Rename Alc to alc

1 files changed, 1798 insertions, 0 deletions

diff --git a/alc/alu.cpp b/alc/alu.cpp
new file mode 100644
index 00000000..cc1a5a98
--- /dev/null
+++ b/alc/alu.cpp
@@ -0,0 +1,1798 @@
+/**
+ * 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 "alu.h"
+
+#include <algorithm>
+#include <array>
+#include <atomic>
+#include <cassert>
+#include <chrono>
+#include <climits>
+#include <cmath>
+#include <cstdarg>
+#include <cstdio>
+#include <cstdlib>
+#include <cstring>
+#include <functional>
+#include <iterator>
+#include <limits>
+#include <memory>
+#include <new>
+#include <numeric>
+#include <utility>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "AL/efx.h"
+
+#include "alAuxEffectSlot.h"
+#include "alBuffer.h"
+#include "alcmain.h"
+#include "alEffect.h"
+#include "alListener.h"
+#include "alcontext.h"
+#include "almalloc.h"
+#include "alnumeric.h"
+#include "alspan.h"
+#include "ambidefs.h"
+#include "atomic.h"
+#include "bformatdec.h"
+#include "bs2b.h"
+#include "cpu_caps.h"
+#include "effects/base.h"
+#include "filters/biquad.h"
+#include "filters/nfc.h"
+#include "filters/splitter.h"
+#include "fpu_modes.h"
+#include "hrtf.h"
+#include "inprogext.h"
+#include "mastering.h"
+#include "math_defs.h"
+#include "mixer/defs.h"
+#include "opthelpers.h"
+#include "ringbuffer.h"
+#include "threads.h"
+#include "uhjfilter.h"
+#include "vecmat.h"
+#include "vector.h"
+
+#include "bsinc_inc.h"
+
+
+namespace {
+
+using namespace std::placeholders;
+
+ALfloat InitConeScale()
+{
+    ALfloat ret{1.0f};
+    const char *str{getenv("__ALSOFT_HALF_ANGLE_CONES")};
+    if(str && (strcasecmp(str, "true") == 0 || strtol(str, nullptr, 0) == 1))
+        ret *= 0.5f;
+    return ret;
+}
+
+ALfloat InitZScale()
+{
+    ALfloat ret{1.0f};
+    const char *str{getenv("__ALSOFT_REVERSE_Z")};
+    if(str && (strcasecmp(str, "true") == 0 || strtol(str, nullptr, 0) == 1))
+        ret *= -1.0f;
+    return ret;
+}
+
+ALboolean InitReverbSOS()
+{
+    ALboolean ret{AL_FALSE};
+    const char *str{getenv("__ALSOFT_REVERB_IGNORES_SOUND_SPEED")};
+    if(str && (strcasecmp(str, "true") == 0 || strtol(str, nullptr, 0) == 1))
+        ret = AL_TRUE;
+    return ret;
+}
+
+} // namespace
+
+/* Cone scalar */
+const ALfloat ConeScale{InitConeScale()};
+
+/* Localized Z scalar for mono sources */
+const ALfloat ZScale{InitZScale()};
+
+/* Force default speed of sound for distance-related reverb decay. */
+const ALboolean OverrideReverbSpeedOfSound{InitReverbSOS()};
+
+
+namespace {
+
+void ClearArray(ALfloat (&f)[MAX_OUTPUT_CHANNELS])
+{
+    std::fill(std::begin(f), std::end(f), 0.0f);
+}
+
+struct ChanMap {
+    Channel channel;
+    ALfloat angle;
+    ALfloat elevation;
+};
+
+HrtfDirectMixerFunc MixDirectHrtf = MixDirectHrtf_<CTag>;
+inline HrtfDirectMixerFunc SelectHrtfMixer(void)
+{
+#ifdef HAVE_NEON
+    if((CPUCapFlags&CPU_CAP_NEON))
+        return MixDirectHrtf_<NEONTag>;
+#endif
+#ifdef HAVE_SSE
+    if((CPUCapFlags&CPU_CAP_SSE))
+        return MixDirectHrtf_<SSETag>;
+#endif
+
+    return MixDirectHrtf_<CTag>;
+}
+
+} // namespace
+
+void aluInit(void)
+{
+    MixDirectHrtf = SelectHrtfMixer();
+}
+
+
+void ProcessHrtf(ALCdevice *device, const ALsizei SamplesToDo)
+{
+    /* HRTF is stereo output only. */
+    const int lidx{device->RealOut.ChannelIndex[FrontLeft]};
+    const int ridx{device->RealOut.ChannelIndex[FrontRight]};
+    ASSUME(lidx >= 0 && ridx >= 0);
+
+    DirectHrtfState *state{device->mHrtfState.get()};
+    MixDirectHrtf(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx], device->Dry.Buffer,
+        device->HrtfAccumData, state, SamplesToDo);
+}
+
+void ProcessAmbiDec(ALCdevice *device, const ALsizei SamplesToDo)
+{
+    BFormatDec *ambidec{device->AmbiDecoder.get()};
+    ambidec->process(device->RealOut.Buffer, device->Dry.Buffer.data(), SamplesToDo);
+}
+
+void ProcessUhj(ALCdevice *device, const ALsizei SamplesToDo)
+{
+    /* UHJ is stereo output only. */
+    const int lidx{device->RealOut.ChannelIndex[FrontLeft]};
+    const int ridx{device->RealOut.ChannelIndex[FrontRight]};
+    ASSUME(lidx >= 0 && ridx >= 0);
+
+    /* Encode to stereo-compatible 2-channel UHJ output. */
+    Uhj2Encoder *uhj2enc{device->Uhj_Encoder.get()};
+    uhj2enc->encode(device->RealOut.Buffer[lidx], device->RealOut.Buffer[ridx],
+        device->Dry.Buffer.data(), SamplesToDo);
+}
+
+void ProcessBs2b(ALCdevice *device, const ALsizei SamplesToDo)
+{
+    /* First, decode the ambisonic mix to the "real" output. */
+    BFormatDec *ambidec{device->AmbiDecoder.get()};
+    ambidec->process(device->RealOut.Buffer, device->Dry.Buffer.data(), SamplesToDo);
+
+    /* BS2B is stereo output only. */
+    const int lidx{device->RealOut.ChannelIndex[FrontLeft]};
+    const int ridx{device->RealOut.ChannelIndex[FrontRight]};
+    ASSUME(lidx >= 0 && ridx >= 0);
+
+    /* Now apply the BS2B binaural/crossfeed filter. */
+    bs2b_cross_feed(device->Bs2b.get(), device->RealOut.Buffer[lidx].data(),
+        device->RealOut.Buffer[ridx].data(), SamplesToDo);
+}
+
+
+/* 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)
+{
+    ALsizei si{BSINC_SCALE_COUNT - 1};
+    ALfloat sf{0.0f};
+
+    if(increment > FRACTIONONE)
+    {
+        sf = static_cast<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 - std::cos(std::asin(sf - si));
+    }
+
+    state->sf = sf;
+    state->m = table->m[si];
+    state->l = (state->m/2) - 1;
+    state->filter = table->Tab + table->filterOffset[si];
+}
+
+
+namespace {
+
+/* 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.
