Files
jpolo1224 90daa091db Android: audio backend options, setting descriptions, RA/haptics polish, and ported GS fixes
Audio
- Optional OpenSL ES output backend for devices where the default AAudio path
  crackles, glitches or won't initialise (Settings -> Audio), plus a lightweight
  SPU2 mode that skips the reverb pipeline to free CPU on low-end devices.
- Keep the audio device alive across the in-game menu pause so Android no longer
  reclaims the idle stream and drops sound after the menu sits open (#333).

Settings
- Restored the per-setting descriptions under every GameDB Fix and Advanced
  Speedhack toggle (lost in the settings redesign).
- Per-game Reset now clears the native per-game INI, so it truly reverts to the
  global values instead of the game keeping stale overrides.
- On-screen display now defaults off; Custom stats appear on boot without a
  reset (#385).

Controls / RetroAchievements
- Vibration Strength slider scaling all rumble and touch haptics 0-200%.
- Achievement Sound Volume slider; points now show in the menu before a game
  loads; unlock sounds play with Do Not Disturb enabled.

Misc
- Drop the compiled GS shader/pipeline cache automatically on app update to
  avoid post-update graphical corruption.
- Animated XMB library-background fallback for GPUs without float-texture
  filtering.

GS correctness (ported from sashkinbro/EmuCoreX)
- Reset per-game hardware-hack HLE state on game change (Burnout bloom,
  IRem/GT channel-shuffle) so it no longer leaks across in-app game switches.
- Fix a non-strict-weak-ordering comparator in SortMultiStretchRects.
- Free the leaked m_expand_vao on the OpenGL device teardown path.
2026-07-21 20:42:35 -04:00

686 lines
20 KiB
C++

// SPDX-FileCopyrightText: 2002-2026 PCSX2 Dev Team
// SPDX-License-Identifier: GPL-3.0+
#include "Host/AudioStream.h"
#include "SPU2/Debug.h"
#include "SPU2/defs.h"
#include "SPU2/spu2.h"
#include "SPU2/interpolate_table.h"
#include "common/Assertions.h"
#if defined(__aarch64__)
#include <arm_neon.h>
#endif
// LOOP/END sets the ENDX bit and sets NAX to LSA, and the voice is muted if LOOP is not set
// LOOP seems to only have any effect on the block with LOOP/END set, where it prevents muting the voice
// (the documented requirement that every block in a loop has the LOOP bit set is nonsense according to tests)
// LOOP/START sets LSA to NAX unless LSA was written manually since sound generation started
// (see LoopMode, the method by which this is achieved on the real SPU2 is unknown)
#define XAFLAG_LOOP_END (1ul << 0)
#define XAFLAG_LOOP (1ul << 1)
#define XAFLAG_LOOP_START (1ul << 2)
#if MULTI_ISA_COMPILE_ONCE
// decoded pcm data, used to cache the decoded data so that it needn't be decoded
// multiple times. Cache chunks are decoded when the mixer requests the blocks, and
// invalided when DMA transfers and memory writes are performed.
PcmCacheEntry pcm_cache_data[pcm_BlockCount];
int g_counter_cache_hits = 0;
int g_counter_cache_misses = 0;
int g_counter_cache_ignores = 0;
#endif
MULTI_ISA_UNSHARED_START
static const s32 tbl_XA_Factor[16][2] =
{
{0, 0},
{60, 0},
{115, -52},
{98, -55},
{122, -60}};
static void __forceinline XA_decode_block(s16* buffer, const s16* block, s32& prev1, s32& prev2)
{
const s32 header = *block;
const s32 shift = (header & 0xF) + 16;
const int id = header >> 4 & 0xF;
if (id > 4 && SPU2::MsgToConsole())
SPU2::ConLog("* SPU2: Unknown ADPCM coefficients table id %d\n", id);
const s32 pred1 = tbl_XA_Factor[id][0];
const s32 pred2 = tbl_XA_Factor[id][1];
const s8* blockbytes = (s8*)&block[1];
const s8* blockend = &blockbytes[13];
for (; blockbytes <= blockend; ++blockbytes)
{
s32 data = ((*blockbytes) << 28) & 0xF0000000;
s32 pcm = (data >> shift) + (((pred1 * prev1) + (pred2 * prev2) + 32) >> 6);
pcm = std::clamp<s32>(pcm, -0x8000, 0x7fff);
*(buffer++) = pcm;
data = ((*blockbytes) << 24) & 0xF0000000;
s32 pcm2 = (data >> shift) + (((pred1 * pcm) + (pred2 * prev1) + 32) >> 6);
pcm2 = std::clamp<s32>(pcm2, -0x8000, 0x7fff);
*(buffer++) = pcm2;
prev2 = pcm;
prev1 = pcm2;
}
}
static void __forceinline IncrementNextA(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
// Important! Both cores signal IRQ when an address is read, regardless of
// which core actually reads the address.
