/** * 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 "almalloc.h" #include "alnumbers.h" #include "alnumeric.h" #include "alspan.h" #include "alstring.h" #include "atomic.h" #include "core/ambidefs.h" #include "core/async_event.h" #include "core/bformatdec.h" #include "core/bs2b.h" #include "core/bsinc_defs.h" #include "core/bsinc_tables.h" #include "core/bufferline.h" #include "core/buffer_storage.h" #include "core/context.h" #include "core/cpu_caps.h" #include "core/cubic_tables.h" #include "core/devformat.h" #include "core/device.h" #include "core/effects/base.h" #include "core/effectslot.h" #include "core/filters/biquad.h" #include "core/filters/nfc.h" #include "core/fpu_ctrl.h" #include "core/hrtf.h" #include "core/mastering.h" #include "core/mixer.h" #include "core/mixer/defs.h" #include "core/mixer/hrtfdefs.h" #include "core/resampler_limits.h" #include "core/uhjfilter.h" #include "core/voice.h" #include "core/voice_change.h" #include "intrusive_ptr.h" #include "opthelpers.h" #include "ringbuffer.h" #include "strutils.h" #include "vecmat.h" #include "vector.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 PointTag; struct LerpTag; struct CubicTag; struct BSincTag; struct FastBSincTag; static_assert(!(MaxResamplerPadding&1), "MaxResamplerPadding is not a multiple of two"); namespace { using uint = unsigned int; using namespace std::chrono; 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; } /* Cone scalar */ const float ConeScale{InitConeScale()}; /* Localized scalars for mono sources (initialized in aluInit, after * configuration is loaded). */ float XScale{1.0f}; float YScale{1.0f}; float ZScale{1.0f}; /* Source distance scale for NFC filters. */ float NfcScale{1.0f}; using HrtfDirectMixerFunc = void(*)(const FloatBufferSpan LeftOut, const FloatBufferSpan RightOut, const al::span InSamples, float2 *AccumSamples, float *TempBuf, HrtfChannelState *ChanState, const size_t IrSize, const size_t BufferSize); HrtfDirectMixerFunc MixDirectHrtf{MixDirectHrtf_}; inline HrtfDirectMixerFunc SelectHrtfMixer() { #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 uint increment, BsincState *state, const BSincTable *table) { size_t si{BSincScaleCount - 1}; float sf{0.0f}; if(increment > MixerFracOne) { sf = MixerFracOne/static_cast(increment) - table->scaleBase; sf = maxf(0.0f, BSincScaleCount*sf*table->scaleRange - 1.0f); 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, uint 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: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Resample_; #endif return Resample_; case Resampler::BSinc12: case Resampler::BSinc24: if(increment > MixerFracOne) { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Resample_; #endif return Resample_; } /* 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_; } return Resample_; } } // namespace void aluInit(CompatFlagBitset flags, const float nfcscale) { MixDirectHrtf = SelectHrtfMixer(); XScale = flags.test(CompatFlags::ReverseX) ? -1.0f : 1.0f; YScale = flags.test(CompatFlags::ReverseY) ? -1.0f : 1.0f; ZScale = flags.test(CompatFlags::ReverseZ) ? -1.0f : 1.0f; NfcScale = clampf(nfcscale, 0.0001f, 10000.0f); } ResamplerFunc PrepareResampler(Resampler resampler, uint increment, InterpState *state) { switch(resampler) { case Resampler::Point: case Resampler::Linear: break; case Resampler::Cubic: state->emplace().filter = gCubicSpline.Tab.data(); break; case Resampler::FastBSinc12: case Resampler::BSinc12: BsincPrepare(increment, &state->emplace(), &gBSinc12); break; case Resampler::FastBSinc24: case Resampler::BSinc24: BsincPrepare(increment, &state->emplace(), &gBSinc24); break; } return SelectResampler(resampler, increment); } void DeviceBase::ProcessHrtf(const size_t SamplesToDo) { /* HRTF is stereo output only. */ const size_t lidx{RealOut.ChannelIndex[FrontLeft]}; const size_t ridx{RealOut.ChannelIndex[FrontRight]}; MixDirectHrtf(RealOut.Buffer[lidx], RealOut.Buffer[ridx], Dry.Buffer, HrtfAccumData.data(), mHrtfState->mTemp.data(), mHrtfState->mChannels.data(), mHrtfState->mIrSize, SamplesToDo); } void