/** * 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 #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "AL/efx.h" #include "al/auxeffectslot.h" #include "al/buffer.h" #include "al/effect.h" #include "al/event.h" #include "al/listener.h" #include "alcmain.h" #include "alcontext.h" #include "almalloc.h" #include "alnumeric.h" #include "alspan.h" #include "alstring.h" #include "ambidefs.h" #include "atomic.h" #include "bformatdec.h" #include "bs2b.h" #include "cpu_caps.h" #include "devformat.h" #include "effects/base.h" #include "filters/biquad.h" #include "filters/nfc.h" #include "filters/splitter.h" #include "fpu_ctrl.h" #include "front_stablizer.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 "strutils.h" #include "threads.h" #include "uhjfilter.h" #include "vecmat.h" #include "voice.h" #include "bsinc_tables.h" struct CTag; #ifdef HAVE_SSE struct SSETag; #endif #ifdef HAVE_SSE2 struct SSE2Tag; #endif #ifdef HAVE_SSE4_1 struct SSE4Tag; #endif #ifdef HAVE_NEON struct NEONTag; #endif struct CopyTag; struct PointTag; struct LerpTag; struct CubicTag; struct BSincTag; struct FastBSincTag; static_assert(!(MAX_RESAMPLER_PADDING&1) && MAX_RESAMPLER_PADDING >= BSINC_POINTS_MAX, "MAX_RESAMPLER_PADDING is not a multiple of two, or is too small"); namespace { using namespace std::placeholders; float InitConeScale() { float ret{1.0f}; if(auto optval = al::getenv("__ALSOFT_HALF_ANGLE_CONES")) { if(al::strcasecmp(optval->c_str(), "true") == 0 || strtol(optval->c_str(), nullptr, 0) == 1) ret *= 0.5f; } return ret; } float InitZScale() { float ret{1.0f}; if(auto optval = al::getenv("__ALSOFT_REVERSE_Z")) { if(al::strcasecmp(optval->c_str(), "true") == 0 || strtol(optval->c_str(), nullptr, 0) == 1) ret *= -1.0f; } return ret; } } // namespace /* Cone scalar */ const float ConeScale{InitConeScale()}; /* Localized Z scalar for mono sources */ const float ZScale{InitZScale()}; namespace { struct ChanMap { Channel channel; float angle; float elevation; }; HrtfDirectMixerFunc MixDirectHrtf{MixDirectHrtf_}; inline HrtfDirectMixerFunc SelectHrtfMixer(void) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixDirectHrtf_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixDirectHrtf_; #endif return MixDirectHrtf_; } inline void BsincPrepare(const ALuint increment, BsincState *state, const BSincTable *table) { size_t si{BSINC_SCALE_COUNT - 1}; float sf{0.0f}; if(increment > FRACTIONONE) { sf = FRACTIONONE / static_cast(increment); sf = maxf(0.0f, (BSINC_SCALE_COUNT-1) * (sf-table->scaleBase) * table->scaleRange); si = float2uint(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 - static_cast(si))); } state->sf = sf; state->m = table->m[si]; state->l = (state->m/2) - 1; state->filter = table->Tab + table->filterOffset[si]; } inline ResamplerFunc SelectResampler(Resampler resampler, ALuint increment) { switch(resampler) { case Resampler::Point: return Resample_; case Resampler::Linear: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE4_1 if((CPUCapFlags&CPU_CAP_SSE4_1)) return Resample_; #endif #ifdef HAVE_SSE2 if((CPUCapFlags&CPU_CAP_SSE2)) return Resample_; #endif return Resample_; case Resampler::Cubic: return Resample_; case Resampler::BSinc12: case Resampler::BSinc24: if(increment <= FRACTIONONE) { /* fall-through */ case Resampler::FastBSinc12: case Resampler::FastBSinc24: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Resample_; #endif return Resample_; } #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Resample_; #endif return Resample_; } return Resample_; } } // namespace void aluInit(void) { MixDirectHrtf = SelectHrtfMixer(); } ResamplerFunc PrepareResampler(Resampler resampler, ALuint increment, InterpState *state) { switch(resampler) { case Resampler::Point: case Resampler::Linear: case Resampler::Cubic: break; case Resampler::FastBSinc12: case Resampler::BSinc12: BsincPrepare(increment, &state->bsinc, &bsinc12); break; case Resampler::FastBSinc24: case Resampler::BSinc24: BsincPrepare(increment, &state->bsinc, &bsinc24); break; } return SelectResampler(resampler, increment); } void ALCdevice::ProcessHrtf(const size_t SamplesToDo) { /* HRTF is stereo output only. */ const ALuint lidx{RealOut.ChannelIndex[FrontLeft]}; const ALuint ridx{RealOut.ChannelIndex[FrontRight]}; MixDirectHrtf(RealOut.Buffer[lidx], RealOut.Buffer[ridx], Dry.Buffer, HrtfAccumData, mHrtfState.get(), SamplesToDo); } void ALCdevice::ProcessAmbiDec(const size_t SamplesToDo) { AmbiDecoder->process(RealOut.Buffer, Dry.Buffer.data(), SamplesToDo); } void ALCdevice::ProcessUhj(const size_t SamplesToDo) { /* UHJ is stereo output only. */ const ALuint lidx{RealOut.ChannelIndex[FrontLeft]}; const ALuint ridx{RealOut.ChannelIndex[FrontRight]}; /* Encode to stereo-compatible 2-channel UHJ output. */ Uhj_Encoder->encode(RealOut.Buffer[lidx], RealOut.Buffer[ridx], Dry.Buffer.data(), SamplesToDo); } void ALCdevice::ProcessBs2b(const size_t SamplesToDo) { /* First, decode the ambisonic mix to the "real" output. */ AmbiDecoder->process(RealOut.Buffer, Dry.Buffer.data(), SamplesToDo); /* BS2B is stereo output only. */ const ALuint lidx{RealOut.ChannelIndex[FrontLeft]}; const ALuint ridx{RealOut.ChannelIndex[FrontRight]}; /* Now apply the BS2B binaural/crossfeed filter. */ bs2b_cross_feed(Bs2b.get(), RealOut.Buffer[lidx].data(), RealOut.Buffer[ridx].data(), SamplesToDo); } 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; } auto GetAmbiScales(AmbiNorm scaletype) noexcept -> const std::array& { if(scaletype == AmbiNorm::FuMa) return AmbiScale::FromFuMa; if(scaletype == AmbiNorm::SN3D) return AmbiScale::FromSN3D; return AmbiScale::FromN3D; } auto GetAmbiLayout(AmbiLayout layouttype) noexcept -> const std::array& { if(layouttype == AmbiLayout::FuMa) return AmbiIndex::FromFuMa; return AmbiIndex::FromACN; } auto GetAmbi2DLayout(AmbiLayout layouttype) noexcept -> const std::array& { if(layouttype == AmbiLayout::FuMa) return AmbiIndex::FromFuMa2D; return AmbiIndex::From2D; } 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 float 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->mUpdate.exchange(nullptr, std::memory_order_acq_rel)}; if(!props) return false; ALlistener &Listener = Context->mListener; Listener.Params.DopplerFactor = props->DopplerFactor; Listener.Params.