Rename Alc to alc

1 files changed, 0 insertions, 1798 deletions

diff --git a/Alc/alu.cpp b/Alc/alu.cpp
deleted file mode 100644
index cc1a5a98..00000000
--- a/Alc/alu.cpp
+++ /dev/null
@@ -1,1798 +0,0 @@
-/**
- * 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);
-    }
-}