/* SPU2-X, A plugin for Emulating the Sound Processing Unit of the Playstation 2
 * Developed and maintained by the Pcsx2 Development Team.
 *
 * Original portions from SPU2ghz are (c) 2008 by David Quintana [gigaherz]
 *
 * SPU2-X is free software: you can redistribute it and/or modify it under the terms
 * of the GNU Lesser General Public License as published by the Free Software Found-
 * ation, either version 3 of the License, or (at your option) any later version.
 *
 * SPU2-X 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 Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public License
 * along with SPU2-X.  If not, see <http://www.gnu.org/licenses/>.
 */

#include "Global.h"

// Games have turned out to be surprisingly sensitive to whether a parked, silent voice is being fully emulated.
// With Silent Hill: Shattered Memories requiring full processing for no obvious reason, we've decided to
// disable the optimisation until we can tie it to the game database.
#define NEVER_SKIP_VOICES 1

void ADMAOutLogWrite(void *lpData, u32 ulSize);

static const s32 tbl_XA_Factor[16][2] =
    {
        {0, 0},
        {60, 0},
        {115, -52},
        {98, -55},
        {122, -60}};


// Performs a 64-bit multiplication between two values and returns the
// high 32 bits as a result (discarding the fractional 32 bits).
// The combined fractional bits of both inputs must be 32 bits for this
// to work properly.
//
// This is meant to be a drop-in replacement for times when the 'div' part
// of a MulDiv is a constant.  (example: 1<<8, or 4096, etc)
//
// [Air] Performance breakdown: This is over 10 times faster than MulDiv in
//   a *worst case* scenario.  It's also more accurate since it forces the
//   caller to  extend the inputs so that they make use of all 32 bits of
//   precision.
//
static __forceinline s32 MulShr32(s32 srcval, s32 mulval)
{
    return (s64)srcval * mulval >> 32;
}

__forceinline s32 clamp_mix(s32 x, u8 bitshift)
{
    assert(bitshift <= 15);
    return GetClamped(x, -(0x8000 << bitshift), 0x7fff << bitshift);
}

#if _MSC_VER
__forceinline
// Without the keyword static, gcc compilation fails on the inlining...
// Unfortunately the function is also used in Reverb.cpp. In order to keep the code
// clean we just disable it.
// We will need link-time code generation / Whole Program optimization to do a clean
// inline. Gcc 4.5 has the experimental options -flto, -fwhopr and -fwhole-program to
// do it but it still experimental...
#endif
    StereoOut32
    clamp_mix(const StereoOut32 &sample, u8 bitshift)
{
    // We should clampify between -0x8000 and 0x7fff, however some audio output
    // modules or sound drivers could (will :p) overshoot with that. So giving it a small safety.

    return StereoOut32(
        GetClamped(sample.Left, -(0x7f00 << bitshift), 0x7f00 << bitshift),
        GetClamped(sample.Right, -(0x7f00 << bitshift), 0x7f00 << bitshift));
}

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 && MsgToConsole())
        ConLog("* SPU2-X: 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);

        Clampify(pcm, -0x8000, 0x7fff);
        *(buffer++) = pcm;

        data = ((*blockbytes) << 24) & 0xF0000000;
        s32 pcm2 = (data >> shift) + (((pred1 * pcm) + (pred2 * prev1) + 32) >> 6);

        Clampify(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;
}

// 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 = NULL;

int g_counter_cache_hits = 0;
int g_counter_cache_misses = 0;
int g_counter_cache_ignores = 0;

// 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)

static __forceinline s32 GetNextDataBuffered(V_Core &thiscore, uint voiceidx)
{
    V_Voice &vc(thiscore.Voices[voiceidx]);

    if ((vc.SCurrent & 3) == 0) {
        IncrementNextA(thiscore, voiceidx);

        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 (MsgVoiceOff())
                            ConLog("* SPU2-X: 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];
        vc.SBuffer = cacheLine.Sampledata;

        if (cacheLine.Validated) {
            // 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-X: 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;

            if (IsDevBuild) {
                if (vc.NextA < SPU2_DYN_MEMLINE)
                    g_counter_cache_ignores++;
                else
                    g_counter_cache_misses++;
            }