for (int i = 0; i < 2; i++)
{
if (Cores[i].IRQEnable && (vc.NextA == Cores[i].IRQA))
{
//if( IsDevBuild )
// ConLog(" * SPU2 Core %d: IRQ Requested (IRQA (%05X) passed; voice %d).\n", i, Cores[i].IRQA, thiscore.Index * 24 + voiceidx);
SetIrqCall(i);
}
}
vc.NextA++;
vc.NextA &= 0xFFFFF;
}
static __forceinline void GetNextDataBuffered(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
if (vc.SBuffer == nullptr)
{
const int cacheIdx = (vc.NextA & 0xFFFF8) / pcm_WordsPerBlock;
PcmCacheEntry& cacheLine = pcm_cache_data[cacheIdx];
vc.SBuffer = cacheLine.Sampledata;
if (cacheLine.Validated && vc.Prev1 == cacheLine.Prev1 && vc.Prev2 == cacheLine.Prev2)
{
// Cached block! Read from the cache directly.
// Make sure to propagate the prev1/prev2 ADPCM:
vc.Prev1 = vc.SBuffer[27];
vc.Prev2 = vc.SBuffer[26];
//ConLog( "* SPU2: Cache Hit! NextA=0x%x, cacheIdx=0x%x\n", vc.NextA, cacheIdx );
if (IsDevBuild)
g_counter_cache_hits++;
}
else
{
// Only flag the cache if it's a non-dynamic memory range.
if (vc.NextA >= SPU2_DYN_MEMLINE)
{
cacheLine.Validated = true;
cacheLine.Prev1 = vc.Prev1;
cacheLine.Prev2 = vc.Prev2;
}
if (IsDevBuild)
{
if (vc.NextA < SPU2_DYN_MEMLINE)
g_counter_cache_ignores++;
else
g_counter_cache_misses++;
}
s16* memptr = GetMemPtr(vc.NextA & 0xFFFF8);
XA_decode_block(vc.SBuffer, memptr, vc.Prev1, vc.Prev2);
}
}
// 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++)
{
vc.DecodeFifo[(vc.DecPosWrite + i) % 32] = vc.SBuffer[sampleIdx + i];
}
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
static __forceinline s32 ApplyVolume(s32 data, s32 volume)
{
return (volume * data) >> 15;
}
#if defined(__aarch64__)
// NEON helper: lanewise `(volume * data) >> 15` on s32x2. Bit-exact with the
// scalar code's `mul w*, w*` + `asr 15` — narrow the s64 product to s32 first
// (matching scalar's implicit s32 truncation) then arithmetic-shift right 15.
static __forceinline int32x2_t ApplyVolumeStereoNEON(int32x2_t data, int32x2_t volume)
{
const int64x2_t prod = vmull_s32(data, volume);
const int32x2_t lo32 = vmovn_s64(prod);
return vshr_n_s32(lo32, 15);
}
#endif
static __forceinline StereoOut32 ApplyVolume(const StereoOut32& data, const V_VolumeLR& volume)
{
#if defined(__aarch64__)
const int32x2_t d = vld1_s32(&data.Left);
const int32x2_t v = vld1_s32(&volume.Left);
StereoOut32 out;
vst1_s32(&out.Left, ApplyVolumeStereoNEON(d, v));
return out;
#else
return StereoOut32(
ApplyVolume(data.Left, volume.Left),
ApplyVolume(data.Right, volume.Right));
#endif
}
static __forceinline StereoOut32 ApplyVolume(const StereoOut32& data, const V_VolumeSlideLR& volume)
{
#if defined(__aarch64__)
// V_VolumeSlide is 12 bytes; .Value is the s32 at offset 8 (Reg_VOL u16 + pad,
// then u32 Counter, then s32 Value). Build {Left.Value, Right.Value} via two
// scalar loads — cheaper than a gather, no aliasing constraints.