+ */
+inline ALuint dither_rng(ALuint *seed) noexcept
+{
+    *seed = (*seed * 96314165) + 907633515;
+    return *seed;
+}
+
+
+inline alu::Vector aluCrossproduct(const alu::Vector &in1, const alu::Vector &in2)
+{
+    return alu::Vector{
+        in1[1]*in2[2] - in1[2]*in2[1],
+        in1[2]*in2[0] - in1[0]*in2[2],
+        in1[0]*in2[1] - in1[1]*in2[0],
+        0.0f
+    };
+}
+
+inline ALfloat aluDotproduct(const alu::Vector &vec1, const alu::Vector &vec2)
+{
+    return vec1[0]*vec2[0] + vec1[1]*vec2[1] + vec1[2]*vec2[2];
+}
+
+
+alu::Vector operator*(const alu::Matrix &mtx, const alu::Vector &vec) noexcept
+{
+    return alu::Vector{
+        vec[0]*mtx[0][0] + vec[1]*mtx[1][0] + vec[2]*mtx[2][0] + vec[3]*mtx[3][0],
+        vec[0]*mtx[0][1] + vec[1]*mtx[1][1] + vec[2]*mtx[2][1] + vec[3]*mtx[3][1],
+        vec[0]*mtx[0][2] + vec[1]*mtx[1][2] + vec[2]*mtx[2][2] + vec[3]*mtx[3][2],
+        vec[0]*mtx[0][3] + vec[1]*mtx[1][3] + vec[2]*mtx[2][3] + vec[3]*mtx[3][3]
+    };
+}
+
+
+bool CalcContextParams(ALCcontext *Context)
+{
+    ALcontextProps *props{Context->Update.exchange(nullptr, std::memory_order_acq_rel)};
+    if(!props) return false;
+
+    ALlistener &Listener = Context->Listener;
+    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.mDistanceModel = props->mDistanceModel;
+
+    AtomicReplaceHead(Context->FreeContextProps, props);
+    return true;
+}
+
+bool CalcListenerParams(ALCcontext *Context)
+{
+    ALlistener &Listener = Context->Listener;
+
+    ALlistenerProps *props{Listener.Update.exchange(nullptr, std::memory_order_acq_rel)};
+    if(!props) return false;
+
+    /* AT then UP */
+    alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
+    N.normalize();
+    alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
+    V.normalize();
+    /* Build and normalize right-vector */
+    alu::Vector U{aluCrossproduct(N, V)};
+    U.normalize();
+
+    Listener.Params.Matrix = alu::Matrix{
+        U[0], V[0], -N[0], 0.0f,
+        U[1], V[1], -N[1], 0.0f,
+        U[2], V[2], -N[2], 0.0f,
+        0.0f, 0.0f,  0.0f, 1.0f
+    };
+
+    const alu::Vector P{Listener.Params.Matrix *
+        alu::Vector{props->Position[0], props->Position[1], props->Position[2], 1.0f}};
+    Listener.Params.Matrix.setRow(3, -P[0], -P[1], -P[2], 1.0f);
+
+    const alu::Vector vel{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f};
+    Listener.Params.Velocity = Listener.Params.Matrix * vel;
+
+    Listener.Params.Gain = props->Gain * Context->GainBoost;
+
+    AtomicReplaceHead(Context->FreeListenerProps, props);
+    return true;
+}
+
+bool CalcEffectSlotParams(ALeffectslot *slot, ALCcontext *context, bool force)
+{
+    ALeffectslotProps *props{slot->Update.exchange(nullptr, std::memory_order_acq_rel)};
+    if(!props && !force) return false;
+
+    EffectState *state;
+    if(!props)
+        state = slot->Params.mEffectState;
+    else
+    {
+        slot->Params.Gain = props->Gain;
+        slot->Params.AuxSendAuto = props->AuxSendAuto;
+        slot->Params.Target = props->Target;
+        slot->Params.EffectType = props->Type;
+        slot->Params.mEffectProps = 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;
+        props->State = nullptr;
+        EffectState *oldstate{slot->Params.mEffectState};
+        slot->Params.mEffectState = state;
+
+        /* Manually decrement the old effect state's refcount if it's greater
+         * than 1. We need to be a bit clever here to avoid the refcount
+         * reaching 0 since it can't be deleted in the mixer.
+         */
+        ALuint oldval{oldstate->mRef.load(std::memory_order_acquire)};
+        while(oldval > 1 && !oldstate->mRef.compare_exchange_weak(oldval, oldval-1,
+            std::memory_order_acq_rel, std::memory_order_acquire))
+        {
+            /* oldval was updated with the current value on failure, so just
+             * try again.
+             */
+        }
+
+        if(oldval < 2)
+        {
+            /* Otherwise, if it would be deleted, send it off with a release
+             * event.
+             */
+            RingBuffer *ring{context->AsyncEvents.get()};
+            auto evt_vec = ring->getWriteVector();
+            if(LIKELY(evt_vec.first.len > 0))
+            {
+                AsyncEvent *evt{new (evt_vec.first.buf) AsyncEvent{EventType_ReleaseEffectState}};
+                evt->u.mEffectState = oldstate;
+                ring->writeAdvance(1);
+                context->EventSem.post();
+            }
+            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 = oldstate;
+            }
+        }
+
+        AtomicReplaceHead(context->FreeEffectslotProps, props);
+    }
+
+    EffectTarget output;
+    if(ALeffectslot *target{slot->Params.Target})
+        output = EffectTarget{&target->Wet, nullptr};
+    else
+    {
+        ALCdevice *device{context->Device};
+        output = EffectTarget{&device->Dry, &device->RealOut};
+    }
+    state->update(context, slot, &slot->Params.mEffectProps, output);
+    return true;
+}
+
+
+/* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
+ * front.
+ */
+inline float ScaleAzimuthFront(float azimuth, float scale)
+{
+    const ALfloat abs_azi{std::fabs(azimuth)};
+    if(!(abs_azi > al::MathDefs<float>::Pi()*0.5f))
+        return minf(abs_azi*scale, al::MathDefs<float>::Pi()*0.5f) * std::copysign(1.0f, azimuth);
+    return azimuth;
+}
+
+void CalcPanningAndFilters(ALvoice *voice, const ALfloat xpos, const ALfloat ypos,
+    const ALfloat zpos, const ALfloat Distance, const ALfloat Spread, const ALfloat DryGain,
+    const ALfloat DryGainHF, const ALfloat DryGainLF, const ALfloat (&WetGain)[MAX_SENDS],
+    const ALfloat (&WetGainLF)[MAX_SENDS], const ALfloat (&WetGainHF)[MAX_SENDS],
+    ALeffectslot *(&SendSlots)[MAX_SENDS], const ALvoicePropsBase *props,
+    const ALlistener &Listener, const ALCdevice *Device)
+{
+    static constexpr 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) }
+    };
+
+    ChanMap StereoMap[2]{
+        { FrontLeft,  Deg2Rad(-30.0f), Deg2Rad(0.0f) },
+        { FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) }
+    };
+
+    const auto Frequency = static_cast<ALfloat>(Device->Frequency);
+    const ALsizei NumSends{Device->NumAuxSends};
+    ASSUME(NumSends >= 0);
+
+    bool DirectChannels{props->DirectChannels != AL_FALSE};
+    const ChanMap *chans{nullptr};
+    ALsizei num_channels{0};
+    bool isbformat{false};
+    ALfloat downmix_gain{1.0f};
+    switch(voice->mFmtChannels)
+    {
+    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;
+    }
+    ASSUME(num_channels > 0);
+
+    std::for_each(voice->mChans.begin(), voice->mChans.begin()+num_channels,
+        [NumSends](ALvoice::ChannelData &chandata) -> void
+        {
+            chandata.mDryParams.Hrtf.Target = HrtfFilter{};
+            ClearArray(chandata.mDryParams.Gains.Target);
+            std::for_each(chandata.mWetParams.begin(), chandata.mWetParams.begin()+NumSends,
+                [](SendParams &params) -> void { ClearArray(params.Gains.Target); });
+        });
+
+    voice->mFlags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC);
+    if(isbformat)
+    {
+        /* Special handling for B-Format sources. */
+
+        if(Distance > std::numeric_limits<float>::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.