            XA_decode_block(vc.SBuffer, memptr, vc.Prev1, vc.Prev2);
        }
    }

    return vc.SBuffer[vc.SCurrent++];
}

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) {
            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 -= 4096 * (4 - (vc.SCurrent & 3));
    vc.SCurrent += 4 - (vc.SCurrent & 3);
}

/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////

static s32 __forceinline GetNoiseValues()
{
    static u16 lfsr = 0xC0FEu;

    u16 bit = lfsr ^ (lfsr << 3) ^ (lfsr << 4) ^ (lfsr << 5);
    lfsr = (lfsr << 1) | (bit >> 15);

    return (s16)lfsr;
}
/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
//                                                                                     //

// Data is expected to be 16 bit signed (typical stuff!).
// volume is expected to be 32 bit signed (31 bits with reverse phase)
// Data is shifted up by 1 bit to give the output an effective 16 bit range.
static __forceinline s32 ApplyVolume(s32 data, s32 volume)
{
    //return (volume * data) >> 15;
    return MulShr32(data << 1, volume);
}

static __forceinline StereoOut32 ApplyVolume(const StereoOut32 &data, const V_VolumeLR &volume)
{
    return StereoOut32(
        ApplyVolume(data.Left, volume.Left),
        ApplyVolume(data.Right, volume.Right));
}

static __forceinline StereoOut32 ApplyVolume(const StereoOut32 &data, const V_VolumeSlideLR &volume)
{
    return StereoOut32(
        ApplyVolume(data.Left, volume.Left.Value),
        ApplyVolume(data.Right, volume.Right.Value));
}

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 = GetClamped((vc.Pitch * (32768 + Cores[coreidx].Voices[voiceidx - 1].OutX)) >> 15, 0, 0x3fff);

    vc.SP += pitch;
}


static __forceinline void CalculateADSR(V_Core &thiscore, uint voiceidx)
{
    V_Voice &vc(thiscore.Voices[voiceidx]);

    if (vc.ADSR.Phase == 0) {
        vc.ADSR.Value = 0;
        return;
    }

    if (!vc.ADSR.Calculate()) {
        if (IsDevBuild) {
            if (MsgVoiceOff())
                ConLog("* SPU2-X: Voice Off by ADSR: %d \n", voiceidx);
        }
        vc.Stop();
    }

    pxAssume(vc.ADSR.Value >= 0); // ADSR should never be negative...
}

/*
   Tension: 65535 is high, 32768 is normal, 0 is low
*/
template <s32 i_tension>
__forceinline static s32 HermiteInterpolate(
    s32 y0, // 16.0
    s32 y1, // 16.0
    s32 y2, // 16.0
    s32 y3, // 16.0
    s32 mu  //  0.12
    )
{
    s32 m00 = ((y1 - y0) * i_tension) >> 16; // 16.0
    s32 m01 = ((y2 - y1) * i_tension) >> 16; // 16.0
    s32 m0 = m00 + m01;

    s32 m10 = ((y2 - y1) * i_tension) >> 16; // 16.0
    s32 m11 = ((y3 - y2) * i_tension) >> 16; // 16.0
    s32 m1 = m10 + m11;

    s32 val = ((2 * y1 + m0 + m1 - 2 * y2) * mu) >> 12;       // 16.0
    val = ((val - 3 * y1 - 2 * m0 - m1 + 3 * y2) * mu) >> 12; // 16.0
    val = ((val + m0) * mu) >> 11;                            // 16.0

    return (val + (y1 << 1));
}

__forceinline static s32 CatmullRomInterpolate(
    s32 y0, // 16.0
    s32 y1, // 16.0
    s32 y2, // 16.0
    s32 y3, // 16.0
    s32 mu  //  0.12
    )
{
    //q(t) = 0.5 *(    	(2 * P1) +
    //	(-P0 + P2) * t +
    //	(2*P0 - 5*P1 + 4*P2 - P3) * t2 +
    //	(-P0 + 3*P1- 3*P2 + P3) * t3)

