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pcsx2/plugins/spu2ghz/src/mixer.cpp

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//GiGaHeRz's SPU2 Driver
//Copyright (c) 2003-2008, David Quintana <gigaherz@gmail.com>
//
//This library is free software; you can redistribute it and/or
//modify it under the terms of the GNU Lesser General Public
//License as published by the Free Software Foundation; either
//version 2.1 of the License, or (at your option) any later version.
//
//This library is distributed in the hope that it will be useful,
//but WITHOUT ANY WARRANTY; without even the implied warranty of
//MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
//Lesser General Public License for more details.
//
//You should have received a copy of the GNU Lesser General Public
//License along with this library; if not, write to the Free Software
//Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
//
// [Air] Notes ----->
// Adding 'static' to the __forceinline methods hints to the linker that it need not
// actually include procedural versions of the methods in the DLL. Under normal circumstances
// the compiler will still generate the procedures even though they are never used (the inline
// code is used instead). Using static reduced the size of my generated .DLL by a few KB.
// (doesn't really make anything faster, but eh... whatever :)
//
#include "spu2.h"
#include <float.h>
#include "lowpass.h"
#undef min
#undef max
#include <algorithm>
using std::min;
using std::max;
extern void spdif_update();
void ADMAOutLogWrite(void *lpData, u32 ulSize);
extern void VoiceStop(int core,int vc);
double pow_2_31 = pow(2.0,31.0);
LPF_data L,R;
extern u32 core;
u32 core, voice;
extern u8 callirq;
double srate_pv=1.0;
static const s32 ADSR_MAX_VOL = 0x7fffffff;
static const s32 f[5][2] =
{
{ 0, 0 },
{ 60, 0 },
{ 115, -52 },
{ 98, -55 },
{ 122, -60 }
};
static const int InvExpOffsets[] = { 0,4,6,8,9,10,11,12 };
static u32 PsxRates[160];
/*=
{
//for +Lin: PsxRates[value+8]
//for -Lin: PsxRates[value+7]
0xFFFFFFFF,0xFFFFFFFF,0xFFFFFFFF,0xFFFFFFFF,0xFFFFFFFF,0xFFFFFFFF,0xFFFFFFFF,0xFFFFFFFF,
0xD744FCCB,0xB504F334,0x9837F052,0x80000000,0x6BA27E65,0x5A82799A,0x4C1BF829,0x40000000,
0x35D13F33,0x2D413CCD,0x260DFC14,0x20000000,0x1AE89F99,0x16A09E66,0x1306FE0A,0x10000000,
0x0D744FCD,0x0B504F33,0x09837F05,0x08000000,0x06BA27E6,0x05A8279A,0x04C1BF83,0x04000000,
0x035D13F3,0x02D413CD,0x0260DFC1,0x02000000,0x01AE89FA,0x016A09E6,0x01306FE1,0x01000000,
0x00D744FD,0x00B504F3,0x009837F0,0x00800000,0x006BA27E,0x005A827A,0x004C1BF8,0x00400000,
0x0035D13F,0x002D413D,0x00260DFC,0x00200000,0x001AE8A0,0x0016A09E,0x001306FE,0x00100000,
0x000D7450,0x000B504F,0x0009837F,0x00080000,0x0006BA28,0x0005A828,0x0004C1C0,0x00040000,
0x00035D14,0x0002D414,0x000260E0,0x00020000,0x0001AE8A,0x00016A0A,0x00013070,0x00010000,
0x0000D745,0x0000B505,0x00009838,0x00008000,0x00006BA2,0x00005A82,0x00004C1C,0x00004000,
0x000035D1,0x00002D41,0x0000260E,0x00002000,0x00001AE9,0x000016A1,0x00001307,0x00001000,
0x00000D74,0x00000B50,0x00000983,0x00000800,0x000006BA,0x000005A8,0x000004C2,0x00000400,
0x0000035D,0x000002D4,0x00000261,0x00000200,0x000001AF,0x0000016A,0x00000130,0x00000100,
0x000000D7,0x000000B5,0x00000098,0x00000080,0x0000006C,0x0000005B,0x0000004C,0x00000040,
0x00000036,0x0000002D,0x00000026,0x00000020,0x0000001B,0x00000017,0x00000013,0x00000010,
0x0000000D,0x0000000B,0x0000000A,0x00000008,0x00000007,0x00000006,0x00000005,0x00000004,
0x00000003,0x00000003,0x00000002,0x00000002,0x00000002,0x00000001,0x00000001,0x00000000,
//128+8
0x00000000,0x00000000,0x00000000,0x00000000,0x00000000,0x00000000,0x00000000,0x00000000,
};*/
// Performs a 64-bit multiplication between two values and returns the
// high 32 bits as a result (discarding the fractional 32 bits).
// The combined fracional bits of both inputs must be 32 bits for this
// to work properly.
//
// This is meant to be a drop-in replacement for times when the 'div' part
// of a MulDiv is a constant. (example: 1<<8, or 4096, etc)
//
// [Air] Performance breakdown: This is over 10 times faster than MulDiv in
// a *worst case* scenario. It's also more accurate since it forces the
// caller to extend the inputs so that they make use of all 32 bits of
// precision.
//
static s32 __forceinline MulShr32( s32 srcval, s32 mulval )
{
s64 tmp = ((s64)srcval * mulval );
return ((s32*)&tmp)[1];
// Performance note: Using the temp var and memory reference
// actually ends up being roughly 2x faster than using a bitshift.