+             */
+
+            if(Device->AvgSpeakerDist > 0.0f)
+            {
+                /* Clamp the distance for really close sources, to prevent
+                 * excessive bass.
+                 */
+                const ALfloat mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)};
+                const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC / (mdist * Frequency)};
+
+                /* Only need to adjust the first channel of a B-Format source. */
+                voice->mChans[0].mDryParams.NFCtrlFilter.adjust(w0);
+
+                voice->mFlags |= VOICE_HAS_NFC;
+            }
+
+            ALfloat coeffs[MAX_AMBI_CHANNELS];
+            if(Device->mRenderMode != StereoPair)
+                CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs);
+            else
+            {
+                /* Clamp Y, in case rounding errors caused it to end up outside
+                 * of -1...+1.
+                 */
+                const ALfloat ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
+                /* Negate Z for right-handed coords with -Z in front. */
+                const ALfloat az{std::atan2(xpos, -zpos)};
+
+                /* 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(ScaleAzimuthFront(az, 1.5f), ev, Spread, coeffs);
+            }
+
+            /* NOTE: W needs to be scaled due to FuMa normalization. */
+            const ALfloat &scale0 = AmbiScale::FromFuMa[0];
+            ComputePanGains(&Device->Dry, coeffs, DryGain*scale0,
+                voice->mChans[0].mDryParams.Gains.Target);
+            for(ALsizei i{0};i < NumSends;i++)
+            {
+                if(const ALeffectslot *Slot{SendSlots[i]})
+                    ComputePanGains(&Slot->Wet, coeffs, WetGain[i]*scale0,
+                        voice->mChans[0].mWetParams[i].Gains.Target);
+            }
+        }
+        else
+        {
+            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.
+                 */
+                voice->mChans[0].mDryParams.NFCtrlFilter.adjust(0.0f);
+
+                voice->mFlags |= VOICE_HAS_NFC;
+            }
+
+            /* Local B-Format sources have their XYZ channels rotated according
+             * to the orientation.
+             */
+            /* AT then UP */
+            alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
+            N.normalize();
+            alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
+            V.normalize();
+            if(!props->HeadRelative)
+            {
+                N = Listener.Params.Matrix * N;
+                V = Listener.Params.Matrix * V;
+            }
+            /* Build and normalize right-vector */
+            alu::Vector U{aluCrossproduct(N, V)};
+            U.normalize();
+
+            /* 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.
+             */
+            const ALfloat &wscale = AmbiScale::FromFuMa[0];
+            const ALfloat &yscale = AmbiScale::FromFuMa[1];
+            const ALfloat &zscale = AmbiScale::FromFuMa[2];
+            const ALfloat &xscale = AmbiScale::FromFuMa[3];
+            const ALfloat matrix[4][MAX_AMBI_CHANNELS]{
+            //      ACN0          ACN1          ACN2          ACN3
+                { wscale,         0.0f,         0.0f,         0.0f }, // FuMa W
+                {   0.0f, -N[0]*xscale,  N[1]*xscale, -N[2]*xscale }, // FuMa X
+                {   0.0f,  U[0]*yscale, -U[1]*yscale,  U[2]*yscale }, // FuMa Y
+                {   0.0f, -V[0]*zscale,  V[1]*zscale, -V[2]*zscale }  // FuMa Z
+            };
+
+            for(ALsizei c{0};c < num_channels;c++)
+            {
+                ComputePanGains(&Device->Dry, matrix[c], DryGain,
+                    voice->mChans[c].mDryParams.Gains.Target);
+
+                for(ALsizei i{0};i < NumSends;i++)
+                {
+                    if(const ALeffectslot *Slot{SendSlots[i]})
+                        ComputePanGains(&Slot->Wet, matrix[c], WetGain[i],
+                            voice->mChans[c].mWetParams[i].Gains.Target);
+                }
+            }
+        }
+    }
+    else if(DirectChannels)
+    {
+        /* Direct source channels always play local. Skip the virtual channels
+         * and write inputs to the matching real outputs.
+         */
+        voice->mDirect.Buffer = Device->RealOut.Buffer;
+
+        for(ALsizei c{0};c < num_channels;c++)
+        {
+            int idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)};
+            if(idx != -1) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain;
+        }
+
+        /* Auxiliary sends still use normal channel panning since they mix to
+         * B-Format, which can't channel-match.
+         */
+        for(ALsizei c{0};c < num_channels;c++)
+        {
+            ALfloat coeffs[MAX_AMBI_CHANNELS];
+            CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs);
+
+            for(ALsizei i{0};i < NumSends;i++)
+            {
+                if(const ALeffectslot *Slot{SendSlots[i]})
+                    ComputePanGains(&Slot->Wet, coeffs, WetGain[i],
+                        voice->mChans[c].mWetParams[i].Gains.Target);
+            }
+        }
+    }
+    else if(Device->mRenderMode == HrtfRender)
+    {
+        /* Full HRTF rendering. Skip the virtual channels and render to the
+         * real outputs.
+         */
+        voice->mDirect.Buffer = Device->RealOut.Buffer;
+
+        if(Distance > std::numeric_limits<float>::epsilon())
+        {
+            const ALfloat ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
+            const ALfloat az{std::atan2(xpos, -zpos)};
+
+            /* Get the HRIR coefficients and delays just once, for the given
+             * source direction.
+             */
+            GetHrtfCoeffs(Device->mHrtf, ev, az, Distance, Spread,
+                voice->mChans[0].mDryParams.Hrtf.Target.Coeffs,
+                voice->mChans[0].mDryParams.Hrtf.Target.Delay);
+            voice->mChans[0].mDryParams.Hrtf.Target.Gain = DryGain * downmix_gain;
+
+            /* Remaining channels use the same results as the first. */
+            for(ALsizei c{1};c < num_channels;c++)
+            {
+                /* Skip LFE */
+                if(chans[c].channel == LFE) continue;
+                voice->mChans[c].mDryParams.Hrtf.Target = voice->mChans[0].mDryParams.Hrtf.Target;
+            }
+
+            /* Calculate the directional coefficients once, which apply to all
+             * input channels of the source sends.
+             */
+            ALfloat coeffs[MAX_AMBI_CHANNELS];
+            CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs);
+
+            for(ALsizei c{0};c < num_channels;c++)
+            {
+                /* Skip LFE */
+                if(chans[c].channel == LFE)
+                    continue;
+                for(ALsizei i{0};i < NumSends;i++)
+                {
+                    if(const ALeffectslot *Slot{SendSlots[i]})
+                        ComputePanGains(&Slot->Wet, coeffs, WetGain[i] * downmix_gain,
+                            voice->mChans[c].mWetParams[i].Gains.Target);
+                }
+            }
+        }
+        else
+        {
+            /* Local sources on HRTF play with each channel panned to its
+             * relative location around the listener, providing "virtual
+             * speaker" responses.
+             */
+            for(ALsizei c{0};c < num_channels;c++)
+            {
+                /* Skip LFE */
+                if(chans[c].channel == LFE)
+                    continue;
+
+                /* Get the HRIR coefficients and delays for this channel
+                 * position.
