mirror of
https://github.com/hrydgard/ppsspp.git
synced 2024-12-03 11:11:25 +00:00
281 lines
9.5 KiB
C++
281 lines
9.5 KiB
C++
// Copyright (c) 2015- PPSSPP Project.
|
|
|
|
// 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, version 2.0 or later versions.
|
|
|
|
// 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 2.0 for more details.
|
|
|
|
// A copy of the GPL 2.0 should have been included with the program.
|
|
// If not, see http://www.gnu.org/licenses/
|
|
|
|
// Official git repository and contact information can be found at
|
|
// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.
|
|
|
|
#include <cstdint>
|
|
#include <cstring>
|
|
|
|
#include "Common/Math/math_util.h"
|
|
#include "Core/Config.h"
|
|
#include "Core/HW/SasReverb.h"
|
|
#include "Core/Util/AudioFormat.h"
|
|
|
|
// This is under the assumption that the reverb used in Sas is the same as the PSX SPU reverb.
|
|
|
|
// Source: http://problemkaputt.de/psx-spx.htm#spureverbformula
|
|
|
|
struct SasReverbData {
|
|
const char *name;
|
|
int32_t size;
|
|
|
|
int16_t dAPF1;
|
|
int16_t dAPF2;
|
|
int16_t vIIR;
|
|
int16_t vCOMB1;
|
|
int16_t vCOMB2;
|
|
int16_t vCOMB3;
|
|
int16_t vCOMB4;
|
|
int16_t vWALL;
|
|
|
|
int16_t vAPF1;
|
|
int16_t vAPF2;
|
|
int16_t mLSAME;
|
|
int16_t mRSAME;
|
|
int16_t mLCOMB1;
|
|
int16_t mRCOMB1;
|
|
int16_t mLCOMB2;
|
|
int16_t mRCOMB2;
|
|
|
|
int16_t dLSAME;
|
|
int16_t dRSAME;
|
|
int16_t mLDIFF;
|
|
int16_t mRDIFF;
|
|
int16_t mLCOMB3;
|
|
int16_t mRCOMB3;
|
|
int16_t mLCOMB4;
|
|
int16_t mRCOMB4;
|
|
|
|
int16_t dLDIFF;
|
|
int16_t dRDIFF;
|
|
int16_t mLAPF1;
|
|
int16_t mRAPF1;
|
|
int16_t mLAPF2;
|
|
int16_t mRAPF2;
|
|
|
|
// These aren't used for anything else than 1.0 in any of the presets so let's drop them.
|
|
// int16_t vLIN;
|
|
// int16_t vRIN;
|
|
};
|
|
|
|
static const SasReverbData presets[10] = {
|
|
{
|
|
"Room",
|
|
0x26C0,
|
|
0x007D,0x005B,0x6D80,0x54B8,(int16_t)0xBED0,0x0000,0x0000,(int16_t)0xBA80,
|
|
0x5800,0x5300,0x04D6,0x0333,0x03F0,0x0227,0x0374,0x01EF,
|
|
0x0334,0x01B5,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
|
|
0x0000,0x0000,0x01B4,0x0136,0x00B8,0x005C, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
{
|
|
"Studio Small",
|
|
0x1F40,
|
|
0x0033,0x0025,0x70F0,0x4FA8,(int16_t)0xBCE0,0x4410,(int16_t)0xC0F0,(int16_t)0x9C00,
|
|
0x5280,0x4EC0,0x03E4,0x031B,0x03A4,0x02AF,0x0372,0x0266,
|
|
0x031C,0x025D,0x025C,0x018E,0x022F,0x0135,0x01D2,0x00B7,
|
|
0x018F,0x00B5,0x00B4,0x0080,0x004C,0x0026, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
{
|
|
"Studio Medium",
|
|
0x4840,
|
|
0x00B1,0x007F,0x70F0,0x4FA8,(int16_t)0xBCE0,0x4510,(int16_t)0xBEF0,(int16_t)0xB4C0,
|
|
0x5280,0x4EC0,0x0904,0x076B,0x0824,0x065F,0x07A2,0x0616,
|
|
0x076C,0x05ED,0x05EC,0x042E,0x050F,0x0305,0x0462,0x02B7,
|
|