    s32 a3 = (-y0 + 3 * y1 - 3 * y2 + y3);
    s32 a2 = (2 * y0 - 5 * y1 + 4 * y2 - y3);
    s32 a1 = (-y0 + y2);
    s32 a0 = (2 * y1);

    s32 val = ((a3)*mu) >> 12;
    val = ((a2 + val) * mu) >> 12;
    val = ((a1 + val) * mu) >> 12;

    return (a0 + val);
}

__forceinline static s32 CubicInterpolate(
    s32 y0, // 16.0
    s32 y1, // 16.0
    s32 y2, // 16.0
    s32 y3, // 16.0
    s32 mu  //  0.12
    )
{
    const s32 a0 = y3 - y2 - y0 + y1;
    const s32 a1 = y0 - y1 - a0;
    const s32 a2 = y2 - y0;

    s32 val = ((a0)*mu) >> 12;
    val = ((val + a1) * mu) >> 12;
    val = ((val + a2) * mu) >> 11;

    return (val + (y1 << 1));
}

// Returns a 16 bit result in Value.
// Uses standard template-style optimization techniques to statically generate five different
// versions of this function (one for each type of interpolation).
template <int InterpType>
static __forceinline s32 GetVoiceValues(V_Core &thiscore, uint voiceidx)
{
    V_Voice &vc(thiscore.Voices[voiceidx]);

    while (vc.SP > 0) {
        if (InterpType >= 2) {
            vc.PV4 = vc.PV3;
            vc.PV3 = vc.PV2;
        }
        vc.PV2 = vc.PV1;
        vc.PV1 = GetNextDataBuffered(thiscore, voiceidx);
        vc.SP -= 4096;
    }

    const s32 mu = vc.SP + 4096;

    switch (InterpType) {
        case 0:
            return vc.PV1 << 1;
        case 1:
            return (vc.PV1 << 1) - (((vc.PV2 - vc.PV1) * vc.SP) >> 11);

        case 2:
            return CubicInterpolate(vc.PV4, vc.PV3, vc.PV2, vc.PV1, mu);
        case 3:
            return HermiteInterpolate<16384>(vc.PV4, vc.PV3, vc.PV2, vc.PV1, mu);
        case 4:
            return CatmullRomInterpolate(vc.PV4, vc.PV3, vc.PV2, vc.PV1, mu);

            jNO_DEFAULT;
    }

    return 0; // technically unreachable!
}

// Noise values need to be mixed without going through interpolation, since it
// can wreak havoc on the noise (causing muffling or popping).  Not that this noise
// generator is accurate in its own right.. but eh, ah well :)
static __forceinline s32 GetNoiseValues(V_Core &thiscore, uint voiceidx)
{
    // V_Voice &vc(thiscore.Voices[voiceidx]);

    s32 retval = GetNoiseValues();

    /*while(vc.SP>=4096)
	{
		retval = GetNoiseValues();
		vc.SP-=4096;
	}*/

    // GetNoiseValues can't set the phase zero on us unexpectedly
    // like GetVoiceValues can.  Better assert just in case though..
    // pxAssume(vc.ADSR.Phase != 0);

    return retval;
}

/////////////////////////////////////////////////////////////////////////////////////////
/////////////////////////////////////////////////////////////////////////////////////////
//                                                                                     //

// 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]);

    // 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
    // methods when needed by checking the flag outside the method here...
    // (Note: Ys 6 : Ark of Nephistm uses these effects)

    vc.Volume.Update();

    // 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.

    if (vc.ADSR.Phase > 0) {
        UpdatePitch(coreidx, voiceidx);

        s32 Value = 0;

        if (vc.Noise)
            Value = GetNoiseValues(thiscore, voiceidx);
        else {
            // Optimization : Forceinline'd Templated Dispatch Table.  Any halfwit compiler will
            // turn this into a clever jump dispatch table (no call/rets, no compares, uber-efficient!)

