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Merge 8328b4e278 into 7fab935c2d

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Ziemas 2025-12-15 01:39:08 -05:00 committed by GitHub
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5 changed files with 111 additions and 147 deletions
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1
pcsx2/SPU2/Debug.cpp
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@ -200,7 +200,6 @@ void SPU2::DoFullDump()
fprintf(dump, " - Sound Start Address: %x\n", Cores[c].Voices[v].StartA); fprintf(dump, " - Sound Start Address: %x\n", Cores[c].Voices[v].StartA);
fprintf(dump, " - Next Data Address: %x\n", Cores[c].Voices[v].NextA); fprintf(dump, " - Next Data Address: %x\n", Cores[c].Voices[v].NextA);
fprintf(dump, " - Play Status: %s\n", (Cores[c].Voices[v].ADSR.Phase > 0) ? "Playing" : "Not Playing"); fprintf(dump, " - Play Status: %s\n", (Cores[c].Voices[v].ADSR.Phase > 0) ? "Playing" : "Not Playing");
fprintf(dump, " - Block Sample: %d\n", Cores[c].Voices[v].SCurrent);
} }
fprintf(dump, "#### END OF DUMP.\n\n"); fprintf(dump, "#### END OF DUMP.\n\n");
} }

213
pcsx2/SPU2/Mixer.cpp
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@ -89,55 +89,13 @@ int g_counter_cache_ignores = 0;
#define XAFLAG_LOOP (1ul << 1) #define XAFLAG_LOOP (1ul << 1)
#define XAFLAG_LOOP_START (1ul << 2) #define XAFLAG_LOOP_START (1ul << 2)
static __forceinline s32 GetNextDataBuffered(V_Core& thiscore, uint voiceidx) static __forceinline void GetNextDataBuffered(V_Core& thiscore, uint voiceidx)
{ {
V_Voice& vc(thiscore.Voices[voiceidx]); V_Voice& vc(thiscore.Voices[voiceidx]);
if ((vc.SCurrent & 3) == 0) if (vc.SBuffer == nullptr)
{ {
IncrementNextA(thiscore, voiceidx); const int cacheIdx = (vc.NextA & 0xFFFF8) / pcm_WordsPerBlock;
if ((vc.NextA & 7) == 0) // vc.SCurrent == 24 equivalent
{
if (vc.LoopFlags & XAFLAG_LOOP_END)
{
thiscore.Regs.ENDX |= (1 << voiceidx);
vc.NextA = vc.LoopStartA | 1;
if (!(vc.LoopFlags & XAFLAG_LOOP))
{
vc.Stop();
if (IsDevBuild)
{
if (SPU2::MsgVoiceOff())
SPU2::ConLog("* SPU2: Voice Off by EndPoint: %d \n", voiceidx);
}
}
}
else
vc.NextA++; // no, don't IncrementNextA here. We haven't read the header yet.
}
}
if (vc.SCurrent == 28)
{
vc.SCurrent = 0;
// We'll need the loop flags and buffer pointers regardless of cache status:
for (int i = 0; i < 2; i++)
if (Cores[i].IRQEnable && Cores[i].IRQA == (vc.NextA & 0xFFFF8))
SetIrqCall(i);
s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
vc.LoopFlags = *memptr >> 8; // grab loop flags from the upper byte.