0x042F,0x0265,0x0264,0x01B2,0x0100,0x0080, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
// Studio Large(size = 6FE0h bytes)
|
|
{
|
|
"Studio Large",
|
|
0x6FE0,
|
|
0x00E3,0x00A9,0x6F60,0x4FA8,(int16_t)0xBCE0,0x4510,(int16_t)0xBEF0,(int16_t)0xA680,
|
|
0x5680,0x52C0,0x0DFB,0x0B58,0x0D09,0x0A3C,0x0BD9,0x0973,
|
|
0x0B59,0x08DA,0x08D9,0x05E9,0x07EC,0x04B0,0x06EF,0x03D2,
|
|
0x05EA,0x031D,0x031C,0x0238,0x0154,0x00AA, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
{
|
|
"Hall",
|
|
0xADE0,
|
|
0x01A5,0x0139,0x6000,0x5000,0x4C00,(int16_t)0xB800,(int16_t)0xBC00,(int16_t)0xC000,
|
|
0x6000,0x5C00,0x15BA,0x11BB,0x14C2,0x10BD,0x11BC,0x0DC1,
|
|
0x11C0,0x0DC3,0x0DC0,0x09C1,0x0BC4,0x07C1,0x0A00,0x06CD,
|
|
0x09C2,0x05C1,0x05C0,0x041A,0x0274,0x013A, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
{
|
|
"Space Echo",
|
|
0xF6C0,
|
|
0x033D,0x0231,0x7E00,0x5000,(int16_t)0xB400,(int16_t)0xB000,0x4C00,(int16_t)0xB000,
|
|
0x6000,0x5400,0x1ED6,0x1A31,0x1D14,0x183B,0x1BC2,0x16B2,
|
|
0x1A32,0x15EF,0x15EE,0x1055,0x1334,0x0F2D,0x11F6,0x0C5D,
|
|
0x1056,0x0AE1,0x0AE0,0x07A2,0x0464,0x0232, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
{
|
|
"Echo (almost infinite)",
|
|
0x18040,
|
|
0x0001,0x0001,0x7FFF,0x7FFF,0x0000,0x0000,0x0000,(int16_t)0xC080,
|
|
0x0000,0x0000,0x1FFF,0x0FFF,0x1005,0x0005,0x0000,0x0000,
|
|
0x1005,0x0005,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
|
|
0x0000,0x0000,0x1004,0x1002,0x0004,0x0002, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
{
|
|
"Delay (one - shot echo)",
|
|
0x18040,
|
|
0x0001,0x0001,0x7FFF,0x7FFF,0x0000,0x0000,0x0000,0x0000,
|
|
0x0000,0x0000,0x1FFF,0x0FFF,0x1005,0x0005,0x0000,0x0000,
|
|
0x1005,0x0005,0x0000,0x0000,0x0000,0x0000,0x0000,0x0000,
|
|
0x0000,0x0000,0x1004,0x1002,0x0004,0x0002, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
|
|
{
|
|
"Half Echo",
|
|
0x3C00,
|
|
0x0017,0x0013,0x70F0,0x4FA8,(int16_t)0xBCE0,0x4510,(int16_t)0xBEF0,(int16_t)0x8500,
|
|
0x5F80,0x54C0,0x0371,0x02AF,0x02E5,0x01DF,0x02B0,0x01D7,
|
|
0x0358,0x026A,0x01D6,0x011E,0x012D,0x00B1,0x011F,0x0059,
|
|
0x01A0,0x00E3,0x0058,0x0040,0x0028,0x0014, //(int16_t)0x8000,(int16_t)0x8000,
|
|
},
|
|
};
|
|
|
|
SasReverb::SasReverb() : preset_(-1), pos_(0) {
|
|
workspace_ = new int16_t[BUFSIZE];
|
|
}
|
|
|
|
SasReverb::~SasReverb() {
|
|
delete[] workspace_;
|
|
}
|
|
|
|
const char *SasReverb::GetPresetName(int preset) {
|
|
if (preset == -1) {
|
|
return "Off";
|
|
}
|
|
return presets[preset].name;
|
|
}
|
|
|
|
void SasReverb::SetPreset(int preset) {
|
|
if (preset < (int)ARRAY_SIZE(presets))
|
|
preset_ = preset;
|
|
if (preset_ != -1) {
|
|
pos_ = BUFSIZE - presets[preset_].size;
|
|
memset(workspace_, 0, sizeof(int16_t) * BUFSIZE);
|
|
} else {
|
|
pos_ = 0;
|
|
}
|
|
}
|
|
|
|
// Wraps around the upper part of a buffer.