            switch (Interpolation) {
                case 0:
                    Value = GetVoiceValues<0>(thiscore, voiceidx);
                    break;
                case 1:
                    Value = GetVoiceValues<1>(thiscore, voiceidx);
                    break;
                case 2:
                    Value = GetVoiceValues<2>(thiscore, voiceidx);
                    break;
                case 3:
                    Value = GetVoiceValues<3>(thiscore, voiceidx);
                    break;
                case 4:
                    Value = GetVoiceValues<4>(thiscore, voiceidx);
                    break;

                    jNO_DEFAULT;
            }
        }

        // Update and Apply ADSR  (applies to normal and noise sources)
        //
        // Note!  It's very important that ADSR stay as accurate as possible.  By the way
        // it is used, various sound effects can end prematurely if we truncate more than
        // one or two bits.  Best result comes from no truncation at all, which is why we
        // use a full 64-bit multiply/result here.

        CalculateADSR(thiscore, voiceidx);
        Value = MulShr32(Value, vc.ADSR.Value);

        // Store Value for eventual modulation later
        // Pseudonym's Crest calculation idea. Actually calculates a crest, unlike the old code which was just peak.
        if (vc.PV1 < vc.NextCrest) {
            vc.OutX = MulShr32(vc.NextCrest, vc.ADSR.Value);
            vc.NextCrest = -0x8000;
        }
        if (vc.PV1 > vc.PV2) {
            vc.NextCrest = vc.PV1;
        }

        if (IsDevBuild)
            DebugCores[coreidx].Voices[voiceidx].displayPeak = std::max(DebugCores[coreidx].Voices[voiceidx].displayPeak, (s32)vc.OutX);

        // Write-back of raw voice data (post ADSR applied)

        if (voiceidx == 1)
            spu2M_WriteFast(((0 == coreidx) ? 0x400 : 0xc00) + OutPos, vc.OutX);
        else if (voiceidx == 3)
            spu2M_WriteFast(((0 == coreidx) ? 0x600 : 0xe00) + OutPos, vc.OutX);

        return ApplyVolume(StereoOut32(Value, Value), vc.Volume);
    } else {
        // Continue processing voice, even if it's "off". Or else we miss interrupts! (Fatal Frame engine died because of this.)
        if (NEVER_SKIP_VOICES || (*GetMemPtr(vc.NextA & 0xFFFF8) >> 8 & 3) != 3 || vc.LoopStartA != (vc.NextA & ~7)    // not in a tight loop
            || (Cores[0].IRQEnable && (Cores[0].IRQA & ~7) == vc.LoopStartA)                                           // or should be interrupting regularly
            || (Cores[1].IRQEnable && (Cores[1].IRQA & ~7) == vc.LoopStartA) || !(thiscore.Regs.ENDX & 1 << voiceidx)) // or isn't currently flagged as having passed the endpoint
        {
            UpdatePitch(coreidx, voiceidx);

            while (vc.SP > 0)
                GetNextDataDummy(thiscore, voiceidx); // Dummy is enough
        }

        // Write-back of raw voice data (some zeros since the voice is "dead")
        if (voiceidx == 1)
            spu2M_WriteFast(((0 == coreidx) ? 0x400 : 0xc00) + OutPos, 0);
        else if (voiceidx == 3)
            spu2M_WriteFast(((0 == coreidx) ? 0x600 : 0xe00) + OutPos, 0);

        return StereoOut32(0, 0);
    }
}

const VoiceMixSet VoiceMixSet::Empty((StereoOut32()), (StereoOut32())); // Don't use SteroOut32::Empty because C++ doesn't make any dep/order checks on global initializers.

static __forceinline void MixCoreVoices(VoiceMixSet &dest, const uint coreidx)
{
    V_Core &thiscore(Cores[coreidx]);

    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;
    }
}

StereoOut32 V_Core::Mix(const VoiceMixSet &inVoices, const StereoOut32 &Input, const StereoOut32 &Ext)
{
    MasterVol.Update();

