diff options
Diffstat (limited to 'plugins/dfsound/reverb.c')
| -rw-r--r-- | plugins/dfsound/reverb.c | 462 |
1 files changed, 462 insertions, 0 deletions
diff --git a/plugins/dfsound/reverb.c b/plugins/dfsound/reverb.c new file mode 100644 index 0000000..92e31fc --- /dev/null +++ b/plugins/dfsound/reverb.c @@ -0,0 +1,462 @@ +/***************************************************************************
+ reverb.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_REVERB
+
+// will be included from spu.c
+#ifdef _IN_SPU
+
+////////////////////////////////////////////////////////////////////////
+// globals
+////////////////////////////////////////////////////////////////////////
+
+// REVERB info and timing vars...
+
+int * sRVBPlay = 0;
+int * sRVBEnd = 0;
+int * sRVBStart = 0;
+int iReverbOff = -1; // some delay factor for reverb
+int iReverbRepeat = 0;
+int iReverbNum = 1;
+
+////////////////////////////////////////////////////////////////////////
+// SET REVERB
+////////////////////////////////////////////////////////////////////////
+
+void SetREVERB(unsigned short val)
+{
+ switch(val)
+ {
+ case 0x0000: iReverbOff=-1; break; // off
+ case 0x007D: iReverbOff=32; iReverbNum=2; iReverbRepeat=128; break; // ok room
+
+ case 0x0033: iReverbOff=32; iReverbNum=2; iReverbRepeat=64; break; // studio small
+ case 0x00B1: iReverbOff=48; iReverbNum=2; iReverbRepeat=96; break; // ok studio medium
+ case 0x00E3: iReverbOff=64; iReverbNum=2; iReverbRepeat=128; break; // ok studio large ok
+
+ case 0x01A5: iReverbOff=128; iReverbNum=4; iReverbRepeat=32; break; // ok hall
+ case 0x033D: iReverbOff=256; iReverbNum=4; iReverbRepeat=64; break; // space echo
+ case 0x0001: iReverbOff=184; iReverbNum=3; iReverbRepeat=128; break; // echo/delay
+ case 0x0017: iReverbOff=128; iReverbNum=2; iReverbRepeat=128; break; // half echo
+ default: iReverbOff=32; iReverbNum=1; iReverbRepeat=0; break;
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+// START REVERB
+////////////////////////////////////////////////////////////////////////
+
+INLINE void StartREVERB(int ch)
+{
+ if(s_chan[ch].bReverb && (spuCtrl&0x80)) // reverb possible?
+ {
+ if(iUseReverb==2) s_chan[ch].bRVBActive=1;
+ else
+ if(iUseReverb==1 && iReverbOff>0) // -> fake reverb used?
+ {
+ s_chan[ch].bRVBActive=1; // -> activate it
+ s_chan[ch].iRVBOffset=iReverbOff*45;
+ s_chan[ch].iRVBRepeat=iReverbRepeat*45;
+ s_chan[ch].iRVBNum =iReverbNum;
+ }
+ }
+ else s_chan[ch].bRVBActive=0; // else -> no reverb
+}
+
+////////////////////////////////////////////////////////////////////////
+// HELPER FOR NEILL'S REVERB: re-inits our reverb mixing buf
+////////////////////////////////////////////////////////////////////////
+
+INLINE void InitREVERB(void)
+{
+ if(iUseReverb==2)
+ {memset(sRVBStart,0,NSSIZE*2*4);}
+}
+
+////////////////////////////////////////////////////////////////////////
+// STORE REVERB
+////////////////////////////////////////////////////////////////////////
+
+INLINE void StoreREVERB(int ch,int ns)
+{
+ if(iUseReverb==0) return;
+ else
+ if(iUseReverb==2) // -------------------------------- // Neil's reverb
+ {
+ const int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x4000;
+ const int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x4000;
+
+ ns<<=1;
+
+ *(sRVBStart+ns) +=iRxl; // -> we mix all active reverb channels into an extra buffer