if ((vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode)
{
vc.LoopStartA = vc.NextA & 0xFFFF8;
}
const int cacheIdx = vc.NextA / pcm_WordsPerBlock;
PcmCacheEntry& cacheLine = pcm_cache_data[cacheIdx]; PcmCacheEntry& cacheLine = pcm_cache_data[cacheIdx];
vc.SBuffer = cacheLine.Sampledata; vc.SBuffer = cacheLine.Sampledata;
@ -172,46 +130,18 @@ static __forceinline s32 GetNextDataBuffered(V_Core& thiscore, uint voiceidx)
g_counter_cache_misses++; g_counter_cache_misses++;
} }
s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
XA_decode_block(vc.SBuffer, memptr, vc.Prev1, vc.Prev2); XA_decode_block(vc.SBuffer, memptr, vc.Prev1, vc.Prev2);
} }
} }
return vc.SBuffer[vc.SCurrent++]; // Get the sample index for NextA, we have to subtract 1 to ignore the loop header
} int sampleIdx = ((vc.NextA % pcm_WordsPerBlock) - 1) * 4;
for (int i = 0; i < 4; i++)
static __forceinline void GetNextDataDummy(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
IncrementNextA(thiscore, voiceidx);
if ((vc.NextA & 7) == 0) // vc.SCurrent == 24 equivalent
{ {
if (vc.LoopFlags & XAFLAG_LOOP_END) vc.DecodeFifo[(vc.DecPosWrite + i) % 32] = vc.SBuffer[sampleIdx + i];
{
thiscore.Regs.ENDX |= (1 << voiceidx);
vc.NextA = vc.LoopStartA | 1;
}
else
vc.NextA++; // no, don't IncrementNextA here. We haven't read the header yet.
} }
if (vc.SCurrent == 28)
{
for (int i = 0; i < 2; i++)
if (Cores[i].IRQEnable && Cores[i].IRQA == (vc.NextA & 0xFFFF8))
SetIrqCall(i);
vc.LoopFlags = *GetMemPtr(vc.NextA & 0xFFFF8) >> 8; // grab loop flags from the upper byte.
if ((vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode)
vc.LoopStartA = vc.NextA & 0xFFFF8;
vc.SCurrent = 0;
}
vc.SP -= 0x1000 * (4 - (vc.SCurrent & 3));
vc.SCurrent += 4 - (vc.SCurrent & 3);
} }
///////////////////////////////////////////////////////////////////////////////////////// /////////////////////////////////////////////////////////////////////////////////////////
@ -237,6 +167,69 @@ static __forceinline StereoOut32 ApplyVolume(const StereoOut32& data, const V_Vo
ApplyVolume(data.Right, volume.Right.Value)); ApplyVolume(data.Right, volume.Right.Value));
} }
static __forceinline void UpdateBlockHeader(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
for (int i = 0; i < 2; i++)
if (Cores[i].IRQEnable && Cores[i].IRQA == (vc.NextA & 0xFFFF8))
SetIrqCall(i);
s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
vc.LoopFlags = *memptr >> 8; // grab loop flags from the upper byte.
if ((vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode)
{
vc.LoopStartA = vc.NextA & 0xFFFF8;
}
}
static __forceinline void DecodeSamples(uint coreidx, uint voiceidx)
{
V_Core& thiscore(Cores[coreidx]);
V_Voice& vc(thiscore.Voices[voiceidx]);
// Update the block header on every audio frame
UpdateBlockHeader(thiscore, voiceidx);
// When a voice is started at 0 pitch, NAX quickly advances to SSA + 5
// So that would mean the decode buffer holds around 12 samples
if (((int)(vc.DecPosWrite - vc.DecPosRead)) > 12) {
// Sufficient data buffered
return;
}
if (vc.ADSR.Phase > V_ADSR::PHASE_STOPPED)
{
GetNextDataBuffered(thiscore, voiceidx);
}
vc.DecPosWrite += 4;
IncrementNextA(thiscore, voiceidx);
if ((vc.NextA & 7) == 0)
{
if (vc.LoopFlags & XAFLAG_LOOP_END)
{
thiscore.Regs.ENDX |= (1 << voiceidx);
vc.NextA = vc.LoopStartA;
if (!(vc.LoopFlags & XAFLAG_LOOP))
{
vc.Stop();
if (IsDevBuild)
{
if (SPU2::MsgVoiceOff())
SPU2::ConLog("* SPU2: Voice Off by EndPoint: %d \n", voiceidx);
}
}
}
IncrementNextA(thiscore, voiceidx);
vc.SBuffer = nullptr;
}
}
static void __forceinline UpdatePitch(uint coreidx, uint voiceidx) static void __forceinline UpdatePitch(uint coreidx, uint voiceidx)
{ {
V_Voice& vc(Cores[coreidx].Voices[voiceidx]); V_Voice& vc(Cores[coreidx].Voices[voiceidx]);
@ -278,33 +271,27 @@ static __forceinline void CalculateADSR(V_Core& thiscore, uint voiceidx)
pxAssume(vc.ADSR.Value >= 0); // ADSR should never be negative... pxAssume(vc.ADSR.Value >= 0); // ADSR should never be negative...
