Convert ALu.c to C++

Required changes to bsincgen to generate C++-friendly structures.

1 files changed, 1884 insertions, 0 deletions

diff --git a/Alc/alu.cpp b/Alc/alu.cpp
new file mode 100644
index 00000000..df857b80
--- /dev/null
+++ b/Alc/alu.cpp
@@ -0,0 +1,1884 @@
+/**
+ * OpenAL cross platform audio library
+ * Copyright (C) 1999-2007 by authors.
+ * This library is free software; you can redistribute it and/or
+ *  modify it under the terms of the GNU Library General Public
+ *  License as published by the Free Software Foundation; either
+ *  version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ *  Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ *  License along with this library; if not, write to the
+ *  Free Software Foundation, Inc.,
+ *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
+ * Or go to http://www.gnu.org/copyleft/lgpl.html
+ */
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+#include <string.h>
+#include <ctype.h>
+#include <assert.h>
+
+#include "alMain.h"
+#include "alSource.h"
+#include "alBuffer.h"
+#include "alListener.h"
+#include "alAuxEffectSlot.h"
+#include "alu.h"
+#include "bs2b.h"
+#include "hrtf.h"
+#include "mastering.h"
+#include "uhjfilter.h"
+#include "bformatdec.h"
+#include "static_assert.h"
+#include "ringbuffer.h"
+#include "filters/splitter.h"
+
+#include "mixer/defs.h"
+#include "fpu_modes.h"
+#include "cpu_caps.h"
+#include "bsinc_inc.h"
+
+
+/* Cone scalar */
+ALfloat ConeScale = 1.0f;
+
+/* Localized Z scalar for mono sources */
+ALfloat ZScale = 1.0f;
+
+/* Force default speed of sound for distance-related reverb decay. */
+ALboolean OverrideReverbSpeedOfSound = AL_FALSE;
+
+const aluMatrixf IdentityMatrixf = {{
+    { 1.0f, 0.0f, 0.0f, 0.0f },
+    { 0.0f, 1.0f, 0.0f, 0.0f },
+    { 0.0f, 0.0f, 1.0f, 0.0f },
+    { 0.0f, 0.0f, 0.0f, 1.0f },
+}};
+
+
+static void ClearArray(ALfloat f[MAX_OUTPUT_CHANNELS])
+{
+    size_t i;
+    for(i = 0;i < MAX_OUTPUT_CHANNELS;i++)
+        f[i] = 0.0f;
+}
+
+struct ChanMap {
+    enum Channel channel;
+    ALfloat angle;
+    ALfloat elevation;
+};
+
+static HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_C;
+
+
+void DeinitVoice(ALvoice *voice)
+{
+    al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice->Update, static_cast<ALvoiceProps*>(nullptr)));
+}
+
+
+static inline HrtfDirectMixerFunc SelectHrtfMixer(void)
+{
+#ifdef HAVE_NEON
+    if((CPUCapFlags&CPU_CAP_NEON))
+        return MixDirectHrtf_Neon;
+#endif
+#ifdef HAVE_SSE
+    if((CPUCapFlags&CPU_CAP_SSE))
+        return MixDirectHrtf_SSE;
+#endif
+
+    return MixDirectHrtf_C;
+}
+
+
+/* This RNG method was created based on the math found in opusdec. It's quick,
+ * and starting with a seed value of 22222, is suitable for generating
+ * whitenoise.
+ */
+static inline ALuint dither_rng(ALuint *seed)
+{
+    *seed = (*seed * 96314165) + 907633515;
+    return *seed;
+}
+
+
+static inline void aluCrossproduct(const ALfloat *inVector1, const ALfloat *inVector2, ALfloat *outVector)
+{
+    outVector[0] = inVector1[1]*inVector2[2] - inVector1[2]*inVector2[1];
+    outVector[1] = inVector1[2]*inVector2[0] - inVector1[0]*inVector2[2];
+    outVector[2] = inVector1[0]*inVector2[1] - inVector1[1]*inVector2[0];
+}
+
+static inline ALfloat aluDotproduct(const aluVector *vec1, const aluVector *vec2)
+{
+    return vec1->v[0]*vec2->v[0] + vec1->v[1]*vec2->v[1] + vec1->v[2]*vec2->v[2];
+}
+
+static ALfloat aluNormalize(ALfloat *vec)
+{
+    ALfloat length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]);
+    if(length > FLT_EPSILON)
+    {
+        ALfloat inv_length = 1.0f/length;
+        vec[0] *= inv_length;
+        vec[1] *= inv_length;
+        vec[2] *= inv_length;
+        return length;
+    }
+    vec[0] = vec[1] = vec[2] = 0.0f;
+    return 0.0f;
+}
+
+static void aluMatrixfFloat3(ALfloat *vec, ALfloat w, const aluMatrixf *mtx)
+{
+    ALfloat v[4] = { vec[0], vec[1], vec[2], w };
+
+    vec[0] = v[0]*mtx->m[0][0] + v[1]*mtx->m[1][0] + v[2]*mtx->m[2][0] + v[3]*mtx->m[3][0];
+    vec[1] = v[0]*mtx->m[0][1] + v[1]*mtx->m[1][1] + v[2]*mtx->m[2][1] + v[3]*mtx->m[3][1];
+    vec[2] = v[0]*mtx->m[0][2] + v[1]*mtx->m[1][2] + v[2]*mtx->m[2][2] + v[3]*mtx->m[3][2];
+}
+
+static aluVector aluMatrixfVector(const aluMatrixf *mtx, const aluVector *vec)
+{
+    aluVector v;
+    v.v[0] = vec->v[0]*mtx->m[0][0] + vec->v[1]*mtx->m[1][0] + vec->v[2]*mtx->m[2][0] + vec->v[3]*mtx->m[3][0];
+    v.v[1] = vec->v[0]*mtx->m[0][1] + vec->v[1]*mtx->m[1][1] + vec->v[2]*mtx->m[2][1] + vec->v[3]*mtx->m[3][1];
+    v.v[2] = vec->v[0]*mtx->m[0][2] + vec->v[1]*mtx->m[1][2] + vec->v[2]*mtx->m[2][2] + vec->v[3]*mtx->m[3][2];
+    v.v[3] = vec->v[0]*mtx->m[0][3] + vec->v[1]*mtx->m[1][3] + vec->v[2]*mtx->m[2][3] + vec->v[3]*mtx->m[3][3];
+    return v;
+}
+
+
+void aluInit(void)
+{
+    MixDirectHrtf = SelectHrtfMixer();
+}
+
+
+static void SendSourceStoppedEvent(ALCcontext *context, ALuint id)
+{
+    AsyncEvent evt = ASYNC_EVENT(EventType_SourceStateChange);
+    ALbitfieldSOFT enabledevt;
+    size_t strpos;
+    ALuint scale;
+
+    enabledevt = ATOMIC_LOAD(&context->EnabledEvts, almemory_order_acquire);
+    if(!(enabledevt&EventType_SourceStateChange)) return;
+
+    evt.u.user.type = AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT;
+    evt.u.user.id = id;
+    evt.u.user.param = AL_STOPPED;
+
+    /* Normally snprintf would be used, but this is called from the mixer and
+     * that function's not real-time safe, so we have to construct it manually.
+     */
+    strcpy(evt.u.user.msg, "Source ID "); strpos = 10;
+    scale = 1000000000;
+    while(scale > 0 && scale > id)
+        scale /= 10;
+    while(scale > 0)
+    {
+        evt.u.user.msg[strpos++] = '0' + ((id/scale)%10);
+        scale /= 10;
+    }
+    strcpy(evt.u.user.msg+strpos, " state changed to AL_STOPPED");
+
+    if(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) == 1)
+        alsem_post(&context->EventSem);
+}
+
+
+static void ProcessHrtf(ALCdevice *device, ALsizei SamplesToDo)
+{
+    DirectHrtfState *state;
+    int lidx, ridx;
+    ALsizei c;
+
+    if(device->AmbiUp)
+        ambiup_process(device->AmbiUp,
+            device->Dry.Buffer, device->Dry.NumChannels, device->FOAOut.Buffer,
+            SamplesToDo
+        );
+
+    lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
+    ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
+    assert(lidx != -1 && ridx != -1);
+
+    state = device->Hrtf;
+    for(c = 0;c < device->Dry.NumChannels;c++)
+    {
+        MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
+            device->Dry.Buffer[c], state->Offset, state->IrSize,
+            state->Chan[c].Coeffs, state->Chan[c].Values, SamplesToDo
+        );
+    }
+    state->Offset += SamplesToDo;
+}
+
+static void ProcessAmbiDec(ALCdevice *device, ALsizei SamplesToDo)
+{
+    if(device->Dry.Buffer != device->FOAOut.Buffer)
+        bformatdec_upSample(device->AmbiDecoder,
+            device->Dry.Buffer, device->FOAOut.Buffer, device->FOAOut.NumChannels,
+            SamplesToDo
+        );
+    bformatdec_process(device->AmbiDecoder,
+        device->RealOut.Buffer, device->RealOut.NumChannels, device->Dry.Buffer,
+        SamplesToDo
+    );
+}
+
+static void ProcessAmbiUp(ALCdevice *device, ALsizei SamplesToDo)
+{
+    ambiup_process(device->AmbiUp,
+        device->RealOut.Buffer, device->RealOut.NumChannels, device->FOAOut.Buffer,
+        SamplesToDo
+    );
+}
+
+static void ProcessUhj(ALCdevice *device, ALsizei SamplesToDo)
+{
+    int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
+    int ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
+    assert(lidx != -1 && ridx != -1);
+
+    /* Encode to stereo-compatible 2-channel UHJ output. */
+    EncodeUhj2(device->Uhj_Encoder,
+        device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
+        device->Dry.Buffer, SamplesToDo
+    );
+}
+
+static void ProcessBs2b(ALCdevice *device, ALsizei SamplesToDo)
+{
+    int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
+    int ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
+    assert(lidx != -1 && ridx != -1);
+
+    /* Apply binaural/crossfeed filter */
+    bs2b_cross_feed(device->Bs2b, device->RealOut.Buffer[lidx],
+                    device->RealOut.Buffer[ridx], SamplesToDo);
+}
+
+void aluSelectPostProcess(ALCdevice *device)
+{
+    if(device->HrtfHandle)
+        device->PostProcess = ProcessHrtf;
+    else if(device->AmbiDecoder)
+        device->PostProcess = ProcessAmbiDec;
+    else if(device->AmbiUp)
+        device->PostProcess = ProcessAmbiUp;
+    else if(device->Uhj_Encoder)
+        device->PostProcess = ProcessUhj;
+    else if(device->Bs2b)
+        device->PostProcess = ProcessBs2b;
+    else
+        device->PostProcess = NULL;
+}
+
+
+/* Prepares the interpolator for a given rate (determined by increment).