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity; Listener.Params.SourceDistanceModel = props->SourceDistanceModel; Listener.Params.mDistanceModel = props->mDistanceModel; AtomicReplaceHead(Context->mFreeContextProps, props); return true; } bool CalcListenerParams(ALCcontext *Context) { ALlistener &Listener = Context->mListener; ALlistenerProps *props{Listener.Params.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->mGainBoost; Listener.Params.MetersPerUnit = props->MetersPerUnit; AtomicReplaceHead(Context->mFreeListenerProps, props); return true; } bool CalcEffectSlotParams(ALeffectslot *slot, ALeffectslot **sorted_slots, ALCcontext *context) { ALeffectslotProps *props{slot->Params.Update.exchange(nullptr, std::memory_order_acq_rel)}; if(!props) return false; /* If the effect slot target changed, clear the first sorted entry to force * a re-sort. */ if(slot->Params.Target != props->Target) *sorted_slots = nullptr; 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 = false; slot->Params.AirAbsorptionGainHF = 1.0f; } EffectState *state{props->State}; props->State = nullptr; EffectState *oldstate{slot->Params.mEffectState}; slot->Params.mEffectState = state; /* Only release the old state if it won't get deleted, since we can't be * deleting/freeing anything in the mixer. */ if(!oldstate->releaseIfNoDelete()) { /* Otherwise, if it would be deleted send it off with a release event. */ RingBuffer *ring{context->mAsyncEvents.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); } 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->mFreeEffectslotProps, props); EffectTarget output; if(ALeffectslot *target{slot->Params.Target}) output = EffectTarget{&target->Wet, nullptr}; else { ALCdevice *device{context->mDevice.get()}; 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 float abs_azi{std::fabs(azimuth)}; if(!(abs_azi >= al::MathDefs::Pi()*0.5f)) return std::copysign(minf(abs_azi*scale, al::MathDefs::Pi()*0.5f), azimuth); return azimuth; } /* Wraps the given value in radians to stay between [-pi,+pi] */ inline float WrapRadians(float r) { constexpr float Pi{al::MathDefs::Pi()}; constexpr float Pi2{al::MathDefs::Tau()}; if(r > Pi) return std::fmod(Pi+r, Pi2) - Pi; if(r < -Pi) return Pi - std::fmod(Pi-r, Pi2); return r; } /* Begin ambisonic rotation helpers. * * Rotating first-order B-Format just needs a straight-forward X/Y/Z rotation * matrix. Higher orders, however, are more complicated. The method implemented * here is a recursive algorithm (the rotation for first-order is used to help * generate the second-order rotation, which helps generate the third-order * rotation, etc). * * Adapted from * , * provided under the BSD 3-Clause license. * * Copyright (c) 2015, Archontis Politis * Copyright (c) 2019, Christopher Robinson * * The u, v, and w coefficients used for generating higher-order rotations are * precomputed since they're constant. The second-order coefficients are * followed by the third-order coefficients, etc. */ struct RotatorCoeffs { float u, v, w; template static std::array ConcatArrays(const std::array &lhs, const std::array &rhs) { std::array ret; auto iter = std::copy(lhs.cbegin(), lhs.cend(), ret.begin()); std::copy(rhs.cbegin(), rhs.cend(), iter); return ret; } template static std::array GenCoeffs() { std::array ret{}; auto coeffs = ret.begin(); for(int m{-l};m <= l;++m) { for(int n{-l};n <= l;++n) { // compute u,v,w terms of Eq.8.1 (Table I) const bool d{m == 0}; // the delta function d_m0 const float denom{static_cast((std::abs(n) == l) ? (2*l) * (2*l - 1) : (l*l - n*n))}; const int abs_m{std::abs(m)}; coeffs->u = std::sqrt(static_cast(l*l - m*m)/denom); coeffs->v = std::sqrt(static_cast(l+abs_m-1) * static_cast(l+abs_m) / denom) * (1.0f+d) * (1.0f - 2.0f*d) * 0.5f; coeffs->w = std::sqrt(static_cast(l-abs_m-1) * static_cast(l-abs_m) / denom) * (1.0f-d) * -0.5f; ++coeffs; } } return ret; } }; const auto RotatorCoeffArray = RotatorCoeffs::ConcatArrays(RotatorCoeffs::GenCoeffs<2>(), RotatorCoeffs::GenCoeffs<3>()); /** * Given the matrix, pre-filled with the (zeroth- and) first-order rotation * coefficients, this fills in the coefficients for the higher orders up to and * including the given order. The matrix is in ACN layout. */ void AmbiRotator(std::array,MAX_AMBI_CHANNELS> &matrix, const int order) { /* Don't do anything for < 2nd order. */ if(order < 2) return; auto P = [](const int i, const int l, const int a, const int n, const size_t last_band, const std::array,MAX_AMBI_CHANNELS> &R) { const float ri1{ R[static_cast(i+2)][ 1+2]}; const float rim1{R[static_cast(i+2)][-1+2]}; const float ri0{ R[static_cast(i+2)][ 0+2]}; auto vec = R[static_cast(a+l-1) + last_band].cbegin() + last_band; if(n == -l) return