// It won't fly on big endian machines though... :)
}
static s32 __forceinline MulShr32su( s32 srcval, u32 mulval )
{
s64 tmp = ((s64)srcval * (s32)mulval );
return ((s32*)&tmp)[1];
}
void InitADSR() // INIT ADSR
{
for (int i=0; i<(32+128); i++)
{
int shift=(i-32)>>2;
s64 rate=(i&3)+4;
if (shift<0)
rate >>= -shift;
else
rate <<= shift;
PsxRates[i] = (int)min( rate, 0x3fffffffLL );
}
}
#define VOL(x) (((s32)x)) //24.8 volume
static void __forceinline XA_decode_block(s16* buffer, const s16* block, s32& prev1, s32& prev2)
{
const s32 header = *block;
s32 shift = ((header>> 0)&0xF)+16;
s32 pred1 = f[(header>> 4)&0xF][0];
s32 pred2 = f[(header>> 4)&0xF][1];
const s8* blockbytes = (s8*)&block[1];
for(int i=0; i<14; i++, blockbytes++)
{
s32 pcm, pcm2;
{
s32 data = ((*blockbytes)<<28) & 0xF0000000;
pcm = data>>shift;
pcm+=((pred1*prev1)+(pred2*prev2))>>6;
if(pcm> 32767) pcm= 32767;
else if(pcm<-32768) pcm=-32768;
*(buffer++) = pcm;
}
//prev2=prev1;
//prev1=pcm;
{
s32 data = ((*blockbytes)<<24) & 0xF0000000;
pcm2 = data>>shift;
pcm2+=((pred1*pcm)+(pred2*prev1))>>6;
if(pcm2> 32767) pcm2= 32767;
else if(pcm2<-32768) pcm2=-32768;
*(buffer++) = pcm2;
}
prev2=pcm;
prev1=pcm2;
}
}
static void __forceinline XA_decode_block_unsaturated(s16* buffer, const s16* block, s32& prev1, s32& prev2)
{
const s32 header = *block;
s32 shift = ((header>> 0)&0xF)+16;
s32 pred1 = f[(header>> 4)&0xF][0];
s32 pred2 = f[(header>> 4)&0xF][1];
const s8* blockbytes = (s8*)&block[1];
for(int i=0; i<14; i++, blockbytes++)
{
s32 pcm, pcm2;
{
s32 data = ((*blockbytes)<<28) & 0xF0000000;
pcm = data>>shift;
pcm+=((pred1*prev1)+(pred2*prev2))>>6;
// [Air] : Fast method, no saturation is performed.
*(buffer++) = pcm;
}
{
s32 data = ((*blockbytes)<<24) & 0xF0000000;
pcm2 = data>>shift;
pcm2+=((pred1*pcm)+(pred2*prev1))>>6;
// [Air] : Fast method, no saturation is performed.
*(buffer++) = pcm2;
}
prev2=pcm;
prev1=pcm2;
}
}
static void __forceinline IncrementNextA( const V_Core& thiscore, V_Voice& vc )
{
// Important! Both cores signal IRQ when an address is read, regardless of
// which core actually reads the address.
for( int i=0; i<2; i++ )
{
if( Cores[i].IRQEnable && (vc.NextA==Cores[i].IRQA ) )
{
#ifndef PUBLIC
ConLog(" * SPU2 Core %d: IRQ Called (IRQ passed).\n", i);
#endif
Spdif.Info=4<<i;
SetIrqCall();
}
}
vc.NextA++;
vc.NextA&=0xFFFFF;
}
// decoded pcm data, used to cache the decoded data so that it needn't be decoded
// multiple times. Cache chunks are decoded when the mixer requests the blocks, and
// invalided when DMA transfers and memory writes are performed.
PcmCacheEntry *pcm_cache_data = NULL;
#ifndef PUBLIC
int g_counter_cache_hits=0;
int g_counter_cache_misses=0;
int g_counter_cache_ignores=0;
#endif
#define XAFLAG_LOOP_END (1ul<<0)
#define XAFLAG_LOOP (1ul<<1)
#define XAFLAG_LOOP_START (1ul<<2)
static void __forceinline __fastcall GetNextDataBuffered( V_Core& thiscore, V_Voice& vc, s32& Data)
{
if (vc.SCurrent<28)
{
// [Air] : skip the increment?
// (witness one of the rare ideal uses of a goto statement!)
if( (vc.SCurrent&3) != 3 ) goto _skipIncrement;
}
else
{
if(vc.LoopFlags & XAFLAG_LOOP_END)
{
if( vc.LoopFlags & XAFLAG_LOOP )
{
vc.NextA=vc.LoopStartA;
thiscore.Regs.ENDX|=1<<voice;
}
else
{
VoiceStop(core,voice);
thiscore.Regs.ENDX|=1<<voice;
#ifndef PUBLIC
if(MsgVoiceOff()) ConLog(" * SPU2: Voice Off by EndPoint: %d \n", voice);
DebugCores[core].Voices[voice].lastStopReason = 1;
#endif
}
}
// We'll need the loop flags and buffer pointers regardless of cache status:
// Note to Self : NextA addresses WORDS (not bytes).
s16* memptr = GetMemPtr(vc.NextA&0xFFFFF);
vc.LoopFlags = *memptr >> 8; // grab loop flags from the upper byte.
const int cacheIdx = vc.NextA / pcm_WordsPerBlock;
PcmCacheEntry& cacheLine = pcm_cache_data[cacheIdx];
vc.SBuffer = cacheLine.Sampledata;
if( cacheLine.Validated )
{
// Cached block! Read from the cache directly.
