mirror of https://github.com/PCSX2/pcsx2.git
421 lines
16 KiB
C
421 lines
16 KiB
C
/***************************************************************************
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reverb.c - description
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-------------------
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begin : Wed May 15 2002
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copyright : (C) 2002 by Pete Bernert
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email : BlackDove@addcom.de
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***************************************************************************/
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/***************************************************************************
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* *
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* This program is free software; you can redistribute it and/or modify *
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* it under the terms of the GNU General Public License as published by *
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* the Free Software Foundation; either version 2 of the License, or *
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* (at your option) any later version. See also the license.txt file for *
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* additional informations. *
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* *
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***************************************************************************/
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//*************************************************************************//
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// History of changes:
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//
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// 2004/04/04 - Pete
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// - changed to SPU2 functionality
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//
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// 2003/01/19 - Pete
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// - added Neill's reverb (see at the end of file)
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//
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// 2002/12/26 - Pete
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// - adjusted reverb handling
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//
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// 2002/08/14 - Pete
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// - added extra reverb
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//
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// 2002/05/15 - Pete
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// - generic cleanup for the Peops release
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//
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//*************************************************************************//
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#include "stdafx.h"
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#define _IN_REVERB
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// will be included from spu.c
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#ifdef _IN_SPU
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////////////////////////////////////////////////////////////////////////
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// globals
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////////////////////////////////////////////////////////////////////////
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// REVERB info and timing vars...
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int * sRVBPlay[2];
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int * sRVBEnd[2];
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int * sRVBStart[2];
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////////////////////////////////////////////////////////////////////////
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// START REVERB
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////////////////////////////////////////////////////////////////////////
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INLINE void StartREVERB(int ch)
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{
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int core=ch/24;
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if((s_chan[ch].bReverbL || s_chan[ch].bReverbR) && (spuCtrl2[core]&0x80)) // reverb possible?
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{
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if(iUseReverb==1) s_chan[ch].bRVBActive=1;
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}
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else s_chan[ch].bRVBActive=0; // else -> no reverb
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}
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////////////////////////////////////////////////////////////////////////
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// HELPER FOR NEILL'S REVERB: re-inits our reverb mixing buf
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////////////////////////////////////////////////////////////////////////
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INLINE void InitREVERB(void)
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{
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if(iUseReverb==1)
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{
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memset(sRVBStart[0],0,NSSIZE*2*4);
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memset(sRVBStart[1],0,NSSIZE*2*4);
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}
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}
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////////////////////////////////////////////////////////////////////////
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// STORE REVERB
