mirror of https://github.com/PCSX2/pcsx2.git
289 lines
6.3 KiB
C++
289 lines
6.3 KiB
C++
/* ZeroSPU2
|
|
* Copyright (C) 2006-2010 zerofrog
|
|
*
|
|
* 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.
|
|
*
|
|
* This program is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
* GNU General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU General Public License
|
|
* along with this program; if not, write to the Free Software
|
|
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
|
|
*/
|
|
|
|
#include <assert.h>
|
|
#include <stdlib.h>
|
|
|
|
#include "zerospu2.h"
|
|
|
|
#include "SoundTouch/SoundTouch.h"
|
|
#include "SoundTouch/WavFile.h"
|
|
|
|
|
|
// VOICE_PROCESSED definitions
|
|
SPU_CONTROL_* VOICE_PROCESSED::GetCtrl()
|
|
{
|
|
return ((SPU_CONTROL_*)(spu2regs+memoffset+REG_C0_CTRL));
|
|
}
|
|
|
|
void VOICE_PROCESSED::SetVolume(s32 iProcessRight)
|
|
{
|
|
u16 vol = iProcessRight ? pvoice->right.word : pvoice->left.word;
|
|
|
|
if (vol&0x8000) // sweep not working
|
|
{
|
|
short sInc=1; // -> sweep up?
|
|
|
|
if (vol&0x2000) sInc=-1; // -> or down?
|
|
if (vol&0x1000) vol^=0xffff; // -> mmm... phase inverted? have to investigate this
|
|
|
|
vol=((vol&0x7f)+1)/2; // -> sweep: 0..127 -> 0..64
|
|
vol+=vol/(2*sInc); // -> HACK: we don't sweep right now, so we just raise/lower the volume by the half!
|
|
vol*=128;
|
|
}
|
|
else // no sweep:
|
|
{
|
|
if (vol&0x4000) vol=0x3fff-(vol&0x3fff); // -> mmm... phase inverted? have to investigate this
|
|
}
|
|
|
|
if ( iProcessRight )
|
|
rightvol = vol&0x3fff;
|
|
else
|
|
leftvol = vol&0x3fff;
|
|
|
|
bVolChanged = true;
|
|
}
|
|
|
|
void VOICE_PROCESSED::StartSound()
|
|
{
|
|
ADSRX.lVolume=1; // and init some adsr vars
|
|
ADSRX.State=0;
|
|
ADSRX.EnvelopeVol=0;
|
|
|
|
if (bReverb && GetCtrl()->reverb)
|
|
{
|
|
// setup the reverb effects
|
|
}
|
|
|
|
pCurr=pStart; // set sample start
|
|
|
|
s_1=0; // init mixing vars
|
|
s_2=0;
|
|
iSBPos=28;
|
|
|
|
bNew=false; // init channel flags
|
|
bStop=false;
|
|
bOn=true;
|
|
SB[29]=0; // init our interpolation helpers
|
|
SB[30]=0;
|
|
|
|
spos=0x10000L;
|
|
SB[31]=0;
|
|
}
|
|
|
|
void VOICE_PROCESSED::VoiceChangeFrequency()
|
|
{
|
|
iUsedFreq=iActFreq; // -> take it and calc steps
|
|
sinc=(u32)pvoice->pitch<<4;
|
|
|
|
if (!sinc) sinc=1;
|
|
|
|
// -> freq change in simle imterpolation mode: set flag
|
|
SB[32]=1;
|
|
}
|
|
|
|
void VOICE_PROCESSED::InterpolateUp()
|
|
{
|
|
if (SB[32]==1) // flag == 1? calc step and set flag... and don't change the value in this pass
|
|
{
|
|
const s32 id1=SB[30]-SB[29]; // curr delta to next val
|
|
const s32 id2=SB[31]-SB[30]; // and next delta to next-next val :)
|
|
|
|
SB[32]=0;
|
|
|
|
if (id1>0) // curr delta positive
|
|
{
|
|
if (id2<id1)
|
|
{
|
|
SB[28]=id1;
|
|
SB[32]=2;
|
|
}
|
|
else if (id2<(id1<<1))
|
|
SB[28]=(id1*sinc)/0x10000L;
|
|
else
|
|
SB[28]=(id1*sinc)/0x20000L;
|
|
}
|
|
else // curr delta negative
|
|
{
|
|
if (id2>id1)
|
|
{
|
|
SB[28]=id1;
|
|
SB[32]=2;
|
|
}
|
|
else if (id2>(id1<<1))
|
|
SB[28]=(id1*sinc)/0x10000L;
|
|
else
|
|
SB[28]=(id1*sinc)/0x20000L;
|
|
}
|
|
}
|
|
else if (SB[32]==2) // flag 1: calc step and set flag... and don't change the value in this pass
|
|
{
|
|
SB[32]=0;
|
|
|
|
SB[28]=(SB[28]*sinc)/0x20000L;
|
|
if (sinc<=0x8000)
|
|
SB[29]=SB[30]-(SB[28]*((0x10000/sinc)-1));
|
|
else
|
|
SB[29]+=SB[28];
|
|
}
|
|
else // no flags? add bigger val (if possible), calc smaller step, set flag1
|
|
SB[29]+=SB[28];
|
|
}
|
|
|
|
//
|
|
// even easier interpolation on downsampling, also no special filter, again just "Pete's common sense" tm
|
|
//
|
|
|
|
void VOICE_PROCESSED::InterpolateDown()
|
|
{
|
|
if (sinc>=0x20000L) // we would skip at least one val?
