1 = (SLONG)(500      * md_mixfreq) / 11000;
+    RVc2 = (SLONG)(507.8125 * md_mixfreq) / 11000;
+    RVc3 = (SLONG)(531.25   * md_mixfreq) / 11000;
+    RVc4 = (SLONG)(570.3125 * md_mixfreq) / 11000;
+    RVc5 = (SLONG)(625      * md_mixfreq) / 11000;
+    RVc6 = (SLONG)(695.3125 * md_mixfreq) / 11000;
+    RVc7 = (SLONG)(781.25   * md_mixfreq) / 11000;
+    RVc8 = (SLONG)(882.8125 * md_mixfreq) / 11000;
+    ReverbPct = 99 - (md_reverb*2);
+    */
+
+    RVc1 = (5000L * md_mixfreq) / REVERBERATION;
+    RVc2 = (5078L * md_mixfreq) / REVERBERATION;
+    RVc3 = (5313L * md_mixfreq) / REVERBERATION;
+    RVc4 = (5703L * md_mixfreq) / REVERBERATION;
+    RVc5 = (6250L * md_mixfreq) / REVERBERATION;
+    RVc6 = (6953L * md_mixfreq) / REVERBERATION;
+    RVc7 = (7813L * md_mixfreq) / REVERBERATION;
+    RVc8 = (8828L * md_mixfreq) / REVERBERATION;
+   
+    if((RVbuf1 = (SLONG *)_mm_calloc((RVc1+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf2 = (SLONG *)_mm_calloc((RVc2+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf3 = (SLONG *)_mm_calloc((RVc3+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf4 = (SLONG *)_mm_calloc((RVc4+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf5 = (SLONG *)_mm_calloc((RVc5+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf6 = (SLONG *)_mm_calloc((RVc6+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf7 = (SLONG *)_mm_calloc((RVc7+1),sizeof(SLONG))) == NULL) return 1;
+    if((RVbuf8 = (SLONG *)_mm_calloc((RVc8+1),sizeof(SLONG))) == NULL) return 1;
+    
+    if(vc_mode & DMODE_STEREO)
+    {   if((RVbuf9 = (SLONG *)_mm_calloc((RVc1+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf10 = (SLONG *)_mm_calloc((RVc2+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf11 = (SLONG *)_mm_calloc((RVc3+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf12 = (SLONG *)_mm_calloc((RVc4+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf13 = (SLONG *)_mm_calloc((RVc5+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf14 = (SLONG *)_mm_calloc((RVc6+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf15 = (SLONG *)_mm_calloc((RVc7+1),sizeof(SLONG))) == NULL) return 1;
+        if((RVbuf16 = (SLONG *)_mm_calloc((RVc8+1),sizeof(SLONG))) == NULL) return 1;
+    }
+    
+    RVRindex = 0;
+    return 0;
+}
+
+
+void VC2_PlayStop(void)
+{
+    if(RVbuf1  != NULL) free(RVbuf1);
+    if(RVbuf2  != NULL) free(RVbuf2);
+    if(RVbuf3  != NULL) free(RVbuf3);
+    if(RVbuf4  != NULL) free(RVbuf4);
+    if(RVbuf5  != NULL) free(RVbuf5);
+    if(RVbuf6  != NULL) free(RVbuf6);
+    if(RVbuf7  != NULL) free(RVbuf7);
+    if(RVbuf8  != NULL) free(RVbuf8);
+    if(RVbuf9  != NULL) free(RVbuf9);
+    if(RVbuf10 != NULL) free(RVbuf10);
+    if(RVbuf11 != NULL) free(RVbuf11);
+    if(RVbuf12 != NULL) free(RVbuf12);
+    if(RVbuf13 != NULL) free(RVbuf13);
+    if(RVbuf14 != NULL) free(RVbuf14);
+    if(RVbuf15 != NULL) free(RVbuf15);
+    if(RVbuf16 != NULL) free(RVbuf16);
+
+    RVbuf1  = NULL;  RVbuf2  = NULL;  RVbuf3  = NULL;  RVbuf4  = NULL;
+    RVbuf5  = NULL;  RVbuf6  = NULL;  RVbuf7  = NULL;  RVbuf8  = NULL;
+    RVbuf9  = NULL;  RVbuf10 = NULL;  RVbuf11 = NULL;  RVbuf12 = NULL;
+    RVbuf13 = NULL;  RVbuf14 = NULL;  RVbuf15 = NULL;  RVbuf16 = NULL;
+}
+
+
+BOOL VC2_Init(void)
+{
+    _mm_errno = MMERR_INITIALIZING_MIXER;
+    if((Samples = (SWORD **)calloc(MAXSAMPLEHANDLES, sizeof(SWORD *))) == NULL) return 1;
+    if(VC2_TICKBUF==NULL) if((VC2_TICKBUF=(SLONG *)malloc((TICKLSIZE+32) * sizeof(SLONG))) == NULL) 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;
+    }
+
+    vc_mode = md_mode;
+
+    _mm_errno = 0;
+    return 0;
+}
+
+
+void VC2_Exit(void)
+
+// Yay, he joys and fruits of C and C++ -
+//                        Deallocation of arrays!