} }
__forceinline static s32 GaussianInterpolate(s32 pv4, s32 pv3, s32 pv2, s32 pv1, s32 i) static __forceinline void ConsumeSamples(V_Core& thiscore, uint voiceidx)
{ {
s32 out = 0; V_Voice& vc(thiscore.Voices[voiceidx]);
out = (interpTable[i][0] * pv4) >> 15;
out += (interpTable[i][1] * pv3) >> 15;
out += (interpTable[i][2] * pv2) >> 15;
out += (interpTable[i][3] * pv1) >> 15;
return out; int consumed = vc.SP >> 12;
vc.SP &= 0xfff;
vc.DecPosRead += consumed;
} }
static __forceinline s32 GetVoiceValues(V_Core& thiscore, uint voiceidx) static __forceinline s32 GetVoiceValues(V_Core& thiscore, uint voiceidx)
{ {
V_Voice& vc(thiscore.Voices[voiceidx]); V_Voice& vc(thiscore.Voices[voiceidx]);
while (vc.SP >= 0) int phase = (vc.SP & 0x0ff0) >> 4;
{ s32 out = 0;
vc.PV4 = vc.PV3; out += (interpTable[phase][0] * vc.DecodeFifo[(vc.DecPosRead + 0) % 32]) >> 15;
vc.PV3 = vc.PV2; out += (interpTable[phase][1] * vc.DecodeFifo[(vc.DecPosRead + 1) % 32]) >> 15;
vc.PV2 = vc.PV1; out += (interpTable[phase][2] * vc.DecodeFifo[(vc.DecPosRead + 2) % 32]) >> 15;
vc.PV1 = GetNextDataBuffered(thiscore, voiceidx); out += (interpTable[phase][3] * vc.DecodeFifo[(vc.DecPosRead + 3) % 32]) >> 15;
vc.SP -= 0x1000;
}
const s32 mu = vc.SP + 0x1000; return out;
return GaussianInterpolate(vc.PV4, vc.PV3, vc.PV2, vc.PV1, (mu & 0x0ff0) >> 4);
} }
// This is Dr. Hell's noise algorithm as implemented in pcsxr // This is Dr. Hell's noise algorithm as implemented in pcsxr
@ -382,21 +369,13 @@ static __forceinline StereoOut32 MixVoice(uint coreidx, uint voiceidx)
V_Core& thiscore(Cores[coreidx]); V_Core& thiscore(Cores[coreidx]);
V_Voice& vc(thiscore.Voices[voiceidx]); V_Voice& vc(thiscore.Voices[voiceidx]);
// If this assertion fails, it mans SCurrent is being corrupted somewhere, or is not initialized
// properly. Invalid values in SCurrent will cause errant IRQs and corrupted audio.
pxAssertMsg((vc.SCurrent <= 28) && (vc.SCurrent != 0), "Current sample should always range from 1->28");
// Most games don't use much volume slide effects. So only call the UpdateVolume // 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... // methods when needed by checking the flag outside the method here...
// (Note: Ys 6 : Ark of Nephistm uses these effects) // (Note: Ys 6 : Ark of Nephistm uses these effects)
vc.Volume.Update(); vc.Volume.Update();
// SPU2 Note: The spu2 continues to process voices for eternity, always, so we DecodeSamples(coreidx, voiceidx);
// have to run through all the motions of updating the voice regardless of it's
// audible status. Otherwise IRQs might not trigger and emulation might fail.