ri1*vec[0] + rim1*vec[static_cast(l-1)*size_t{2}]; if(n == l) return ri1*vec[static_cast(l-1)*size_t{2}] - rim1*vec[0]; return ri0*vec[static_cast(n+l-1)]; }; auto U = [P](const int l, const int m, const int n, const size_t last_band, const std::array,MAX_AMBI_CHANNELS> &R) { return P(0, l, m, n, last_band, R); }; auto V = [P](const int l, const int m, const int n, const size_t last_band, const std::array,MAX_AMBI_CHANNELS> &R) { if(m > 0) { const bool d{m == 1}; const float p0{P( 1, l, m-1, n, last_band, R)}; const float p1{P(-1, l, -m+1, n, last_band, R)}; return d ? p0*std::sqrt(2.0f) : (p0 - p1); } const bool d{m == -1}; const float p0{P( 1, l, m+1, n, last_band, R)}; const float p1{P(-1, l, -m-1, n, last_band, R)}; return d ? p1*std::sqrt(2.0f) : (p0 + p1); }; auto W = [P](const int l, const int m, const int n, const size_t last_band, const std::array,MAX_AMBI_CHANNELS> &R) { assert(m != 0); if(m > 0) { const float p0{P( 1, l, m+1, n, last_band, R)}; const float p1{P(-1, l, -m-1, n, last_band, R)}; return p0 + p1; } const float p0{P( 1, l, m-1, n, last_band, R)}; const float p1{P(-1, l, -m+1, n, last_band, R)}; return p0 - p1; }; // compute rotation matrix of each subsequent band recursively auto coeffs = RotatorCoeffArray.cbegin(); size_t band_idx{4}, last_band{1}; for(int l{2};l <= order;++l) { size_t y{band_idx}; for(int m{-l};m <= l;++m,++y) { size_t x{band_idx}; for(int n{-l};n <= l;++n,++x) { float r{0.0f}; // computes Eq.8.1 const float u{coeffs->u}; if(u != 0.0f) r += u * U(l, m, n, last_band, matrix); const float v{coeffs->v}; if(v != 0.0f) r += v * V(l, m, n, last_band, matrix); const float w{coeffs->w}; if(w != 0.0f) r += w * W(l, m, n, last_band, matrix); matrix[y][x] = r; ++coeffs; } } last_band = band_idx; band_idx += static_cast(l)*size_t{2} + 1; } } /* End ambisonic rotation helpers. */ struct GainTriplet { float Base, HF, LF; }; void CalcPanningAndFilters(Voice *voice, const float xpos, const float ypos, const float zpos, const float Distance, const float Spread, const GainTriplet &DryGain, const al::span WetGain, ALeffectslot *(&SendSlots)[MAX_SENDS], const VoiceProps *props, const ALlistener &Listener, const ALCdevice *Device) { static const 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(Device->Frequency); const ALuint NumSends{Device->NumAuxSends}; const size_t num_channels{voice->mChans.size()}; ASSUME(num_channels > 0); for(auto &chandata : voice->mChans) { chandata.mDryParams.Hrtf.Target = HrtfFilter{}; chandata.mDryParams.Gains.Target.fill(0.0f); std::for_each(chandata.mWetParams.begin(), chandata.mWetParams.begin()+NumSends, [](SendParams ¶ms) -> void { params.Gains.Target.fill(0.0f); }); } DirectMode DirectChannels{props->DirectChannels}; const ChanMap *chans{nullptr}; float downmix_gain{1.0f}; switch(voice->mFmtChannels) { case FmtMono: chans = MonoMap; /* Mono buffers are never played direct. */ DirectChannels = DirectMode::Off; break; case FmtStereo: if(DirectChannels == DirectMode::Off) { /* Convert counter-clockwise to clock-wise, and wrap between * [-pi,+pi]. */ StereoMap[0].angle = WrapRadians(-props->StereoPan[0]); StereoMap[1].angle = WrapRadians(-props->StereoPan[1]); } chans = StereoMap; downmix_gain = 1.0f / 2.0f; break; case FmtRear: chans = RearMap; downmix_gain = 1.0f / 2.0f; break; case FmtQuad: chans = QuadMap; downmix_gain = 1.0f / 4.0f; break; case FmtX51: chans = X51Map; /* NOTE: Excludes LFE. */ downmix_gain = 1.0f / 5.0f; break; case FmtX61: chans = X61Map; /* NOTE: Excludes LFE. */ downmix_gain = 1.0f / 6.0f; break; case FmtX71: chans = X71Map; /* NOTE: Excludes LFE. */ downmix_gain = 1.0f / 7.0f; break; case FmtBFormat2D: case FmtBFormat3D: DirectChannels = DirectMode::Off; break; } voice->mFlags &= ~(VOICE_HAS_HRTF | VOICE_HAS_NFC); if(voice->mFmtChannels == FmtBFormat2D || voice->mFmtChannels == FmtBFormat3D) { /* Special handling for B-Format sources. */ if(Distance > std::numeric_limits::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 float mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)}; const float 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; } float 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 float ev{std::asin(clampf(ypos, -1.0f, 1.0f))}; /* Negate Z for right-handed coords with -Z in front. */ const float 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 according to channel scaling. */ const float scale0{GetAmbiScales(voice->mAmbiScaling)[0]}; ComputePanGains(&Device->Dry, coeffs, DryGain.Base*scale0, voice->mChans[0].mDryParams.Gains.Target); for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base*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 rotation matrix. Manually fill the zeroth- and first- * order elements, then construct the rotation for the higher * orders. */ std::array,MAX_AMBI_CHANNELS> shrot{}; shrot[0][0] = 1.0f; shrot[1][1] = U[0]; shrot[1][2] = -V[0]; shrot[1][3] = -N[0]; shrot[2][1] = -U[1]; shrot[2][2] = V[1]; shrot[2][3] = N[1]; shrot[3][1] = U[2]; shrot[3][2] = -V[2]; shrot[3][3] = -N[2]; AmbiRotator(shrot, static_cast(minu(voice->mAmbiOrder, Device->mAmbiOrder))); /* Convert the rotation matrix for input ordering and scaling, and * whether input is 2D or 3D. */ const uint8_t *index_map{(voice->mFmtChannels == FmtBFormat2D) ? GetAmbi2DLayout(voice->mAmbiLayout).data() : GetAmbiLayout(voice->mAmbiLayout).data()}; const float *scales{GetAmbiScales(voice->mAmbiScaling).data()}; static const uint8_t ChansPerOrder[MAX_AMBI_ORDER+1]{1, 3, 5, 7,}; static const uint8_t OrderOffset[MAX_AMBI_ORDER+1]{0, 