+ *
+ * With a bit of work, and a trade of memory for CPU cost, this could be
+ * modified for use with an interpolated increment for buttery-smooth pitch
+ * changes.
+ */
+void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *table)
+{
+    ALfloat sf = 0.0f;
+    ALsizei si = BSINC_SCALE_COUNT-1;
+
+    if(increment > FRACTIONONE)
+    {
+        sf = (ALfloat)FRACTIONONE / increment;
+        sf = maxf(0.0f, (BSINC_SCALE_COUNT-1) * (sf-table->scaleBase) * table->scaleRange);
+        si = float2int(sf);
+        /* The interpolation factor is fit to this diagonally-symmetric curve
+         * to reduce the transition ripple caused by interpolating different
+         * scales of the sinc function.
+         */
+        sf = 1.0f - cosf(asinf(sf - si));
+    }
+
+    state->sf = sf;
+    state->m = table->m[si];
+    state->l = (state->m/2) - 1;
+    state->filter = table->Tab + table->filterOffset[si];
+}
+
+
+static bool CalcContextParams(ALCcontext *Context)
+{
+    ALlistener *Listener = Context->Listener;
+    struct ALcontextProps *props;
+
+    props = static_cast<ALcontextProps*>(ATOMIC_EXCHANGE_PTR(&Context->Update,
+        static_cast<ALcontextProps*>(nullptr), almemory_order_acq_rel));
+    if(!props) return false;
+
+    Listener->Params.MetersPerUnit = props->MetersPerUnit;
+
+    Listener->Params.DopplerFactor = props->DopplerFactor;
+    Listener->Params.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity;
+    if(!OverrideReverbSpeedOfSound)
+        Listener->Params.ReverbSpeedOfSound = Listener->Params.SpeedOfSound *
+                                              Listener->Params.MetersPerUnit;
+
+    Listener->Params.SourceDistanceModel = props->SourceDistanceModel;
+    Listener->Params.DistanceModel = props->DistanceModel;
+
+    ATOMIC_REPLACE_HEAD(struct ALcontextProps*, &Context->FreeContextProps, props);
+    return true;
+}
+
+static bool CalcListenerParams(ALCcontext *Context)
+{
+    ALlistener *Listener = Context->Listener;
+    ALfloat N[3], V[3], U[3], P[3];
+    struct ALlistenerProps *props;
+    aluVector vel;
+
+    props = static_cast<ALlistenerProps*>(ATOMIC_EXCHANGE_PTR(&Listener->Update,
+        static_cast<ALlistenerProps*>(nullptr), almemory_order_acq_rel));
+    if(!props) return false;
+
+    /* AT then UP */
+    N[0] = props->Forward[0];
+    N[1] = props->Forward[1];
+    N[2] = props->Forward[2];
+    aluNormalize(N);
+    V[0] = props->Up[0];
+    V[1] = props->Up[1];
+    V[2] = props->Up[2];
+    aluNormalize(V);
+    /* Build and normalize right-vector */
+    aluCrossproduct(N, V, U);
+    aluNormalize(U);
+
+    aluMatrixfSet(&Listener->Params.Matrix,
+        U[0], V[0], -N[0], 0.0,
+        U[1], V[1], -N[1], 0.0,
+        U[2], V[2], -N[2], 0.0,
+         0.0,  0.0,   0.0, 1.0
+    );
+
+    P[0] = props->Position[0];
+    P[1] = props->Position[1];
+    P[2] = props->Position[2];
+    aluMatrixfFloat3(P, 1.0, &Listener->Params.Matrix);
+    aluMatrixfSetRow(&Listener->Params.Matrix, 3, -P[0], -P[1], -P[2], 1.0f);
+
+    aluVectorSet(&vel, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f);
+    Listener->Params.Velocity = aluMatrixfVector(&Listener->Params.Matrix, &vel);
+
+    Listener->Params.Gain = props->Gain * Context->GainBoost;
+
+    ATOMIC_REPLACE_HEAD(struct ALlistenerProps*, &Context->FreeListenerProps, props);
+    return true;
+}
+
+static bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool force)
+{
+    struct ALeffectslotProps *props;
+    ALeffectState *state;
+
+    props = static_cast<ALeffectslotProps*>(ATOMIC_EXCHANGE_PTR(&slot->Update,
+        static_cast<ALeffectslotProps*>(nullptr), almemory_order_acq_rel));
+    if(!props && !force) return false;
+
+    if(props)
+    {
+        slot->Params.Gain = props->Gain;
+        slot->Params.AuxSendAuto = props->AuxSendAuto;
+        slot->Params.EffectType = props->Type;
+        slot->Params.EffectProps = props->Props;
+        if(IsReverbEffect(props->Type))
+        {
+            slot->Params.RoomRolloff = props->Props.Reverb.RoomRolloffFactor;
+            slot->Params.DecayTime = props->Props.Reverb.DecayTime;
+            slot->Params.DecayLFRatio = props->Props.Reverb.DecayLFRatio;
+            slot->Params.DecayHFRatio = props->Props.Reverb.DecayHFRatio;
+            slot->Params.DecayHFLimit = props->Props.Reverb.DecayHFLimit;
+            slot->Params.AirAbsorptionGainHF = props->Props.Reverb.AirAbsorptionGainHF;
+        }
+        else
+        {
+            slot->Params.RoomRolloff = 0.0f;
+            slot->Params.DecayTime = 0.0f;
+            slot->Params.DecayLFRatio = 0.0f;
+            slot->Params.DecayHFRatio = 0.0f;
+            slot->Params.DecayHFLimit = AL_FALSE;
+            slot->Params.AirAbsorptionGainHF = 1.0f;
+        }
+
+        state = props->State;
+
+        if(state == slot->Params.EffectState)
+        {
+            /* If the effect state is the same as current, we can decrement its
+             * count safely to remove it from the update object (it can't reach
+             * 0 refs since the current params also hold a reference).
+             */
+            DecrementRef(&state->Ref);
+            props->State = NULL;
+        }
+        else
+        {
+            /* Otherwise, replace it and send off the old one with a release
+             * event.
+             */
+            AsyncEvent evt = ASYNC_EVENT(EventType_ReleaseEffectState);
+            evt.u.EffectState = slot->Params.EffectState;
+
+            slot->Params.EffectState = state;
+            props->State = NULL;
+
+            if(LIKELY(ll_ringbuffer_write(context->AsyncEvents, (const char*)&evt, 1) != 0))
+                alsem_post(&context->EventSem);
+            else
+            {
+                /* If writing the event failed, the queue was probably full.
+                 * Store the old state in the property object where it can
+                 * eventually be cleaned up sometime later (not ideal, but
+                 * better than blocking or leaking).