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////////////////////////////////////////////////////////////////////////
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INLINE void StoreREVERB(int ch,int ns)
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{
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int core=ch/24;
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if(iUseReverb==0) return;
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else
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if(iUseReverb==1) // -------------------------------- // Neil's reverb
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{
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const int iRxl=(s_chan[ch].sval*s_chan[ch].iLeftVolume*s_chan[ch].bReverbL)/0x4000;
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const int iRxr=(s_chan[ch].sval*s_chan[ch].iRightVolume*s_chan[ch].bReverbR)/0x4000;
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ns<<=1;
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*(sRVBStart[core]+ns) +=iRxl; // -> we mix all active reverb channels into an extra buffer
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*(sRVBStart[core]+ns+1)+=iRxr;
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}
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}
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////////////////////////////////////////////////////////////////////////
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INLINE int g_buffer(int iOff,int core) // get_buffer content helper: takes care about wraps
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{
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short * p=(short *)spuMem;
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iOff=(iOff)+rvb[core].CurrAddr;
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while(iOff>rvb[core].EndAddr) iOff=rvb[core].StartAddr+(iOff-(rvb[core].EndAddr+1));
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while(iOff<rvb[core].StartAddr) iOff=rvb[core].EndAddr-(rvb[core].StartAddr-iOff);
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return (int)*(p+iOff);
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}
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////////////////////////////////////////////////////////////////////////
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INLINE void s_buffer(int iOff,int iVal,int core) // set_buffer content helper: takes care about wraps and clipping
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{
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short * p=(short *)spuMem;
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iOff=(iOff)+rvb[core].CurrAddr;
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while(iOff>rvb[core].EndAddr) iOff=rvb[core].StartAddr+(iOff-(rvb[core].EndAddr+1));
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while(iOff<rvb[core].StartAddr) iOff=rvb[core].EndAddr-(rvb[core].StartAddr-iOff);
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if(iVal<-33768L) iVal=-33768L;if(iVal>33768L) iVal=33768L;
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*(p+iOff)=(short)iVal;
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}
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////////////////////////////////////////////////////////////////////////
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INLINE void s_buffer1(int iOff,int iVal,int core) // set_buffer (+1 sample) content helper: takes care about wraps and clipping
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{
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short * p=(short *)spuMem;
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iOff=(iOff)+rvb[core].CurrAddr+1;
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while(iOff>rvb[core].EndAddr) iOff=rvb[core].StartAddr+(iOff-(rvb[core].EndAddr+1));
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while(iOff<rvb[core].StartAddr) iOff=rvb[core].EndAddr-(rvb[core].StartAddr-iOff);
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if(iVal<-33768L) iVal=-33768L;if(iVal>33768L) iVal=33768L;
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*(p+iOff)=(short)iVal;
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}
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////////////////////////////////////////////////////////////////////////
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INLINE int MixREVERBLeft(int ns,int core)
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{
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if(iUseReverb==1)
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{
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if(!rvb[core].StartAddr || !rvb[core].EndAddr ||
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rvb[core].StartAddr>=rvb[core].EndAddr) // reverb is off
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{
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rvb[core].iLastRVBLeft=rvb[core].iLastRVBRight=rvb[core].iRVBLeft=rvb[core].iRVBRight=0;
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return 0;
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}
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rvb[core].iCnt++;
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if(rvb[core].iCnt&1) // we work on every second left value: downsample to 22 khz
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{
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if((spuCtrl2[core]&0x80)) // -> reverb on? oki
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{
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int ACC0,ACC1,FB_A0,FB_A1,FB_B0,FB_B1;
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const int INPUT_SAMPLE_L=*(sRVBStart[core]+(ns<<1));
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const int INPUT_SAMPLE_R=*(sRVBStart[core]+(ns<<1)+1);
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const int IIR_INPUT_A0 = (g_buffer(rvb[core].IIR_SRC_A0,core) * rvb[core].IIR_COEF)/33768L + (INPUT_SAMPLE_L * rvb[core].IN_COEF_L)/33768L;