1, 4, 9,}; for(size_t c{0};c < num_channels;c++) { const size_t acn{index_map[c]}; const size_t order{AmbiIndex::OrderFromChannel[acn]}; const size_t tocopy{ChansPerOrder[order]}; const size_t offset{OrderOffset[order]}; const float scale{scales[acn]}; auto in = shrot.cbegin() + offset; float coeffs[MAX_AMBI_CHANNELS]{}; for(size_t x{0};x < tocopy;++x) coeffs[offset+x] = in[x][acn] * scale; ComputePanGains(&Device->Dry, coeffs, DryGain.Base, voice->mChans[c].mDryParams.Gains.Target); for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[c].mWetParams[i].Gains.Target); } } } } else if(DirectChannels != DirectMode::Off && Device->FmtChans != DevFmtAmbi3D) { /* 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(size_t c{0};c < num_channels;c++) { ALuint idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)}; if(idx != INVALID_CHANNEL_INDEX) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base; else if(DirectChannels == DirectMode::RemixMismatch) { auto match_channel = [chans,c](const InputRemixMap &map) noexcept -> bool { return chans[c].channel == map.channel; }; auto remap = std::find_if(Device->RealOut.RemixMap.cbegin(), Device->RealOut.RemixMap.cend(), match_channel); if(remap != Device->RealOut.RemixMap.cend()) for(const auto &target : remap->targets) { idx = GetChannelIdxByName(Device->RealOut, target.channel); if(idx != INVALID_CHANNEL_INDEX) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base * target.mix; } } } /* Auxiliary sends still use normal channel panning since they mix to * B-Format, which can't channel-match. */ for(size_t c{0};c < num_channels;c++) { float coeffs[MAX_AMBI_CHANNELS]; CalcAngleCoeffs(chans[c].angle, chans[c].elevation, 0.0f, coeffs); for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, 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::epsilon()) { const float ev{std::asin(clampf(ypos, -1.0f, 1.0f))}; const float az{std::atan2(xpos, -zpos)}; /* Get the HRIR coefficients and delays just once, for the given * source direction. */ GetHrtfCoeffs(Device->mHrtf.get(), 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.Base * downmix_gain; /* Remaining channels use the same results as the first. */ for(size_t 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. */ float coeffs[MAX_AMBI_CHANNELS]; CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs); for(size_t c{0};c < num_channels;c++) { /* Skip LFE */ if(chans[c].channel == LFE) continue; for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * 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(size_t 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.get(), chans[c].elevation, chans[c].angle, std::numeric_limits::infinity(), Spread, voice->mChans[c].mDryParams.Hrtf.Target.Coeffs, voice->mChans[c].mDryParams.Hrtf.Target.Delay); voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain.Base; /* Normal panning for auxiliary sends. */ float coeffs[MAX_AMBI_CHANNELS]; CalcAngleCoeffs(chans[c].angle, chans[c].elevation, Spread, coeffs); for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, 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::epsilon()) { /* Calculate NFC filter coefficient if needed. */ if(Device->AvgSpeakerDist > 0.0f) { /* Clamp the distance for really close sources, to prevent * excessive bass. */ const float mdist{maxf(Distance, Device->AvgSpeakerDist/4.0f)}; const float w0{SPEEDOFSOUNDMETRESPERSEC / (mdist * Frequency)}; /* Adjust NFC filters. */ for(size_t 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. */ float coeffs[MAX_AMBI_CHANNELS]; if(Device->mRenderMode != StereoPair) CalcDirectionCoeffs({xpos, ypos, zpos}, Spread, coeffs); else { const float ev{std::asin(clampf(ypos, -1.0f, 1.0f))}; const float az{std::atan2(xpos, -zpos)}; CalcAngleCoeffs(ScaleAzimuthFront(az, 1.5f), ev, Spread, coeffs); } for(size_t c{0};c < num_channels;c++) { /* Special-case LFE */ if(chans[c].channel == LFE) { if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data()) { const ALuint idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)}; if(idx != INVALID_CHANNEL_INDEX) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base; } continue; } ComputePanGains(&Device->Dry, coeffs, DryGain.Base * downmix_gain, voice->mChans[c].mDryParams.Gains.Target); for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * downmix_gain, voice->mChans[c].mWetParams[i].Gains.Target); } } } else { if(Device->AvgSpeakerDist > 0.0f) { /* If the source distance is 0, simulate a plane-wave by using * infinite distance, which results in a w0 of 0. */ constexpr float w0{0.0f}; for(size_t c{0};c < num_channels;c++) voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0); voice->mFlags |= VOICE_HAS_NFC; } for(size_t c{0};c < num_channels;c++) { /* Special-case LFE */ if(chans[c].channel == LFE) { if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data()) { const ALuint idx{GetChannelIdxByName(Device->RealOut, chans[c].channel)}; if(idx != INVALID_CHANNEL_INDEX) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base; } continue; } float 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.Base, voice->mChans[c].mDryParams.Gains.Target); for(ALuint i{0};i < NumSends;i++) { if(const ALeffectslot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[c].mWetParams[i].Gains.Target); } } } } { const float hfNorm{props->Direct.HFReference / Frequency}; const float lfNorm{props->Direct.LFReference / Frequency}; voice->mDirect.FilterType = AF_None; if(DryGain.HF != 1.0f) voice->mDirect.FilterType |= AF_LowPass; if(DryGain.LF != 1.0f) voice->mDirect.FilterType |= AF_HighPass; auto &lowpass = voice->mChans[0].mDryParams.LowPass; auto &highpass = voice->mChans[0].mDryParams.HighPass; lowpass.setParamsFromSlope(BiquadType::HighShelf, hfNorm, DryGain.HF, 1.0f); highpass.setParamsFromSlope(BiquadType::LowShelf, lfNorm, DryGain.LF, 1.0f); for(size_t c{1};c < num_channels;c++) { voice->mChans[c].mDryParams.LowPass.copyParamsFrom(lowpass); voice->mChans[c].mDryParams.HighPass.copyParamsFrom(highpass); } } for(ALuint i{0};i < NumSends;i++) { const float hfNorm{props->Send[i].HFReference / Frequency}; const float lfNorm{props->Send[i].LFReference / Frequency}; voice->mSend[i].FilterType = AF_None; if(WetGain[i].HF != 1.0f) voice->mSend[i].FilterType |= AF_LowPass; if(WetGain[i].LF != 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.setParamsFromSlope(BiquadType::HighShelf, hfNorm, WetGain[i].HF, 1.0f); highpass.setParamsFromSlope(BiquadType::LowShelf, lfNorm, WetGain[i].LF, 1.0f); for(size_t c{1};c < num_channels;c++) { voice->mChans[c].mWetParams[i].LowPass.copyParamsFrom(lowpass); voice->mChans[c].mWetParams[i].HighPass.copyParamsFrom(highpass); } } } void CalcNonAttnSourceParams(Voice *voice, const VoiceProps *props, const ALCcontext *ALContext) { const ALCdevice *Device{ALContext->mDevice.get()}; ALeffectslot *SendSlots[MAX_SENDS]; voice->mDirect.Buffer = Device->Dry.Buffer; for(ALuint i{0};i < Device->NumAuxSends;i++) { SendSlots[i] = props->Send[i].Slot; if(!SendSlots[i] && i == 0) SendSlots[i] = ALContext->mDefaultSlot.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(voice->mFrequency) / static_cast(Device->Frequency) * props->Pitch; if(Pitch > float{MAX_PITCH}) voice->mStep = MAX_PITCH<mStep = maxu(fastf2u(Pitch * FRACTIONONE), 1); voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState); /* Calculate gains */ const ALlistener &Listener = ALContext->mListener; GainTriplet DryGain; DryGain.Base = minf(clampf(props->Gain, props->MinGain, props->MaxGain) * props->Direct.Gain * Listener.Params.Gain, GAIN_MIX_MAX); DryGain.HF = props->Direct.GainHF; DryGain.LF = props->Direct.GainLF; GainTriplet WetGain[MAX_SENDS]; for(ALuint i{0};i < Device->NumAuxSends;i++) { WetGain[i].Base = minf(clampf(props->Gain, props->MinGain, props->MaxGain) * props->Send[i].Gain * Listener.Params.Gain, GAIN_MIX_MAX); WetGain[i].HF = props->Send[i].GainHF; WetGain[i].LF = props->Send[i].GainLF; } CalcPanningAndFilters(voice, 0.0f, 0.0f, -1.0f, 0.0f, 0.0f, DryGain, WetGain, SendSlots, props, Listener, Device); } void CalcAttnSourceParams(Voice *voice, const VoiceProps *props, const ALCcontext *ALContext) { const ALCdevice *Device{ALContext->mDevice.get()}; const ALuint NumSends{Device->NumAuxSends}; const ALlistener &Listener = ALContext->mListener; /* Set mixing buffers and get send parameters. */ voice->mDirect.Buffer = Device->Dry.Buffer; ALeffectslot *SendSlots[MAX_SENDS]; float RoomRolloff[MAX_SENDS]; GainTriplet DecayDistance[MAX_SENDS]; for(ALuint i{0};i < NumSends;i++) { SendSlots[i] = props->Send[i].Slot; if(!SendSlots[i] && i == 0) SendSlots[i] = ALContext->mDefaultSlot.get(); if(!SendSlots[i] || SendSlots[i]->Params.EffectType == AL_EFFECT_NULL) { SendSlots[i] = nullptr; RoomRolloff[i] = 0.0f; DecayDistance[i].Base = 0.0f; DecayDistance[i].LF = 0.0f; DecayDistance[i].HF = 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].Base = SendSlots[i]->Params.DecayTime * SPEEDOFSOUNDMETRESPERSEC; DecayDistance[i].LF = DecayDistance[i].Base * SendSlots[i]->Params.DecayLFRatio; DecayDistance[i].HF = DecayDistance[i].Base * SendSlots[i]->Params.DecayHFRatio; if(SendSlots[i]->Params.DecayHFLimit) { const float 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). */ const float absorb_dist{std::log10(REVERB_DECAY_GAIN) / std::log10(airAbsorption)}; DecayDistance[i].HF = minf(absorb_dist, DecayDistance[i].HF); } } } 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].Base = 0.0f; DecayDistance[i].LF = 0.0f; DecayDistance[i].HF = 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 float Distance{ToSource.normalize()}; /* Initial source gain */ GainTriplet DryGain{props->Gain, 1.0f, 1.0f}; GainTriplet WetGain[MAX_SENDS]; for(ALuint i{0};i < NumSends;i++) WetGain[i] = DryGain; /* Calculate distance attenuation */ float 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 { float dist{lerp(props->RefDistance, ClampedDist, props->RolloffFactor)}; if(dist > 0.0f) DryGain.Base *= props->RefDistance / dist; for(ALuint i{0};i < NumSends;i++) { dist = lerp(props->RefDistance, ClampedDist, RoomRolloff[i]); if(dist > 0.0f) WetGain[i].Base *= 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 { float attn{props->RolloffFactor * (ClampedDist-props->RefDistance) / (props->MaxDistance-props->RefDistance)}; DryGain.Base *= maxf(1.0f - attn, 0.0f); for(ALuint i{0};i < NumSends;i++) { attn = RoomRolloff[i] * (ClampedDist-props->RefDistance) / (props->MaxDistance-props->RefDistance); WetGain[i].Base *= 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 { const float dist_ratio{ClampedDist/props->RefDistance}; DryGain.Base *= std::pow(dist_ratio, -props->RolloffFactor); for(ALuint i{0};i < NumSends;i++) WetGain[i].Base *= std::pow(dist_ratio, -RoomRolloff[i]); } break; case DistanceModel::Disable: ClampedDist = props->RefDistance; break; } /* Calculate directional soundcones */ if(directional && props->InnerAngle < 360.0f) { const float Angle{Rad2Deg(std::acos(-aluDotproduct(Direction, ToSource)) * ConeScale * 2.0f)}; float ConeGain, ConeHF; if(!