mirror of
https://github.com/PCSX2/pcsx2.git
synced 2025-12-16 04:08:48 +00:00
Merge 8328b4e278 into 7fab935c2d
This commit is contained in:
Ziemas
GitHub
commit
09715b10e0
@ -200,7 +200,6 @@ void SPU2::DoFullDump()
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fprintf(dump, " - Sound Start Address: %x\n", Cores[c].Voices[v].StartA);
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fprintf(dump, " - Next Data Address: %x\n", Cores[c].Voices[v].NextA);
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fprintf(dump, " - Play Status: %s\n", (Cores[c].Voices[v].ADSR.Phase > 0) ? "Playing" : "Not Playing");
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fprintf(dump, " - Block Sample: %d\n", Cores[c].Voices[v].SCurrent);
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}
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fprintf(dump, "#### END OF DUMP.\n\n");
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}
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@ -89,55 +89,13 @@ int g_counter_cache_ignores = 0;
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#define XAFLAG_LOOP (1ul << 1)
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#define XAFLAG_LOOP_START (1ul << 2)
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static __forceinline s32 GetNextDataBuffered(V_Core& thiscore, uint voiceidx)
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static __forceinline void GetNextDataBuffered(V_Core& thiscore, uint voiceidx)
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{
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V_Voice& vc(thiscore.Voices[voiceidx]);
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if ((vc.SCurrent & 3) == 0)
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if (vc.SBuffer == nullptr)
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{
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IncrementNextA(thiscore, voiceidx);
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if ((vc.NextA & 7) == 0) // vc.SCurrent == 24 equivalent
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{
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if (vc.LoopFlags & XAFLAG_LOOP_END)
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{
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thiscore.Regs.ENDX |= (1 << voiceidx);
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vc.NextA = vc.LoopStartA | 1;
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if (!(vc.LoopFlags & XAFLAG_LOOP))
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{
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vc.Stop();
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if (IsDevBuild)
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{
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if (SPU2::MsgVoiceOff())
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SPU2::ConLog("* SPU2: Voice Off by EndPoint: %d \n", voiceidx);
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}
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}
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}
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else
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vc.NextA++; // no, don't IncrementNextA here. We haven't read the header yet.
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}
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}
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if (vc.SCurrent == 28)
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{
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vc.SCurrent = 0;
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// We'll need the loop flags and buffer pointers regardless of cache status:
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for (int i = 0; i < 2; i++)
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if (Cores[i].IRQEnable && Cores[i].IRQA == (vc.NextA & 0xFFFF8))
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SetIrqCall(i);
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s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
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vc.LoopFlags = *memptr >> 8; // grab loop flags from the upper byte.
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if ((vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode)
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{
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vc.LoopStartA = vc.NextA & 0xFFFF8;
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}
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const int cacheIdx = vc.NextA / pcm_WordsPerBlock;
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const int cacheIdx = (vc.NextA & 0xFFFF8) / pcm_WordsPerBlock;
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PcmCacheEntry& cacheLine = pcm_cache_data[cacheIdx];
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vc.SBuffer = cacheLine.Sampledata;
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@ -172,46 +130,18 @@ static __forceinline s32 GetNextDataBuffered(V_Core& thiscore, uint voiceidx)
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g_counter_cache_misses++;
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}
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s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
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XA_decode_block(vc.SBuffer, memptr, vc.Prev1, vc.Prev2);
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}
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}
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return vc.SBuffer[vc.SCurrent++];
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}
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static __forceinline void GetNextDataDummy(V_Core& thiscore, uint voiceidx)
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{
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V_Voice& vc(thiscore.Voices[voiceidx]);
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IncrementNextA(thiscore, voiceidx);
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if ((vc.NextA & 7) == 0) // vc.SCurrent == 24 equivalent
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// Get the sample index for NextA, we have to subtract 1 to ignore the loop header
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int sampleIdx = ((vc.NextA % pcm_WordsPerBlock) - 1) * 4;
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for (int i = 0; i < 4; i++)
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{
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if (vc.LoopFlags & XAFLAG_LOOP_END)
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{
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thiscore.Regs.ENDX |= (1 << voiceidx);
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vc.NextA = vc.LoopStartA | 1;
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vc.DecodeFifo[(vc.DecPosWrite + i) % 32] = vc.SBuffer[sampleIdx + i];
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}
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else
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vc.NextA++; // no, don't IncrementNextA here. We haven't read the header yet.