static_assert(sizeof(V_VolumeSlide) == 12, "V_VolumeSlide layout assumed by NEON ApplyVolume");
const int32x2_t d = vld1_s32(&data.Left);
const int32x2_t v = { volume.Left.Value, volume.Right.Value };
StereoOut32 out;
vst1_s32(&out.Left, ApplyVolumeStereoNEON(d, v));
return out;
#else
return StereoOut32(
ApplyVolume(data.Left, volume.Left.Value),
ApplyVolume(data.Right, volume.Right.Value));
#endif
}
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)
{
V_Voice& vc(Cores[coreidx].Voices[voiceidx]);
s32 pitch;
// [Air] : re-ordered comparisons: Modulated is much more likely to be zero than voice,
// and so the way it was before it's have to check both voice and modulated values
// most of the time. Now it'll just check Modulated and short-circuit past the voice
// check (not that it amounts to much, but eh every little bit helps).
if ((vc.Modulated == 0) || (voiceidx == 0))
pitch = vc.Pitch;
else
pitch = std::clamp((vc.Pitch * (32768 + Cores[coreidx].Voices[voiceidx - 1].OutX)) >> 15, 0, 0x3fff);
pitch = std::min(pitch, 0x3FFF);
vc.SP += pitch;
}
static __forceinline void CalculateADSR(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
if (vc.ADSR.Phase == V_ADSR::PHASE_STOPPED)
{
vc.ADSR.Value = 0;
return;
}
if (!vc.ADSR.Calculate(thiscore.Index | (voiceidx << 1)))
{
if (IsDevBuild)
{
if (SPU2::MsgVoiceOff())
SPU2::ConLog("* SPU2: Voice Off by ADSR: %d \n", voiceidx);
}
vc.Stop();
}
pxAssume(vc.ADSR.Value >= 0); // ADSR should never be negative...
}
static __forceinline void ConsumeSamples(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
int consumed = vc.SP >> 12;
vc.SP &= 0xfff;
vc.DecPosRead += consumed;
}
static __forceinline s32 GetVoiceValues(V_Core& thiscore, uint voiceidx)
{
V_Voice& vc(thiscore.Voices[voiceidx]);
int phase = (vc.SP & 0x0ff0) >> 4;
s32 out = 0;
out += (interpTable[phase][0] * vc.DecodeFifo[(vc.DecPosRead + 0) % 32]) >> 15;
out += (interpTable[phase][1] * vc.DecodeFifo[(vc.DecPosRead + 1) % 32]) >> 15;
out += (interpTable[phase][2] * vc.DecodeFifo[(vc.DecPosRead + 2) % 32]) >> 15;
out += (interpTable[phase][3] * vc.DecodeFifo[(vc.DecPosRead + 3) % 32]) >> 15;
return out;
}
// This is Dr. Hell's noise algorithm as implemented in pcsxr
// Supposedly this is 100% accurate
static __forceinline void UpdateNoise(V_Core& thiscore)
{
static const uint8_t noise_add[64] = {
1, 0, 0, 1, 0, 1, 1, 0,
1, 0, 0, 1, 0, 1, 1, 0,
1, 0, 0, 1, 0, 1, 1, 0,
1, 0, 0, 1, 0, 1, 1, 0,
0, 1, 1, 0, 1, 0, 0, 1,
0, 1, 1, 0, 1, 0, 0, 1,
0, 1, 1, 0, 1, 0, 0, 1,
0, 1, 1, 0, 1, 0, 0, 1};
static const uint16_t noise_freq_add[5] = {
0, 84, 140, 180, 210};
u32 level = 0x8000 >> (thiscore.NoiseClk >> 2);
level <<= 16;
thiscore.NoiseCnt += 0x10000;
thiscore.NoiseCnt += noise_freq_add[thiscore.NoiseClk & 3];
if ((thiscore.NoiseCnt & 0xffff) >= noise_freq_add[4])
{
thiscore.NoiseCnt += 0x10000;
thiscore.NoiseCnt -= noise_freq_add[thiscore.NoiseClk & 3];
}
if (thiscore.NoiseCnt >= level)
{
while (thiscore.NoiseCnt >= level)
thiscore.NoiseCnt -= level;
thiscore.NoiseOut = (thiscore.NoiseOut << 1) | noise_add[(thiscore.NoiseOut >> 10) & 63];