|
|
template<int bufsize>
|
|
class BufferWrapper {
|
|
public:
|
|
BufferWrapper(int16_t *buffer, int position, int usedSize) : buf_(buffer), pos_(position), end_(bufsize), base_(bufsize - usedSize), size_(usedSize) {}
|
|
int16_t &operator [](int index) {
|
|
int addr = pos_ + index;
|
|
if (addr >= end_) { addr -= size_; }
|
|
if (addr < base_) { addr += size_; }
|
|
return buf_[addr];
|
|
}
|
|
|
|
int GetPosition() { return pos_; }
|
|
void Next() {
|
|
pos_++;
|
|
if (pos_ >= end_) {
|
|
pos_ -= size_;
|
|
}
|
|
}
|
|
|
|
private:
|
|
int16_t *buf_;
|
|
int pos_;
|
|
int end_;
|
|
int base_;
|
|
int size_;
|
|
};
|
|
|
|
void SasReverb::ProcessReverb(int16_t *output, const int16_t *input, size_t inputSize, uint16_t volLeft, uint16_t volRight) {
|
|
// This means replicate the input signal in the processed buffer.
|
|
// Can also be used to verify that the error is in here...
|
|
if (preset_ == -1) {
|
|
// Strangely, OFF is not filled with zeroes every other. Seems special cased.
|
|
for (size_t i = 0; i < inputSize; ++i) {
|
|
output[i * 4 + 0] = clamp_s16((int)input[i * 2 + 0] * volLeft >> 15);
|
|
output[i * 4 + 1] = clamp_s16((int)input[i * 2 + 1] * volRight >> 15);
|
|
output[i * 4 + 2] = clamp_s16((int)input[i * 2 + 0] * volLeft >> 15);
|
|
output[i * 4 + 3] = clamp_s16((int)input[i * 2 + 1] * volRight >> 15);
|
|
}
|
|
return;
|
|
}
|
|
|
|
const uint8_t reverbVolume = Clamp(g_Config.iReverbVolume, 0, 25);
|
|
// Standard volume is 10, which pairs with a normal shift of 15.
|
|
const uint8_t finalShift = 25 - reverbVolume;
|
|
if (reverbVolume == 0) {
|
|
// Force to zero output, which is not the same as "Off."
|
|
memset(output, 0, inputSize * 4);
|
|
return;
|
|
}
|
|
|
|
const SasReverbData &d = presets[preset_];
|
|
|
|
// We put this on the stack instead of in the object to let the compiler optimize better (avoid mem r/w).
|
|
BufferWrapper<BUFSIZE> b(workspace_, pos_, d.size);
|
|
|
|
// This runs at 22khz.
|
|
// Very unoptimized, straight from the description. Can probably be reformulated into something way more efficient.
|
|
// Or we could actually template the whole thing with the parameters as template arguments, as the presets are fixed.
|
|
for (size_t i = 0; i < inputSize; i++) {
|
|
// Dividing by two here is an incorrect hack. Some multiplication factor is needed to prevent the reverb from getting too loud, though.