+                 */
+                props->State = evt.u.EffectState;
+            }
+        }
+
+        ATOMIC_REPLACE_HEAD(struct ALeffectslotProps*, &context->FreeEffectslotProps, props);
+    }
+    else
+        state = slot->Params.EffectState;
+
+    V(state,update)(context, slot, &slot->Params.EffectProps);
+    return true;
+}
+
+
+static const struct ChanMap MonoMap[1] = {
+    { FrontCenter, 0.0f, 0.0f }
+}, RearMap[2] = {
+    { BackLeft,  DEG2RAD(-150.0f), DEG2RAD(0.0f) },
+    { BackRight, DEG2RAD( 150.0f), DEG2RAD(0.0f) }
+}, QuadMap[4] = {
+    { FrontLeft,  DEG2RAD( -45.0f), DEG2RAD(0.0f) },
+    { FrontRight, DEG2RAD(  45.0f), DEG2RAD(0.0f) },
+    { BackLeft,   DEG2RAD(-135.0f), DEG2RAD(0.0f) },
+    { BackRight,  DEG2RAD( 135.0f), DEG2RAD(0.0f) }
+}, X51Map[6] = {
+    { FrontLeft,   DEG2RAD( -30.0f), DEG2RAD(0.0f) },
+    { FrontRight,  DEG2RAD(  30.0f), DEG2RAD(0.0f) },
+    { FrontCenter, DEG2RAD(   0.0f), DEG2RAD(0.0f) },
+    { LFE, 0.0f, 0.0f },
+    { SideLeft,    DEG2RAD(-110.0f), DEG2RAD(0.0f) },
+    { SideRight,   DEG2RAD( 110.0f), DEG2RAD(0.0f) }
+}, X61Map[7] = {
+    { FrontLeft,    DEG2RAD(-30.0f), DEG2RAD(0.0f) },
+    { FrontRight,   DEG2RAD( 30.0f), DEG2RAD(0.0f) },
+    { FrontCenter,  DEG2RAD(  0.0f), DEG2RAD(0.0f) },
+    { LFE, 0.0f, 0.0f },
+    { BackCenter,   DEG2RAD(180.0f), DEG2RAD(0.0f) },
+    { SideLeft,     DEG2RAD(-90.0f), DEG2RAD(0.0f) },
+    { SideRight,    DEG2RAD( 90.0f), DEG2RAD(0.0f) }
+}, X71Map[8] = {
+    { FrontLeft,   DEG2RAD( -30.0f), DEG2RAD(0.0f) },
+    { FrontRight,  DEG2RAD(  30.0f), DEG2RAD(0.0f) },
+    { FrontCenter, DEG2RAD(   0.0f), DEG2RAD(0.0f) },
+    { LFE, 0.0f, 0.0f },
+    { BackLeft,    DEG2RAD(-150.0f), DEG2RAD(0.0f) },
+    { BackRight,   DEG2RAD( 150.0f), DEG2RAD(0.0f) },
+    { SideLeft,    DEG2RAD( -90.0f), DEG2RAD(0.0f) },
+    { SideRight,   DEG2RAD(  90.0f), DEG2RAD(0.0f) }
+};
+
+static void CalcPanningAndFilters(ALvoice *voice, const ALfloat Azi, const ALfloat Elev,
+                                  const ALfloat Distance, const ALfloat Spread,
+                                  const ALfloat DryGain, const ALfloat DryGainHF,
+                                  const ALfloat DryGainLF, const ALfloat *WetGain,
+                                  const ALfloat *WetGainLF, const ALfloat *WetGainHF,
+                                  ALeffectslot **SendSlots, const ALbuffer *Buffer,
+                                  const struct ALvoiceProps *props, const ALlistener *Listener,
+                                  const ALCdevice *Device)
+{
+    struct ChanMap StereoMap[2] = {
+        { FrontLeft,  DEG2RAD(-30.0f), DEG2RAD(0.0f) },
+        { FrontRight, DEG2RAD( 30.0f), DEG2RAD(0.0f) }
+    };
+    bool DirectChannels = props->DirectChannels;
+    const ALsizei NumSends = Device->NumAuxSends;
+    const ALuint Frequency = Device->Frequency;
+    const struct ChanMap *chans = NULL;
+    ALsizei num_channels = 0;
+    bool isbformat = false;
+    ALfloat downmix_gain = 1.0f;
+    ALsizei c, i;
+
+    switch(Buffer->FmtChannels)
+    {
+    case FmtMono:
+        chans = MonoMap;
+        num_channels = 1;
+        /* Mono buffers are never played direct. */
+        DirectChannels = false;
+        break;
+
+    case FmtStereo:
+        /* Convert counter-clockwise to clockwise. */
+        StereoMap[0].angle = -props->StereoPan[0];
+        StereoMap[1].angle = -props->StereoPan[1];
+
+        chans = StereoMap;
+        num_channels = 2;
+        downmix_gain = 1.0f / 2.0f;
+        break;
+
+    case FmtRear:
+        chans = RearMap;
+        num_channels = 2;
+        downmix_gain = 1.0f / 2.0f;
+        break;
+
+    case FmtQuad:
+        chans = QuadMap;
+        num_channels = 4;
+        downmix_gain = 1.0f / 4.0f;
+        break;
+
+    case FmtX51:
+        chans = X51Map;
+        num_channels = 6;
+        /* NOTE: Excludes LFE. */
+        downmix_gain = 1.0f / 5.0f;
+        break;
+
+    case FmtX61:
+        chans = X61Map;
+        num_channels = 7;
+        /* NOTE: Excludes LFE. */
+        downmix_gain = 1.0f / 6.0f;
+        break;
+
+    case FmtX71:
+        chans = X71Map;
+        num_channels = 8;
+        /* NOTE: Excludes LFE. */
+        downmix_gain = 1.0f / 7.0f;
+        break;
+
+    case FmtBFormat2D:
+        num_channels = 3;
+        isbformat = true;
+        DirectChannels = false;
+        break;
+
+    case FmtBFormat3D:
+        num_channels = 4;
+        isbformat = true;
+        DirectChannels = false;
+        break;
+    }
+
+    for(c = 0;c < num_channels;c++)
+    {
+        memset(&voice->Direct.Params[c].Hrtf.Target, 0,
+               sizeof(voice->Direct.Params[c].Hrtf.Target));
+        ClearArray(voice->Direct.Params[c].Gains.Target);
+    }
+    for(i = 0;i < NumSends;i++)
+    {
+        for(c = 0;c < num_channels;c++)
+            ClearArray(voice->Send[i].Params[c].Gains.Target);
+    }
+
+    voice->Flags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC);
+    if(isbformat)
+    {
+        /* Special handling for B-Format sources. */
+
+        if(Distance > FLT_EPSILON)
+        {
+            /* Panning a B-Format sound toward some direction is easy. Just pan
+             * the first (W) channel as a normal mono sound and silence the
+             * others.
+             */
+            ALfloat coeffs[MAX_AMBI_COEFFS];
+
+            if(Device->AvgSpeakerDist > 0.0f)
+            {
+                ALfloat mdist = Distance * Listener->Params.MetersPerUnit;
+                ALfloat w0 = SPEEDOFSOUNDMETRESPERSEC /
+                             (mdist * (ALfloat)Device->Frequency);
+                ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC /
+                             (Device->AvgSpeakerDist * (ALfloat)Device->Frequency);
+                /* Clamp w0 for really close distances, to prevent excessive
+                 * bass.
+                 */
+                w0 = minf(w0, w1*4.0f);
+
+                /* Only need to adjust the first channel of a B-Format source. */
+                NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, w0);
+
+                for(i = 0;i < MAX_AMBI_ORDER+1;i++)
+                    voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i];
+                voice->Flags |= VOICE_HAS_NFC;
+            }
+
+            /* A scalar of 1.5 for plain stereo results in +/-60 degrees being
+             * moved to +/-90 degrees for direct right and left speaker
+             * responses.
+             */
+            CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi,
+                            Elev, Spread, coeffs);
+
+            /* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */
+            ComputePanGains(&Device->Dry, coeffs, DryGain*SQRTF_2,
+                               voice->Direct.Params[0].Gains.Target);
+            for(i = 0;i < NumSends;i++)
+            {
+                const ALeffectslot *Slot = SendSlots[i];
+                if(Slot)
+                    ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels, coeffs,
+                        WetGain[i]*SQRTF_2, voice->Send[i].Params[0].Gains.Target
+                    );
+            }
+        }
+        else
+        {
+            /* Local B-Format sources have their XYZ channels rotated according
+             * to the orientation.
+             */
+            ALfloat N[3], V[3], U[3];
+            aluMatrixf matrix;
+
+            if(Device->AvgSpeakerDist > 0.0f)
+            {
+                /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
+                 * is what we want for FOA input. The first channel may have
+                 * been previously re-adjusted if panned, so reset it.
+                 */
+                NfcFilterAdjust(&voice->Direct.Params[0].NFCtrlFilter, 0.0f);
+
+                voice->Direct.ChannelsPerOrder[0] = 1;
+                voice->Direct.ChannelsPerOrder[1] = mini(voice->Direct.Channels-1, 3);
+                for(i = 2;i < MAX_AMBI_ORDER+1;i++)
+                    voice->Direct.ChannelsPerOrder[i] = 0;
+                voice->Flags |= VOICE_HAS_NFC;
+            }
+
+            /* AT then UP */
+            N[0] = props->Orientation[0][0];
+            N[1] = props->Orientation[0][1];
+            N[2] = props->Orientation[0][2];
+            aluNormalize(N);
+            V[0] = props->Orientation[1][0];
+            V[1] = props->Orientation[1][1];
+            V[2] = props->Orientation[1][2];
+            aluNormalize(V);
+            if(!props->HeadRelative)
+            {
+                const aluMatrixf *lmatrix = &Listener->Params.Matrix;
+                aluMatrixfFloat3(N, 0.0f, lmatrix);
+                aluMatrixfFloat3(V, 0.0f, lmatrix);
+            }
+            /* Build and normalize right-vector */
+            aluCrossproduct(N, V, U);
+            aluNormalize(U);
+
+            /* Build a rotate + conversion matrix (FuMa -> ACN+N3D). NOTE: This
+             * matrix is transposed, for the inputs to align on the rows and
+             * outputs on the columns.
+             */
+            aluMatrixfSet(&matrix,
+                // ACN0           ACN1           ACN2           ACN3
+                SQRTF_2,          0.0f,          0.0f,          0.0f, // Ambi W
+                   0.0f, -N[0]*SQRTF_3,  N[1]*SQRTF_3, -N[2]*SQRTF_3, // Ambi X
+                   0.0f,  U[0]*SQRTF_3, -U[1]*SQRTF_3,  U[2]*SQRTF_3, // Ambi Y
+                   0.0f, -V[0]*SQRTF_3,  V[1]*SQRTF_3, -V[2]*SQRTF_3  // Ambi Z
+            );
+
+            voice->Direct.Buffer = Device->FOAOut.Buffer;
+            voice->Direct.Channels = Device->FOAOut.NumChannels;
+            for(c = 0;c < num_channels;c++)
+                ComputePanGains(&Device->FOAOut, matrix.m[c], DryGain,
+                                voice->Direct.Params[c].Gains.Target);
+            for(i = 0;i < NumSends;i++)
+            {
+                const ALeffectslot *Slot = SendSlots[i];
+                if(Slot)
+                {
+                    for(c = 0;c < num_channels;c++)
+                        ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
+                            matrix.m[c], WetGain[i], voice->Send[i].Params[c].Gains.Target
+                        );
+                }
+            }
+        }
+    }
+    else if(DirectChannels)
+    {
+        /* Direct source channels always play local. Skip the virtual channels
+         * and write inputs to the matching real outputs.