    // 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 == Index) ? 0x1000 : 0x1800) + OutPos, Voices.Dry.Left);
    spu2M_WriteFast(((0 == Index) ? 0x1200 : 0x1A00) + OutPos, Voices.Dry.Right);
    spu2M_WriteFast(((0 == Index) ? 0x1400 : 0x1C00) + OutPos, Voices.Wet.Left);
    spu2M_WriteFast(((0 == Index) ? 0x1600 : 0x1E00) + OutPos, Voices.Wet.Right);

    // Write mixed results to logfile (if enabled)

    WaveDump::WriteCore(Index, CoreSrc_DryVoiceMix, Voices.Dry);
    WaveDump::WriteCore(Index, CoreSrc_WetVoiceMix, Voices.Wet);

    // Mix in the Input data

    StereoOut32 TD(
        Input.Left & DryGate.InpL,
        Input.Right & DryGate.InpR);

    // Mix in the Voice data
    TD.Left += Voices.Dry.Left & DryGate.SndL;
    TD.Right += Voices.Dry.Right & DryGate.SndR;

    // Mix in the External (nothing/core0) data
    TD.Left += Ext.Left & DryGate.ExtL;
    TD.Right += Ext.Right & DryGate.ExtR;

    // User-level Effects disabling.  Nice speedup but breaks games that depend on
    // reverb IRQs (very few -- if you find one name it here!).
    if (EffectsDisabled)
        return TD;

    // ----------------------------------------------------------------------------
    //    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. ;)

    Reverb_AdvanceBuffer(); // Updates the reverb work area as well, if needed.

    // ToDo:
    // Bad EndA causes memory corruption. Bad for us, unknown on PS2!
    // According to no$psx, effects always run but don't always write back, so the FxEnable check may be wrong
    if (!FxEnable || EffectsEndA >= 0x100000)
        return TD;

    StereoOut32 TW;

    // Mix Input, Voice, and External data:

    TW.Left = Input.Left & WetGate.InpL;
    TW.Right = Input.Right & WetGate.InpR;

    TW.Left += Voices.Wet.Left & WetGate.SndL;
    TW.Right += Voices.Wet.Right & WetGate.SndR;
    TW.Left += Ext.Left & WetGate.ExtL;
    TW.Right += Ext.Right & WetGate.ExtR;

    WaveDump::WriteCore(Index, CoreSrc_PreReverb, TW);

    StereoOut32 RV = DoReverb(TW);

    WaveDump::WriteCore(Index, CoreSrc_PostReverb, RV);

    // Mix Dry + Wet
    // (master volume is applied later to the result of both outputs added together).
    return TD + ApplyVolume(RV, FxVol);
}

// Filters that work on the final output to de-alias and equlize it.
// Taken from http://nenolod.net/projects/upse/
#define OVERALL_SCALE (0.87f)

StereoOut32 Apply_Frequency_Response_Filter(StereoOut32 &SoundStream)
{
    static FrequencyResponseFilter FRF = FrequencyResponseFilter();

    s32 in, out;
    s32 l, r;
    s32 mid, side;

    l = SoundStream.Left;
    r = SoundStream.Right;

    mid = l + r;
    side = l - r;

    in = mid;
    out = FRF.la0 * in + FRF.la1 * FRF.lx1 + FRF.la2 * FRF.lx2 - FRF.lb1 * FRF.ly1 - FRF.lb2 * FRF.ly2;

    FRF.lx2 = FRF.lx1;
    FRF.lx1 = in;

    FRF.ly2 = FRF.ly1;
    FRF.ly1 = out;

    mid = out;

    l = ((0.5) * (OVERALL_SCALE)) * (mid + side);
    r = ((0.5) * (OVERALL_SCALE)) * (mid - side);

    in = l;
    out = FRF.ha0 * in + FRF.ha1 * FRF.History_One_In.Left + FRF.ha2 * FRF.History_Two_In.Left - FRF.hb1 * FRF.History_One_Out.Left - FRF.hb2 * FRF.History_Two_Out.Left;
    FRF.History_Two_In.Left = FRF.History_One_In.Left;
    FRF.History_One_In.Left = in;
    FRF.History_Two_Out.Left = FRF.History_One_Out.Left;
    FRF.History_One_Out.Left = out;
    l = out;

    in = r;
    out = FRF.ha0 * in + FRF.ha1 * FRF.History_One_In.Right + FRF.ha2 * FRF.History_Two_In.Right - FRF.hb1 * FRF.History_One_Out.Right - FRF.hb2 * FRF.History_Two_Out.Right;
    FRF.History_Two_In.Right = FRF.History_One_In.Right;
    FRF.History_One_In.Right = in;
    FRF.History_Two_Out.Right = FRF.History_One_Out.Right;
    FRF.History_One_Out.Right = out;
    r = out;