// Make sure to propagate the prev1/prev2 ADPCM:
vc.Prev1 = vc.SBuffer[27];
vc.Prev2 = vc.SBuffer[26];
//ConLog( " * SPU2 : Cache Hit! NextA=0x%x, cacheIdx=0x%x\n", vc.NextA, cacheIdx );
#ifndef PUBLIC
g_counter_cache_hits++;
#endif
}
else
{
// Only flag the cache if it's a non-dynamic memory range.
if( vc.NextA >= SPU2_DYN_MEMLINE )
cacheLine.Validated = true;
#ifndef PUBLIC
if( vc.NextA < SPU2_DYN_MEMLINE )
g_counter_cache_ignores++;
else
g_counter_cache_misses++;
#endif
s16* sbuffer = cacheLine.Sampledata;
// saturated decoder
XA_decode_block( sbuffer, memptr, vc.Prev1, vc.Prev2 );
// [Air]: Testing use of a new unsaturated decoder. (benchmark needed)
// Chances are the saturation isn't needed, but for a very few exception games.
// This is definitely faster than the above version, but is it by enough to
// merit possible lower compatibility? Especially now that games that make
// heavy use of the SPU2 via music or sfx will mostly use the cache anyway.
//XA_decode_block_unsaturated( vc.SBuffer, memptr, vc.Prev1, vc.Prev2 );
}
vc.SCurrent = 0;
if( (vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode )
vc.LoopStartA = vc.NextA;
}
IncrementNextA( thiscore, vc );
_skipIncrement:
Data = vc.SBuffer[vc.SCurrent++];
}
// Returns the linear slide value for AR and SR inputs.
static int GetLinearSrAr( uint SrAr )
{
// The Sr/Ar settings work in quarter steps, which means
// the bottom 2 bits go on the left side of the shift, and
// the right side of the shift gets divided by 4:
const uint newSr = 0x7f - SrAr;
return ((1|(newSr&3)) << (newSr>>2));
}
static void __forceinline CalculateADSR( V_Voice& vc )
{
V_ADSR& env(vc.ADSR);
jASSUME( env.Phase != 0 );
if(env.Releasing && (env.Phase < 5))
env.Phase = 5;
switch (env.Phase)
{
case 1: // attack
if( env.Value == ADSR_MAX_VOL )
{
// Already maxed out. Progress phase and nothing more:
env.Phase++;
break;
}
// Case 1 below is for pseudo exponential below 75%.
// Pseudo Exp > 75% and Linear are the same.
if (env.Am && (env.Value>=0x60000000))
env.Value += PsxRates[(env.Ar^0x7f)-0x18+32];
else //if( env.Ar < 0x7f )
env.Value+=PsxRates[(env.Ar^0x7f)-0x10+32];
//env.Value += GetLinearSrAr( env.Ar );
if( env.Value < 0 )
{
// We hit the ceiling.
env.Phase++;
env.Value = ADSR_MAX_VOL;
}
break;
case 2: // decay
{
u32 off = InvExpOffsets[(env.Value>>28)&7];
env.Value-=PsxRates[((env.Dr^0x1f)*4)-0x18+off+32];
// calculate sustain level by mirroring the bits
// of the sustain var into the lower bits as we shift up
// (total shift, 27 bits)
s32 suslev = (env.Sl << 4) | env.Sl;
suslev = (suslev << 8) | suslev; // brings us to 12 bits!
suslev = (suslev << 12) | suslev; // 24 bits!
if( env.Value <= (suslev<<3) )
{
if (env.Value < 0)
env.Value = 0;
env.Phase++;
}
}
break;
case 3: // sustain
{
// 0x7f disables sustain (infinite sustain)
if( env.Sr == 0x7f ) return;
if (env.Sm&2) // decreasing
{
if (env.Sm&4) // exponential
{
u32 off = InvExpOffsets[(env.Value>>28)&7];
env.Value-=PsxRates[(env.Sr^0x7f)-0x1b+off+32];
}
else // linear
{
env.Value-=PsxRates[(env.Sr^0x7f)-0xf+32];
//env.Value -= GetLinearSrAr( env.Sr );
}
if( env.Value <= 0 )
{
env.Value = 0;
env.Phase++;
}
}
else // increasing
{
if( (env.Sm&4) && (env.Value>=0x60000000) )
env.Value+=PsxRates[(env.Sr^0x7f)-0x18+32];
else
{
// linear / Pseudo below 75% (they're the same)
env.Value+=PsxRates[(env.Sr^0x7f)-0x10+32];
//env.Value += GetLinearSrAr( env.Sr );
}
if( env.Value < 0 )
{
env.Value = ADSR_MAX_VOL;
env.Phase++;
}
}
}
break;
case 4: // sustain end
env.Value = (env.Sm&2) ? 0 : ADSR_MAX_VOL;
if(env.Value==0)
env.Phase=6;
break;
case 5: // release
if (env.Rm) // exponential
{
u32 off=InvExpOffsets[(env.Value>>28)&7];
env.Value-=PsxRates[((env.Rr^0x1f)*4)-0x18+off+32];
}
else // linear
{
//env.Value-=PsxRates[((env.Rr^0x1f)*4)-0xc+32];
if( env.Rr != 0x1f )
env.Value -= (1 << (0x1f-env.Rr));
}
if( env.Value <= 0 )
{
env.Value=0;
env.Phase++;
}
break;
case 6: // release end
env.Value=0;
break;
jNO_DEFAULT
}
if (env.Phase==6) {
#ifndef PUBLIC
if(MsgVoiceOff()) ConLog(" * SPU2: Voice Off by ADSR: %d \n", voice);
DebugCores[core].Voices[voice].lastStopReason = 2;
#endif
VoiceStop(core,voice);
Cores[core].Regs.ENDX|=(1<<voice);
env.Phase=0;
}
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
static void __forceinline GetNoiseValues(s32& VD)
{
static s32 Seed = 0x41595321;
if(Seed&0x100) VD = (s32)((Seed&0xff)<<8);
else if(!(Seed&0xffff)) VD = (s32)0x8000;
else VD = (s32)0x7fff;
__asm {
MOV eax,Seed
ROR eax,5
XOR eax,0x9a
MOV ebx,eax
ROL eax,2
ADD eax,ebx
XOR eax,ebx
ROR eax,3
MOV Seed,eax
}
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
void LowPassFilterInit()
{
LPF_init(&L,11000,48000);
LPF_init(&R,11000,48000);
}
void LowPass(s32& VL, s32& VR)
{
VL = (s32)(LPF(&L,(VL)/pow_2_31)*pow_2_31);
VR = (s32)(LPF(&R,(VR)/pow_2_31)*pow_2_31);
}
// Data is expected to be 16 bit signed (typical stuff!).