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}
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if (vc.SCurrent == 28)
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{
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for (int i = 0; i < 2; i++)
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if (Cores[i].IRQEnable && Cores[i].IRQA == (vc.NextA & 0xFFFF8))
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SetIrqCall(i);
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vc.LoopFlags = *GetMemPtr(vc.NextA & 0xFFFF8) >> 8; // grab loop flags from the upper byte.
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if ((vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode)
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vc.LoopStartA = vc.NextA & 0xFFFF8;
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vc.SCurrent = 0;
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}
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vc.SP -= 0x1000 * (4 - (vc.SCurrent & 3));
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vc.SCurrent += 4 - (vc.SCurrent & 3);
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}
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/////////////////////////////////////////////////////////////////////////////////////////
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@ -237,6 +167,69 @@ static __forceinline StereoOut32 ApplyVolume(const StereoOut32& data, const V_Vo
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ApplyVolume(data.Right, volume.Right.Value));
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}
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static __forceinline void UpdateBlockHeader(V_Core& thiscore, uint voiceidx)
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{
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V_Voice& vc(thiscore.Voices[voiceidx]);
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for (int i = 0; i < 2; i++)
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if (Cores[i].IRQEnable && Cores[i].IRQA == (vc.NextA & 0xFFFF8))
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SetIrqCall(i);
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s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
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vc.LoopFlags = *memptr >> 8; // grab loop flags from the upper byte.
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if ((vc.LoopFlags & XAFLAG_LOOP_START) && !vc.LoopMode)
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{
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vc.LoopStartA = vc.NextA & 0xFFFF8;
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}
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}
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static __forceinline void DecodeSamples(uint coreidx, uint voiceidx)
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{
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V_Core& thiscore(Cores[coreidx]);
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V_Voice& vc(thiscore.Voices[voiceidx]);
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// Update the block header on every audio frame
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UpdateBlockHeader(thiscore, voiceidx);
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// When a voice is started at 0 pitch, NAX quickly advances to SSA + 5
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// So that would mean the decode buffer holds around 12 samples
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if (((int)(vc.DecPosWrite - vc.DecPosRead)) > 12) {
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// Sufficient data buffered
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return;
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}
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if (vc.ADSR.Phase > V_ADSR::PHASE_STOPPED)
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{
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GetNextDataBuffered(thiscore, voiceidx);
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}
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vc.DecPosWrite += 4;
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IncrementNextA(thiscore, voiceidx);
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if ((vc.NextA & 7) == 0)
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{
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if (vc.LoopFlags & XAFLAG_LOOP_END)
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{
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thiscore.Regs.ENDX |= (1 << voiceidx);
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vc.NextA = vc.LoopStartA;
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if (!(vc.LoopFlags & XAFLAG_LOOP))
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{
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vc.Stop();
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if (IsDevBuild)
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{
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if (SPU2::MsgVoiceOff())
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SPU2::ConLog("* SPU2: Voice Off by EndPoint: %d \n", voiceidx);
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}
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}
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}
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IncrementNextA(thiscore, voiceidx);
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vc.SBuffer = nullptr;
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}
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}
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static void __forceinline UpdatePitch(uint coreidx, uint voiceidx)
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{
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V_Voice& vc(Cores[coreidx].Voices[voiceidx]);
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@ -278,33 +271,27 @@ static __forceinline void CalculateADSR(V_Core& thiscore, uint voiceidx)
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pxAssume(vc.ADSR.Value >= 0); // ADSR should never be negative...