+ *(sRVBStart+ns+1)+=iRxr;
+ }
+ else // --------------------------------------------- // Pete's easy fake reverb
+ {
+ int * pN;int iRn,iRr=0;
+
+ // we use the half channel volume (/0x8000) for the first reverb effects, quarter for next and so on
+
+ int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume)/0x8000;
+ int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume)/0x8000;
+
+ for(iRn=1;iRn<=s_chan[ch].iRVBNum;iRn++,iRr+=s_chan[ch].iRVBRepeat,iRxl/=2,iRxr/=2)
+ {
+ pN=sRVBPlay+((s_chan[ch].iRVBOffset+iRr+ns)<<1);
+ if(pN>=sRVBEnd) pN=sRVBStart+(pN-sRVBEnd);
+
+ (*pN)+=iRxl;
+ pN++;
+ (*pN)+=iRxr;
+ }
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int g_buffer(int iOff) // get_buffer content helper: takes care about wraps
+{
+ short * p=(short *)spuMem;
+ iOff=(iOff*4)+rvb.CurrAddr;
+ while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);
+ while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);
+ return (int)*(p+iOff);
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE void s_buffer(int iOff,int iVal) // set_buffer content helper: takes care about wraps and clipping
+{
+ short * p=(short *)spuMem;
+ iOff=(iOff*4)+rvb.CurrAddr;
+ while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);
+ while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);
+ if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L;
+ *(p+iOff)=(short)iVal;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE void s_buffer1(int iOff,int iVal) // set_buffer (+1 sample) content helper: takes care about wraps and clipping
+{
+ short * p=(short *)spuMem;
+ iOff=(iOff*4)+rvb.CurrAddr+1;
+ while(iOff>0x3FFFF) iOff=rvb.StartAddr+(iOff-0x40000);
+ while(iOff<rvb.StartAddr) iOff=0x3ffff-(rvb.StartAddr-iOff);
+ if(iVal<-32768L) iVal=-32768L;if(iVal>32767L) iVal=32767L;
+ *(p+iOff)=(short)iVal;
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int MixREVERBLeft(int ns)
+{
+ if(iUseReverb==0) return 0;
+ else
+ if(iUseReverb==2)
+ {
+ static int iCnt=0; // this func will be called with 44.1 khz
+
+ if(!rvb.StartAddr) // reverb is off
+ {
+ rvb.iLastRVBLeft=rvb.iLastRVBRight=rvb.iRVBLeft=rvb.iRVBRight=0;
+ return 0;
+ }
+
+ iCnt++;
+
+ if(iCnt&1) // we work on every second left value: downsample to 22 khz
+ {
+ if(spuCtrl&0x80) // -> reverb on? oki
+ {
+ int ACC0,ACC1,FB_A0,FB_A1,FB_B0,FB_B1;
+
+ const int INPUT_SAMPLE_L=*(sRVBStart+(ns<<1));
+ const int INPUT_SAMPLE_R=*(sRVBStart+(ns<<1)+1);
+
+ const int IIR_INPUT_A0 = (g_buffer(rvb.IIR_SRC_A0) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb.IN_COEF_L)/32768L;
+ const int IIR_INPUT_A1 = (g_buffer(rvb.IIR_SRC_A1) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb.IN_COEF_R)/32768L;
+ const int IIR_INPUT_B0 = (g_buffer(rvb.IIR_SRC_B0) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_L * rvb.IN_COEF_L)/32768L;
+ const int IIR_INPUT_B1 = (g_buffer(rvb.IIR_SRC_B1) * rvb.IIR_COEF)/32768L + (INPUT_SAMPLE_R * rvb.IN_COEF_R)/32768L;
+
+ const int IIR_A0 = (IIR_INPUT_A0 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_A0) * (32768L - rvb.IIR_ALPHA))/32768L;
+ const int IIR_A1 = (IIR_INPUT_A1 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_A1) * (32768L - rvb.IIR_ALPHA))/32768L;
+ const int IIR_B0 = (IIR_INPUT_B0 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_B0) * (32768L - rvb.IIR_ALPHA))/32768L;