+                 */
+                GetHrtfCoeffs(Device->mHrtf, chans[c].elevation, chans[c].angle,
+                    std::numeric_limits<float>::infinity(), Spread,
+                    voice->mChans[c].mDryParams.Hrtf.Target.Coeffs,
+                    voice->mChans[c].mDryParams.Hrtf.Target.Delay);
+                voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain;
+
+                /* Normal panning for auxiliary sends. */
+                ALfloat coeffs[MAX_AMBI_CHANNELS];
+                CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs);
+
+                for(ALsizei i{0};i < NumSends;i++)
+                {
+                    if(const ALeffectslot *Slot{SendSlots[i]})
+                        ComputePanGains(&Slot->Wet, coeffs, WetGain[i],
+                            voice->mChans[c].mWetParams[i].Gains.Target);
+                }
+            }
+        }
+
+        voice->mFlags |= VOICE_HAS_HRTF;
+    }
+    else
+    {
+        /* Non-HRTF rendering. Use normal panning to the output. */
+
+        if(Distance > std::numeric_limits<float>::epsilon())
+        {
+            /* Calculate NFC filter coefficient if needed. */
+            if(Device->AvgSpeakerDist > 0.0f)
+            {
+                /* Clamp the distance for really close sources, to prevent
+                 * excessive bass.
+                 */
+                const ALfloat mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)};
+                const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC / (mdist * Frequency)};
+
+                /* Adjust NFC filters. */
+                for(ALsizei c{0};c < num_channels;c++)
+                    voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
+
+                voice->mFlags |= VOICE_HAS_NFC;
+            }
+
+            /* Calculate the directional coefficients once, which apply to all
+             * input channels.
+             */
+            ALfloat coeffs[MAX_AMBI_CHANNELS];
+            if(Device->mRenderMode != StereoPair)
+                CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs);
+            else
+            {
+                const ALfloat ev{std::asin(clampf(ypos, -1.0f, 1.0f))};
+                const ALfloat az{std::atan2(xpos, -zpos)};
+                CalcAngleCoeffs(ScaleAzimuthFront(az, 1.5f), ev, Spread, coeffs);
+            }
+
+            for(ALsizei c{0};c < num_channels;c++)
+            {
+                /* Special-case LFE */
+                if(chans[c].channel == LFE)
+                {
+                    if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
+                    {
+                        int idx = GetChannelIdxByName(Device->RealOut, chans[c].channel);
+                        if(idx != -1) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain;
+                    }
+                    continue;
+                }
+
+                ComputePanGains(&Device->Dry, coeffs, DryGain * downmix_gain,
+                    voice->mChans[c].mDryParams.Gains.Target);
+            }
+
+            for(ALsizei c{0};c < num_channels;c++)
+            {
+                /* Skip LFE */
+                if(chans[c].channel == LFE)
+                    continue;
+                for(ALsizei i{0};i < NumSends;i++)
+                {
+                    if(const ALeffectslot *Slot{SendSlots[i]})
+                        ComputePanGains(&Slot->Wet, coeffs, WetGain[i] * downmix_gain,
+                            voice->mChans[c].mWetParams[i].Gains.Target);
+                }
+            }
+        }
+        else
+        {
+            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.
+                 */
+                const ALfloat w0{SPEEDOFSOUNDMETRESPERSEC / (Device->AvgSpeakerDist * Frequency)};
+
+                for(ALsizei c{0};c < num_channels;c++)
+                    voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
+
+                voice->mFlags |= VOICE_HAS_NFC;
+            }
+
+            for(ALsizei c{0};c < num_channels;c++)
+            {
+                /* Special-case LFE */
+                if(chans[c].channel == LFE)
+                {
+                    if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
+                    {
+                        int idx = GetChannelIdxByName(Device->RealOut, chans[c].channel);
+                        if(idx != -1) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain;
+                    }
+                    continue;
+                }
+
+                ALfloat coeffs[MAX_AMBI_CHANNELS];
+                CalcAngleCoeffs(
+                    (Device->mRenderMode==StereoPair) ? ScaleAzimuthFront(chans[c].angle, 3.0f)
+                                                      : chans[c].angle,
+                    chans[c].elevation, Spread, coeffs
+                );
+
+                ComputePanGains(&Device->Dry, coeffs, DryGain,
+                    voice->mChans[c].mDryParams.Gains.Target);
+                for(ALsizei i{0};i < NumSends;i++)
+                {
+                    if(const ALeffectslot *Slot{SendSlots[i]})
+                        ComputePanGains(&Slot->Wet, coeffs, WetGain[i],
+                            voice->mChans[c].mWetParams[i].Gains.Target);
+                }
+            }
+        }
+    }
+
+    {
+        const ALfloat hfScale{props->Direct.HFReference / Frequency};
+        const ALfloat lfScale{props->Direct.LFReference / Frequency};
+        const ALfloat gainHF{maxf(DryGainHF, 0.001f)}; /* Limit -60dB */
+        const ALfloat gainLF{maxf(DryGainLF, 0.001f)};
+
+        voice->mDirect.FilterType = AF_None;
+        if(gainHF != 1.0f) voice->mDirect.FilterType |= AF_LowPass;
+        if(gainLF != 1.0f) voice->mDirect.FilterType |= AF_HighPass;
+        auto &lowpass = voice->mChans[0].mDryParams.LowPass;
+        auto &highpass = voice->mChans[0].mDryParams.HighPass;
+        lowpass.setParams(BiquadType::HighShelf, gainHF, hfScale,
+            lowpass.rcpQFromSlope(gainHF, 1.0f));
+        highpass.setParams(BiquadType::LowShelf, gainLF, lfScale,
+            highpass.rcpQFromSlope(gainLF, 1.0f));
+        for(ALsizei c{1};c < num_channels;c++)
+        {
+            voice->mChans[c].mDryParams.LowPass.copyParamsFrom(lowpass);
+            voice->mChans[c].mDryParams.HighPass.copyParamsFrom(highpass);
+        }
+    }
+    for(ALsizei i{0};i < NumSends;i++)
+    {
+        const ALfloat hfScale{props->Send[i].HFReference / Frequency};
+        const ALfloat lfScale{props->Send[i].LFReference / Frequency};
+        const ALfloat gainHF{maxf(WetGainHF[i], 0.001f)};
+        const ALfloat gainLF{maxf(WetGainLF[i], 0.001f)};
+
+        voice->mSend[i].FilterType = AF_None;
+        if(gainHF != 1.0f) voice->mSend[i].FilterType |= AF_LowPass;
+        if(gainLF != 1.0f) voice->mSend[i].FilterType |= AF_HighPass;
+
+        auto &lowpass = voice->mChans[0].mWetParams[i].LowPass;
+        auto &highpass = voice->mChans[0].mWetParams[i].HighPass;
+        lowpass.setParams(BiquadType::HighShelf, gainHF, hfScale,
+            lowpass.rcpQFromSlope(gainHF, 1.0f));
+        highpass.setParams(BiquadType::LowShelf, gainLF, lfScale,
+            highpass.rcpQFromSlope(gainLF, 1.0f));
+        for(ALsizei c{1};c < num_channels;c++)
+        {
+            voice->mChans[c].mWetParams[i].LowPass.copyParamsFrom(lowpass);