(Angle > props->InnerAngle)) { ConeGain = 1.0f; ConeHF = 1.0f; } else if(Angle < props->OuterAngle) { const float scale{(Angle-props->InnerAngle) / (props->OuterAngle-props->InnerAngle)}; ConeGain = lerp(1.0f, props->OuterGain, scale); ConeHF = lerp(1.0f, props->OuterGainHF, scale); } else { ConeGain = props->OuterGain; ConeHF = props->OuterGainHF; } DryGain.Base *= ConeGain; if(props->DryGainHFAuto) DryGain.HF *= ConeHF; if(props->WetGainAuto) std::for_each(std::begin(WetGain), std::begin(WetGain)+NumSends, [ConeGain](GainTriplet &gain) noexcept -> void { gain.Base *= ConeGain; }); if(props->WetGainHFAuto) std::for_each(std::begin(WetGain), std::begin(WetGain)+NumSends, [ConeHF](GainTriplet &gain) noexcept -> void { gain.HF *= ConeHF; }); } /* Apply gain and frequency filters */ DryGain.Base = minf(clampf(DryGain.Base, props->MinGain, props->MaxGain) * props->Direct.Gain * Listener.Params.Gain, GAIN_MIX_MAX); DryGain.HF *= props->Direct.GainHF; DryGain.LF *= props->Direct.GainLF; for(ALuint i{0};i < NumSends;i++) { WetGain[i].Base = minf(clampf(WetGain[i].Base, props->MinGain, props->MaxGain) * props->Send[i].Gain * Listener.Params.Gain, GAIN_MIX_MAX); WetGain[i].HF *= props->Send[i].GainHF; WetGain[i].LF *= props->Send[i].GainLF; } /* Distance-based air absorption and initial send decay. */ if(ClampedDist > props->RefDistance && props->RolloffFactor > 0.0f) { const float meters_base{(ClampedDist-props->RefDistance) * props->RolloffFactor * Listener.Params.MetersPerUnit}; if(props->AirAbsorptionFactor > 0.0f) { const float hfattn{std::pow(AIRABSORBGAINHF, meters_base*props->AirAbsorptionFactor)}; DryGain.HF *= hfattn; std::for_each(std::begin(WetGain), std::begin(WetGain)+NumSends, [hfattn](GainTriplet &gain) noexcept -> void { gain.HF *= 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(ALuint i{0};i < NumSends;i++) { if(!(DecayDistance[i].Base > 0.0f)) continue; const float gain{std::pow(REVERB_DECAY_GAIN, meters_base/DecayDistance[i].Base)}; WetGain[i].Base *= gain; /* Yes, the wet path's air absorption is applied with * WetGainAuto on, rather than WetGainHFAuto. */ if(gain > 0.0f) { float gainhf{std::pow(REVERB_DECAY_GAIN, meters_base/DecayDistance[i].HF)}; WetGain[i].HF *= minf(gainhf / gain, 1.0f); float gainlf{std::pow(REVERB_DECAY_GAIN, meters_base/DecayDistance[i].LF)}; WetGain[i].LF *= minf(gainlf / gain, 1.0f); } } } } /* Initial source pitch */ float Pitch{props->Pitch}; /* Calculate velocity-based doppler effect */ float DopplerFactor{props->DopplerFactor * Listener.Params.DopplerFactor}; if(DopplerFactor > 0.0f) { const alu::Vector &lvelocity = Listener.Params.Velocity; float vss{aluDotproduct(Velocity, ToSource) * -DopplerFactor}; float vls{aluDotproduct(lvelocity, ToSource) * -DopplerFactor}; const float 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::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(voice->mFrequency) / static_cast(Device->Frequency); if(Pitch > float{MAX_PITCH}) voice->mStep = MAX_PITCH<mStep = maxu(fastf2u(Pitch * FRACTIONONE), 1); voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState); float spread{0.0f}; if(props->Radius > Distance) spread = al::MathDefs::Tau() - Distance/props->Radius*al::MathDefs::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, WetGain, SendSlots, props, Listener, Device); } void CalcSourceParams(Voice *voice, ALCcontext *context, bool force) { VoicePropsItem *props{voice->mUpdate.exchange(nullptr, std::memory_order_acq_rel)}; if(!props && !force) return; if(props) { voice->mProps = *props; AtomicReplaceHead(context->mFreeVoiceProps, props); } if((voice->mProps.DirectChannels != DirectMode::Off && voice->mFmtChannels != FmtMono && voice->mFmtChannels != FmtBFormat2D && voice->mFmtChannels != FmtBFormat3D) || voice->mProps.mSpatializeMode == SpatializeOff || (voice->mProps.mSpatializeMode == SpatializeAuto && voice->mFmtChannels != FmtMono)) CalcNonAttnSourceParams(voice, &voice->mProps, context); else CalcAttnSourceParams(voice, &voice->mProps, context); } void SendSourceStateEvent(ALCcontext *context, ALuint id, ALenum state) { RingBuffer *ring{context->mAsyncEvents.get()}; auto evt_vec = ring->getWriteVector(); if(evt_vec.first.len < 1) return; AsyncEvent *evt{new (evt_vec.first.buf) AsyncEvent{EventType_SourceStateChange}}; evt->u.srcstate.id = id; evt->u.srcstate.state = state; ring->writeAdvance(1); } void ProcessVoiceChanges(ALCcontext *ctx) { VoiceChange *cur{ctx->mCurrentVoiceChange.load(std::memory_order_acquire)}; VoiceChange *next{cur->mNext.load(std::memory_order_acquire)}; if(!next) return; const ALbitfieldSOFT enabledevt{ctx->mEnabledEvts.load(std::memory_order_acquire)}; do { cur = next; bool sendevt{false}; if(cur->mState == AL_INITIAL || cur->mState == AL_STOPPED) { if(Voice *voice{cur->mVoice}) { 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::State oldvstate{Voice::Playing}; sendevt = voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping, std::memory_order_relaxed, std::memory_order_acquire); voice->mPendingChange.store(false, std::memory_order_release); } /* AL_INITIAL state change events are always sent, even if the * voice is already stopped or even if there is no voice. */ sendevt |= (cur->mState == AL_INITIAL); } else if(cur->mState == AL_PAUSED) { Voice *voice{cur->mVoice}; Voice::State oldvstate{Voice::Playing}; sendevt = voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping, std::memory_order_release, std::memory_order_acquire); } else if(cur->mState == AL_PLAYING) { /* NOTE: When playing a voice, sending a source state change event * depends if there's an old voice to stop and if that stop is * successful. If there is no old voice, a playing event is always * sent. If there is an old voice, an event is sent only if the * voice is already stopped. */ if(Voice *oldvoice{cur->mOldVoice}) { oldvoice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed); oldvoice->mLoopBuffer.store(nullptr, std::memory_order_relaxed); oldvoice->mSourceID.store(0u, std::memory_order_relaxed); Voice::State oldvstate{Voice::Playing}; sendevt = !oldvoice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping, std::memory_order_relaxed, std::memory_order_acquire); oldvoice->mPendingChange.store(false, std::memory_order_release); } else sendevt = true; Voice *voice{cur->mVoice}; voice->mPlayState.store(Voice::Playing, std::memory_order_release); } else if(cur->mState == AL_SAMPLE_OFFSET) { /* Changing a voice offset never sends a source change event. */ Voice *oldvoice{cur->mOldVoice}; oldvoice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed); oldvoice->mLoopBuffer.store(nullptr, std::memory_order_relaxed); /* If there's no sourceID, the old voice finished so don't start * the new one at its new offset. */ if(oldvoice->mSourceID.exchange(0u, std::memory_order_relaxed) != 0u) { /* Otherwise, set the voice to stopping if it's not already (it * might already be, if paused), and play the new voice as * appropriate. */ Voice::State oldvstate{Voice::Playing}; oldvoice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping, std::memory_order_relaxed, std::memory_order_acquire); Voice *voice{cur->mVoice}; voice->mPlayState.store((oldvstate == Voice::Playing) ? Voice::Playing : Voice::Stopped, std::memory_order_release); } oldvoice->mPendingChange.store(false, std::memory_order_release); } if(sendevt && (enabledevt&EventType_SourceStateChange)) SendSourceStateEvent(ctx, cur->mSourceID, cur->mState); next = cur->mNext.load(std::memory_order_acquire); } while(next); ctx->mCurrentVoiceChange.store(cur, std::memory_order_release); } void ProcessParamUpdates(ALCcontext *ctx, const ALeffectslotArray &slots, const al::span voices) { ProcessVoiceChanges(ctx); IncrementRef(ctx->mUpdateCount); if LIKELY(!ctx->mHoldUpdates.load(std::memory_order_acquire)) { bool force{CalcContextParams(ctx)}; force |= CalcListenerParams(ctx); auto sorted_slots = const_cast(slots.data() + slots.size()); for(ALeffectslot *slot : slots) force |= CalcEffectSlotParams(slot, sorted_slots, ctx); for(Voice *voice : voices) { /* Only update voices that have a source. */ if(voice->mSourceID.load(std::memory_order_relaxed) != 0) CalcSourceParams(voice, ctx, force); } } IncrementRef(ctx->mUpdateCount); } void ProcessContexts(ALCdevice *device, const ALuint SamplesToDo) { ASSUME(SamplesToDo > 0); for(ALCcontext *ctx : *device->mContexts.load(std::memory_order_acquire)) { const ALeffectslotArray &auxslots = *ctx->mActiveAuxSlots.load(std::memory_order_acquire); const al::span voices{ctx->getVoicesSpanAcquired()}; /* Process pending propery updates for objects on the context. */ ProcessParamUpdates(ctx, auxslots, voices); /* Clear auxiliary effect slot mixing buffers. */ for(ALeffectslot *slot : auxslots) { for(auto &buffer : slot->MixBuffer) std::fill_n(buffer.begin(), SamplesToDo, 0.0f); } /* Process voices that have a playing source. */ for(Voice *voice : voices) { const Voice::State vstate{voice->mPlayState.load(std::memory_order_acquire)}; if(vstate != Voice::Stopped && vstate != Voice::Pending) voice->mix(vstate, ctx, SamplesToDo); } /* Process effects. */ if(const size_t num_slots{auxslots.size()}) { auto slots = auxslots.data(); auto slots_end = slots + num_slots; /* First sort the slots into extra storage, so that effects come * before their effect target (or their targets' target). */ auto sorted_slots = const_cast(slots_end); auto sorted_slots_end = sorted_slots; if(*sorted_slots) { /* Skip sorting if it has already been done. */ sorted_slots_end += num_slots; goto skip_sorting; } *sorted_slots_end = *slots; ++sorted_slots_end; while(++slots != slots_end) { auto in_chain = [](const ALeffectslot *s1, const ALeffectslot *s2) noexcept -> bool { while((s1=s1->Params.Target) != nullptr) { if(s1 == s2) return true; } return false; }; /* 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; } skip_sorting: auto process_effect = [SamplesToDo](const ALeffectslot *slot) -> void { EffectState *state{slot->Params.mEffectState}; state->process(SamplesToDo, slot->Wet.Buffer, state->mOutTarget); }; std::for_each(sorted_slots, sorted_slots_end, process_effect); } /* Signal the event handler if there are any events to read. */ RingBuffer *ring{ctx->mAsyncEvents.get()}; if(ring->readSpace() > 0) ctx->mEventSem.post(); } } void ApplyStablizer(FrontStablizer *Stablizer, const al::span Buffer, const ALuint lidx, const ALuint ridx, const ALuint cidx, const ALuint 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 >= ALuint{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)); } } float (&lsplit)[2][BUFFERSIZE] = Stablizer->LSplit; float (&rsplit)[2][BUFFERSIZE] = Stablizer->RSplit; const al::span tmpbuf{Stablizer->TempBuf, SamplesToDo+FrontStablizer::DelayLength}; /* 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 &InBuf, const al::span DelayBuf, BandSplitter &Filter, float (&splitbuf)[2][BUFFERSIZE]) -> void { /* Combine the input and delayed samples into a temp buffer in reverse, * then copy the final samples into the delay