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}
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__forceinline static s32 GaussianInterpolate(s32 pv4, s32 pv3, s32 pv2, s32 pv1, s32 i)
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static __forceinline void ConsumeSamples(V_Core& thiscore, uint voiceidx)
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{
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s32 out = 0;
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out = (interpTable[i][0] * pv4) >> 15;
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out += (interpTable[i][1] * pv3) >> 15;
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out += (interpTable[i][2] * pv2) >> 15;
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out += (interpTable[i][3] * pv1) >> 15;
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V_Voice& vc(thiscore.Voices[voiceidx]);
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return out;
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int consumed = vc.SP >> 12;
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vc.SP &= 0xfff;
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vc.DecPosRead += consumed;
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}
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static __forceinline s32 GetVoiceValues(V_Core& thiscore, uint voiceidx)
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{
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V_Voice& vc(thiscore.Voices[voiceidx]);
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while (vc.SP >= 0)
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{
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vc.PV4 = vc.PV3;
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vc.PV3 = vc.PV2;
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vc.PV2 = vc.PV1;
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vc.PV1 = GetNextDataBuffered(thiscore, voiceidx);
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vc.SP -= 0x1000;
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}
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int phase = (vc.SP & 0x0ff0) >> 4;
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s32 out = 0;
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out += (interpTable[phase][0] * vc.DecodeFifo[(vc.DecPosRead + 0) % 32]) >> 15;
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out += (interpTable[phase][1] * vc.DecodeFifo[(vc.DecPosRead + 1) % 32]) >> 15;
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out += (interpTable[phase][2] * vc.DecodeFifo[(vc.DecPosRead + 2) % 32]) >> 15;
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out += (interpTable[phase][3] * vc.DecodeFifo[(vc.DecPosRead + 3) % 32]) >> 15;
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const s32 mu = vc.SP + 0x1000;
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return GaussianInterpolate(vc.PV4, vc.PV3, vc.PV2, vc.PV1, (mu & 0x0ff0) >> 4);
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return out;
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}
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// This is Dr. Hell's noise algorithm as implemented in pcsxr
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@ -382,21 +369,13 @@ static __forceinline StereoOut32 MixVoice(uint coreidx, uint voiceidx)
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V_Core& thiscore(Cores[coreidx]);
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V_Voice& vc(thiscore.Voices[voiceidx]);
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// If this assertion fails, it mans SCurrent is being corrupted somewhere, or is not initialized
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// properly. Invalid values in SCurrent will cause errant IRQs and corrupted audio.
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pxAssertMsg((vc.SCurrent <= 28) && (vc.SCurrent != 0), "Current sample should always range from 1->28");
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// Most games don't use much volume slide effects. So only call the UpdateVolume
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// methods when needed by checking the flag outside the method here...
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// (Note: Ys 6 : Ark of Nephistm uses these effects)
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vc.Volume.Update();
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// SPU2 Note: The spu2 continues to process voices for eternity, always, so we
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// have to run through all the motions of updating the voice regardless of it's
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// audible status. Otherwise IRQs might not trigger and emulation might fail.
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UpdatePitch(coreidx, voiceidx);
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DecodeSamples(coreidx, voiceidx);
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StereoOut32 voiceOut(0, 0);
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s32 Value = 0;
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@ -419,11 +398,14 @@ static __forceinline StereoOut32 MixVoice(uint coreidx, uint voiceidx)
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voiceOut = ApplyVolume(StereoOut32(Value, Value), vc.Volume);
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}
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else
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{
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while (vc.SP >= 0)
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GetNextDataDummy(thiscore, voiceidx); // Dummy is enough
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}
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// SPU2 Note: The spu2 continues to process voices for eternity, always, so we
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// have to run through all the motions of updating the voice regardless of it's
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// audible status. Otherwise IRQs might not trigger and emulation might fail.
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UpdatePitch(coreidx, voiceidx);
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ConsumeSamples(thiscore, voiceidx);
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// Write-back of raw voice data (post ADSR applied)
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if (voiceidx == 1)
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@ -533,7 +515,8 @@ StereoOut32 V_Core::Mix(const VoiceMixSet& inVoices, const StereoOut32& Input, c
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return TD + ApplyVolume(RV, FxVol);
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}
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static StereoOut32 DCFilter(StereoOut32 input) {
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static StereoOut32 DCFilter(StereoOut32 input)
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{
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// A simple DC blocking high-pass filter
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// Implementation from http://peabody.sapp.org/class/dmp2/lab/dcblock/
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// The magic number 0x7f5c is ceil(INT16_MAX * 0.995)
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@ -256,29 +256,16 @@ struct V_Voice
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// Sample pointer (19:12 bit fixed point)
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s32 SP;
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// Sample pointer for Cubic Interpolation
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// Cubic interpolation mixes a sample behind Linear, so that it
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// can have sample data to either side of the end points from which
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// to extrapolate. This SP represents that late sample position.