// Volume is the SPU2 register value, usually ranged from 0 to 0x7fff.
static __forceinline s32 ApplyVolume(s32 data, s32 volume)
{
return (volume * data) >> 15;
}
static void __forceinline UpdatePitch( V_Voice& vc )
{
s32 pitch;
// [Air] : re-ordered comparisons: Modulated is much more likely to be zero than voice,
// and so the way it was before it's have to check both voice and modulated values
// most of the time. Now it'll just check Modulated and short-circuit past the voice
// check (not that it amounts to much, but eh every little bit helps).
if( (vc.Modulated==0) || (voice==0) )
pitch = vc.Pitch;
else
pitch = (vc.Pitch*(32768 + abs(Cores[core].Voices[voice-1].OutX)))>>15;
vc.SP+=pitch;
}
// Returns a 16 bit result in Value.
static void __forceinline GetVoiceValues_Linear(V_Core& thiscore, V_Voice& vc, s32& Value)
{
while( vc.SP > 0 )
{
vc.PV2=vc.PV1;
GetNextDataBuffered( thiscore, vc, vc.PV1 );
vc.SP-=4096;
}
if( vc.ADSR.Phase==0 )
{
Value = 0;
return;
}
CalculateADSR( vc );
jASSUME( vc.ADSR.Value >= 0 ); // ADSR should never be negative...
// Note! It's very important that ADSR stay as accurate as possible. By the way
// it is used, various sound effects can end prematurely if we truncate more than
// one or two bits.
if(Interpolation==0)
{
Value = MulShr32( vc.PV1<<1, vc.ADSR.Value );
}
else //if(Interpolation==1) //must be linear
{
s32 t0 = vc.PV2 - vc.PV1;
Value = MulShr32( (vc.PV1<<1) - ((t0*vc.SP)>>11), vc.ADSR.Value );
}
}
// Returns a 16 bit result in Value.
static void __forceinline GetVoiceValues_Cubic(V_Core& thiscore, V_Voice& vc, s32& Value)
{
while( vc.SP > 0 )
{
vc.PV4=vc.PV3;
vc.PV3=vc.PV2;
vc.PV2=vc.PV1;
GetNextDataBuffered( thiscore, vc, vc.PV1 );
vc.PV1<<=2;
vc.SPc = vc.SP&4095; // just the fractional part, please!
vc.SP-=4096;
}
if( vc.ADSR.Phase==0 )
{
Value = 0;
return;
}
CalculateADSR( vc );
jASSUME( vc.ADSR.Value >= 0 ); // ADSR should never be negative...
s32 z0 = vc.PV3 - vc.PV4 + vc.PV1 - vc.PV2;
s32 z1 = (vc.PV4 - vc.PV3 - z0);
s32 z2 = (vc.PV2 - vc.PV4);
s32 mu = vc.SPc;
s32 val = (z0 * mu) >> 12;
val = ((val + z1) * mu) >> 12;
val = ((val + z2) * mu) >> 12;
val += vc.PV2;
// Note! It's very important that ADSR stay as accurate as possible. By the way
// it is used, various sound effects can end prematurely if we truncate more than
// one or two bits.
Value = MulShr32( val, vc.ADSR.Value>>1 );
}
// Noise values need to be mixed without going through interpolation, since it
// can wreak havoc on the noise (causing muffling or popping). Not that this noise
// generator is accurate in its own right.. but eh, ah well :)
static void __forceinline __fastcall GetNoiseValues(V_Core& thiscore, V_Voice& vc, s32& Data)
{
while(vc.SP>=4096)
{
GetNoiseValues( Data );
vc.SP-=4096;
}
// GetNoiseValues can't set the phase zero on us unexpectedly
// like GetVoiceValues can. Better assert just in case though..
jASSUME( vc.ADSR.Phase != 0 );
CalculateADSR( vc );
// Yup, ADSR applies even to noise sources...
Data = MulShr32( Data, vc.ADSR.Value );
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
void __fastcall ReadInput(V_Core& thiscore, s32& PDataL,s32& PDataR)
{
if((thiscore.AutoDMACtrl&(core+1))==(core+1))
{
s32 tl,tr;
if((core==1)&&((PlayMode&8)==8))
{
thiscore.InputPos&=~1;
// CDDA mode
// Source audio data is 32 bits.
// We don't yet have the capability to handle this high res input data
// so we just downgrade it to 16 bits for now.