+ const int IIR_B1 = (IIR_INPUT_B1 * rvb.IIR_ALPHA)/32768L + (g_buffer(rvb.IIR_DEST_B1) * (32768L - rvb.IIR_ALPHA))/32768L;
+
+ s_buffer1(rvb.IIR_DEST_A0, IIR_A0);
+ s_buffer1(rvb.IIR_DEST_A1, IIR_A1);
+ s_buffer1(rvb.IIR_DEST_B0, IIR_B0);
+ s_buffer1(rvb.IIR_DEST_B1, IIR_B1);
+
+ ACC0 = (g_buffer(rvb.ACC_SRC_A0) * rvb.ACC_COEF_A)/32768L +
+ (g_buffer(rvb.ACC_SRC_B0) * rvb.ACC_COEF_B)/32768L +
+ (g_buffer(rvb.ACC_SRC_C0) * rvb.ACC_COEF_C)/32768L +
+ (g_buffer(rvb.ACC_SRC_D0) * rvb.ACC_COEF_D)/32768L;
+ ACC1 = (g_buffer(rvb.ACC_SRC_A1) * rvb.ACC_COEF_A)/32768L +
+ (g_buffer(rvb.ACC_SRC_B1) * rvb.ACC_COEF_B)/32768L +
+ (g_buffer(rvb.ACC_SRC_C1) * rvb.ACC_COEF_C)/32768L +
+ (g_buffer(rvb.ACC_SRC_D1) * rvb.ACC_COEF_D)/32768L;
+
+ FB_A0 = g_buffer(rvb.MIX_DEST_A0 - rvb.FB_SRC_A);
+ FB_A1 = g_buffer(rvb.MIX_DEST_A1 - rvb.FB_SRC_A);
+ FB_B0 = g_buffer(rvb.MIX_DEST_B0 - rvb.FB_SRC_B);
+ FB_B1 = g_buffer(rvb.MIX_DEST_B1 - rvb.FB_SRC_B);
+
+ s_buffer(rvb.MIX_DEST_A0, ACC0 - (FB_A0 * rvb.FB_ALPHA)/32768L);
+ s_buffer(rvb.MIX_DEST_A1, ACC1 - (FB_A1 * rvb.FB_ALPHA)/32768L);
+
+ s_buffer(rvb.MIX_DEST_B0, (rvb.FB_ALPHA * ACC0)/32768L - (FB_A0 * (int)(rvb.FB_ALPHA^0xFFFF8000))/32768L - (FB_B0 * rvb.FB_X)/32768L);
+ s_buffer(rvb.MIX_DEST_B1, (rvb.FB_ALPHA * ACC1)/32768L - (FB_A1 * (int)(rvb.FB_ALPHA^0xFFFF8000))/32768L - (FB_B1 * rvb.FB_X)/32768L);
+
+ rvb.iLastRVBLeft = rvb.iRVBLeft;
+ rvb.iLastRVBRight = rvb.iRVBRight;
+
+ rvb.iRVBLeft = (g_buffer(rvb.MIX_DEST_A0)+g_buffer(rvb.MIX_DEST_B0))/3;
+ rvb.iRVBRight = (g_buffer(rvb.MIX_DEST_A1)+g_buffer(rvb.MIX_DEST_B1))/3;
+
+ rvb.iRVBLeft = (rvb.iRVBLeft * rvb.VolLeft) / 0x4000;
+ rvb.iRVBRight = (rvb.iRVBRight * rvb.VolRight) / 0x4000;
+
+ rvb.CurrAddr++;
+ if(rvb.CurrAddr>0x3ffff) rvb.CurrAddr=rvb.StartAddr;
+
+ return rvb.iLastRVBLeft+(rvb.iRVBLeft-rvb.iLastRVBLeft)/2;
+ }
+ else // -> reverb off
+ {
+ rvb.iLastRVBLeft=rvb.iLastRVBRight=rvb.iRVBLeft=rvb.iRVBRight=0;
+ }
+
+ rvb.CurrAddr++;
+ if(rvb.CurrAddr>0x3ffff) rvb.CurrAddr=rvb.StartAddr;
+ }
+
+ return rvb.iLastRVBLeft;
+ }
+ else // easy fake reverb:
+ {
+ const int iRV=*sRVBPlay; // -> simply take the reverb mix buf value
+ *sRVBPlay++=0; // -> init it after
+ if(sRVBPlay>=sRVBEnd) sRVBPlay=sRVBStart; // -> and take care about wrap arounds
+ return iRV; // -> return reverb mix buf val
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+INLINE int MixREVERBRight(void)
+{
+ if(iUseReverb==0) return 0;
+ else
+ if(iUseReverb==2) // Neill's reverb:
+ {
+ int i=rvb.iLastRVBRight+(rvb.iRVBRight-rvb.iLastRVBRight)/2;
+ rvb.iLastRVBRight=rvb.iRVBRight;
+ return i; // -> just return the last right reverb val (little bit scaled by the previous right val)
+ }
+ else // easy fake reverb:
+ {
+ const int iRV=*sRVBPlay; // -> simply take the reverb mix buf value
+ *sRVBPlay++=0; // -> init it after
+ if(sRVBPlay>=sRVBEnd) sRVBPlay=sRVBStart; // -> and take care about wrap arounds
+ return iRV; // -> return reverb mix buf val
+ }
+}
+
+////////////////////////////////////////////////////////////////////////
+
+#endif
+
+/*
+-----------------------------------------------------------------------------
+PSX reverb hardware notes
+by Neill Corlett
+-----------------------------------------------------------------------------
+
+Yadda yadda disclaimer yadda probably not perfect yadda well it's okay anyway
+yadda yadda.