+
+{
+    //if(VC2_TICKBUF!=NULL) free(VC2_TICKBUF);
+    if(vinf!=NULL) free(vinf);
+    if(Samples!=NULL) free(Samples);
+
+//    VC2_TICKBUF     = NULL;
+    vinf            = NULL;
+    Samples         = NULL;
+}
+
+
+BOOL VC2_SetNumVoices(void)
+{
+    int t;
+
+    if((vc_softchn = md_softchn) == 0) return 0;
+
+    if(vinf!=NULL) free(vinf);
+    if((vinf = _mm_calloc(sizeof(VINFO),vc_softchn)) == NULL) return 1;
+    
+    for(t=0; t<vc_softchn; t++)
+    {   vinf[t].frq = 10000;
+        vinf[t].pan = (t & 1) ? 0 : 255;
+    }
+
+    return 0;
+}
+
+
+void VC2_VoiceSetVolume(UBYTE voice, UWORD vol)
+{
+    vinf[voice].vol = vol;
+}
+
+
+void VC2_VoiceSetFrequency(UBYTE voice, ULONG frq)
+{
+    vinf[voice].frq = frq;
+}
+
+
+void VC2_VoiceSetPanning(UBYTE voice, ULONG pan)
+{
+    vinf[voice].pan = pan;
+}
+
+
+void VC2_VoicePlay(UBYTE voice, SWORD handle, ULONG start, ULONG size, ULONG reppos, ULONG repend, UWORD flags)
+{
+    vinf[voice].flags    = flags;
+    vinf[voice].handle   = handle;
+    vinf[voice].start    = start;
+    vinf[voice].size     = size;
+    vinf[voice].reppos   = reppos;
+    vinf[voice].repend   = repend;
+    vinf[voice].kick     = 1;
+}
+
+
+void VC2_VoiceStop(UBYTE voice)
+{
+    vinf[voice].active = 0;
+}  
+
+
+BOOL VC2_VoiceStopped(UBYTE voice)
+{
+    return(vinf[voice].active==0);
+}
+
+
+void VC2_VoiceReleaseSustain(UBYTE voice)
+{
+
+}
+
+
+SLONG VC2_VoiceGetPosition(UBYTE voice)
+{
+    return(vinf[voice].current>>FRACBITS);
+}
+
+
+/**************************************************
+***************************************************
+***************************************************
+**************************************************/
+
+
+void VC2_SampleUnload(SWORD handle)
+{
+    void *sampleadr = Samples[handle];
+
+    free(sampleadr);
+    Samples[handle] = NULL;
+}
+
+
+SWORD VC2_SampleLoad(SAMPLOAD *sload, int type, FILE *fp)
+{
+    SAMPLE *s = sload->sample;
+    int    handle;
+    ULONG  t, length,loopstart,loopend;
+
+    if(type==MD_HARDWARE) return -1;
+
+    // Find empty slot to put sample address in
+    for(handle=0; handle<MAXSAMPLEHANDLES; handle++)
+        if(Samples[handle]==NULL) break;
+
+    if(handle==MAXSAMPLEHANDLES)
+    {   _mm_errno = MMERR_OUT_OF_HANDLES;
+        return -1;
+    }
+
+    length    = s->length;
+    loopstart = s->loopstart;
+    loopend   = s->loopend;
+
+    SL_SampleSigned(sload);
+
+    SL_Sample8to16(sload);
+    if((Samples[handle]=(SWORD *)malloc((length+16)<<1))==NULL)
+    {   _mm_errno = MMERR_SAMPLE_TOO_BIG;
+        return -1;
+    }
+    // read sample into buffer.
+    SL_Load(Samples[handle],sload,length);
+
+    // Unclick samples:
+
+    if(s->flags & SF_LOOP)
+    {   if(s->flags & SF_BIDI)
+            for(t=0; t<16; t++) Samples[handle][loopend+t] = Samples[handle][(loopend-t)-1];
+        else
+            for(t=0; t<16; t++) Samples[handle][loopend+t] = Samples[handle][t+loopstart];
+    } else
+        for(t=0; t<16; t++) Samples[handle][t+length] = 0;
+
+    return handle;
+}
+
+
+ULONG VC2_SampleSpace(int type)
+{
+    return vc_memory;
+}
+
+
+ULONG VC2_SampleLength(int type, SAMPLE *s)
+{
+    return (s->length * ((s->flags & SF_16BITS) ? 2 : 1)) + 16;
+}
+
+
+/**************************************************
+***************************************************
+***************************************************
+**************************************************/
+