#ifdef PCM24_S1_INTERLEAVE
*PDataL=*(((s32*)(thiscore.ADMATempBuffer+(thiscore.InputPos<<1))));
*PDataR=*(((s32*)(thiscore.ADMATempBuffer+(thiscore.InputPos<<1)+2)));
#else
s32 *pl=(s32*)&(thiscore.ADMATempBuffer[thiscore.InputPos]);
s32 *pr=(s32*)&(thiscore.ADMATempBuffer[thiscore.InputPos+0x200]);
PDataL=*pl;
PDataR=*pr;
#endif
PDataL>>=1; //give 31 bit data (SndOut downsamples the rest of the way)
PDataR>>=1;
thiscore.InputPos+=2;
if((thiscore.InputPos==0x100)||(thiscore.InputPos>=0x200)) {
thiscore.AdmaInProgress=0;
if(thiscore.InputDataLeft>=0x200)
{
u8 k=thiscore.InputDataLeft>=thiscore.InputDataProgress;
#ifdef PCM24_S1_INTERLEAVE
AutoDMAReadBuffer(core,1);
#else
AutoDMAReadBuffer(core,0);
#endif
thiscore.AdmaInProgress=1;
thiscore.TSA=(core<<10)+thiscore.InputPos;
if (thiscore.InputDataLeft<0x200)
{
FileLog("[%10d] AutoDMA%c block end.\n",Cycles, (core==0)?'4':'7');
#ifndef PUBLIC
if(thiscore.InputDataLeft>0)
{
if(MsgAutoDMA()) ConLog("WARNING: adma buffer didn't finish with a whole block!!\n");
}
#endif
thiscore.InputDataLeft=0;
thiscore.DMAICounter=1;
}
}
thiscore.InputPos&=0x1ff;
}
}
else if((core==0)&&((PlayMode&4)==4))
{
thiscore.InputPos&=~1;
s32 *pl=(s32*)&(thiscore.ADMATempBuffer[thiscore.InputPos]);
s32 *pr=(s32*)&(thiscore.ADMATempBuffer[thiscore.InputPos+0x200]);
PDataL=*pl;
PDataR=*pr;
thiscore.InputPos+=2;
if(thiscore.InputPos>=0x200) {
thiscore.AdmaInProgress=0;
if(thiscore.InputDataLeft>=0x200)
{
u8 k=thiscore.InputDataLeft>=thiscore.InputDataProgress;
AutoDMAReadBuffer(core,0);
thiscore.AdmaInProgress=1;
thiscore.TSA=(core<<10)+thiscore.InputPos;
if (thiscore.InputDataLeft<0x200)
{
FileLog("[%10d] Spdif AutoDMA%c block end.\n",Cycles, (core==0)?'4':'7');
#ifndef PUBLIC
if(thiscore.InputDataLeft>0)
{
if(MsgAutoDMA()) ConLog("WARNING: adma buffer didn't finish with a whole block!!\n");
}
#endif
thiscore.InputDataLeft=0;
thiscore.DMAICounter=1;
}
}
thiscore.InputPos&=0x1ff;
}
}
else
{
if((core==1)&&((PlayMode&2)!=0))
{
tl=0;
tr=0;
}
else
{
// Using the temporary buffer because this area gets overwritten by some other code.
//*PDataL=(s32)*(s16*)(spu2mem+0x2000+(core<<10)+thiscore.InputPos);
//*PDataR=(s32)*(s16*)(spu2mem+0x2200+(core<<10)+thiscore.InputPos);
tl=(s32)thiscore.ADMATempBuffer[thiscore.InputPos];
tr=(s32)thiscore.ADMATempBuffer[thiscore.InputPos+0x200];
}
PDataL=tl;
PDataR=tr;
thiscore.InputPos++;
if((thiscore.InputPos==0x100)||(thiscore.InputPos>=0x200)) {
thiscore.AdmaInProgress=0;
if(thiscore.InputDataLeft>=0x200)
{
u8 k=thiscore.InputDataLeft>=thiscore.InputDataProgress;
AutoDMAReadBuffer(core,0);
thiscore.AdmaInProgress=1;
thiscore.TSA=(core<<10)+thiscore.InputPos;
if (thiscore.InputDataLeft<0x200)
{
thiscore.AutoDMACtrl |= ~3;
#ifndef PUBLIC
FileLog("[%10d] AutoDMA%c block end.\n",Cycles, (core==0)?'4':'7');
if(thiscore.InputDataLeft>0)
{
if(MsgAutoDMA()) ConLog("WARNING: adma buffer didn't finish with a whole block!!\n");
}
#endif
thiscore.InputDataLeft=0;
thiscore.DMAICounter=1;
}
}
thiscore.InputPos&=0x1ff;
}
}
}
else {
PDataL=0;
PDataR=0;
}
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
static void __forceinline __fastcall ReadInputPV(V_Core& thiscore, s32& ValL,s32& ValR)
{
s32 DL=0, DR=0;
u32 pitch=AutoDMAPlayRate[core];
if(pitch==0) pitch=48000;
thiscore.ADMAPV+=pitch;
while(thiscore.ADMAPV>=48000)
{
ReadInput(thiscore, DL,DR);
thiscore.ADMAPV-=48000;
thiscore.ADMAPL=DL;
thiscore.ADMAPR=DR;
}
ValL=thiscore.ADMAPL;
ValR=thiscore.ADMAPR;
#ifndef PUBLIC
s32 InputPeak = max(abs(ValL),abs(ValR));
if(DebugCores[core].AutoDMAPeak<InputPeak) DebugCores[core].AutoDMAPeak=InputPeak;
#endif
// Apply volumes:
ValL = ApplyVolume( ValL, thiscore.InpL );
ValR = ApplyVolume( ValR, thiscore.InpR );
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
#define VOLFLAG_REVERSE_PHASE (1ul<<0)
#define VOLFLAG_DECREMENT (1ul<<1)
#define VOLFLAG_EXPONENTIAL (1ul<<2)
#define VOLFLAG_SLIDE_ENABLE (1ul<<3)
static void __fastcall UpdateVolume(V_Volume& Vol)
{
// TODO : Vol.Value should be a 32 bit internal value, same as ADSR.