UpdatePitch(coreidx, voiceidx);
StereoOut32 voiceOut(0, 0); StereoOut32 voiceOut(0, 0);
s32 Value = 0; s32 Value = 0;
@ -419,11 +398,14 @@ static __forceinline StereoOut32 MixVoice(uint coreidx, uint voiceidx)
voiceOut = ApplyVolume(StereoOut32(Value, Value), vc.Volume); voiceOut = ApplyVolume(StereoOut32(Value, Value), vc.Volume);
} }
else
{ // SPU2 Note: The spu2 continues to process voices for eternity, always, so we
while (vc.SP >= 0) // have to run through all the motions of updating the voice regardless of it's
GetNextDataDummy(thiscore, voiceidx); // Dummy is enough // audible status. Otherwise IRQs might not trigger and emulation might fail.
}
UpdatePitch(coreidx, voiceidx);
ConsumeSamples(thiscore, voiceidx);
// Write-back of raw voice data (post ADSR applied) // Write-back of raw voice data (post ADSR applied)
if (voiceidx == 1) if (voiceidx == 1)
@ -533,7 +515,8 @@ StereoOut32 V_Core::Mix(const VoiceMixSet& inVoices, const StereoOut32& Input, c
return TD + ApplyVolume(RV, FxVol); return TD + ApplyVolume(RV, FxVol);
} }
static StereoOut32 DCFilter(StereoOut32 input) { static StereoOut32 DCFilter(StereoOut32 input)
{
// A simple DC blocking high-pass filter // A simple DC blocking high-pass filter
// Implementation from http://peabody.sapp.org/class/dmp2/lab/dcblock/ // Implementation from http://peabody.sapp.org/class/dmp2/lab/dcblock/
// The magic number 0x7f5c is ceil(INT16_MAX * 0.995) // The magic number 0x7f5c is ceil(INT16_MAX * 0.995)
@ -634,9 +617,9 @@ __forceinline void spu2Mix()
if (SPU2::MsgCache()) if (SPU2::MsgCache())
{ {
SPU2::ConLog(" * SPU2 > CacheStats > Hits: %d Misses: %d Ignores: %d\n", SPU2::ConLog(" * SPU2 > CacheStats > Hits: %d Misses: %d Ignores: %d\n",
g_counter_cache_hits, g_counter_cache_hits,
g_counter_cache_misses, g_counter_cache_misses,
g_counter_cache_ignores); g_counter_cache_ignores);
} }
g_counter_cache_hits = g_counter_cache_hits =

21
pcsx2/SPU2/defs.h
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@ -256,29 +256,16 @@ struct V_Voice
// Sample pointer (19:12 bit fixed point) // Sample pointer (19:12 bit fixed point)
s32 SP; s32 SP;
// Sample pointer for Cubic Interpolation
// Cubic interpolation mixes a sample behind Linear, so that it
// can have sample data to either side of the end points from which
// to extrapolate. This SP represents that late sample position.
s32 SPc;
// Previous sample values - used for interpolation
// Inverted order of these members to match the access order in the
// code (might improve cache hits).
s32 PV4;
s32 PV3;
s32 PV2;
s32 PV1;
// Last outputted audio value, used for voice modulation. // Last outputted audio value, used for voice modulation.
s32 OutX; s32 OutX;
s32 NextCrest; // temp value for Crest calculation
// SBuffer now points directly to an ADPCM cache entry. // SBuffer now points directly to an ADPCM cache entry.
s16* SBuffer; s16* SBuffer;
// sample position within the current decoded packet. // Each voice has a buffer of decoded samples
s32 SCurrent; s32 DecodeFifo[32];
u32 DecPosWrite;
u32 DecPosRead;
// it takes a few ticks for voices to start on the real SPU2? // it takes a few ticks for voices to start on the real SPU2?