+         */
+        voice->Direct.Buffer = Device->RealOut.Buffer;
+        voice->Direct.Channels = Device->RealOut.NumChannels;
+
+        for(c = 0;c < num_channels;c++)
+        {
+            int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel);
+            if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain;
+        }
+
+        /* Auxiliary sends still use normal channel panning since they mix to
+         * B-Format, which can't channel-match.
+         */
+        for(c = 0;c < num_channels;c++)
+        {
+            ALfloat coeffs[MAX_AMBI_COEFFS];
+            CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs);
+
+            for(i = 0;i < NumSends;i++)
+            {
+                const ALeffectslot *Slot = SendSlots[i];
+                if(Slot)
+                    ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
+                        coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target
+                    );
+            }
+        }
+    }
+    else if(Device->Render_Mode == HrtfRender)
+    {
+        /* Full HRTF rendering. Skip the virtual channels and render to the
+         * real outputs.
+         */
+        voice->Direct.Buffer = Device->RealOut.Buffer;
+        voice->Direct.Channels = Device->RealOut.NumChannels;
+
+        if(Distance > FLT_EPSILON)
+        {
+            ALfloat coeffs[MAX_AMBI_COEFFS];
+
+            /* Get the HRIR coefficients and delays just once, for the given
+             * source direction.
+             */
+            GetHrtfCoeffs(Device->HrtfHandle, Elev, Azi, Spread,
+                          voice->Direct.Params[0].Hrtf.Target.Coeffs,
+                          voice->Direct.Params[0].Hrtf.Target.Delay);
+            voice->Direct.Params[0].Hrtf.Target.Gain = DryGain * downmix_gain;
+
+            /* Remaining channels use the same results as the first. */
+            for(c = 1;c < num_channels;c++)
+            {
+                /* Skip LFE */
+                if(chans[c].channel != LFE)
+                    voice->Direct.Params[c].Hrtf.Target = voice->Direct.Params[0].Hrtf.Target;
+            }
+
+            /* Calculate the directional coefficients once, which apply to all
+             * input channels of the source sends.
+             */
+            CalcAngleCoeffs(Azi, Elev, Spread, coeffs);
+
+            for(i = 0;i < NumSends;i++)
+            {
+                const ALeffectslot *Slot = SendSlots[i];
+                if(Slot)
+                    for(c = 0;c < num_channels;c++)
+                    {
+                        /* Skip LFE */
+                        if(chans[c].channel != LFE)
+                            ComputePanningGainsBF(Slot->ChanMap,
+                                Slot->NumChannels, coeffs, WetGain[i] * downmix_gain,
+                                voice->Send[i].Params[c].Gains.Target
+                            );
+                    }
+            }
+        }
+        else
+        {
+            /* Local sources on HRTF play with each channel panned to its
+             * relative location around the listener, providing "virtual
+             * speaker" responses.
+             */
+            for(c = 0;c < num_channels;c++)
+            {
+                ALfloat coeffs[MAX_AMBI_COEFFS];
+
+                if(chans[c].channel == LFE)
+                {
+                    /* Skip LFE */
+                    continue;
+                }
+
+                /* Get the HRIR coefficients and delays for this channel
+                 * position.
+                 */
+                GetHrtfCoeffs(Device->HrtfHandle,
+                    chans[c].elevation, chans[c].angle, Spread,
+                    voice->Direct.Params[c].Hrtf.Target.Coeffs,
+                    voice->Direct.Params[c].Hrtf.Target.Delay
+                );
+                voice->Direct.Params[c].Hrtf.Target.Gain = DryGain;
+
+                /* Normal panning for auxiliary sends. */
+                CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs);
+
+                for(i = 0;i < NumSends;i++)
+                {
+                    const ALeffectslot *Slot = SendSlots[i];
+                    if(Slot)
+                        ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
+                            coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target
+                        );
+                }
+            }
+        }
+
+        voice->Flags |= VOICE_HAS_HRTF;
+    }
+    else
+    {
+        /* Non-HRTF rendering. Use normal panning to the output. */
+
+        if(Distance > FLT_EPSILON)
+        {
+            ALfloat coeffs[MAX_AMBI_COEFFS];
+            ALfloat w0 = 0.0f;
+
+            /* Calculate NFC filter coefficient if needed. */
+            if(Device->AvgSpeakerDist > 0.0f)
+            {
+                ALfloat mdist = Distance * Listener->Params.MetersPerUnit;
+                ALfloat w1 = SPEEDOFSOUNDMETRESPERSEC /
+                             (Device->AvgSpeakerDist * (ALfloat)Device->Frequency);
+                w0 = SPEEDOFSOUNDMETRESPERSEC /
+                     (mdist * (ALfloat)Device->Frequency);
+                /* Clamp w0 for really close distances, to prevent excessive
+                 * bass.
+                 */
+                w0 = minf(w0, w1*4.0f);
+
+                /* Adjust NFC filters. */
+                for(c = 0;c < num_channels;c++)
+                    NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0);
+
+                for(i = 0;i < MAX_AMBI_ORDER+1;i++)
+                    voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i];
+                voice->Flags |= VOICE_HAS_NFC;
+            }
+
+            /* Calculate the directional coefficients once, which apply to all
+             * input channels.
+             */
+            CalcAngleCoeffs((Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(Azi, 1.5f) : Azi,
+                            Elev, Spread, coeffs);
+
+            for(c = 0;c < num_channels;c++)
+            {
+                /* Special-case LFE */
+                if(chans[c].channel == LFE)
+                {
+                    if(Device->Dry.Buffer == Device->RealOut.Buffer)
+                    {
+                        int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel);
+                        if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain;
+                    }
+                    continue;
+                }
+
+                ComputePanGains(&Device->Dry, coeffs, DryGain * downmix_gain,
+                                voice->Direct.Params[c].Gains.Target);
+            }
+
+            for(i = 0;i < NumSends;i++)
+            {
+                const ALeffectslot *Slot = SendSlots[i];
+                if(Slot)
+                    for(c = 0;c < num_channels;c++)
+                    {
+                        /* Skip LFE */
+                        if(chans[c].channel != LFE)
+                            ComputePanningGainsBF(Slot->ChanMap,
+                                Slot->NumChannels, coeffs, WetGain[i] * downmix_gain,
+                                voice->Send[i].Params[c].Gains.Target
+                            );
+                    }
+            }
+        }
+        else
+        {
+            ALfloat w0 = 0.0f;
+
+            if(Device->AvgSpeakerDist > 0.0f)
+            {
+                /* If the source distance is 0, set w0 to w1 to act as a pass-
+                 * through. We still want to pass the signal through the
+                 * filters so they keep an appropriate history, in case the
+                 * source moves away from the listener.