// Volume slides use the same basic logic as ADSR, but simplified (single-stage
// instead of multi-stage)
if (Vol.Mode & VOLFLAG_DECREMENT)
{
// Decrement
if(Vol.Mode & VOLFLAG_EXPONENTIAL)
{
u32 off = InvExpOffsets[(Vol.Value>>12)&7];
Vol.Value -= PsxRates[(Vol.Increment^0x7f)-0x1b+off+32] >> 16;
}
else
Vol.Value -= Vol.Increment;
if (Vol.Value < 0)
{
Vol.Value = 0;
Vol.Mode = 0; // disable slide
}
}
else
{
// Increment
// Pseudo-exponential increments, as done by the SPU2 (really!)
// Above 75% slides slow, below 75% slides fast. It's exponential, pseudoly speaking.
if( (Vol.Mode & VOLFLAG_EXPONENTIAL) && (Vol.Value>=0x6000))
Vol.Value += PsxRates[(Vol.Increment^0x7f)-0x18+32] >> 16;
else
Vol.Value += Vol.Increment;
if( Vol.Value < 0 ) // wrapped around the "top"?
{
Vol.Value = 0x7fff;
Vol.Mode = 0; // disable slide
}
}
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
static s32 __forceinline clamp(s32 x)
{
if (x>0x00ffffff) return 0x00ffffff;
if (x<0xff000000) return 0xff000000;
return x;
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
static void DoReverb( V_Core& thiscore, s32& OutL, s32& OutR, s32 InL, s32 InR)
{
static s32 INPUT_SAMPLE_L,INPUT_SAMPLE_R;
static s32 OUTPUT_SAMPLE_L,OUTPUT_SAMPLE_R;
if(!(thiscore.FxEnable&&EffectsEnabled))
{
OUTPUT_SAMPLE_L=0;
OUTPUT_SAMPLE_R=0;
}
else if((Cycles&1)==0)
{
INPUT_SAMPLE_L=InL;
INPUT_SAMPLE_R=InR;
}
else
{
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
s32 IIR_INPUT_A0,IIR_INPUT_A1,IIR_INPUT_B0,IIR_INPUT_B1;
s32 ACC0,ACC1;
s32 FB_A0,FB_A1,FB_B0,FB_B1;
s32 buffsize=thiscore.EffectsEndA-thiscore.EffectsStartA+1;
if(buffsize<0)
{
buffsize = thiscore.EffectsEndA;
thiscore.EffectsEndA=thiscore.EffectsStartA;
thiscore.EffectsStartA=buffsize;
buffsize=thiscore.EffectsEndA-thiscore.EffectsStartA+1;
}
//filter the 2 samples (prev then current)
LowPass(INPUT_SAMPLE_L, INPUT_SAMPLE_R);
LowPass(InL, InR);
INPUT_SAMPLE_L=(INPUT_SAMPLE_L+InL)>>9;
INPUT_SAMPLE_R=(INPUT_SAMPLE_R+InR)>>9;
#define BUFFER(x) ((s32)(*GetMemPtr(thiscore.EffectsStartA + ((thiscore.ReverbX + buffsize-((x)<<2))%buffsize))))
#define SBUFFER(x) (*GetMemPtr(thiscore.EffectsStartA + ((thiscore.ReverbX + buffsize-((x)<<2))%buffsize)))
thiscore.ReverbX=((thiscore.ReverbX + 4)%buffsize);
IIR_INPUT_A0 = (BUFFER(thiscore.Revb.IIR_SRC_A0) * thiscore.Revb.IIR_COEF + INPUT_SAMPLE_L * thiscore.Revb.IN_COEF_L)>>16;
IIR_INPUT_A1 = (BUFFER(thiscore.Revb.IIR_SRC_A1) * thiscore.Revb.IIR_COEF + INPUT_SAMPLE_R * thiscore.Revb.IN_COEF_R)>>16;
IIR_INPUT_B0 = (BUFFER(thiscore.Revb.IIR_SRC_B0) * thiscore.Revb.IIR_COEF + INPUT_SAMPLE_L * thiscore.Revb.IN_COEF_L)>>16;
IIR_INPUT_B1 = (BUFFER(thiscore.Revb.IIR_SRC_B1) * thiscore.Revb.IIR_COEF + INPUT_SAMPLE_R * thiscore.Revb.IN_COEF_R)>>16;
SBUFFER(thiscore.Revb.IIR_DEST_A0 + 4) = clamp((IIR_INPUT_A0 * thiscore.Revb.IIR_ALPHA + BUFFER(thiscore.Revb.IIR_DEST_A0) * (65535 - thiscore.Revb.IIR_ALPHA))>>16);
SBUFFER(thiscore.Revb.IIR_DEST_A1 + 4) = clamp((IIR_INPUT_A1 * thiscore.Revb.IIR_ALPHA + BUFFER(thiscore.Revb.IIR_DEST_A1) * (65535 - thiscore.Revb.IIR_ALPHA))>>16);
SBUFFER(thiscore.Revb.IIR_DEST_B0 + 4) = clamp((IIR_INPUT_B0 * thiscore.Revb.IIR_ALPHA + BUFFER(thiscore.Revb.IIR_DEST_B0) * (65535 - thiscore.Revb.IIR_ALPHA))>>16);
SBUFFER(thiscore.Revb.IIR_DEST_B1 + 4) = clamp((IIR_INPUT_B1 * thiscore.Revb.IIR_ALPHA + BUFFER(thiscore.Revb.IIR_DEST_B1) * (65535 - thiscore.Revb.IIR_ALPHA))>>16);
ACC0 = (s32)(BUFFER(thiscore.Revb.ACC_SRC_A0) * thiscore.Revb.ACC_COEF_A +
BUFFER(thiscore.Revb.ACC_SRC_B0) * thiscore.Revb.ACC_COEF_B +
BUFFER(thiscore.Revb.ACC_SRC_C0) * thiscore.Revb.ACC_COEF_C +