void Start(); void Start();

21
pcsx2/SPU2/spu2sys.cpp
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@ -181,7 +181,6 @@ void V_Core::Init(int index)
VoiceGates[v].WetR = -1; VoiceGates[v].WetR = -1;
Voices[v].Volume = V_VolumeSlideLR(0, 0); // V_VolumeSlideLR::Max; Voices[v].Volume = V_VolumeSlideLR(0, 0); // V_VolumeSlideLR::Max;
Voices[v].SCurrent = 28;
Voices[v].ADSR.Counter = 0; Voices[v].ADSR.Counter = 0;
Voices[v].ADSR.Value = 0; Voices[v].ADSR.Value = 0;
@ -190,6 +189,10 @@ void V_Core::Init(int index)
Voices[v].NextA = 0x2801; Voices[v].NextA = 0x2801;
Voices[v].StartA = 0x2800; Voices[v].StartA = 0x2800;
Voices[v].LoopStartA = 0x2800; Voices[v].LoopStartA = 0x2800;
memset(Voices[v].DecodeFifo, 0, sizeof(Voices[v].DecodeFifo));
Voices[v].DecPosRead = 0;
Voices[v].DecPosWrite = 0;
} }
DMAICounter = 0; DMAICounter = 0;
@ -212,23 +215,18 @@ void V_Voice::Start()
} }
ADSR.Attack(); ADSR.Attack();
SCurrent = 28;
LoopMode = 0; LoopMode = 0;
// When SP >= 0 the next sample will be grabbed, we don't want this to happen SP = 0;
// instantly because in the case of pitch being 0 we want to delay getting
// the next block header. This is a hack to work around the fact that unlike
// the HW we don't update the block header on every cycle.
SP = -1;
LoopFlags = 0; LoopFlags = 0;
NextA = StartA | 1; NextA = StartA | 1;
Prev1 = 0; Prev1 = 0;
Prev2 = 0; Prev2 = 0;
PV1 = PV2 = 0; SBuffer = nullptr;
PV3 = PV4 = 0; DecPosRead = 0;
NextCrest = -0x8000; DecPosWrite = 0;
} }
void V_Voice::Stop() void V_Voice::Stop()
@ -989,12 +987,10 @@ static void RegWrite_VoiceAddr(u16 value)
// Wallace And Gromit: Curse Of The Were-Rabbit. // Wallace And Gromit: Curse Of The Were-Rabbit.
thisvoice.NextA = ((u32)(value & 0x0F) << 16) | (thisvoice.NextA & 0xFFF8) | 1; thisvoice.NextA = ((u32)(value & 0x0F) << 16) | (thisvoice.NextA & 0xFFF8) | 1;
thisvoice.SCurrent = 28;
break; break;
case 5: case 5:
thisvoice.NextA = (thisvoice.NextA & 0x0F0000) | (value & 0xFFF8) | 1; thisvoice.NextA = (thisvoice.NextA & 0x0F0000) | (value & 0xFFF8) | 1;
thisvoice.SCurrent = 28;
break; break;
} }
} }
@ -1212,7 +1208,6 @@ static void RegWrite_Core(u16 value)
for (uint v = 0; v < 24; ++v) for (uint v = 0; v < 24; ++v)
{ {
Cores[1].Voices[v].Volume = V_VolumeSlideLR(0, 0); // V_VolumeSlideLR::Max; Cores[1].Voices[v].Volume = V_VolumeSlideLR(0, 0); // V_VolumeSlideLR::Max;
Cores[1].Voices[v].SCurrent = 28;
Cores[1].Voices[v].ADSR.Value = 0; Cores[1].Voices[v].ADSR.Value = 0;
Cores[1].Voices[v].ADSR.Phase = 0; Cores[1].Voices[v].ADSR.Phase = 0;

2
pcsx2/SaveState.h
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@ -26,7 +26,7 @@ enum class FreezeAction
// [SAVEVERSION+] // [SAVEVERSION+]
// This informs the auto updater that the users savestates will be invalidated. // This informs the auto updater that the users savestates will be invalidated.
static const u32 g_SaveVersion = (0x9A55 << 16) | 0x0000; static const u32 g_SaveVersion = (0x9A57 << 16) | 0x0000;
// the freezing data between submodules and core // the freezing data between submodules and core