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const int IIR_INPUT_A1 = (g_buffer(rvb[core].IIR_SRC_A1,core) * rvb[core].IIR_COEF)/33768L + (INPUT_SAMPLE_R * rvb[core].IN_COEF_R)/33768L;
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const int IIR_INPUT_B0 = (g_buffer(rvb[core].IIR_SRC_B0,core) * rvb[core].IIR_COEF)/33768L + (INPUT_SAMPLE_L * rvb[core].IN_COEF_L)/33768L;
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const int IIR_INPUT_B1 = (g_buffer(rvb[core].IIR_SRC_B1,core) * rvb[core].IIR_COEF)/33768L + (INPUT_SAMPLE_R * rvb[core].IN_COEF_R)/33768L;
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const int IIR_A0 = (IIR_INPUT_A0 * rvb[core].IIR_ALPHA)/33768L + (g_buffer(rvb[core].IIR_DEST_A0,core) * (33768L - rvb[core].IIR_ALPHA))/33768L;
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const int IIR_A1 = (IIR_INPUT_A1 * rvb[core].IIR_ALPHA)/33768L + (g_buffer(rvb[core].IIR_DEST_A1,core) * (33768L - rvb[core].IIR_ALPHA))/33768L;
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const int IIR_B0 = (IIR_INPUT_B0 * rvb[core].IIR_ALPHA)/33768L + (g_buffer(rvb[core].IIR_DEST_B0,core) * (33768L - rvb[core].IIR_ALPHA))/33768L;
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const int IIR_B1 = (IIR_INPUT_B1 * rvb[core].IIR_ALPHA)/33768L + (g_buffer(rvb[core].IIR_DEST_B1,core) * (33768L - rvb[core].IIR_ALPHA))/33768L;
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s_buffer1(rvb[core].IIR_DEST_A0, IIR_A0,core);
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s_buffer1(rvb[core].IIR_DEST_A1, IIR_A1,core);
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s_buffer1(rvb[core].IIR_DEST_B0, IIR_B0,core);
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s_buffer1(rvb[core].IIR_DEST_B1, IIR_B1,core);
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ACC0 = (g_buffer(rvb[core].ACC_SRC_A0,core) * rvb[core].ACC_COEF_A)/33768L +
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(g_buffer(rvb[core].ACC_SRC_B0,core) * rvb[core].ACC_COEF_B)/33768L +
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(g_buffer(rvb[core].ACC_SRC_C0,core) * rvb[core].ACC_COEF_C)/33768L +
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(g_buffer(rvb[core].ACC_SRC_D0,core) * rvb[core].ACC_COEF_D)/33768L;
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ACC1 = (g_buffer(rvb[core].ACC_SRC_A1,core) * rvb[core].ACC_COEF_A)/33768L +
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(g_buffer(rvb[core].ACC_SRC_B1,core) * rvb[core].ACC_COEF_B)/33768L +
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(g_buffer(rvb[core].ACC_SRC_C1,core) * rvb[core].ACC_COEF_C)/33768L +
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(g_buffer(rvb[core].ACC_SRC_D1,core) * rvb[core].ACC_COEF_D)/33768L;
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FB_A0 = g_buffer(rvb[core].MIX_DEST_A0 - rvb[core].FB_SRC_A,core);
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FB_A1 = g_buffer(rvb[core].MIX_DEST_A1 - rvb[core].FB_SRC_A,core);
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FB_B0 = g_buffer(rvb[core].MIX_DEST_B0 - rvb[core].FB_SRC_B,core);
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FB_B1 = g_buffer(rvb[core].MIX_DEST_B1 - rvb[core].FB_SRC_B,core);
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s_buffer(rvb[core].MIX_DEST_A0, ACC0 - (FB_A0 * rvb[core].FB_ALPHA)/33768L,core);
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s_buffer(rvb[core].MIX_DEST_A1, ACC1 - (FB_A1 * rvb[core].FB_ALPHA)/33768L,core);
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s_buffer(rvb[core].MIX_DEST_B0, (rvb[core].FB_ALPHA * ACC0)/33768L - (FB_A0 * (int)(rvb[core].FB_ALPHA^0xFFFF8000))/33768L - (FB_B0 * rvb[core].FB_X)/33768L,core);
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s_buffer(rvb[core].MIX_DEST_B1, (rvb[core].FB_ALPHA * ACC1)/33768L - (FB_A1 * (int)(rvb[core].FB_ALPHA^0xFFFF8000))/33768L - (FB_B1 * rvb[core].FB_X)/33768L,core);
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rvb[core].iLastRVBLeft = rvb[core].iRVBLeft;
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rvb[core].iLastRVBRight = rvb[core].iRVBRight;
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rvb[core].iRVBLeft = (g_buffer(rvb[core].MIX_DEST_A0,core)+g_buffer(rvb[core].MIX_DEST_B0,core))/3;
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rvb[core].iRVBRight = (g_buffer(rvb[core].MIX_DEST_A1,core)+g_buffer(rvb[core].MIX_DEST_B1,core))/3;
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rvb[core].iRVBLeft = (rvb[core].iRVBLeft * rvb[core].VolLeft) / 0x4000;
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rvb[core].iRVBRight = (rvb[core].iRVBRight * rvb[core].VolRight) / 0x4000;
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rvb[core].CurrAddr++;
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if(rvb[core].CurrAddr>rvb[core].EndAddr) rvb[core].CurrAddr=rvb[core].StartAddr;
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return rvb[core].iLastRVBLeft+(rvb[core].iRVBLeft-rvb[core].iLastRVBLeft)/2;
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}
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else // -> reverb off
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{
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rvb[core].iLastRVBLeft=rvb[core].iLastRVBRight=rvb[core].iRVBLeft=rvb[core].iRVBRight=0;
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}
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rvb[core].CurrAddr++;
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if(rvb[core].CurrAddr>rvb[core].EndAddr) rvb[core].CurrAddr=rvb[core].StartAddr;
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}
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return rvb[core].iLastRVBLeft;
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}
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return 0;
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}
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////////////////////////////////////////////////////////////////////////
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INLINE int MixREVERBRight(int core)
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{
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if(iUseReverb==1) // Neill's reverb:
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{
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int i=rvb[core].iLastRVBRight+(rvb[core].iRVBRight-rvb[core].iLastRVBRight)/2;
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rvb[core].iLastRVBRight=rvb[core].iRVBRight;
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return i; // -> just return the last right reverb val (little bit scaled by the previous right val)
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}
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return 0;
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}
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////////////////////////////////////////////////////////////////////////
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#endif
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/*
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-----------------------------------------------------------------------------
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PSX reverb hardware notes
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by Neill Corlett
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-----------------------------------------------------------------------------
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Yadda yadda disclaimer yadda probably not perfect yadda well it's okay anyway
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yadda yadda.
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-----------------------------------------------------------------------------
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Basics
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------
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- The reverb buffer is 22khz 16-bit mono PCM.
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- It starts at the reverb address given by 1DA2, extends to
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the end of sound RAM, and wraps back to the 1DA2 address.
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Setting the address at 1DA2 resets the current reverb work address.
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This work address ALWAYS increments every 1/22050 sec., regardless of
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whether reverb is enabled (bit 7 of 1DAA set).
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And the contents of the reverb buffer ALWAYS play, scaled by the
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"reverberation depth left/right" volumes (1D84/1D86).
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(which, by the way, appear to be scaled so 3FFF=approx. 1.0, 4000=-1.0)
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-----------------------------------------------------------------------------
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Register names
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--------------
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These are probably not their real names.
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These are probably not even correct names.
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We will use them anyway, because we can.
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1DC0: FB_SRC_A (offset)
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1DC2: FB_SRC_B (offset)
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1DC4: IIR_ALPHA (coef.)
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1DC6: ACC_COEF_A (coef.)
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1DC8: ACC_COEF_B (coef.)
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1DCA: ACC_COEF_C (coef.)
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1DCC: ACC_COEF_D (coef.)
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1DCE: IIR_COEF (coef.)
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1DD0: FB_ALPHA (coef.)
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1DD2: FB_X (coef.)
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1DD4: IIR_DEST_A0 (offset)
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1DD6: IIR_DEST_A1 (offset)
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1DD8: ACC_SRC_A0 (offset)
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1DDA: ACC_SRC_A1 (offset)
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1DDC: ACC_SRC_B0 (offset)
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1DDE: ACC_SRC_B1 (offset)
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1DE0: IIR_SRC_A0 (offset)
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1DE2: IIR_SRC_A1 (offset)
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1DE4: IIR_DEST_B0 (offset)
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1DE6: IIR_DEST_B1 (offset)
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1DE8: ACC_SRC_C0 (offset)
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1DEA: ACC_SRC_C1 (offset)
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1DEC: ACC_SRC_D0 (offset)
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1DEE: ACC_SRC_D1 (offset)
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1DF0: IIR_SRC_B1 (offset)
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1DF2: IIR_SRC_B0 (offset)
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1DF4: MIX_DEST_A0 (offset)
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1DF6: MIX_DEST_A1 (offset)
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1DF8: MIX_DEST_B0 (offset)
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1DFA: MIX_DEST_B1 (offset)
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1DFC: IN_COEF_L (coef.)
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1DFE: IN_COEF_R (coef.)
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The coefficients are signed fractional values.
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-32768 would be -1.0
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32768 would be 1.0 (if it were possible... the highest is of course 32767)
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The offsets are (byte/8) offsets into the reverb buffer.