|
|
{
|
|
SB[29]+=(SB[30]-SB[29])/2; // add easy weight
|
|
if (sinc>=0x30000L) // we would skip even more vals?
|
|
SB[29]+=(SB[31]-SB[30])/2; // add additional next weight
|
|
}
|
|
}
|
|
|
|
void VOICE_PROCESSED::FModChangeFrequency(s32 ns)
|
|
{
|
|
s32 NP=pvoice->pitch;
|
|
|
|
NP=((32768L+iFMod[ns])*NP)/32768L;
|
|
|
|
if (NP>0x3fff) NP=0x3fff;
|
|
if (NP<0x1) NP=0x1;
|
|
|
|
NP = (SAMPLE_RATE * NP) / (4096L); // calc frequency
|
|
|
|
iActFreq=NP;
|
|
iUsedFreq=NP;
|
|
sinc=(((NP/10)<<16)/4800);
|
|
if (!sinc) sinc=1;
|
|
|
|
// freq change in simple interpolation mode
|
|
SB[32]=1;
|
|
|
|
iFMod[ns]=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;
|
|
|
|
#ifdef _WIN32
|
|
__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
|
|
}
|
|
#else
|
|
__asm__ (
|
|
".intel_syntax\n"
|
|
"MOV %%eax,%1\n"
|
|
"ROR %%eax,5\n"
|
|
"XOR %%eax,0x9a\n"
|
|
"MOV %%ebx,%%eax\n"
|
|
"ROL %%eax,2\n"
|
|
"ADD %%eax,%%ebx\n"
|
|
"XOR %%eax,%%ebx\n"
|
|
"ROR %%eax,3\n"
|
|
"MOV %0,%%eax\n"
|
|
".att_syntax\n" : "r="(Seed) :"r"(Seed));
|
|
#endif
|
|
}
|
|
|
|
// fixme - noise handler... just produces some noise data
|
|
// surely wrong... and no noise frequency (spuCtrl&0x3f00) will be used...
|
|
// and sometimes the noise will be used as fmod modulation... pfff
|
|
int VOICE_PROCESSED::iGetNoiseVal()
|
|
{
|
|
s32 fa;
|
|
|
|
/*if ((dwNoiseVal<<=1)&0x80000000L)
|
|
{
|
|
dwNoiseVal^=0x0040001L;
|
|
fa = ((dwNoiseVal>>2)&0x7fff);
|
|
fa = -fa;
|
|
}
|
|
else
|
|
fa=(dwNoiseVal>>2)&0x7fff;*/
|
|
GetNoiseValues(fa);
|
|
|
|
// mmm... depending on the noise freq we allow bigger/smaller changes to the previous val
|
|
fa=iOldNoise + ((fa - iOldNoise) / ((0x001f - (GetCtrl()->noiseFreq)) + 1));
|
|
|
|
clamp16(fa);
|
|
|
|
iOldNoise=fa;
|
|
SB[29] = fa; // -> store noise val in "current sample" slot
|
|
return fa;
|
|
}
|
|
|
|
void VOICE_PROCESSED::StoreInterpolationVal(s32 fa)
|
|
{
|
|
if (bFMod==2) // fmod freq channel
|
|
SB[29]=fa;
|
|
else
|
|
{
|
|
if (!GetCtrl()->spuUnmute)
|
|
fa=0; // muted?
|
|
else // else adjust
|
|
{
|
|
clamp16(fa);
|
|
}
|
|
|
|
SB[28] = 0;
|
|
SB[29] = SB[30]; // -> helpers for simple linear interpolation: delay real val for two slots, and calc the two deltas, for a 'look at the future behaviour'
|
|
SB[30] = SB[31];
|
|
SB[31] = fa;
|
|
SB[32] = 1; // -> flag: calc new interolation
|
|
}
|
|
}
|
|
|
|
s32 VOICE_PROCESSED::iGetInterpolationVal()
|
|
{
|
|
s32 fa;
|
|
|
|
if (bFMod==2) return SB[29];
|
|
|
|
if (sinc<0x10000L) // -> upsampling?
|
|
InterpolateUp(); // --> interpolate up
|
|
else
|
|
InterpolateDown(); // --> else down
|
|
|
|
fa=SB[29];
|
|
return fa;
|
|
}
|
|
|
|
void VOICE_PROCESSED::Stop()
|
|
{
|
|
}
|