BUFFER(thiscore.Revb.ACC_SRC_D0) * thiscore.Revb.ACC_COEF_D)>>16;
ACC1 = (s32)(BUFFER(thiscore.Revb.ACC_SRC_A1) * thiscore.Revb.ACC_COEF_A +
BUFFER(thiscore.Revb.ACC_SRC_B1) * thiscore.Revb.ACC_COEF_B +
BUFFER(thiscore.Revb.ACC_SRC_C1) * thiscore.Revb.ACC_COEF_C +
BUFFER(thiscore.Revb.ACC_SRC_D1) * thiscore.Revb.ACC_COEF_D)>>16;
FB_A0 = BUFFER(thiscore.Revb.MIX_DEST_A0 - thiscore.Revb.FB_SRC_A);
FB_A1 = BUFFER(thiscore.Revb.MIX_DEST_A1 - thiscore.Revb.FB_SRC_A);
FB_B0 = BUFFER(thiscore.Revb.MIX_DEST_B0 - thiscore.Revb.FB_SRC_B);
FB_B1 = BUFFER(thiscore.Revb.MIX_DEST_B1 - thiscore.Revb.FB_SRC_B);
SBUFFER(thiscore.Revb.MIX_DEST_A0) = clamp((ACC0 - FB_A0 * thiscore.Revb.FB_ALPHA)>>16);
SBUFFER(thiscore.Revb.MIX_DEST_A1) = clamp((ACC1 - FB_A1 * thiscore.Revb.FB_ALPHA)>>16);
SBUFFER(thiscore.Revb.MIX_DEST_B0) = clamp(((thiscore.Revb.FB_ALPHA * ACC0) - FB_A0 * (65535 - thiscore.Revb.FB_ALPHA) - FB_B0 * thiscore.Revb.FB_X)>>16);
SBUFFER(thiscore.Revb.MIX_DEST_B1) = clamp(((thiscore.Revb.FB_ALPHA * ACC1) - FB_A1 * (65535 - thiscore.Revb.FB_ALPHA) - FB_B1 * thiscore.Revb.FB_X)>>16);
OUTPUT_SAMPLE_L=clamp((BUFFER(thiscore.Revb.MIX_DEST_A0)+BUFFER(thiscore.Revb.MIX_DEST_B0))>>2);
OUTPUT_SAMPLE_R=clamp((BUFFER(thiscore.Revb.MIX_DEST_B1)+BUFFER(thiscore.Revb.MIX_DEST_B1))>>2);
}
OutL=OUTPUT_SAMPLE_L;
OutR=OUTPUT_SAMPLE_R;
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
// writes a signed value to the SPU2 ram
// Performs no cache invalidation -- use only for dynamic memory ranges
// of the SPU2 (between 0x0000 and SPU2_DYN_MEMLINE)
static __forceinline void spu2M_WriteFast( u32 addr, s16 value )
{
// throw an assertion if the memory range is invalid:
jASSUME( addr < SPU2_DYN_MEMLINE );
*GetMemPtr( addr ) = value;
}
static __forceinline void MixVoice( V_Core& thiscore, V_Voice& vc, s32& VValL, s32& VValR )
{
s32 Value=0;
VValL = 0;
VValR = 0;
// Most games don't use much volume slide effects. So only call the UpdateVolume
// methods when needed by checking the flag outside the method here...
if( vc.VolumeL.Mode & VOLFLAG_SLIDE_ENABLE ) UpdateVolume( vc.VolumeL );
if( vc.VolumeR.Mode & VOLFLAG_SLIDE_ENABLE ) UpdateVolume( vc.VolumeR );
if( vc.ADSR.Phase > 0 )
{
UpdatePitch( vc );
if( vc.Noise )
GetNoiseValues( thiscore, vc, Value );
else
{
if( Interpolation == 2 )
GetVoiceValues_Cubic( thiscore, vc, Value );
else
GetVoiceValues_Linear( thiscore, vc, Value );
}
// Record the output (used for modulation effects)
vc.OutX = Value;
#ifndef PUBLIC
DebugCores[core].Voices[voice].displayPeak = max(DebugCores[core].Voices[voice].displayPeak,abs(Value));
#endif
// TODO : Implement this using high-def MulShr32.
// vc.VolumeL/R are 15 bits. Value should be 32 bits (but is currently 16)
VValL = ApplyVolume(Value,vc.VolumeL.Value);
VValR = ApplyVolume(Value,vc.VolumeR.Value);
}
// Write-back of raw voice data (post ADSR applied)
if (voice==1) spu2M_WriteFast( 0x400 + (core<<12) + OutPos, (s16)Value );
else if (voice==3) spu2M_WriteFast( 0x600 + (core<<12) + OutPos, (s16)Value );
}
static void __fastcall MixCore(s32& OutL, s32& OutR, s32 ExtL, s32 ExtR)
{
s32 RVL,RVR;
s32 SDL=0,SDR=0;
s32 SWL=0,SWR=0;
V_Core& thiscore( Cores[core] );
for (voice=0;voice<24;voice++)
{
s32 VValL,VValR;
V_Voice& vc( thiscore.Voices[voice] );
MixVoice( thiscore, vc, VValL, VValR );
// Note: Results from MixVoice are ranged at 16 bits.
// Following muls are toggles only (0 or 1)
SDL += VValL & vc.DryL;
SDR += VValR & vc.DryR;
SWL += VValL & vc.WetL;
SWR += VValR & vc.WetR;
}
// Saturate final result to standard 16 bit range.