+            voice->mChans[c].mWetParams[i].HighPass.copyParamsFrom(highpass);
+        }
+    }
+}
+
+void CalcNonAttnSourceParams(ALvoice *voice, const ALvoicePropsBase *props, const ALCcontext *ALContext)
+{
+    const ALCdevice *Device{ALContext->Device};
+    ALeffectslot *SendSlots[MAX_SENDS];
+
+    voice->mDirect.Buffer = Device->Dry.Buffer;
+    for(ALsizei i{0};i < Device->NumAuxSends;i++)
+    {
+        SendSlots[i] = props->Send[i].Slot;
+        if(!SendSlots[i] && i == 0)
+            SendSlots[i] = ALContext->DefaultSlot.get();
+        if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL)
+        {
+            SendSlots[i] = nullptr;
+            voice->mSend[i].Buffer = {};
+        }
+        else
+            voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
+    }
+
+    /* Calculate the stepping value */
+    const auto Pitch = static_cast<ALfloat>(voice->mFrequency) /
+        static_cast<ALfloat>(Device->Frequency) * props->Pitch;
+    if(Pitch > static_cast<ALfloat>(MAX_PITCH))
+        voice->mStep = MAX_PITCH<<FRACTIONBITS;
+    else
+        voice->mStep = maxi(fastf2i(Pitch * FRACTIONONE), 1);
+    if(props->mResampler == BSinc24Resampler)
+        BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc24);
+    else if(props->mResampler == BSinc12Resampler)
+        BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc12);
+    voice->mResampler = SelectResampler(props->mResampler);
+
+    /* Calculate gains */
+    const ALlistener &Listener = ALContext->Listener;
+    ALfloat DryGain{clampf(props->Gain, props->MinGain, props->MaxGain)};
+    DryGain *= props->Direct.Gain * Listener.Params.Gain;
+    DryGain  = minf(DryGain, GAIN_MIX_MAX);
+    ALfloat DryGainHF{props->Direct.GainHF};
+    ALfloat DryGainLF{props->Direct.GainLF};
+    ALfloat WetGain[MAX_SENDS], WetGainHF[MAX_SENDS], WetGainLF[MAX_SENDS];
+    for(ALsizei 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, -1.0f, 0.0f, 0.0f, DryGain, DryGainHF, DryGainLF,
+        WetGain, WetGainLF, WetGainHF, SendSlots, props, Listener, Device);
+}
+
+void CalcAttnSourceParams(ALvoice *voice, const ALvoicePropsBase *props, const ALCcontext *ALContext)
+{
+    const ALCdevice *Device{ALContext->Device};
+    const ALsizei NumSends{Device->NumAuxSends};
+    const ALlistener &Listener = ALContext->Listener;
+
+    /* Set mixing buffers and get send parameters. */
+    voice->mDirect.Buffer = Device->Dry.Buffer;
+    ALeffectslot *SendSlots[MAX_SENDS];
+    ALfloat RoomRolloff[MAX_SENDS];
+    ALfloat DecayDistance[MAX_SENDS];
+    ALfloat DecayLFDistance[MAX_SENDS];
+    ALfloat DecayHFDistance[MAX_SENDS];
+    for(ALsizei i{0};i < NumSends;i++)
+    {
+        SendSlots[i] = props->Send[i].Slot;
+        if(!SendSlots[i] && i == 0)
+            SendSlots[i] = ALContext->DefaultSlot.get();
+        if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL)
+        {
+            SendSlots[i] = nullptr;
+            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{std::log10(REVERB_DECAY_GAIN) / std::log10(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->mSend[i].Buffer = {};
+        else
+            voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
+    }
+
+    /* Transform source to listener space (convert to head relative) */
+    alu::Vector Position{props->Position[0], props->Position[1], props->Position[2], 1.0f};
+    alu::Vector Velocity{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f};
+    alu::Vector Direction{props->Direction[0], props->Direction[1], props->Direction[2], 0.0f};
+    if(props->HeadRelative == AL_FALSE)
+    {
+        /* Transform source vectors */
+        Position = Listener.Params.Matrix * Position;
+        Velocity = Listener.Params.Matrix * Velocity;
+        Direction = Listener.Params.Matrix * Direction;
+    }
+    else
+    {
+        /* Offset the source velocity to be relative of the listener velocity */
+        Velocity += Listener.Params.Velocity;
+    }
+
+    const bool directional{Direction.normalize() > 0.0f};
+    alu::Vector ToSource{Position[0], Position[1], Position[2], 0.0f};
+    const ALfloat Distance{ToSource.normalize()};
+
+    /* Initial source gain */
+    ALfloat DryGain{props->Gain};
+    ALfloat DryGainHF{1.0f};
+    ALfloat DryGainLF{1.0f};
+    ALfloat WetGain[MAX_SENDS], WetGainHF[MAX_SENDS], WetGainLF[MAX_SENDS];
+    for(ALsizei i{0};i < NumSends;i++)
+    {
+        WetGain[i] = props->Gain;
+        WetGainHF[i] = 1.0f;
+        WetGainLF[i] = 1.0f;
+    }
+
+    /* Calculate distance attenuation */
+    ALfloat ClampedDist{Distance};
+
+    switch(Listener.Params.SourceDistanceModel ?
+           props->mDistanceModel : Listener.Params.mDistanceModel)
+    {
+        case DistanceModel::InverseClamped:
+            ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
+            if(props->MaxDistance < props->RefDistance) break;
+            /*fall-through*/
+        case DistanceModel::Inverse:
+            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(ALsizei i{0};i < NumSends;i++)
+                {
+                    dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]);
+                    if(dist > 0.0f) WetGain[i] *= props->RefDistance / dist;
+                }
+            }
+            break;
+
+        case DistanceModel::LinearClamped:
+            ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
+            if(props->MaxDistance < props->RefDistance) break;
+            /*fall-through*/
+        case DistanceModel::Linear:
+            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(ALsizei 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 DistanceModel::ExponentClamped:
+            ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance);
+            if(props->MaxDistance < props->RefDistance) break;
+            /*fall-through*/
+        case DistanceModel::Exponent:
+            if(!(ClampedDist > 0.0f && props->RefDistance > 0.0f))
+                ClampedDist = props->RefDistance;
+            else
+            {
+                DryGain *= std::pow(ClampedDist/props->RefDistance, -props->RolloffFactor);
+                for(ALsizei i{0};i < NumSends;i++)
+                    WetGain[i] *= std::pow(ClampedDist/props->RefDistance, -RoomRolloff[i]);
+            }
+            break;
+
+        case DistanceModel::Disable:
+            ClampedDist = props->RefDistance;
+            break;
+    }
+
+    /* Calculate directional soundcones */
+    if(directional && props->InnerAngle < 360.0f)
+    {
+        const ALfloat Angle{Rad2Deg(std::acos(-aluDotproduct(Direction, ToSource)) *
+            ConeScale * 2.0f)};
+
+        ALfloat ConeVolume, ConeHF;
+        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)
+            std::transform(std::begin(WetGain), std::begin(WetGain)+NumSends, std::begin(WetGain),
+                [ConeVolume](ALfloat gain) noexcept -> ALfloat { return gain * ConeVolume; }
+            );
+        if(props->WetGainHFAuto)
+            std::transform(std::begin(WetGainHF), std::begin(WetGainHF)+NumSends,
+                std::begin(WetGainHF),
+                [ConeHF](ALfloat gain) noexcept -> ALfloat { return gain * 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(ALsizei 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{std::pow(AIRABSORBGAINHF, meters_base * props->AirAbsorptionFactor)};