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s32 SPc;
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// Previous sample values - used for interpolation
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// Inverted order of these members to match the access order in the
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// code (might improve cache hits).
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s32 PV4;
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s32 PV3;
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s32 PV2;
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s32 PV1;
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// Last outputted audio value, used for voice modulation.
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s32 OutX;
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s32 NextCrest; // temp value for Crest calculation
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// SBuffer now points directly to an ADPCM cache entry.
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s16* SBuffer;
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// sample position within the current decoded packet.
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s32 SCurrent;
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// Each voice has a buffer of decoded samples
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s32 DecodeFifo[32];
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u32 DecPosWrite;
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u32 DecPosRead;
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// it takes a few ticks for voices to start on the real SPU2?
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void Start();
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@ -181,7 +181,6 @@ void V_Core::Init(int index)
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VoiceGates[v].WetR = -1;
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Voices[v].Volume = V_VolumeSlideLR(0, 0); // V_VolumeSlideLR::Max;
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Voices[v].SCurrent = 28;
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Voices[v].ADSR.Counter = 0;
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Voices[v].ADSR.Value = 0;
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@ -190,6 +189,10 @@ void V_Core::Init(int index)
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Voices[v].NextA = 0x2801;
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Voices[v].StartA = 0x2800;
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Voices[v].LoopStartA = 0x2800;
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memset(Voices[v].DecodeFifo, 0, sizeof(Voices[v].DecodeFifo));
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Voices[v].DecPosRead = 0;
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Voices[v].DecPosWrite = 0;
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}
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DMAICounter = 0;
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@ -212,23 +215,18 @@ void V_Voice::Start()
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}
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ADSR.Attack();
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SCurrent = 28;
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LoopMode = 0;
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// When SP >= 0 the next sample will be grabbed, we don't want this to happen
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// instantly because in the case of pitch being 0 we want to delay getting
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// the next block header. This is a hack to work around the fact that unlike
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// the HW we don't update the block header on every cycle.
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SP = -1;
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SP = 0;
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LoopFlags = 0;
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NextA = StartA | 1;
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Prev1 = 0;
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Prev2 = 0;
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PV1 = PV2 = 0;
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PV3 = PV4 = 0;
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NextCrest = -0x8000;
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SBuffer = nullptr;
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DecPosRead = 0;
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DecPosWrite = 0;
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}
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void V_Voice::Stop()
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@ -989,12 +987,10 @@ static void RegWrite_VoiceAddr(u16 value)
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// Wallace And Gromit: Curse Of The Were-Rabbit.
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thisvoice.NextA = ((u32)(value & 0x0F) << 16) | (thisvoice.NextA & 0xFFF8) | 1;
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thisvoice.SCurrent = 28;
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break;
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case 5:
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thisvoice.NextA = (thisvoice.NextA & 0x0F0000) | (value & 0xFFF8) | 1;
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thisvoice.SCurrent = 28;
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break;
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}
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}
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@ -1212,7 +1208,6 @@ static void RegWrite_Core(u16 value)
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for (uint v = 0; v < 24; ++v)
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{
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Cores[1].Voices[v].Volume = V_VolumeSlideLR(0, 0); // V_VolumeSlideLR::Max;
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Cores[1].Voices[v].SCurrent = 28;
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Cores[1].Voices[v].ADSR.Value = 0;
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Cores[1].Voices[v].ADSR.Phase = 0;
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@ -26,7 +26,7 @@ enum class FreezeAction
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// [SAVEVERSION+]
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// This informs the auto updater that the users savestates will be invalidated.
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static const u32 g_SaveVersion = (0x9A55 << 16) | 0x0000;
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static const u32 g_SaveVersion = (0x9A57 << 16) | 0x0000;
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// the freezing data between submodules and core
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