}
}
static __forceinline s32 GetNoiseValues(V_Core& thiscore)
{
return (s16)thiscore.NoiseOut;
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
// //
// writes a signed value to the SPU2 ram
// Performs no cache invalidation -- use only for dynamic memory ranges
// of the SPU2 (between 0x0000 and SPU2_DYN_MEMLINE)
static __forceinline void spu2M_WriteFast(u32 addr, s16 value)
{
// Fixes some of the oldest hangs in pcsx2's history! :p
for (int i = 0; i < 2; i++)
{
if (Cores[i].IRQEnable && Cores[i].IRQA == addr)
{
//printf("Core %d special write IRQ Called (IRQ passed). IRQA = %x\n",i,addr);
SetIrqCall(i);
}
}
// throw an assertion if the memory range is invalid:
#ifndef DEBUG_FAST
pxAssume(addr < SPU2_DYN_MEMLINE);
#endif
*GetMemPtr(addr) = value;
}
static __forceinline StereoOut32 MixVoice(uint coreidx, uint voiceidx)
{
V_Core& thiscore(Cores[coreidx]);
V_Voice& vc(thiscore.Voices[voiceidx]);
// 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...
// (Note: Ys 6 : Ark of Nephistm uses these effects)
vc.Volume.Update();
DecodeSamples(coreidx, voiceidx);
StereoOut32 voiceOut(0, 0);
s32 Value = 0;
if (vc.ADSR.Phase > V_ADSR::PHASE_STOPPED)
{
if (vc.Noise)
Value = GetNoiseValues(thiscore);
else
Value = GetVoiceValues(thiscore, voiceidx);
// Update and Apply ADSR (applies to normal and noise sources)
CalculateADSR(thiscore, voiceidx);
Value = ApplyVolume(Value, vc.ADSR.Value);
vc.OutX = Value;
if (IsDevBuild)
DebugCores[coreidx].Voices[voiceidx].displayPeak = std::max(DebugCores[coreidx].Voices[voiceidx].displayPeak, (s32)vc.OutX);
voiceOut = ApplyVolume(StereoOut32(Value, Value), vc.Volume);
}
// SPU2 Note: The spu2 continues to process voices for eternity, always, so we
// 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);
ConsumeSamples(thiscore, voiceidx);
// Write-back of raw voice data (post ADSR applied)
if (voiceidx == 1)
spu2M_WriteFast(((0 == coreidx) ? 0x400 : 0xc00) + OutPos, Value);
else if (voiceidx == 3)
spu2M_WriteFast(((0 == coreidx) ? 0x600 : 0xe00) + OutPos, Value);
return voiceOut;
}
static __forceinline void MixCoreVoices(VoiceMixSet& dest, const uint coreidx)
{
V_Core& thiscore(Cores[coreidx]);
#if defined(__aarch64__)
// dest is {Dry.L, Dry.R, Wet.L, Wet.R} = 4 contiguous s32, and each
// V_VoiceGates entry is the same {DryL, DryR, WetL, WetR} contiguous s32x4.
// Per voice: vval = {VVal.L, VVal.R, VVal.L, VVal.R}, accum += vval & gates.
// Bit-identical to the scalar version below.
int32x4_t accum = vld1q_s32(&dest.Dry.Left);
for (uint voiceidx = 0; voiceidx < V_Core::NumVoices; ++voiceidx)
{
const StereoOut32 VVal(MixVoice(coreidx, voiceidx));
const int32x2_t lr = vld1_s32(&VVal.Left);
const int32x4_t vval = vcombine_s32(lr, lr);
const int32x4_t gate = vld1q_s32(&thiscore.VoiceGates[voiceidx].DryL);
accum = vaddq_s32(accum, vandq_s32(vval, gate));
}
vst1q_s32(&dest.Dry.Left, accum);
#else
for (uint voiceidx = 0; voiceidx < V_Core::NumVoices; ++voiceidx)
{
StereoOut32 VVal(MixVoice(coreidx, voiceidx));
// Note: Results from MixVoice are ranged at 16 bits.