buffer for next time. * Note that the delay buffer's samples are stored backwards here. */ auto tmp_iter = std::reverse_copy(InBuf.cbegin(), InBuf.cbegin()+SamplesToDo, tmpbuf.begin()); std::copy(DelayBuf.cbegin(), DelayBuf.cend(), tmp_iter); std::copy_n(tmpbuf.cbegin(), DelayBuf.size(), DelayBuf.begin()); /* 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); std::reverse(tmpbuf.begin(), tmpbuf.end()); /* Now apply the band-splitter, combining its phase shift with the * reversed phase shift, restoring the original phase on the split * signal. */ Filter.process(tmpbuf.first(SamplesToDo), splitbuf[1], splitbuf[0]); }; apply_splitter(Buffer[lidx], Stablizer->DelayBuf[lidx], Stablizer->LFilter, lsplit); apply_splitter(Buffer[ridx], Stablizer->DelayBuf[ridx], Stablizer->RFilter, rsplit); for(ALuint i{0};i < SamplesToDo;i++) { float lfsum{lsplit[0][i] + rsplit[0][i]}; float hfsum{lsplit[1][i] + rsplit[1][i]}; float 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. */ float m{lfsum*std::cos(1.0f/3.0f * (al::MathDefs::Pi()*0.5f)) + hfsum*std::cos(1.0f/4.0f * (al::MathDefs::Pi()*0.5f))}; float c{lfsum*std::sin(1.0f/3.0f * (al::MathDefs::Pi()*0.5f)) + hfsum*std::sin(1.0f/4.0f * (al::MathDefs::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 Samples, const ALuint SamplesToDo, const DistanceComp::DistData *distcomp) { ASSUME(SamplesToDo > 0); for(auto &chanbuffer : Samples) { const float gain{distcomp->Gain}; const ALuint base{distcomp->Length}; float *distbuf{al::assume_aligned<16>(distcomp->Buffer)}; ++distcomp; if(base < 1) continue; float *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{}, _1, gain)); } } void ApplyDither(const al::span Samples, ALuint *dither_seed, const float quant_scale, const ALuint 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 float 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 float sample) noexcept -> float { float val{sample * quant_scale}; ALuint rng0{dither_rng(&seed)}; ALuint rng1{dither_rng(&seed)}; val += static_cast(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 inline T SampleConv(float) noexcept; template<> inline float SampleConv(float val) noexcept { return val; } template<> inline int32_t SampleConv(float 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 int16_t SampleConv(float val) noexcept { return static_cast(fastf2i(clampf(val*32768.0f, -32768.0f, 32767.0f))); } template<> inline int8_t SampleConv(float val) noexcept { return static_cast(fastf2i(clampf(val*128.0f, -128.0f, 127.0f))); } /* Define unsigned output variations. */ template<> inline uint32_t SampleConv(float val) noexcept { return static_cast(SampleConv(val)) + 2147483648u; } template<> inline uint16_t SampleConv(float val) noexcept { return static_cast(SampleConv(val) + 32768); } template<> inline uint8_t SampleConv(float val) noexcept { return static_cast(SampleConv(val) + 128); } template void Write(const al::span InBuffer, void *OutBuffer, const size_t Offset, const ALuint SamplesToDo, const size_t FrameStep) { using SampleType = typename DevFmtTypeTraits::Type; ASSUME(FrameStep > 0); SampleType *outbase = static_cast(OutBuffer) + Offset*FrameStep; auto conv_channel = [&outbase,SamplesToDo,FrameStep](const FloatBufferLine &inbuf) -> void { ASSUME(SamplesToDo > 0); SampleType *out{outbase++}; auto conv_sample = [FrameStep,&out](const float s) noexcept -> void { *out = SampleConv(s); out += FrameStep; }; std::for_each(inbuf.begin(), inbuf.begin()+SamplesToDo, conv_sample); }; std::for_each(InBuffer.cbegin(), InBuffer.cend(), conv_channel); } } // namespace void aluMixData(ALCdevice *device, void *OutBuffer, const ALuint NumSamples, const size_t FrameStep) { FPUCtl mixer_mode{}; for(ALuint SamplesDone{0u};SamplesDone < NumSamples;) { const ALuint SamplesToDo{minu(NumSamples-SamplesDone, BUFFERSIZE)}; /* Clear main mixing buffers. */ std::for_each(device->MixBuffer.begin(), device->MixBuffer.end(), [SamplesToDo](FloatBufferLine &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); /* Process and mix each context's sources and effects. */ ProcessContexts(device, 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). */ device->postProcess(SamplesToDo); const al::span RealOut{device->RealOut.Buffer}; /* Apply front image stablization for surround sound, if applicable. */ if(FrontStablizer *stablizer{device->Stablizer.get()}) { const ALuint lidx{GetChannelIdxByName(device->RealOut, FrontLeft)}; const ALuint ridx{GetChannelIdxByName(device->RealOut, FrontRight)}; const ALuint cidx{GetChannelIdxByName(device->RealOut, FrontCenter)}; ApplyStablizer(stablizer, 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(RealOut, OutBuffer, SamplesDone, SamplesToDo, FrameStep); 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(msglen) >= sizeof(evt.u.user.msg)) evt.u.user.msg[sizeof(evt.u.user.msg)-1] = 0; IncrementRef(device->MixCount); for(ALCcontext *ctx : *device->mContexts.load()) { const ALbitfieldSOFT enabledevt{ctx->mEnabledEvts.load(std::memory_order_acquire)}; if((enabledevt&EventType_Disconnected)) { RingBuffer *ring{ctx->mAsyncEvents.get()}; auto evt_data = ring->getWriteVector().first; if(evt_data.len > 0) { ::new (evt_data.buf) AsyncEvent{evt}; ring->writeAdvance(1); ctx->mEventSem.post(); } } auto voicelist = ctx->getVoicesSpanAcquired(); auto stop_voice = [](Voice *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(Voice::Stopped, std::memory_order_release); }; std::for_each(voicelist.begin(), voicelist.end(), stop_voice); } IncrementRef(device->MixCount); }