    //clamp_mix(l);
    //clamp_mix(r);

    SoundStream.Left = l;
    SoundStream.Right = r;

    return SoundStream;
}

StereoOut32 Apply_Dealias_Filter(StereoOut32 &SoundStream)
{
    static StereoOut32 Old = StereoOut32::Empty;

    s32 l, r;

    l = SoundStream.Left;
    r = SoundStream.Right;

    l += (l - Old.Left);
    r += (r - Old.Right);

    Old.Left = SoundStream.Left;
    Old.Right = SoundStream.Right;

    SoundStream.Left = l;
    SoundStream.Right = r;

    return SoundStream;
}

// used to throttle the output rate of cache stat reports
static int p_cachestat_counter = 0;

// Gcc does not want to inline it when lto is enabled because some functions growth too much.
// The function is big enought to see any speed impact. -- Gregory
#ifndef __POSIX__
__forceinline
#endif
    void
    Mix()
{
    // 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)};

    WaveDump::WriteCore(0, CoreSrc_Input, InputData[0]);
    WaveDump::WriteCore(1, CoreSrc_Input, InputData[1]);

    // Todo: Replace me with memzero initializer!
    VoiceMixSet VoiceData[2] = {VoiceMixSet::Empty, VoiceMixSet::Empty}; // mixed voice data for each core.
    MixCoreVoices(VoiceData[0], 0);
    MixCoreVoices(VoiceData[1], 1);

    StereoOut32 Ext(Cores[0].Mix(VoiceData[0], InputData[0], StereoOut32::Empty));

    if ((PlayMode & 4) || (Cores[0].Mute != 0))
        Ext = StereoOut32::Empty;
    else {
        Ext = clamp_mix(ApplyVolume(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);

    WaveDump::WriteCore(0, CoreSrc_External, Ext);

    Ext = ApplyVolume(Ext, Cores[1].ExtVol);
    StereoOut32 Out(Cores[1].Mix(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.Left = MulShr32(Out.Left << (SndOutVolumeShift + 1), Cores[1].MasterVol.Left.Value);
        Out.Right = MulShr32(Out.Right << (SndOutVolumeShift + 1), Cores[1].MasterVol.Right.Value);

#ifdef DEBUG_KEYS
        if (postprocess_filter_enabled)
#endif
        {
            if (postprocess_filter_dealias) {
                // Dealias filter emphasizes the highs too much.
                Out = Apply_Dealias_Filter(Out);
            }
            Out = Apply_Frequency_Response_Filter(Out);
        }

        // Final Clamp!
        // Like any good audio system, the PS2 pumps the volume and incurs some distortion in its
        // output, giving us a nice thumpy sound at times.  So we add 1 above (2x volume pump) and
        // then clamp it all here.

        // Edit: I'm sorry Jake, but I know of no good audio system that arbitrary distorts and clips
        // output by design.
        // Good thing though that this code gets the volume exactly right, as per tests :)
        Out = clamp_mix(Out, SndOutVolumeShift);
    }

    // Configurable output volume
    Out.Left *= FinalVolume;
    Out.Right *= FinalVolume;

    SndBuffer::Write(Out);

    // Update AutoDMA output positioning
    OutPos++;
    if (OutPos >= 0x200)
        OutPos = 0;

    if (IsDevBuild) {
        p_cachestat_counter++;
        if (p_cachestat_counter > (48000 * 10)) {
            p_cachestat_counter = 0;
            if (MsgCache())
                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;
        }
    }
}