dest.Dry.Left += VVal.Left & thiscore.VoiceGates[voiceidx].DryL;
dest.Dry.Right += VVal.Right & thiscore.VoiceGates[voiceidx].DryR;
dest.Wet.Left += VVal.Left & thiscore.VoiceGates[voiceidx].WetL;
dest.Wet.Right += VVal.Right & thiscore.VoiceGates[voiceidx].WetR;
}
#endif
}
static __forceinline StereoOut32 MixCore(const uint coreidx, const VoiceMixSet& inVoices, const StereoOut32& Input, const StereoOut32& Ext)
{
V_Core& thiscore(Cores[coreidx]);
thiscore.MasterVol.Update();
UpdateNoise(thiscore);
// Saturate final result to standard 16 bit range.
const VoiceMixSet Voices(clamp_mix(inVoices.Dry), clamp_mix(inVoices.Wet));
// Write Mixed results To Output Area
spu2M_WriteFast(((0 == thiscore.Index) ? 0x1000 : 0x1800) + OutPos, Voices.Dry.Left);
spu2M_WriteFast(((0 == thiscore.Index) ? 0x1200 : 0x1A00) + OutPos, Voices.Dry.Right);
spu2M_WriteFast(((0 == thiscore.Index) ? 0x1400 : 0x1C00) + OutPos, Voices.Wet.Left);
spu2M_WriteFast(((0 == thiscore.Index) ? 0x1600 : 0x1E00) + OutPos, Voices.Wet.Right);
// Write mixed results to logfile (if enabled)
#ifdef PCSX2_DEVBUILD
WaveDump::WriteCore(thiscore.Index, CoreSrc_DryVoiceMix, Voices.Dry);
WaveDump::WriteCore(thiscore.Index, CoreSrc_WetVoiceMix, Voices.Wet);
#endif
// Mix in the Input data
StereoOut32 TD(
Input.Left & thiscore.DryGate.InpL,
Input.Right & thiscore.DryGate.InpR);
// Mix in the Voice data
TD.Left += Voices.Dry.Left & thiscore.DryGate.SndL;
TD.Right += Voices.Dry.Right & thiscore.DryGate.SndR;
// Mix in the External (nothing/core0) data
TD.Left += Ext.Left & thiscore.DryGate.ExtL;
TD.Right += Ext.Right & thiscore.DryGate.ExtR;
// ----------------------------------------------------------------------------
// Reverberation Effects Processing
// ----------------------------------------------------------------------------
// SPU2 has an FxEnable bit which seems to disable all reverb processing *and*
// output, but does *not* disable the advancing buffers. IRQs are not triggered
// and reverb is rendered silent.
//
// Technically we should advance the buffers even when fx are disabled. However
// there are two things that make this very unlikely to matter:
//
// 1. Any SPU2 app wanting to avoid noise or pops needs to clear the reverb buffers
// when adjusting settings anyway; so the read/write positions in the reverb
// buffer after FxEnabled is set back to 1 doesn't really matter.
//
// 2. Writes to ESA (and possibly EEA) reset the buffer pointers to 0.
//
// On the other hand, updating the buffer is cheap and easy, so might as well. ;)
StereoOut32 TW;
// Mix Input, Voice, and External data:
TW.Left = Input.Left & thiscore.WetGate.InpL;
TW.Right = Input.Right & thiscore.WetGate.InpR;
TW.Left += Voices.Wet.Left & thiscore.WetGate.SndL;
TW.Right += Voices.Wet.Right & thiscore.WetGate.SndR;
TW.Left += Ext.Left & thiscore.WetGate.ExtL;
TW.Right += Ext.Right & thiscore.WetGate.ExtR;
// Lightweight audio mode (low-end Android CPU lever): keep wet-routed voices
// audible but skip the considerably heavier SPU2 reverb pipeline (the
// ReverbDownsample/Upsample FIR resamplers + comb/all-pass network in
// DoReverb). Trades all echo/spatial reverb for CPU; off by default.