|
|
int16_t LeftInput = input[i * 2] >> 1;
|
|
int16_t RightInput = input[i * 2 + 1] >> 1;
|
|
|
|
int16_t Lin = LeftInput; // (d.vLIN * LeftInput) >> 15;
|
|
int16_t Rin = RightInput; // (d.vRIN * RightInput) >> 15;
|
|
|
|
// ____Same Side Reflection(left - to - left and right - to - right)___________________
|
|
b[d.mLSAME] = clamp_s16(Lin + (b[d.dLSAME] * d.vWALL >> 15) - (b[d.mLSAME - 1]*d.vIIR >> 15) + b[d.mLSAME - 1]); // L - to - L
|
|
b[d.mRSAME] = clamp_s16(Rin + (b[d.dRSAME] * d.vWALL >> 15) - (b[d.mRSAME - 1]*d.vIIR >> 15) + b[d.mRSAME - 1]); // R - to - R
|
|
// ___Different Side Reflection(left - to - right and right - to - left)_______________
|
|
b[d.mLDIFF] = clamp_s16(Lin + (b[d.dRDIFF] * d.vWALL >> 15) - (b[d.mLDIFF - 1]*d.vIIR >> 15) + b[d.mLDIFF - 1]); // R - to - L
|
|
b[d.mRDIFF] = clamp_s16(Rin + (b[d.dLDIFF] * d.vWALL >> 15) - (b[d.mRDIFF - 1]*d.vIIR >> 15) + b[d.mRDIFF - 1]); // L - to - R
|
|
// ___Early Echo(Comb Filter, with input from buffer)__________________________
|
|
int32_t Lout = ((d.vCOMB1*b[d.mLCOMB1] + d.vCOMB2*b[d.mLCOMB2] + d.vCOMB3*b[d.mLCOMB3] + d.vCOMB4*b[d.mLCOMB4]) >> 15);
|
|
int32_t Rout = ((d.vCOMB1*b[d.mRCOMB1] + d.vCOMB2*b[d.mRCOMB2] + d.vCOMB3*b[d.mRCOMB3] + d.vCOMB4*b[d.mRCOMB4]) >> 15);
|
|
// ___Late Reverb APF1(All Pass Filter 1, with input from COMB)________________
|
|
b[d.mLAPF1] = clamp_s16(Lout - (d.vAPF1*b[(d.mLAPF1 - d.dAPF1)] >> 15));
|
|
Lout = b[(d.mLAPF1 - d.dAPF1)] + (b[d.mLAPF1] * d.vAPF1 >> 15);
|
|
b[d.mRAPF1] = clamp_s16(Rout - (d.vAPF1*b[(d.mRAPF1 - d.dAPF1)] >> 15));
|
|
Rout = b[(d.mRAPF1 - d.dAPF1)] + (b[d.mRAPF1] * d.vAPF1 >> 15);
|
|
// ___Late Reverb APF2(All Pass Filter 2, with input from APF1)________________
|
|
b[d.mLAPF2] = clamp_s16(Lout - (d.vAPF2*b[(d.mLAPF2 - d.dAPF2)] >> 15));
|
|
Lout = b[(d.mLAPF2 - d.dAPF2)] + (b[d.mLAPF2] * d.vAPF2 >> 15);
|
|
b[d.mRAPF2] = clamp_s16(Rout - (d.vAPF2*b[(d.mRAPF2 - d.dAPF2)] >> 15));
|
|
Rout = b[(d.mRAPF2 - d.dAPF2)] + (b[d.mRAPF2] * d.vAPF2 >> 15);
|
|
// ___Output to Mixer(Output volume multiplied with input from APF2)___________
|
|
output[i * 4 + 0] = clamp_s16((Lout * volLeft) >> finalShift);
|
|
output[i * 4 + 1] = clamp_s16((Rout * volRight) >> finalShift);
|
|
output[i * 4 + 2] = 0;
|
|
output[i * 4 + 3] = 0;
|
|
|
|
b.Next();
|
|
}
|
|
|
|
// Save the state in the object.
|
|
pos_ = b.GetPosition();
|
|
}
|
|
|