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i.e. you multiply them by 8, you get byte offsets.
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You can also think of them as (samples/4) offsets.
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They appear to be signed. They can be negative.
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None of the documented presets make them negative, though.
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Yes, 1DF0 and 1DF2 appear to be backwards. Not a typo.
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-----------------------------------------------------------------------------
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What it does
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------------
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We take all reverb sources:
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- regular channels that have the reverb bit on
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- cd and external sources, if their reverb bits are on
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and mix them into one stereo 44100hz signal.
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Lowpass/downsample that to 22050hz. The PSX uses a proper bandlimiting
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algorithm here, but I haven't figured out the hysterically exact specifics.
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I use an 8-tap filter with these coefficients, which are nice but probably
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not the real ones:
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0.037828187894
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0.157538631280
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0.321159685278
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0.449322115345
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0.449322115345
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0.321159685278
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0.157538631280
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0.037828187894
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So we have two input samples (INPUT_SAMPLE_L, INPUT_SAMPLE_R) every 22050hz.
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* IN MY EMULATION, I divide these by 2 to make it clip less.
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(and of course the L/R output coefficients are adjusted to compensate)
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The real thing appears to not do this.
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At every 22050hz tick:
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- If the reverb bit is enabled (bit 7 of 1DAA), execute the reverb
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steady-state algorithm described below
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- AFTERWARDS, retrieve the "wet out" L and R samples from the reverb buffer
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(This part may not be exactly right and I guessed at the coefs. TODO: check later.)
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L is: 0.333 * (buffer[MIX_DEST_A0] + buffer[MIX_DEST_B0])
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R is: 0.333 * (buffer[MIX_DEST_A1] + buffer[MIX_DEST_B1])
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- Advance the current buffer position by 1 sample
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The wet out L and R are then upsampled to 44100hz and played at the
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"reverberation depth left/right" (1D84/1D86) volume, independent of the main
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volume.
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-----------------------------------------------------------------------------
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Reverb steady-state
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-------------------
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The reverb steady-state algorithm is fairly clever, and of course by
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"clever" I mean "batshit insane".
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buffer[x] is relative to the current buffer position, not the beginning of
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the buffer. Note that all buffer offsets must wrap around so they're
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contained within the reverb work area.
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Clipping is performed at the end... maybe also sooner, but definitely at
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the end.
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IIR_INPUT_A0 = buffer[IIR_SRC_A0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;
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IIR_INPUT_A1 = buffer[IIR_SRC_A1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;
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IIR_INPUT_B0 = buffer[IIR_SRC_B0] * IIR_COEF + INPUT_SAMPLE_L * IN_COEF_L;
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IIR_INPUT_B1 = buffer[IIR_SRC_B1] * IIR_COEF + INPUT_SAMPLE_R * IN_COEF_R;
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IIR_A0 = IIR_INPUT_A0 * IIR_ALPHA + buffer[IIR_DEST_A0] * (1.0 - IIR_ALPHA);
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IIR_A1 = IIR_INPUT_A1 * IIR_ALPHA + buffer[IIR_DEST_A1] * (1.0 - IIR_ALPHA);
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IIR_B0 = IIR_INPUT_B0 * IIR_ALPHA + buffer[IIR_DEST_B0] * (1.0 - IIR_ALPHA);
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IIR_B1 = IIR_INPUT_B1 * IIR_ALPHA + buffer[IIR_DEST_B1] * (1.0 - IIR_ALPHA);
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buffer[IIR_DEST_A0 + 1sample] = IIR_A0;
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buffer[IIR_DEST_A1 + 1sample] = IIR_A1;
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buffer[IIR_DEST_B0 + 1sample] = IIR_B0;
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buffer[IIR_DEST_B1 + 1sample] = IIR_B1;
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ACC0 = buffer[ACC_SRC_A0] * ACC_COEF_A +
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buffer[ACC_SRC_B0] * ACC_COEF_B +
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|
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;
|
|
|
|
-----------------------------------------------------------------------------
|
|
*/
|
|
|