if (EmuConfig.SPU2.LightweightMode)
return TD + TW;
#ifdef PCSX2_DEVBUILD
WaveDump::WriteCore(thiscore.Index, CoreSrc_PreReverb, TW);
#endif
StereoOut32 RV = thiscore.DoReverb(TW);
#ifdef PCSX2_DEVBUILD
WaveDump::WriteCore(thiscore.Index, CoreSrc_PostReverb, RV);
#endif
// Mix Dry + Wet
// (master volume is applied later to the result of both outputs added together).
return TD + ApplyVolume(RV, thiscore.FxVol);
}
void spu2Mix()
{
// Note: Playmode 4 is SPDIF, which overrides other inputs.
StereoOut32 InputData[2] =
{
// SPDIF is on Core 0:
// Fixme:
// 1. We do not have an AC3 decoder for the bitstream.
// 2. Games usually provide a normal ADMA stream as well and want to see it getting read!
/*(PlayMode&4) ? StereoOut32::Empty : */ ApplyVolume(Cores[0].ReadInput(), Cores[0].InpVol),
// CDDA is on Core 1:
(PlayMode & 8) ? StereoOut32::Empty : ApplyVolume(Cores[1].ReadInput(), Cores[1].InpVol)};
#ifdef PCSX2_DEVBUILD
WaveDump::WriteCore(0, CoreSrc_Input, InputData[0]);
WaveDump::WriteCore(1, CoreSrc_Input, InputData[1]);
#endif
// Todo: Replace me with memzero initializer!
VoiceMixSet VoiceData[2] = {{StereoOut32(), StereoOut32()}, {StereoOut32(), StereoOut32()}}; // mixed voice data for each core.
MixCoreVoices(VoiceData[0], 0);
MixCoreVoices(VoiceData[1], 1);
StereoOut32 Ext(MixCore(0, VoiceData[0], InputData[0], StereoOut32::Empty));
if ((PlayMode & 4) || (Cores[0].Mute != 0))
Ext = StereoOut32::Empty;
else
{
Ext = ApplyVolume(clamp_mix(Ext), Cores[0].MasterVol);
}
// Commit Core 0 output to ram before mixing Core 1:
spu2M_WriteFast(0x800 + OutPos, Ext.Left);
spu2M_WriteFast(0xA00 + OutPos, Ext.Right);
#ifdef PCSX2_DEVBUILD
WaveDump::WriteCore(0, CoreSrc_External, Ext);
#endif
Ext = ApplyVolume(Ext, Cores[1].ExtVol);
StereoOut32 Out(MixCore(1, VoiceData[1], InputData[1], Ext));
if (PlayMode & 8)
{
// Experimental CDDA support
// The CDDA overrides all other mixer output. It's a direct feed!
Out = Cores[1].ReadInput_HiFi();
//WaveLog::WriteCore( 1, "CDDA-32", OutL, OutR );
}
else
{
Out = ApplyVolume(clamp_mix(Out), Cores[1].MasterVol);
}
#ifdef PCSX2_DEVBUILD
// Log final output to wavefile.
WaveDump::WriteCore(1, CoreSrc_External, Out);
#endif
// Optional stereo L<->R swap for devices forced into reverse-landscape (flipped
// speakers). Applied to the final mixed sample, so it covers every output path.
if (SPU2::IsSwapChannels())
{
const s32 tmp = Out.Left;
Out.Left = Out.Right;
Out.Right = tmp;
}
spu2Output(Out);
// Update AutoDMA output positioning
OutPos++;
if (OutPos >= 0x200)
OutPos = 0;
if constexpr (IsDevBuild)
{
// used to throttle the output rate of cache stat reports
static int p_cachestat_counter = 0;
p_cachestat_counter++;
if (p_cachestat_counter > (48000 * 10))
{
p_cachestat_counter = 0;
if (SPU2::MsgCache())
{
SPU2::ConLog(" * SPU2 > CacheStats > Hits: %d Misses: %d Ignores: %d\n",
g_counter_cache_hits,
g_counter_cache_misses,
g_counter_cache_ignores);
}
g_counter_cache_hits =
g_counter_cache_misses =
g_counter_cache_ignores = 0;
}
}
}
MULTI_ISA_UNSHARED_END