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Audio interpolation (#1176)

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add audio interpolation (emulation improvement)
This commit is contained in:
Arisotura
2021-08-08 14:27:57 +02:00
committed by GitHub
parent b28a9e4d24
commit 2df6b4fdc3
14 changed files with 180 additions and 45 deletions
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83
src/SPU.cpp
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@ -18,6 +18,7 @@
#include <stdio.h>
#include <string.h>
#include <cmath>
#include "Platform.h"
#include "NDS.h"
#include "DSi.h"
@ -27,7 +28,6 @@
// SPU TODO
// * capture addition modes, overflow bugs
// * channel hold
// * 'length less than 4' glitch
namespace SPU
{
@ -62,6 +62,12 @@ const s16 PSGTable[8][8] =
{-0x7FFF, -0x7FFF, -0x7FFF, -0x7FFF, -0x7FFF, -0x7FFF, -0x7FFF, -0x7FFF}
};
// audio interpolation is an improvement upon the original hardware
// (which performs no interpolation)
int InterpType;
s16 InterpCos[0x100];
s16 InterpCubic[0x100][4];
const u32 OutputBufferSize = 2*2048;
s16 OutputBackbuffer[2 * OutputBufferSize];
u32 OutputBackbufferWritePosition;
@ -90,6 +96,32 @@ bool Init()
AudioLock = Platform::Mutex_Create();
InterpType = 0;
// generate interpolation tables
// values are 1:1:14 fixed-point
float m_pi = std::acos(-1.0f);
for (int i = 0; i < 0x100; i++)
{
float ratio = (i * m_pi) / 255.0f;
ratio = 1.0f - std::cos(ratio);
InterpCos[i] = (s16)(ratio * 0x2000);
}
for (int i = 0; i < 0x100; i++)
{
s32 i1 = i << 6;
s32 i2 = (i * i) >> 2;
s32 i3 = (i * i * i) >> 10;
InterpCubic[i][0] = -i3 + 2*i2 - i1;
InterpCubic[i][1] = i3 - 2*i2 + 0x4000;
InterpCubic[i][2] = -i3 + i2 + i1;
InterpCubic[i][3] = i3 - i2;
}
return true;
}
@ -148,6 +180,11 @@ void DoSavestate(Savestate* file)
}
void SetInterpolation(int type)
{
InterpType = type;
}
void SetBias(u16 bias)
{
Bias = bias;
@ -202,6 +239,7 @@ void Channel::DoSavestate(Savestate* file)
file->Var8((u8*)&KeyOn);
file->Var32(&Timer);
file->Var32((u32*)&Pos);
file->VarArray(PrevSample, sizeof(PrevSample));
file->Var16((u16*)&CurSample);
file->Var16(&NoiseVal);
@ -215,7 +253,7 @@ void Channel::DoSavestate(Savestate* file)
file->Var32(&FIFOWritePos);
file->Var32(&FIFOReadOffset);
file->Var32(&FIFOLevel);
file->VarArray(FIFO, 8*4);
file->VarArray(FIFO, sizeof(FIFO));
}
void Channel::FIFO_BufferData()
@ -269,6 +307,9 @@ void Channel::Start()
Pos = -3;
NoiseVal = 0x7FFF;
PrevSample[0] = 0;
PrevSample[1] = 0;
PrevSample[2] = 0;
CurSample = 0;
FIFOReadPos = 0;
@ -444,6 +485,16 @@ s32 Channel::Run()
{
Timer = TimerReload + (Timer - 0x10000);
// for optional interpolation: save previous samples
// the interpolated audio will be delayed by a couple samples,
// but it's easier to deal with this way
if ((type < 3) && (InterpType != 0))
{
PrevSample[2] = PrevSample[1];
PrevSample[1] = PrevSample[0];
PrevSample[0] = CurSample;
}
switch (type)
{
case 0: NextSample_PCM8(); break;
@ -455,6 +506,34 @@ s32 Channel::Run()
}
s32 val = (s32)CurSample;
// interpolation (emulation improvement, not a hardware feature)
if ((type < 3) && (InterpType != 0))
{
s32 samplepos = ((Timer - TimerReload) * 0x100) / (0x10000 - TimerReload);
if (samplepos > 0xFF) samplepos = 0xFF;
switch (InterpType)
{
case 1: // linear
val = ((val * samplepos) +
(PrevSample[0] * (0xFF-samplepos))) >> 8;
break;
case 2: // cosine
val = ((val * InterpCos[samplepos]) +
(PrevSample[0] * InterpCos[0xFF-samplepos])) >> 14;
break;
case 3: // cubic
val = ((PrevSample[2] * InterpCubic[samplepos][0]) +
(PrevSample[1] * InterpCubic[samplepos][1]) +
(PrevSample[0] * InterpCubic[samplepos][2]) +
(val * InterpCubic[samplepos][3])) >> 14;
break;
}
}
val <<= VolumeShift;
val *= Volume;
return val;