+            DryGainHF *= hfattn;
+            std::transform(std::begin(WetGainHF), std::begin(WetGainHF)+NumSends,
+                std::begin(WetGainHF),
+                [hfattn](ALfloat gain) noexcept -> ALfloat { return gain * 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(ALsizei i{0};i < NumSends;i++)
+            {
+                if(!(DecayDistance[i] > 0.0f))
+                    continue;
+
+                const ALfloat gain{std::pow(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)
+                {
+                    ALfloat gainhf{std::pow(REVERB_DECAY_GAIN, meters_base/DecayHFDistance[i])};
+                    WetGainHF[i] *= minf(gainhf / gain, 1.0f);
+                    ALfloat gainlf{std::pow(REVERB_DECAY_GAIN, meters_base/DecayLFDistance[i])};
+                    WetGainLF[i] *= minf(gainlf / gain, 1.0f);
+                }
+            }
+        }
+    }
+
+
+    /* Initial source pitch */
+    ALfloat Pitch{props->Pitch};
+
+    /* Calculate velocity-based doppler effect */
+    ALfloat DopplerFactor{props->DopplerFactor * Listener.Params.DopplerFactor};
+    if(DopplerFactor > 0.0f)
+    {
+        const alu::Vector &lvelocity = Listener.Params.Velocity;
+        ALfloat vss{aluDotproduct(Velocity, ToSource) * -DopplerFactor};
+        ALfloat vls{aluDotproduct(lvelocity, ToSource) * -DopplerFactor};
+
+        const ALfloat SpeedOfSound{Listener.Params.SpeedOfSound};
+        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 = std::numeric_limits<float>::infinity();
+        }
+        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 *= static_cast<ALfloat>(voice->mFrequency)/static_cast<ALfloat>(Device->Frequency);
+    if(Pitch > static_cast<ALfloat>(MAX_PITCH))
+        voice->mStep = MAX_PITCH<<FRACTIONBITS;
+    else
+        voice->mStep = maxi(fastf2i(Pitch * FRACTIONONE), 1);
+    if(props->mResampler == BSinc24Resampler)
+        BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc24);
+    else if(props->mResampler == BSinc12Resampler)
+        BsincPrepare(voice->mStep, &voice->mResampleState.bsinc, &bsinc12);
+    voice->mResampler = SelectResampler(props->mResampler);
+
+    ALfloat spread{0.0f};
+    if(props->Radius > Distance)
+        spread = al::MathDefs<float>::Tau() - Distance/props->Radius*al::MathDefs<float>::Pi();
+    else if(Distance > 0.0f)
+        spread = std::asin(props->Radius/Distance) * 2.0f;
+
+    CalcPanningAndFilters(voice, ToSource[0], ToSource[1], ToSource[2]*ZScale,
+        Distance*Listener.Params.MetersPerUnit, spread, DryGain, DryGainHF, DryGainLF, WetGain,
+        WetGainLF, WetGainHF, SendSlots, props, Listener, Device);
+}
+
+void CalcSourceParams(ALvoice *voice, ALCcontext *context, bool force)
+{
+    ALvoiceProps *props{voice->mUpdate.exchange(nullptr, std::memory_order_acq_rel)};
+    if(!props && !force) return;
+
+    if(props)
+    {
+        voice->mProps = *props;
+
+        AtomicReplaceHead(context->FreeVoiceProps, props);
+    }
+
+    if((voice->mProps.mSpatializeMode == SpatializeAuto && voice->mFmtChannels == FmtMono) ||
+       voice->mProps.mSpatializeMode == SpatializeOn)
+        CalcAttnSourceParams(voice, &voice->mProps, context);
+    else
+        CalcNonAttnSourceParams(voice, &voice->mProps, context);
+}
+
+
+void ProcessParamUpdates(ALCcontext *ctx, const ALeffectslotArray *slots)
+{
+    IncrementRef(&ctx->UpdateCount);
+    if(LIKELY(!ctx->HoldUpdates.load(std::memory_order_acquire)))
+    {
+        bool cforce{CalcContextParams(ctx)};
+        bool force{CalcListenerParams(ctx) || cforce};
+        force = std::accumulate(slots->begin(), slots->end(), force,
+            [ctx,cforce](bool force, ALeffectslot *slot) -> bool
+            { return CalcEffectSlotParams(slot, ctx, cforce) | force; }
+        );
+
+        std::for_each(ctx->Voices->begin(),
+            ctx->Voices->begin() + ctx->VoiceCount.load(std::memory_order_acquire),
+            [ctx,force](ALvoice &voice) -> void
+            {
+                ALuint sid{voice.mSourceID.load(std::memory_order_acquire)};
+                if(sid) CalcSourceParams(&voice, ctx, force);
+            }
+        );
+    }
+    IncrementRef(&ctx->UpdateCount);
+}
+
+void ProcessContext(ALCcontext *ctx, const ALsizei SamplesToDo)
+{
+    ASSUME(SamplesToDo > 0);
+
+    const ALeffectslotArray *auxslots{ctx->ActiveAuxSlots.load(std::memory_order_acquire)};
+
+    /* Process pending propery updates for objects on the context. */
+    ProcessParamUpdates(ctx, auxslots);
+
+    /* Clear auxiliary effect slot mixing buffers. */
+    std::for_each(auxslots->begin(), auxslots->end(),
+        [SamplesToDo](ALeffectslot *slot) -> void
+        {
+            for(auto &buffer : slot->MixBuffer)
+                std::fill_n(buffer.begin(), SamplesToDo, 0.0f);
+        }
+    );
+
+    /* Process voices that have a playing source. */
+    std::for_each(ctx->Voices->begin(),
+        ctx->Voices->begin() + ctx->VoiceCount.load(std::memory_order_acquire),
+        [SamplesToDo,ctx](ALvoice &voice) -> void
+        {
+            const ALvoice::State vstate{voice.mPlayState.load(std::memory_order_acquire)};
+            if(vstate == ALvoice::Stopped) return;
+            const ALuint sid{voice.mSourceID.load(std::memory_order_relaxed)};
+            if(voice.mStep < 1) return;
+
+            MixVoice(&voice, vstate, sid, ctx, SamplesToDo);
+        }
+    );
+
+    /* Process effects. */
+    if(auxslots->size() < 1) return;
+    auto slots = auxslots->data();
+    auto slots_end = slots + auxslots->size();
+
+    /* First sort the slots into scratch storage, so that effects come before
+     * their effect target (or their targets' target).
+     */
+    auto sorted_slots = const_cast<ALeffectslot**>(slots_end);
+    auto sorted_slots_end = sorted_slots;
+    auto in_chain = [](const ALeffectslot *slot1, const ALeffectslot *slot2) noexcept -> bool
+    {
+        while((slot1=slot1->Params.Target) != nullptr) {
+            if(slot1 == slot2) return true;
+        }
+        return false;
+    };
+
+    *sorted_slots_end = *slots;
+    ++sorted_slots_end;
+    while(++slots != slots_end)
+    {
+        /* If this effect slot targets an effect slot already in the list (i.e.
+         * slots outputs to something in sorted_slots), directly or indirectly,
+         * insert it prior to that element.
+         */
+        auto checker = sorted_slots;
+        do {
+            if(in_chain(*slots, *checker)) break;
+        } while(++checker != sorted_slots_end);
+
+        checker = std::move_backward(checker, sorted_slots_end, sorted_slots_end+1);
+        *--checker = *slots;
+        ++sorted_slots_end;
+    }
+
+    std::for_each(sorted_slots, sorted_slots_end,
+        [SamplesToDo](const ALeffectslot *slot) -> void
+        {
+            EffectState *state{slot->Params.mEffectState};
+            state->process(SamplesToDo, slot->Wet.Buffer.data(),
+                static_cast<ALsizei>(slot->Wet.Buffer.size()), state->mOutTarget);
+        }
+    );
+}
+
+
+void ApplyStablizer(FrontStablizer *Stablizer, const al::span<FloatBufferLine> Buffer,
+    const ALuint lidx, const ALuint ridx, const ALuint cidx, const ALsizei SamplesToDo)
+{
+    ASSUME(SamplesToDo > 0);
+
+    /* Apply a delay to all channels, except the front-left and front-right, so
+     * they maintain correct timing.