+
+-----------------------------------------------------------------------------
+
+Basics
+------
+
+- The reverb buffer is 22khz 16-bit mono PCM.
+- It starts at the reverb address given by 1DA2, extends to
+ the end of sound RAM, and wraps back to the 1DA2 address.
+
+Setting the address at 1DA2 resets the current reverb work address.
+
+This work address ALWAYS increments every 1/22050 sec., regardless of
+whether reverb is enabled (bit 7 of 1DAA set).
+
+And the contents of the reverb buffer ALWAYS play, scaled by the
+"reverberation depth left/right" volumes (1D84/1D86).
+(which, by the way, appear to be scaled so 3FFF=approx. 1.0, 4000=-1.0)
+
+-----------------------------------------------------------------------------
+
+Register names
+--------------
+
+These are probably not their real names.
+These are probably not even correct names.
+We will use them anyway, because we can.
+
+1DC0: FB_SRC_A (offset)
+1DC2: FB_SRC_B (offset)
+1DC4: IIR_ALPHA (coef.)
+1DC6: ACC_COEF_A (coef.)
+1DC8: ACC_COEF_B (coef.)
+1DCA: ACC_COEF_C (coef.)
+1DCC: ACC_COEF_D (coef.)
+1DCE: IIR_COEF (coef.)
+1DD0: FB_ALPHA (coef.)
+1DD2: FB_X (coef.)
+1DD4: IIR_DEST_A0 (offset)
+1DD6: IIR_DEST_A1 (offset)
+1DD8: ACC_SRC_A0 (offset)
+1DDA: ACC_SRC_A1 (offset)
+1DDC: ACC_SRC_B0 (offset)
+1DDE: ACC_SRC_B1 (offset)
+1DE0: IIR_SRC_A0 (offset)
+1DE2: IIR_SRC_A1 (offset)
+1DE4: IIR_DEST_B0 (offset)
+1DE6: IIR_DEST_B1 (offset)
+1DE8: ACC_SRC_C0 (offset)
+1DEA: ACC_SRC_C1 (offset)
+1DEC: ACC_SRC_D0 (offset)
+1DEE: ACC_SRC_D1 (offset)
+1DF0: IIR_SRC_B1 (offset)
+1DF2: IIR_SRC_B0 (offset)
+1DF4: MIX_DEST_A0 (offset)
+1DF6: MIX_DEST_A1 (offset)
+1DF8: MIX_DEST_B0 (offset)
+1DFA: MIX_DEST_B1 (offset)
+1DFC: IN_COEF_L (coef.)
+1DFE: IN_COEF_R (coef.)
+
+The coefficients are signed fractional values.
+-32768 would be -1.0
+ 32768 would be 1.0 (if it were possible... the highest is of course 32767)
+
+The offsets are (byte/8) offsets into the reverb buffer.
+i.e. you multiply them by 8, you get byte offsets.
+You can also think of them as (samples/4) offsets.
+They appear to be signed. They can be negative.
+None of the documented presets make them negative, though.
+
+Yes, 1DF0 and 1DF2 appear to be backwards. Not a typo.
+
+-----------------------------------------------------------------------------
+
+What it does
+------------
+
+We take all reverb sources:
+- regular channels that have the reverb bit on
+- cd and external sources, if their reverb bits are on
+and mix them into one stereo 44100hz signal.
+
+Lowpass/downsample that to 22050hz. The PSX uses a proper bandlimiting
+algorithm here, but I haven't figured out the hysterically exact specifics.
+I use an 8-tap filter with these coefficients, which are nice but probably
+not the real ones:
+
+0.037828187894
+0.157538631280
+0.321159685278
+0.449322115345
+0.449322115345
+0.321159685278
+0.157538631280
+0.037828187894
+
+So we have two input samples (INPUT_SAMPLE_L, INPUT_SAMPLE_R) every 22050hz.