+                 */
+                w0 = SPEEDOFSOUNDMETRESPERSEC /
+                     (Device->AvgSpeakerDist * (ALfloat)Device->Frequency);
+
+                for(c = 0;c < num_channels;c++)
+                    NfcFilterAdjust(&voice->Direct.Params[c].NFCtrlFilter, w0);
+
+                for(i = 0;i < MAX_AMBI_ORDER+1;i++)
+                    voice->Direct.ChannelsPerOrder[i] = Device->NumChannelsPerOrder[i];
+                voice->Flags |= VOICE_HAS_NFC;
+            }
+
+            for(c = 0;c < num_channels;c++)
+            {
+                ALfloat coeffs[MAX_AMBI_COEFFS];
+
+                /* Special-case LFE */
+                if(chans[c].channel == LFE)
+                {
+                    if(Device->Dry.Buffer == Device->RealOut.Buffer)
+                    {
+                        int idx = GetChannelIdxByName(&Device->RealOut, chans[c].channel);
+                        if(idx != -1) voice->Direct.Params[c].Gains.Target[idx] = DryGain;
+                    }
+                    continue;
+                }
+
+                CalcAngleCoeffs(
+                    (Device->Render_Mode==StereoPair) ? ScaleAzimuthFront(chans[c].angle, 3.0f)
+                                                      : chans[c].angle,
+                    chans[c].elevation, Spread, coeffs
+                );
+
+                ComputePanGains(&Device->Dry, coeffs, DryGain,
+                                voice->Direct.Params[c].Gains.Target);
+                for(i = 0;i < NumSends;i++)
+                {
+                    const ALeffectslot *Slot = SendSlots[i];
+                    if(Slot)
+                        ComputePanningGainsBF(Slot->ChanMap, Slot->NumChannels,
+                            coeffs, WetGain[i], voice->Send[i].Params[c].Gains.Target
+                        );
+                }
+            }
+        }
+    }
+
+    {
+        ALfloat hfScale = props->Direct.HFReference / Frequency;
+        ALfloat lfScale = props->Direct.LFReference / Frequency;
+        ALfloat gainHF = maxf(DryGainHF, 0.001f); /* Limit -60dB */
+        ALfloat gainLF = maxf(DryGainLF, 0.001f);
+
+        voice->Direct.FilterType = AF_None;
+        if(gainHF != 1.0f) voice->Direct.FilterType |= AF_LowPass;
+        if(gainLF != 1.0f) voice->Direct.FilterType |= AF_HighPass;
+        BiquadFilter_setParams(
+            &voice->Direct.Params[0].LowPass, BiquadType_HighShelf,
+            gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f)
+        );
+        BiquadFilter_setParams(
+            &voice->Direct.Params[0].HighPass, BiquadType_LowShelf,
+            gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f)
+        );
+        for(c = 1;c < num_channels;c++)
+        {
+            BiquadFilter_copyParams(&voice->Direct.Params[c].LowPass,
+                                    &voice->Direct.Params[0].LowPass);
+            BiquadFilter_copyParams(&voice->Direct.Params[c].HighPass,
+                                    &voice->Direct.Params[0].HighPass);
+        }
+    }
+    for(i = 0;i < NumSends;i++)
+    {
+        ALfloat hfScale = props->Send[i].HFReference / Frequency;
+        ALfloat lfScale = props->Send[i].LFReference / Frequency;
+        ALfloat gainHF = maxf(WetGainHF[i], 0.001f);
+        ALfloat gainLF = maxf(WetGainLF[i], 0.001f);
+
+        voice->Send[i].FilterType = AF_None;
+        if(gainHF != 1.0f) voice->Send[i].FilterType |= AF_LowPass;
+        if(gainLF != 1.0f) voice->Send[i].FilterType |= AF_HighPass;
+        BiquadFilter_setParams(
+            &voice->Send[i].Params[0].LowPass, BiquadType_HighShelf,
+            gainHF, hfScale, calc_rcpQ_from_slope(gainHF, 1.0f)
+        );
+        BiquadFilter_setParams(
+            &voice->Send[i].Params[0].HighPass, BiquadType_LowShelf,
+            gainLF, lfScale, calc_rcpQ_from_slope(gainLF, 1.0f)
+        );
+        for(c = 1;c < num_channels;c++)
+        {
+            BiquadFilter_copyParams(&voice->Send[i].Params[c].LowPass,
+                                    &voice->Send[i].Params[0].LowPass);
+            BiquadFilter_copyParams(&voice->Send[i].Params[c].HighPass,
+                                    &voice->Send[i].Params[0].HighPass);
+        }
+    }
+}
+
+static void CalcNonAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext)
+{
+    const ALCdevice *Device = ALContext->Device;
+    const ALlistener *Listener = ALContext->Listener;
+    ALfloat DryGain, DryGainHF, DryGainLF;
+    ALfloat WetGain[MAX_SENDS];
+    ALfloat WetGainHF[MAX_SENDS];
+    ALfloat WetGainLF[MAX_SENDS];
+    ALeffectslot *SendSlots[MAX_SENDS];
+    ALfloat Pitch;
+    ALsizei i;
+
+    voice->Direct.Buffer = Device->Dry.Buffer;
+    voice->Direct.Channels = Device->Dry.NumChannels;
+    for(i = 0;i < Device->NumAuxSends;i++)
+    {
+        SendSlots[i] = props->Send[i].Slot;
+        if(!SendSlots[i] && i == 0)
+            SendSlots[i] = ALContext->DefaultSlot;
+        if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL)
+        {
+            SendSlots[i] = NULL;
+            voice->Send[i].Buffer = NULL;
+            voice->Send[i].Channels = 0;
+        }
+        else
+        {
+            voice->Send[i].Buffer = SendSlots[i]->WetBuffer;
+            voice->Send[i].Channels = SendSlots[i]->NumChannels;
+        }
+    }
+
+    /* Calculate the stepping value */
+    Pitch = (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency * props->Pitch;
+    if(Pitch > (ALfloat)MAX_PITCH)
+        voice->Step = MAX_PITCH<<FRACTIONBITS;
+    else
+        voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1);
+    if(props->Resampler == BSinc24Resampler)
+        BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24);
+    else if(props->Resampler == BSinc12Resampler)
+        BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12);
+    voice->Resampler = SelectResampler(props->Resampler);
+
+    /* Calculate gains */
+    DryGain  = clampf(props->Gain, props->MinGain, props->MaxGain);
+    DryGain *= props->Direct.Gain * Listener->Params.Gain;
+    DryGain  = minf(DryGain, GAIN_MIX_MAX);
+    DryGainHF = props->Direct.GainHF;
+    DryGainLF = props->Direct.GainLF;
+    for(i = 0;i < Device->NumAuxSends;i++)
+    {
+        WetGain[i]  = clampf(props->Gain, props->MinGain, props->MaxGain);
+        WetGain[i] *= props->Send[i].Gain * Listener->Params.Gain;
+        WetGain[i]  = minf(WetGain[i], GAIN_MIX_MAX);
+        WetGainHF[i] = props->Send[i].GainHF;
+        WetGainLF[i] = props->Send[i].GainLF;
+    }
+
+    CalcPanningAndFilters(voice, 0.0f, 0.0f, 0.0f, 0.0f, DryGain, DryGainHF, DryGainLF, WetGain,
+                          WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device);
+}
+
+static void CalcAttnSourceParams(ALvoice *voice, const struct ALvoiceProps *props, const ALbuffer *ALBuffer, const ALCcontext *ALContext)
+{
+    const ALCdevice *Device = ALContext->Device;
+    const ALlistener *Listener = ALContext->Listener;
+    const ALsizei NumSends = Device->NumAuxSends;
+    aluVector Position, Velocity, Direction, SourceToListener;
+    ALfloat Distance, ClampedDist, DopplerFactor;
+    ALeffectslot *SendSlots[MAX_SENDS];
+    ALfloat RoomRolloff[MAX_SENDS];
+    ALfloat DecayDistance[MAX_SENDS];
+    ALfloat DecayLFDistance[MAX_SENDS];
+    ALfloat DecayHFDistance[MAX_SENDS];
+    ALfloat DryGain, DryGainHF, DryGainLF;
+    ALfloat WetGain[MAX_SENDS];
+    ALfloat WetGainHF[MAX_SENDS];
+    ALfloat WetGainLF[MAX_SENDS];
+    bool directional;
+    ALfloat ev, az;
+    ALfloat spread;
+    ALfloat Pitch;
+    ALint i;
+
+    /* Set mixing buffers and get send parameters. */
+    voice->Direct.Buffer = Device->Dry.Buffer;
+    voice->Direct.Channels = Device->Dry.NumChannels;
+    for(i = 0;i < NumSends;i++)
+    {
+        SendSlots[i] = props->Send[i].Slot;
+        if(!SendSlots[i] && i == 0)
+            SendSlots[i] = ALContext->DefaultSlot;
+        if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL)
+        {
+            SendSlots[i] = NULL;
+            RoomRolloff[i] = 0.0f;
+            DecayDistance[i] = 0.0f;
+            DecayLFDistance[i] = 0.0f;
+            DecayHFDistance[i] = 0.0f;
+        }
+        else if(SendSlots[i]->Params.AuxSendAuto)
+        {
+            RoomRolloff[i] = SendSlots[i]->Params.RoomRolloff + props->RoomRolloffFactor;
+            /* Calculate the distances to where this effect's decay reaches
+             * -60dB.
+             */
+            DecayDistance[i] = SendSlots[i]->Params.DecayTime *
+                               Listener->Params.ReverbSpeedOfSound;
+            DecayLFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayLFRatio;
+            DecayHFDistance[i] = DecayDistance[i] * SendSlots[i]->Params.DecayHFRatio;
+            if(SendSlots[i]->Params.DecayHFLimit)
+            {
+                ALfloat airAbsorption = SendSlots[i]->Params.AirAbsorptionGainHF;
+                if(airAbsorption < 1.0f)
+                {
+                    /* Calculate the distance to where this effect's air
+                     * absorption reaches -60dB, and limit the effect's HF
+                     * decay distance (so it doesn't take any longer to decay
+                     * than the air would allow).
+                     */
+                    ALfloat absorb_dist = log10f(REVERB_DECAY_GAIN) / log10f(airAbsorption);
+                    DecayHFDistance[i] = minf(absorb_dist, DecayHFDistance[i]);
+                }
+            }
+        }
+        else
+        {
+            /* If the slot's auxiliary send auto is off, the data sent to the
+             * effect slot is the same as the dry path, sans filter effects */
+            RoomRolloff[i] = props->RolloffFactor;
+            DecayDistance[i] = 0.0f;
+            DecayLFDistance[i] = 0.0f;
+            DecayHFDistance[i] = 0.0f;
+        }
+
+        if(!SendSlots[i])
+        {
+            voice->Send[i].Buffer = NULL;
+            voice->Send[i].Channels = 0;
+        }
+        else
+        {
+            voice->Send[i].Buffer = SendSlots[i]->WetBuffer;
+            voice->Send[i].Channels = SendSlots[i]->NumChannels;
+        }
+    }
+
+    /* Transform source to listener space (convert to head relative) */
+    aluVectorSet(&Position, props->Position[0], props->Position[1], props->Position[2], 1.0f);
+    aluVectorSet(&Direction, props->Direction[0], props->Direction[1], props->Direction[2], 0.0f);
+    aluVectorSet(&Velocity, props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f);
+    if(props->HeadRelative == AL_FALSE)
+    {
+        const aluMatrixf *Matrix = &Listener->Params.Matrix;
+        /* Transform source vectors */
+        Position = aluMatrixfVector(Matrix, &Position);
+        Velocity = aluMatrixfVector(Matrix, &Velocity);
+        Direction = aluMatrixfVector(Matrix, &Direction);
+    }
+    else
+    {
+        const aluVector *lvelocity = &Listener->Params.Velocity;
+        /* Offset the source velocity to be relative of the listener velocity */
+        Velocity.v[0] += lvelocity->v[0];
+        Velocity.v[1] += lvelocity->v[1];
+        Velocity.v[2] += lvelocity->v[2];
+    }
+
+    directional = aluNormalize(Direction.v) > 0.0f;
+    SourceToListener.v[0] = -Position.v[0];
+    SourceToListener.v[1] = -Position.v[1];
+    SourceToListener.v[2] = -Position.v[2];
+    SourceToListener.v[3] = 0.0f;
+    Distance = aluNormalize(SourceToListener.v);
+
+    /* Initial source gain */
+    DryGain = props->Gain;
+    DryGainHF = 1.0f;
+    DryGainLF = 1.0f;
+    for(i = 0;i < NumSends;i++)
+    {
+        WetGain[i] = props->Gain;
+        WetGainHF[i] = 1.0f;
+        WetGainLF[i] = 1.0f;
+    }
+
+    /* Calculate distance attenuation */
+    ClampedDist = Distance;
+
+    switch(Listener->Params.SourceDistanceModel ?