+     */
+    const size_t NumChannels{Buffer.size()};
+    for(size_t i{0u};i < NumChannels;i++)
+    {
+        if(i == lidx || i == ridx)
+            continue;
+
+        auto &DelayBuf = Stablizer->DelayBuf[i];
+        auto buffer_end = Buffer[i].begin() + SamplesToDo;
+        if(LIKELY(SamplesToDo >= ALsizei{FrontStablizer::DelayLength}))
+        {
+            auto delay_end = std::rotate(Buffer[i].begin(),
+                buffer_end - FrontStablizer::DelayLength, buffer_end);
+            std::swap_ranges(Buffer[i].begin(), delay_end, std::begin(DelayBuf));
+        }
+        else
+        {
+            auto delay_start = std::swap_ranges(Buffer[i].begin(), buffer_end,
+                std::begin(DelayBuf));
+            std::rotate(std::begin(DelayBuf), delay_start, std::end(DelayBuf));
+        }
+    }
+
+    ALfloat (&lsplit)[2][BUFFERSIZE] = Stablizer->LSplit;
+    ALfloat (&rsplit)[2][BUFFERSIZE] = Stablizer->RSplit;
+    auto &tmpbuf = Stablizer->TempBuf;
+
+    /* This applies the band-splitter, preserving phase at the cost of some
+     * delay. The shorter the delay, the more error seeps into the result.
+     */
+    auto apply_splitter = [&tmpbuf,SamplesToDo](const FloatBufferLine &Buffer,
+        ALfloat (&DelayBuf)[FrontStablizer::DelayLength], BandSplitter &Filter,
+        ALfloat (&splitbuf)[2][BUFFERSIZE]) -> void
+    {
+        /* Combine the delayed samples and the input samples into the temp
+         * buffer, in reverse. Then copy the final samples back into the delay
+         * buffer for next time. Note that the delay buffer's samples are
+         * stored backwards here.
+         */
+        auto tmpbuf_end = std::begin(tmpbuf) + SamplesToDo;
+        std::copy_n(std::begin(DelayBuf), FrontStablizer::DelayLength, tmpbuf_end);
+        std::reverse_copy(Buffer.begin(), Buffer.begin()+SamplesToDo, std::begin(tmpbuf));
+        std::copy_n(std::begin(tmpbuf), FrontStablizer::DelayLength, std::begin(DelayBuf));
+
+        /* Apply an all-pass on the reversed signal, then reverse the samples
+         * to get the forward signal with a reversed phase shift.
+         */
+        Filter.applyAllpass(tmpbuf, SamplesToDo+FrontStablizer::DelayLength);
+        std::reverse(std::begin(tmpbuf), tmpbuf_end+FrontStablizer::DelayLength);
+
+        /* Now apply the band-splitter, combining its phase shift with the
+         * reversed phase shift, restoring the original phase on the split
+         * signal.
+         */
+        Filter.process(splitbuf[1], splitbuf[0], tmpbuf, SamplesToDo);
+    };
+    apply_splitter(Buffer[lidx], Stablizer->DelayBuf[lidx], Stablizer->LFilter, lsplit);
+    apply_splitter(Buffer[ridx], Stablizer->DelayBuf[ridx], Stablizer->RFilter, rsplit);
+
+    for(ALsizei i{0};i < SamplesToDo;i++)
+    {
+        ALfloat lfsum{lsplit[0][i] + rsplit[0][i]};
+        ALfloat hfsum{lsplit[1][i] + rsplit[1][i]};
+        ALfloat 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.
+         */
+        ALfloat m{lfsum*std::cos(1.0f/3.0f * (al::MathDefs<float>::Pi()*0.5f)) +
+            hfsum*std::cos(1.0f/4.0f * (al::MathDefs<float>::Pi()*0.5f))};
+        ALfloat c{lfsum*std::sin(1.0f/3.0f * (al::MathDefs<float>::Pi()*0.5f)) +
+            hfsum*std::sin(1.0f/4.0f * (al::MathDefs<float>::Pi()*0.5f))};
+
+        /* 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;
+    }
+}
+
+void ApplyDistanceComp(const al::span<FloatBufferLine> Samples, const ALsizei SamplesToDo,
+    const DistanceComp::DistData *distcomp)
+{
+    ASSUME(SamplesToDo > 0);
+
+    for(auto &chanbuffer : Samples)
+    {
+        const ALfloat gain{distcomp->Gain};
+        const ALsizei base{distcomp->Length};
+        ALfloat *distbuf{al::assume_aligned<16>(distcomp->Buffer)};
+        ++distcomp;
+
+        if(base < 1)
+            continue;
+
+        ALfloat *inout{al::assume_aligned<16>(chanbuffer.data())};
+        auto inout_end = inout + SamplesToDo;
+        if(LIKELY(SamplesToDo >= base))
+        {
+            auto delay_end = std::rotate(inout, inout_end - base, inout_end);
+            std::swap_ranges(inout, delay_end, distbuf);
+        }
+        else
+        {
+            auto delay_start = std::swap_ranges(inout, inout_end, distbuf);
+            std::rotate(distbuf, delay_start, distbuf + base);
+        }
+        std::transform(inout, inout_end, inout, std::bind(std::multiplies<float>{}, _1, gain));
+    }
+}
+
+void ApplyDither(const al::span<FloatBufferLine> Samples, ALuint *dither_seed,
+    const ALfloat quant_scale, const ALsizei SamplesToDo)
+{
+    /* Dithering. Generate whitenoise (uniform distribution of random values
+     * between -1 and +1) and add it to the sample values, after scaling up to
+     * the desired quantization depth amd before rounding.
+     */
+    const ALfloat invscale{1.0f / quant_scale};
+    ALuint seed{*dither_seed};
+    auto dither_channel = [&seed,invscale,quant_scale,SamplesToDo](FloatBufferLine &input) -> void
+    {
+        ASSUME(SamplesToDo > 0);
+        auto dither_sample = [&seed,invscale,quant_scale](const ALfloat sample) noexcept -> ALfloat
+        {
+            ALfloat val{sample * quant_scale};
+            ALuint rng0{dither_rng(&seed)};
+            ALuint rng1{dither_rng(&seed)};
+            val += static_cast<ALfloat>(rng0*(1.0/UINT_MAX) - rng1*(1.0/UINT_MAX));
+            return fast_roundf(val) * invscale;
+        };
+        std::transform(input.begin(), input.begin()+SamplesToDo, input.begin(), dither_sample);
+    };
+    std::for_each(Samples.begin(), Samples.end(), dither_channel);
+    *dither_seed = seed;
+}
+
+
+/* Base template left undefined. Should be marked =delete, but Clang 3.8.1
+ * chokes on that given the inline specializations.
+ */
+template<typename T>
+inline T SampleConv(ALfloat) noexcept;
+
+template<> inline ALfloat SampleConv(ALfloat val) noexcept
+{ return val; }
+template<> inline ALint SampleConv(ALfloat val) noexcept
+{
+    /* Floats have a 23-bit mantissa, plus an implied 1 bit and a sign bit.
+     * This means a normalized float has at most 25 bits of signed precision.
+     * When scaling and clamping for a signed 32-bit integer, these following
+     * values are the best a float can give.