+
+* IN MY EMULATION, I divide these by 2 to make it clip less.
+ (and of course the L/R output coefficients are adjusted to compensate)
+ The real thing appears to not do this.
+
+At every 22050hz tick:
+- If the reverb bit is enabled (bit 7 of 1DAA), execute the reverb
+ steady-state algorithm described below
+- AFTERWARDS, retrieve the "wet out" L and R samples from the reverb buffer
+ (This part may not be exactly right and I guessed at the coefs. TODO: check later.)
+ L is: 0.333 * (buffer[MIX_DEST_A0] + buffer[MIX_DEST_B0])
+ R is: 0.333 * (buffer[MIX_DEST_A1] + buffer[MIX_DEST_B1])
+- Advance the current buffer position by 1 sample
+
+The wet out L and R are then upsampled to 44100hz and played at the
+"reverberation depth left/right" (1D84/1D86) volume, independent of the main
+volume.
+
+-----------------------------------------------------------------------------
+
+Reverb steady-state
+-------------------
+
+The reverb steady-state algorithm is fairly clever, and of course by
+"clever" I mean "batshit insane".
+
+buffer[x] is relative to the current buffer position, not the beginning of
+the buffer. Note that all buffer offsets must wrap around so they're
+contained within the reverb work area.
+
+Clipping is performed at the end... maybe also sooner, but definitely at
+the end.
+
+IIR_INPUT_A0 = buffer[IIR_SRC_A0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;
+IIR_INPUT_A1 = buffer[IIR_SRC_A1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;
+IIR_INPUT_B0 = buffer[IIR_SRC_B0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;
+IIR_INPUT_B1 = buffer[IIR_SRC_B1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;
+
+IIR_A0 = IIR_INPUT_A0 * IIR_ALPHA + buffer[IIR_DEST_A0] * (1.0 - IIR_ALPHA);
+IIR_A1 = IIR_INPUT_A1 * IIR_ALPHA + buffer[IIR_DEST_A1] * (1.0 - IIR_ALPHA);
+IIR_B0 = IIR_INPUT_B0 * IIR_ALPHA + buffer[IIR_DEST_B0] * (1.0 - IIR_ALPHA);
+IIR_B1 = IIR_INPUT_B1 * IIR_ALPHA + buffer[IIR_DEST_B1] * (1.0 - IIR_ALPHA);
+
+buffer[IIR_DEST_A0 + 1sample] = IIR_A0;
+buffer[IIR_DEST_A1 + 1sample] = IIR_A1;
+buffer[IIR_DEST_B0 + 1sample] = IIR_B0;
+buffer[IIR_DEST_B1 + 1sample] = IIR_B1;
+
+ACC0 = buffer[ACC_SRC_A0] * ACC_COEF_A +
+ buffer[ACC_SRC_B0] * ACC_COEF_B +
+ buffer[ACC_SRC_C0] * ACC_COEF_C +
+ buffer[ACC_SRC_D0] * ACC_COEF_D;
+ACC1 = buffer[ACC_SRC_A1] * ACC_COEF_A +
+ buffer[ACC_SRC_B1] * ACC_COEF_B +
+ buffer[ACC_SRC_C1] * ACC_COEF_C +
+ buffer[ACC_SRC_D1] * ACC_COEF_D;
+
+FB_A0 = buffer[MIX_DEST_A0 - FB_SRC_A];
+FB_A1 = buffer[MIX_DEST_A1 - FB_SRC_A];
+FB_B0 = buffer[MIX_DEST_B0 - FB_SRC_B];
+FB_B1 = buffer[MIX_DEST_B1 - FB_SRC_B];
+
+buffer[MIX_DEST_A0] = ACC0 - FB_A0 * FB_ALPHA;
+buffer[MIX_DEST_A1] = ACC1 - FB_A1 * FB_ALPHA;
+buffer[MIX_DEST_B0] = (FB_ALPHA * ACC0) - FB_A0 * (FB_ALPHA^0x8000) - FB_B0 * FB_X;
+buffer[MIX_DEST_B1] = (FB_ALPHA * ACC1) - FB_A1 * (FB_ALPHA^0x8000) - FB_B1 * FB_X;
+
+-----------------------------------------------------------------------------
+*/
+
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