+           props->DistanceModel : Listener->Params.DistanceModel)
+    {
+        case InverseDistanceClamped:
+            ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
+            if(props->MaxDistance < props->RefDistance)
+                break;
+            /*fall-through*/
+        case InverseDistance:
+            if(!(props->RefDistance > 0.0f))
+                ClampedDist = props->RefDistance;
+            else
+            {
+                ALfloat dist = lerp(props->RefDistance, ClampedDist, props->RolloffFactor);
+                if(dist > 0.0f) DryGain *= props->RefDistance / dist;
+                for(i = 0;i < NumSends;i++)
+                {
+                    dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]);
+                    if(dist > 0.0f) WetGain[i] *= props->RefDistance / dist;
+                }
+            }
+            break;
+
+        case LinearDistanceClamped:
+            ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
+            if(props->MaxDistance < props->RefDistance)
+                break;
+            /*fall-through*/
+        case LinearDistance:
+            if(!(props->MaxDistance != props->RefDistance))
+                ClampedDist = props->RefDistance;
+            else
+            {
+                ALfloat attn = props->RolloffFactor * (ClampedDist-props->RefDistance) /
+                               (props->MaxDistance-props->RefDistance);
+                DryGain *= maxf(1.0f - attn, 0.0f);
+                for(i = 0;i < NumSends;i++)
+                {
+                    attn = RoomRolloff[i] * (ClampedDist-props->RefDistance) /
+                           (props->MaxDistance-props->RefDistance);
+                    WetGain[i] *= maxf(1.0f - attn, 0.0f);
+                }
+            }
+            break;
+
+        case ExponentDistanceClamped:
+            ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
+            if(props->MaxDistance < props->RefDistance)
+                break;
+            /*fall-through*/
+        case ExponentDistance:
+            if(!(ClampedDist > 0.0f && props->RefDistance > 0.0f))
+                ClampedDist = props->RefDistance;
+            else
+            {
+                DryGain *= powf(ClampedDist/props->RefDistance, -props->RolloffFactor);
+                for(i = 0;i < NumSends;i++)
+                    WetGain[i] *= powf(ClampedDist/props->RefDistance, -RoomRolloff[i]);
+            }
+            break;
+
+        case DisableDistance:
+            ClampedDist = props->RefDistance;
+            break;
+    }
+
+    /* Calculate directional soundcones */
+    if(directional && props->InnerAngle < 360.0f)
+    {
+        ALfloat ConeVolume;
+        ALfloat ConeHF;
+        ALfloat Angle;
+
+        Angle = acosf(aluDotproduct(&Direction, &SourceToListener));
+        Angle = RAD2DEG(Angle * ConeScale * 2.0f);
+        if(!(Angle > props->InnerAngle))
+        {
+            ConeVolume = 1.0f;
+            ConeHF = 1.0f;
+        }
+        else if(Angle < props->OuterAngle)
+        {
+            ALfloat scale = (            Angle-props->InnerAngle) /
+                            (props->OuterAngle-props->InnerAngle);
+            ConeVolume = lerp(1.0f, props->OuterGain, scale);
+            ConeHF = lerp(1.0f, props->OuterGainHF, scale);
+        }
+        else
+        {
+            ConeVolume = props->OuterGain;
+            ConeHF = props->OuterGainHF;
+        }
+
+        DryGain *= ConeVolume;
+        if(props->DryGainHFAuto)
+            DryGainHF *= ConeHF;
+        if(props->WetGainAuto)
+        {
+            for(i = 0;i < NumSends;i++)
+                WetGain[i] *= ConeVolume;
+        }
+        if(props->WetGainHFAuto)
+        {
+            for(i = 0;i < NumSends;i++)
+                WetGainHF[i] *= ConeHF;
+        }
+    }
+
+    /* Apply gain and frequency filters */
+    DryGain  = clampf(DryGain, props->MinGain, props->MaxGain);
+    DryGain  = minf(DryGain*props->Direct.Gain*Listener->Params.Gain, GAIN_MIX_MAX);
+    DryGainHF *= props->Direct.GainHF;
+    DryGainLF *= props->Direct.GainLF;
+    for(i = 0;i < NumSends;i++)
+    {
+        WetGain[i]  = clampf(WetGain[i], props->MinGain, props->MaxGain);
+        WetGain[i]  = minf(WetGain[i]*props->Send[i].Gain*Listener->Params.Gain, GAIN_MIX_MAX);
+        WetGainHF[i] *= props->Send[i].GainHF;
+        WetGainLF[i] *= props->Send[i].GainLF;
+    }
+
+    /* Distance-based air absorption and initial send decay. */
+    if(ClampedDist > props->RefDistance && props->RolloffFactor > 0.0f)
+    {
+        ALfloat meters_base = (ClampedDist-props->RefDistance) * props->RolloffFactor *
+                              Listener->Params.MetersPerUnit;
+        if(props->AirAbsorptionFactor > 0.0f)
+        {
+            ALfloat hfattn = powf(AIRABSORBGAINHF, meters_base * props->AirAbsorptionFactor);
+            DryGainHF *= hfattn;
+            for(i = 0;i < NumSends;i++)
+                WetGainHF[i] *= hfattn;
+        }
+
+        if(props->WetGainAuto)
+        {
+            /* Apply a decay-time transformation to the wet path, based on the
+             * source distance in meters. The initial decay of the reverb
+             * effect is calculated and applied to the wet path.
+             */
+            for(i = 0;i < NumSends;i++)
+            {
+                ALfloat gain, gainhf, gainlf;
+
+                if(!(DecayDistance[i] > 0.0f))
+                    continue;
+
+                gain = powf(REVERB_DECAY_GAIN, meters_base/DecayDistance[i]);
+                WetGain[i] *= gain;
+                /* Yes, the wet path's air absorption is applied with
+                 * WetGainAuto on, rather than WetGainHFAuto.
+                 */
+                if(gain > 0.0f)
+                {
+                    gainhf = powf(REVERB_DECAY_GAIN, meters_base/DecayHFDistance[i]);
+                    WetGainHF[i] *= minf(gainhf / gain, 1.0f);
+                    gainlf = powf(REVERB_DECAY_GAIN, meters_base/DecayLFDistance[i]);
+                    WetGainLF[i] *= minf(gainlf / gain, 1.0f);
+                }
+            }
+        }
+    }
+
+
+    /* Initial source pitch */
+    Pitch = props->Pitch;
+
+    /* Calculate velocity-based doppler effect */
+    DopplerFactor = props->DopplerFactor * Listener->Params.DopplerFactor;
+    if(DopplerFactor > 0.0f)
+    {
+        const aluVector *lvelocity = &Listener->Params.Velocity;
+        const ALfloat SpeedOfSound = Listener->Params.SpeedOfSound;
+        ALfloat vss, vls;
+
+        vss = aluDotproduct(&Velocity, &SourceToListener) * DopplerFactor;
+        vls = aluDotproduct(lvelocity, &SourceToListener) * DopplerFactor;
+
+        if(!(vls < SpeedOfSound))
+        {
+            /* Listener moving away from the source at the speed of sound.
+             * Sound waves can't catch it.
+             */
+            Pitch = 0.0f;
+        }
+        else if(!(vss < SpeedOfSound))
+        {
+            /* Source moving toward the listener at the speed of sound. Sound
+             * waves bunch up to extreme frequencies.
+             */
+            Pitch = HUGE_VALF;
+        }
+        else
+        {
+            /* Source and listener movement is nominal. Calculate the proper
+             * doppler shift.
+             */
+            Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss);
+        }
+    }
+
+    /* Adjust pitch based on the buffer and output frequencies, and calculate
+     * fixed-point stepping value.
+     */
+    Pitch *= (ALfloat)ALBuffer->Frequency/(ALfloat)Device->Frequency;
+    if(Pitch > (ALfloat)MAX_PITCH)
+        voice->Step = MAX_PITCH<<FRACTIONBITS;
+    else
+        voice->Step = maxi(fastf2i(Pitch * FRACTIONONE), 1);
+    if(props->Resampler == BSinc24Resampler)
+        BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc24);
+    else if(props->Resampler == BSinc12Resampler)
+        BsincPrepare(voice->Step, &voice->ResampleState.bsinc, &bsinc12);
+    voice->Resampler = SelectResampler(props->Resampler);
+
+    if(Distance > 0.0f)
+    {
+        /* Clamp Y, in case rounding errors caused it to end up outside of
+         * -1...+1.