+     */
+    return fastf2i(clampf(val*2147483648.0f, -2147483648.0f, 2147483520.0f));
+}
+template<> inline ALshort SampleConv(ALfloat val) noexcept
+{ return fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f)); }
+template<> inline ALbyte SampleConv(ALfloat val) noexcept
+{ return fastf2i(clampf(val*128.0f, -128.0f, 127.0f)); }
+
+/* Define unsigned output variations. */
+template<> inline ALuint SampleConv(ALfloat val) noexcept
+{ return SampleConv<ALint>(val) + 2147483648u; }
+template<> inline ALushort SampleConv(ALfloat val) noexcept
+{ return SampleConv<ALshort>(val) + 32768; }
+template<> inline ALubyte SampleConv(ALfloat val) noexcept
+{ return SampleConv<ALbyte>(val) + 128; }
+
+template<DevFmtType T>
+void Write(const al::span<const FloatBufferLine> InBuffer, ALvoid *OutBuffer, const size_t Offset,
+    const ALsizei SamplesToDo)
+{
+    using SampleType = typename DevFmtTypeTraits<T>::Type;
+
+    const size_t numchans{InBuffer.size()};
+    ASSUME(numchans > 0);
+
+    SampleType *outbase = static_cast<SampleType*>(OutBuffer) + Offset*numchans;
+    auto conv_channel = [&outbase,SamplesToDo,numchans](const FloatBufferLine &inbuf) -> void
+    {
+        ASSUME(SamplesToDo > 0);
+        SampleType *out{outbase++};
+        auto conv_sample = [numchans,&out](const ALfloat s) noexcept -> void
+        {
+            *out = SampleConv<SampleType>(s);
+            out += numchans;
+        };
+        std::for_each(inbuf.begin(), inbuf.begin()+SamplesToDo, conv_sample);
+    };
+    std::for_each(InBuffer.cbegin(), InBuffer.cend(), conv_channel);
+}
+
+} // namespace
+
+void aluMixData(ALCdevice *device, ALvoid *OutBuffer, ALsizei NumSamples)
+{
+    FPUCtl mixer_mode{};
+    for(ALsizei SamplesDone{0};SamplesDone < NumSamples;)
+    {
+        const ALsizei SamplesToDo{mini(NumSamples-SamplesDone, BUFFERSIZE)};
+
+        /* Clear main mixing buffers. */
+        std::for_each(device->MixBuffer.begin(), device->MixBuffer.end(),
+            [SamplesToDo](std::array<ALfloat,BUFFERSIZE> &buffer) -> void
+            { std::fill_n(buffer.begin(), SamplesToDo, 0.0f); }
+        );
+
+        /* Increment the mix count at the start (lsb should now be 1). */
+        IncrementRef(&device->MixCount);
+
+        /* For each context on this device, process and mix its sources and
+         * effects.
+         */
+        for(ALCcontext *ctx : *device->mContexts.load(std::memory_order_acquire))
+            ProcessContext(ctx, SamplesToDo);
+
+        /* 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 a stable
+         * conversion.
+         */
+        device->SamplesDone += SamplesToDo;
+        device->ClockBase += std::chrono::seconds{device->SamplesDone / device->Frequency};
+        device->SamplesDone %= device->Frequency;
+
+        /* Increment the mix count at the end (lsb should now be 0). */
+        IncrementRef(&device->MixCount);
+
+        /* Apply any needed post-process for finalizing the Dry mix to the
+         * RealOut (Ambisonic decode, UHJ encode, etc).
+         */
+        if(LIKELY(device->PostProcess))
+            device->PostProcess(device, SamplesToDo);
+        const al::span<FloatBufferLine> RealOut{device->RealOut.Buffer};
+
+        /* Apply front image stablization for surround sound, if applicable. */
+        if(device->Stablizer)
+        {
+            const int lidx{GetChannelIdxByName(device->RealOut, FrontLeft)};
+            const int ridx{GetChannelIdxByName(device->RealOut, FrontRight)};
+            const int cidx{GetChannelIdxByName(device->RealOut, FrontCenter)};
+            assert(lidx >= 0 && ridx >= 0 && cidx >= 0);
+
+            ApplyStablizer(device->Stablizer.get(), RealOut, lidx, ridx, cidx, SamplesToDo);
+        }
+
+        /* Apply compression, limiting sample amplitude if needed or desired. */
+        if(Compressor *comp{device->Limiter.get()})
+            comp->process(SamplesToDo, RealOut.data());
+
+        /* Apply delays and attenuation for mismatched speaker distances. */
+        ApplyDistanceComp(RealOut, SamplesToDo, device->ChannelDelay.as_span().cbegin());
+
+        /* Apply dithering. The compressor should have left enough headroom for
+         * the dither noise to not saturate.
+         */
+        if(device->DitherDepth > 0.0f)
+            ApplyDither(RealOut, &device->DitherSeed, device->DitherDepth, SamplesToDo);
+
+        if(LIKELY(OutBuffer))
+        {
+            /* Finally, interleave and convert samples, writing to the device's
+             * output buffer.
+             */
+            switch(device->FmtType)
+            {
+#define HANDLE_WRITE(T) case T:                                            \
+    Write<T>(RealOut, OutBuffer, SamplesDone, SamplesToDo); break;
+                HANDLE_WRITE(DevFmtByte)
+                HANDLE_WRITE(DevFmtUByte)
+                HANDLE_WRITE(DevFmtShort)
+                HANDLE_WRITE(DevFmtUShort)
+                HANDLE_WRITE(DevFmtInt)
+                HANDLE_WRITE(DevFmtUInt)
+                HANDLE_WRITE(DevFmtFloat)
+#undef HANDLE_WRITE
+            }
+        }
+
+        SamplesDone += SamplesToDo;
+    }
+}
+
+
+void aluHandleDisconnect(ALCdevice *device, const char *msg, ...)
+{
+    if(!device->Connected.exchange(false, std::memory_order_acq_rel))
+        return;
+
+    AsyncEvent evt{EventType_Disconnected};
+    evt.u.user.type = AL_EVENT_TYPE_DISCONNECTED_SOFT;
+    evt.u.user.id = 0;
+    evt.u.user.param = 0;
+
+    va_list args;
+    va_start(args, msg);
+    int msglen{vsnprintf(evt.u.user.msg, sizeof(evt.u.user.msg), msg, args)};
+    va_end(args);
+
+    if(msglen < 0 || static_cast<size_t>(msglen) >= sizeof(evt.u.user.msg))
+        evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0;
+
+    for(ALCcontext *ctx : *device->mContexts.load())
+    {
+        const ALbitfieldSOFT enabledevt{ctx->EnabledEvts.load(std::memory_order_acquire)};
+        if((enabledevt&EventType_Disconnected))
+        {
+            RingBuffer *ring{ctx->AsyncEvents.get()};
+            auto evt_data = ring->getWriteVector().first;
+            if(evt_data.len > 0)
+            {
+                new (evt_data.buf) AsyncEvent{evt};
+                ring->writeAdvance(1);
+                ctx->EventSem.post();
+            }
+        }
+
+        auto stop_voice = [](ALvoice &voice) -> void
+        {
+            voice.mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
+            voice.mLoopBuffer.store(nullptr, std::memory_order_relaxed);
+            voice.mSourceID.store(0u, std::memory_order_relaxed);
+            voice.mPlayState.store(ALvoice::Stopped, std::memory_order_release);
+        };
+        std::for_each(ctx->Voices->begin(),
+            ctx->Voices->begin() + ctx->VoiceCount.load(std::memory_order_acquire),
+            stop_voice);
+    }
+}