SDL = min( max( SDL, -0x8000 ), 0x7fff );
SDR = min( max( SDR, -0x8000 ), 0x7fff );
SWL = min( max( SWL, -0x8000 ), 0x7fff );
SWR = min( max( SWR, -0x8000 ), 0x7fff );
// Write Mixed results To Output Area
spu2M_WriteFast( 0x1000 + (core<<12) + OutPos, (s16)SDL );
spu2M_WriteFast( 0x1200 + (core<<12) + OutPos, (s16)SDR );
spu2M_WriteFast( 0x1400 + (core<<12) + OutPos, (s16)SWL );
spu2M_WriteFast( 0x1600 + (core<<12) + OutPos, (s16)SWR );
// Write mixed results to logfile (if enabled)
WaveDump::WriteCore( core, CoreSrc_DryVoiceMix, SDL, SDR );
WaveDump::WriteCore( core, CoreSrc_WetVoiceMix, SWL, SWR );
s32 TDL,TDR;
// Mix in the Input data
TDL = OutL & thiscore.InpDryL;
TDR = OutR & thiscore.InpDryR;
// Mix in the Voice data
TDL += SDL & thiscore.SndDryL;
TDR += SDR & thiscore.SndDryR;
// Mix in the External (nothing/core0) data
TDL += ExtL & thiscore.ExtDryL;
TDR += ExtR & thiscore.ExtDryR;
if(EffectsEnabled)
{
s32 TWL,TWR;
// Mix Input, Voice, and External data:
TWL = OutL & thiscore.InpWetL;
TWR = OutR & thiscore.InpWetR;
TWL += SWL & thiscore.SndWetL;
TWR += SWR & thiscore.SndWetR;
TWL += ExtL & thiscore.ExtWetL;
TWR += ExtR & thiscore.ExtWetR;
WaveDump::WriteCore( core, CoreSrc_PreReverb, TWL, TWR );
//Apply Effects
DoReverb( thiscore, RVL, RVR, TWL, TWR );
TWL = ApplyVolume(RVL,VOL(thiscore.FxL));
TWR = ApplyVolume(RVR,VOL(thiscore.FxR));
WaveDump::WriteCore( core, CoreSrc_PostReverb, TWL, TWR );
//Mix Wet,Dry
OutL = (TDL + TWL);
OutR = (TDR + TWR);
}
else
{
OutL = TDL;
OutR = TDR;
}
//Apply Master Volume
if( thiscore.MasterL.Mode & VOLFLAG_SLIDE_ENABLE ) UpdateVolume(thiscore.MasterL);
if( thiscore.MasterR.Mode & VOLFLAG_SLIDE_ENABLE ) UpdateVolume(thiscore.MasterR);
if (thiscore.Mute==0)
{
// Final output value -- We don't use ApplyVolume sot hat we can leave the 15 bits of
// fixed point accuracy in place for SoundTouch and other post processing.
OutL = OutL * thiscore.MasterL.Value;
OutR = OutR * thiscore.MasterR.Value;
}
else
{
OutL=0;
OutR=0;
}
}
// used to throttle the output rate of cache stat reports
static int p_cachestat_counter=0;
void __fastcall Mix()
{
s32 ExtL=0, ExtR=0, OutL, OutR;
// **** CORE ZERO ****
core=0;
if( (PlayMode&4) == 0 )
{
// get input data from input buffers
ReadInputPV(Cores[0], ExtL, ExtR);
WaveDump::WriteCore( 0, CoreSrc_Input, ExtL, ExtR );
}
MixCore( ExtL, ExtR, 0, 0 );
if( PlayMode & 4 )
{
ExtL=0;
ExtR=0;
}
// Commit Core 0 output to ram before mixing Core 1:
ExtL>>=15;
ExtR>>=15;
ExtL = min( max( ExtL, -0x8000 ), 0x7fff );
ExtR = min( max( ExtR, -0x8000 ), 0x7fff );
spu2M_WriteFast( 0x800 + OutPos, ExtL );
spu2M_WriteFast( 0xA00 + OutPos, ExtR );
WaveDump::WriteCore( 0, CoreSrc_External, ExtL, ExtR );
// **** CORE ONE ****
core=1;
if( (PlayMode&8) != 8 )
{
ReadInputPV(Cores[1], OutL, OutR); // get input data from input buffers
WaveDump::WriteCore( 1, CoreSrc_Input, OutL, OutR );
}
// Apply volume to the external (Core 0) input data.
MixCore( OutL, OutR, ApplyVolume( ExtL, Cores[1].ExtL), ApplyVolume( ExtR, Cores[1].ExtR) );
if( PlayMode & 8 )
{
// Experimental CDDA support
// The CDDA overrides all other mixer output. It's a direct feed!
ReadInput(Cores[1], OutL, OutR);
//WaveLog::WriteCore( 1, "CDDA-32", OutL, OutR );
}
#ifndef PUBLIC
static s32 Peak0,Peak1;
static s32 PCount;
Peak0 = max(Peak0,max(ExtL,ExtR));
Peak1 = max(Peak1,max(OutL,OutR));
#endif
// Update spdif (called each sample)
if(PlayMode&4)
spdif_update();
OutL >>= 6;
OutR >>= 6;
// AddToBuffer
SndWrite(OutL, OutR);
OutPos++;
if (OutPos>=0x200) OutPos=0;
#ifndef PUBLIC
// [TODO]: Create an INI option to enable/disable this particular log.
p_cachestat_counter++;
if(p_cachestat_counter > (48000*10) )
{
p_cachestat_counter = 0;
if( MsgCache() ) ConLog( " * SPU2 > CacheStats > Hits: %d Misses: %d Ignores: %d\n",
g_counter_cache_hits,
g_counter_cache_misses,
g_counter_cache_ignores );
g_counter_cache_hits =
g_counter_cache_misses =
g_counter_cache_ignores = 0;
}
#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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