+         */
+        ev = asinf(clampf(-SourceToListener.v[1], -1.0f, 1.0f));
+        /* Double negation on Z cancels out; negate once for changing source-
+         * to-listener to listener-to-source, and again for right-handed coords
+         * with -Z in front.
+         */
+        az = atan2f(-SourceToListener.v[0], SourceToListener.v[2]*ZScale);
+    }
+    else
+        ev = az = 0.0f;
+
+    if(props->Radius > Distance)
+        spread = F_TAU - Distance/props->Radius*F_PI;
+    else if(Distance > 0.0f)
+        spread = asinf(props->Radius / Distance) * 2.0f;
+    else
+        spread = 0.0f;
+
+    CalcPanningAndFilters(voice, az, ev, Distance, spread, DryGain, DryGainHF, DryGainLF, WetGain,
+                          WetGainLF, WetGainHF, SendSlots, ALBuffer, props, Listener, Device);
+}
+
+static void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force)
+{
+    ALbufferlistitem *BufferListItem;
+    struct ALvoiceProps *props;
+
+    props = static_cast<ALvoiceProps*>(ATOMIC_EXCHANGE_PTR(&voice->Update,
+        static_cast<ALvoiceProps*>(nullptr), almemory_order_acq_rel));
+    if(!props && !force) return;
+
+    if(props)
+    {
+        memcpy(voice->Props, props,
+            FAM_SIZE(struct ALvoiceProps, Send, context->Device->NumAuxSends)
+        );
+
+        ATOMIC_REPLACE_HEAD(struct ALvoiceProps*, &context->FreeVoiceProps, props);
+    }
+    props = voice->Props;
+
+    BufferListItem = ATOMIC_LOAD(&voice->current_buffer, almemory_order_relaxed);
+    while(BufferListItem != NULL)
+    {
+        const ALbuffer *buffer = NULL;
+        ALsizei i = 0;
+        while(!buffer && i < BufferListItem->num_buffers)
+            buffer = BufferListItem->buffers[i];
+        if(LIKELY(buffer))
+        {
+            if(props->SpatializeMode == SpatializeOn ||
+               (props->SpatializeMode == SpatializeAuto && buffer->FmtChannels == FmtMono))
+                CalcAttnSourceParams(voice, props, buffer, context);
+            else
+                CalcNonAttnSourceParams(voice, props, buffer, context);
+            break;
+        }
+        BufferListItem = ATOMIC_LOAD(&BufferListItem->next, almemory_order_acquire);
+    }
+}
+
+
+static void ProcessParamUpdates(ALCcontext *ctx, const struct ALeffectslotArray *slots)
+{
+    ALvoice **voice, **voice_end;
+    ALsource *source;
+    ALsizei i;
+
+    IncrementRef(&ctx->UpdateCount);
+    if(!ATOMIC_LOAD(&ctx->HoldUpdates, almemory_order_acquire))
+    {
+        bool cforce = CalcContextParams(ctx);
+        bool force = CalcListenerParams(ctx) | cforce;
+        for(i = 0;i < slots->count;i++)
+            force |= CalcEffectSlotParams(slots->slot[i], ctx, cforce);
+
+        voice = ctx->Voices;
+        voice_end = voice + ctx->VoiceCount;
+        for(;voice != voice_end;++voice)
+        {
+            source = ATOMIC_LOAD(&(*voice)->Source, almemory_order_acquire);
+            if(source) CalcSourceParams(*voice, ctx, force);
+        }
+    }
+    IncrementRef(&ctx->UpdateCount);
+}
+
+
+static void ApplyStablizer(FrontStablizer *Stablizer, ALfloat (*RESTRICT Buffer)[BUFFERSIZE],
+                           int lidx, int ridx, int cidx, ALsizei SamplesToDo,
+                           ALsizei NumChannels)
+{
+    ALfloat (*RESTRICT lsplit)[BUFFERSIZE] = Stablizer->LSplit;
+    ALfloat (*RESTRICT rsplit)[BUFFERSIZE] = Stablizer->RSplit;
+    ALsizei i;
+
+    /* Apply an all-pass to all channels, except the front-left and front-
+     * right, so they maintain the same relative phase.
+     */
+    for(i = 0;i < NumChannels;i++)
+    {
+        if(i == lidx || i == ridx)
+            continue;
+        splitterap_process(&Stablizer->APFilter[i], Buffer[i], SamplesToDo);
+    }
+
+    bandsplit_process(&Stablizer->LFilter, lsplit[1], lsplit[0], Buffer[lidx], SamplesToDo);
+    bandsplit_process(&Stablizer->RFilter, rsplit[1], rsplit[0], Buffer[ridx], SamplesToDo);
+
+    for(i = 0;i < SamplesToDo;i++)
+    {
+        ALfloat lfsum, hfsum;
+        ALfloat m, s, c;
+
+        lfsum = lsplit[0][i] + rsplit[0][i];
+        hfsum = lsplit[1][i] + rsplit[1][i];
+        s = lsplit[0][i] + lsplit[1][i] - rsplit[0][i] - rsplit[1][i];
+
+        /* This pans the separate low- and high-frequency sums between being on
+         * the center channel and the left/right channels. The low-frequency
+         * sum is 1/3rd toward center (2/3rds on left/right) and the high-
+         * frequency sum is 1/4th toward center (3/4ths on left/right). These
+         * values can be tweaked.
+         */
+        m = lfsum*cosf(1.0f/3.0f * F_PI_2) + hfsum*cosf(1.0f/4.0f * F_PI_2);
+        c = lfsum*sinf(1.0f/3.0f * F_PI_2) + hfsum*sinf(1.0f/4.0f * F_PI_2);
+
+        /* The generated center channel signal adds to the existing signal,
+         * while the modified left and right channels replace.
+         */
+        Buffer[lidx][i] = (m + s) * 0.5f;
+        Buffer[ridx][i] = (m - s) * 0.5f;
+        Buffer[cidx][i] += c * 0.5f;
+    }
+}
+
+static void ApplyDistanceComp(ALfloat (*RESTRICT Samples)[BUFFERSIZE], DistanceComp *distcomp,
+                              ALfloat *RESTRICT Values, ALsizei SamplesToDo, ALsizei numchans)
+{
+    ALsizei i, c;
+
+    for(c = 0;c < numchans;c++)
+    {
+        ALfloat *RESTRICT inout = Samples[c];
+        const ALfloat gain = distcomp[c].Gain;
+        const ALsizei base = distcomp[c].Length;
+        ALfloat *RESTRICT distbuf = distcomp[c].Buffer;
+
+        if(base == 0)
+        {
+            if(gain < 1.0f)
+            {
+                for(i = 0;i < SamplesToDo;i++)
+                    inout[i] *= gain;
+            }
+            continue;
+        }
+
+        if(LIKELY(SamplesToDo >= base))
+        {
+            for(i = 0;i < base;i++)
+                Values[i] = distbuf[i];
+            for(;i < SamplesToDo;i++)
+                Values[i] = inout[i-base];
+            memcpy(distbuf, &inout[SamplesToDo-base], base*sizeof(ALfloat));
+        }
+        else
+        {
+            for(i = 0;i < SamplesToDo;i++)
+                Values[i] = distbuf[i];
+            memmove(distbuf, distbuf+SamplesToDo, (base-SamplesToDo)*sizeof(ALfloat));
+            memcpy(distbuf+base-SamplesToDo, inout, SamplesToDo*sizeof(ALfloat));
+        }
+        for(i = 0;i < SamplesToDo;i++)
+            inout[i] = Values[i]*gain;
+    }
+}
+
+static void ApplyDither(ALfloat (*RESTRICT Samples)[BUFFERSIZE], ALuint *dither_seed,
+                        const ALfloat quant_scale, const ALsizei SamplesToDo,
+                        const ALsizei numchans)
+{
+    const ALfloat invscale = 1.0f / quant_scale;
+    ALuint seed = *dither_seed;
+    ALsizei c, i;
+
+    ASSUME(numchans > 0);
+    ASSUME(SamplesToDo > 0);
+
+    /* Dithering. Step 1, generate whitenoise (uniform distribution of random
+     * values between -1 and +1). Step 2 is to add the noise to the samples,
+     * before rounding and after scaling up to the desired quantization depth.
+     */
+    for(c = 0;c < numchans;c++)
+    {
+        ALfloat *RESTRICT samples = Samples[c];
+        for(i = 0;i < SamplesToDo;i++)
+        {
+            ALfloat val = samples[i] * quant_scale;
+            ALuint rng0 = dither_rng(&seed);
+            ALuint rng1 = dither_rng(&seed);
+            val += (ALfloat)(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
+            samples[i] = fast_roundf(val) * invscale;
+        }
+    }
+    *dither_seed = seed;
+}
+
+
+static inline ALfloat Conv_ALfloat(ALfloat val)
+{ return val; }
+static inline ALint Conv_ALint(ALfloat val)
+{
+    /* Floats have a 23-bit mantissa. There is an implied 1 bit in the mantissa
+     * along with the sign bit, giving 25 bits total, so [-16777216, +16777216]
+     * is the max value a normalized float can be scaled to before losing
+     * precision.
+     */
+    return fastf2i(clampf(val*16777216.0f, -16777216.0f, 16777215.0f))<<7;
+}
+static inline ALshort Conv_ALshort(ALfloat val)
+{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); }
+static inline ALbyte Conv_ALbyte(ALfloat val)
+{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); }
+
+/* Define unsigned output variations. */
+#define DECL_TEMPLATE(T, func, O)                             \
+static inline T Conv_##T(ALfloat val) { return func(val)+O; }
+
+DECL_TEMPLATE(ALubyte, Conv_ALbyte, 128)
+DECL_TEMPLATE(ALushort, Conv_ALshort, 32768)
+DECL_TEMPLATE(ALuint, Conv_ALint, 2147483648u)
+
+#undef DECL_TEMPLATE
+
+#define DECL_TEMPLATE(T, A)                                                   \
+static void Write##A(const ALfloat (*RESTRICT InBuffer)[BUFFERSIZE],          \
+                     ALvoid *OutBuffer, ALsizei Offset, ALsizei SamplesToDo,  \
+                     ALsizei numchans)                                        \
+{                                                                             \
+    ALsizei i, j;                                                             \
+                                                                              \
+    ASSUME(numchans > 0);                                                     \
+    ASSUME(SamplesToDo > 0);                                                  \
+                                                                              \
+    for(j = 0;j < numchans;j++)                                               \
+    {                                                                         \
+        const ALfloat *RESTRICT in = InBuffer[j];                             \
+        T *RESTRICT out = (T*)OutBuffer + Offset*numchans + j;                \
+                                                                              \
+        for(i = 0;i < SamplesToDo;i++)                                        \
+            out[i*numchans] = Conv_##T(in[i]);                                \
+    }                                                                         \
+}
+
+DECL_TEMPLATE(ALfloat, F32)
+DECL_TEMPLATE(ALuint, UI32)
+DECL_TEMPLATE(ALint, I32)
+DECL_TEMPLATE(ALushort, UI16)
+DECL_TEMPLATE(ALshort, I16)
+DECL_TEMPLATE(ALubyte, UI8)
+DECL_TEMPLATE(ALbyte, I8)
+
+#undef DECL_TEMPLATE
+
+
+void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples)
+{
+    ALsizei SamplesToDo;
+    ALsizei SamplesDone;
+    ALCcontext *ctx;
+    ALsizei i, c;
+
+    START_MIXER_MODE();
+    for(SamplesDone = 0;SamplesDone < NumSamples;)
+    {
+        SamplesToDo = mini(NumSamples-SamplesDone, BUFFERSIZE);
+        for(c = 0;c < device->Dry.NumChannels;c++)
+            memset(device->Dry.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
+        if(device->Dry.Buffer != device->FOAOut.Buffer)
+            for(c = 0;c < device->FOAOut.NumChannels;c++)
+                memset(device->FOAOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
+        if(device->Dry.Buffer != device->RealOut.Buffer)
+            for(c = 0;c < device->RealOut.NumChannels;c++)
+                memset(device->RealOut.Buffer[c], 0, SamplesToDo*sizeof(ALfloat));
+
+        IncrementRef(&device->MixCount);
+
+        ctx = ATOMIC_LOAD(&device->ContextList, almemory_order_acquire);
+        while(ctx)
+        {
+            const struct ALeffectslotArray *auxslots;
+
+            auxslots = ATOMIC_LOAD(&ctx->ActiveAuxSlots, almemory_order_acquire);
+            ProcessParamUpdates(ctx, auxslots);
+
+            for(i = 0;i < auxslots->count;i++)
+            {
+                ALeffectslot *slot = auxslots->slot[i];
+                for(c = 0;c < slot->NumChannels;c++)
+                    memset(slot->WetBuffer[c], 0, SamplesToDo*sizeof(ALfloat));
+            }
+
+            /* source processing */
+            for(i = 0;i < ctx->VoiceCount;i++)
+            {
+                ALvoice *voice = ctx->Voices[i];
+                ALsource *source = ATOMIC_LOAD(&voice->Source, almemory_order_acquire);
+                if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed) &&
+                   voice->Step > 0)
+                {
+                    if(!MixSource(voice, source->id, ctx, SamplesToDo))
+                    {
+                        ATOMIC_STORE(&voice->Source, static_cast<ALsource*>(nullptr),
+                                     almemory_order_relaxed);
+                        ATOMIC_STORE(&voice->Playing, false, almemory_order_release);
+                        SendSourceStoppedEvent(ctx, source->id);
+                    }
+                }
+            }
+
+            /* effect slot processing */
+            for(i = 0;i < auxslots->count;i++)
+            {
+                const ALeffectslot *slot = auxslots->slot[i];
+                ALeffectState *state = slot->Params.EffectState;
+                V(state,process)(SamplesToDo, slot->WetBuffer, state->OutBuffer,
+                                 state->OutChannels);
+            }
+
+            ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed);
+        }
+
+        /* Increment the clock time. Every second's worth of samples is
+         * converted and added to clock base so that large sample counts don't
+         * overflow during conversion. This also guarantees an exact, stable
+         * conversion. */
+        device->SamplesDone += SamplesToDo;
+        device->ClockBase += (device->SamplesDone/device->Frequency) * DEVICE_CLOCK_RES;
+        device->SamplesDone %= device->Frequency;
+        IncrementRef(&device->MixCount);
+
+        /* Apply post-process for finalizing the Dry mix to the RealOut
+         * (Ambisonic decode, UHJ encode, etc).
+         */
+        if(LIKELY(device->PostProcess))
+            device->PostProcess(device, SamplesToDo);
+
+        if(device->Stablizer)
+        {
+            int lidx = GetChannelIdxByName(&device->RealOut, FrontLeft);
+            int ridx = GetChannelIdxByName(&device->RealOut, FrontRight);
+            int cidx = GetChannelIdxByName(&device->RealOut, FrontCenter);
+            assert(lidx >= 0 && ridx >= 0 && cidx >= 0);
+
+            ApplyStablizer(device->Stablizer, device->RealOut.Buffer, lidx, ridx, cidx,
+                           SamplesToDo, device->RealOut.NumChannels);
+        }
+
+        ApplyDistanceComp(device->RealOut.Buffer, device->ChannelDelay, device->TempBuffer[0],
+                          SamplesToDo, device->RealOut.NumChannels);
+
+        if(device->Limiter)
+            ApplyCompression(device->Limiter, SamplesToDo, device->RealOut.Buffer);
+
+        if(device->DitherDepth > 0.0f)
+            ApplyDither(device->RealOut.Buffer, &device->DitherSeed, device->DitherDepth,
+                        SamplesToDo, device->RealOut.NumChannels);
+
+        if(LIKELY(OutBuffer))
+        {
+            ALfloat (*Buffer)[BUFFERSIZE] = device->RealOut.Buffer;
+            ALsizei Channels = device->RealOut.NumChannels;
+
+            switch(device->FmtType)
+            {
+#define HANDLE_WRITE(T, S) case T:                                            \
+    Write##S(Buffer, OutBuffer, SamplesDone, SamplesToDo, Channels); break;
+                HANDLE_WRITE(DevFmtByte, I8)
+                HANDLE_WRITE(DevFmtUByte, UI8)
+                HANDLE_WRITE(DevFmtShort, I16)
+                HANDLE_WRITE(DevFmtUShort, UI16)
+                HANDLE_WRITE(DevFmtInt, I32)
+                HANDLE_WRITE(DevFmtUInt, UI32)
+                HANDLE_WRITE(DevFmtFloat, F32)
+#undef HANDLE_WRITE
+            }
+        }
+
+        SamplesDone += SamplesToDo;
+    }
+    END_MIXER_MODE();
+}
+
+
+void aluHandleDisconnect(ALCdevice *device, const char *msg, ...)
+{
+    AsyncEvent evt = ASYNC_EVENT(EventType_Disconnected);
+    ALCcontext *ctx;
+    va_list args;
+    int msglen;
+
+    if(!ATOMIC_EXCHANGE(&device->Connected, AL_FALSE, almemory_order_acq_rel))
+        return;
+
+    evt.u.user.type = AL_EVENT_TYPE_DISCONNECTED_SOFT;
+    evt.u.user.id = 0;
+    evt.u.user.param = 0;
+
+    va_start(args, msg);
+    msglen = vsnprintf(evt.u.user.msg, sizeof(evt.u.user.msg), msg, args);
+    va_end(args);
+
+    if(msglen < 0 || (size_t)msglen >= sizeof(evt.u.user.msg))
+        evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0;
+
+    ctx = ATOMIC_LOAD_SEQ(&device->ContextList);
+    while(ctx)
+    {
+        ALbitfieldSOFT enabledevt = ATOMIC_LOAD(&ctx->EnabledEvts, almemory_order_acquire);
+        ALsizei i;
+
+        if((enabledevt&EventType_Disconnected) &&
+           ll_ringbuffer_write(ctx->AsyncEvents, (const char*)&evt, 1) == 1)
+            alsem_post(&ctx->EventSem);
+
+        for(i = 0;i < ctx->VoiceCount;i++)
+        {
+            ALvoice *voice = ctx->Voices[i];
+            ALsource *source;
+
+            source = static_cast<ALsource*>(ATOMIC_EXCHANGE_PTR(&voice->Source,
+                static_cast<ALsource*>(nullptr), almemory_order_relaxed));
+            if(source && ATOMIC_LOAD(&voice->Playing, almemory_order_relaxed))
+            {
+                /* If the source's voice was playing, it's now effectively
+                 * stopped (the source state will be updated the next time it's
+                 * checked).
+                 */
+                SendSourceStoppedEvent(ctx, source->id);
+            }
+            ATOMIC_STORE(&voice->Playing, false, almemory_order_release);
+        }
+
+        ctx = ATOMIC_LOAD(&ctx->next, almemory_order_relaxed);
+    }
+}