DeviceBase::ProcessAmbiDec(const size_t SamplesToDo) { AmbiDecoder->process(RealOut.Buffer, Dry.Buffer.data(), SamplesToDo); } void DeviceBase::ProcessAmbiDecStablized(const size_t SamplesToDo) { /* Decode with front image stablization. */ const size_t lidx{RealOut.ChannelIndex[FrontLeft]}; const size_t ridx{RealOut.ChannelIndex[FrontRight]}; const size_t cidx{RealOut.ChannelIndex[FrontCenter]}; AmbiDecoder->processStablize(RealOut.Buffer, Dry.Buffer.data(), lidx, ridx, cidx, SamplesToDo); } void DeviceBase::ProcessUhj(const size_t SamplesToDo) { /* UHJ is stereo output only. */ const size_t lidx{RealOut.ChannelIndex[FrontLeft]}; const size_t ridx{RealOut.ChannelIndex[FrontRight]}; /* Encode to stereo-compatible 2-channel UHJ output. */ mUhjEncoder->encode(RealOut.Buffer[lidx].data(), RealOut.Buffer[ridx].data(), {{Dry.Buffer[0].data(), Dry.Buffer[1].data(), Dry.Buffer[2].data()}}, SamplesToDo); } void DeviceBase::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 size_t lidx{RealOut.ChannelIndex[FrontLeft]}; const size_t ridx{RealOut.ChannelIndex[FrontRight]}; /* Now apply the BS2B binaural/crossfeed filter. */ Bs2b->cross_feed(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 uint dither_rng(uint *seed) noexcept { *seed = (*seed * 96314165) + 907633515; return *seed; } /* Ambisonic upsampler function. It's effectively a matrix multiply. It takes * an 'upsampler' and 'rotator' as the input matrices, and creates a matrix * that behaves as if the B-Format input was first decoded to a speaker array * at its input order, encoded back into the higher order mix, then finally * rotated. */ void UpsampleBFormatTransform( const al::span,MaxAmbiChannels> output, const al::span> upsampler, const al::span,MaxAmbiChannels> rotator, size_t coeffs_order) { const size_t num_chans{AmbiChannelsFromOrder(coeffs_order)}; for(size_t i{0};i < upsampler.size();++i) output[i].fill(0.0f); for(size_t i{0};i < upsampler.size();++i) { for(size_t k{0};k < num_chans;++k) { float *RESTRICT out{output[i].data()}; /* Write the full number of channels. The compiler will have an * easier time optimizing if it has a fixed length. */ for(size_t j{0};j < MaxAmbiChannels;++j) out[j] += upsampler[i][k] * rotator[k][j]; } } } constexpr auto GetAmbiScales(AmbiScaling scaletype) noexcept { switch(scaletype) { case AmbiScaling::FuMa: return al::span{AmbiScale::FromFuMa}; case AmbiScaling::SN3D: return al::span{AmbiScale::FromSN3D}; case AmbiScaling::UHJ: return al::span{AmbiScale::FromUHJ}; case AmbiScaling::N3D: break; } return al::span{AmbiScale::FromN3D}; } constexpr auto GetAmbiLayout(AmbiLayout layouttype) noexcept { if(layouttype == AmbiLayout::FuMa) return al::span{AmbiIndex::FromFuMa}; return al::span{AmbiIndex::FromACN}; } constexpr auto GetAmbi2DLayout(AmbiLayout layouttype) noexcept { if(layouttype == AmbiLayout::FuMa) return al::span{AmbiIndex::FromFuMa2D}; return al::span{AmbiIndex::FromACN2D}; } bool CalcContextParams(ContextBase *ctx) { ContextProps *props{ctx->mParams.ContextUpdate.exchange(nullptr, std::memory_order_acq_rel)}; if(!props) return false; const alu::Vector pos{props->Position[0], props->Position[1], props->Position[2], 1.0f}; ctx->mParams.Position = pos; /* 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{N.cross_product(V)}; U.normalize(); const alu::Matrix rot{ U[0], V[0], -N[0], 0.0, U[1], V[1], -N[1], 0.0, U[2], V[2], -N[2], 0.0, 0.0, 0.0, 0.0, 1.0}; const alu::Vector vel{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0}; ctx->mParams.Matrix = rot; ctx->mParams.Velocity = rot * vel; ctx->mParams.Gain = props->Gain * ctx->mGainBoost; ctx->mParams.MetersPerUnit = props->MetersPerUnit; ctx->mParams.AirAbsorptionGainHF = props->AirAbsorptionGainHF; ctx->mParams.DopplerFactor = props->DopplerFactor; ctx->mParams.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity; ctx->mParams.SourceDistanceModel = props->SourceDistanceModel; ctx->mParams.mDistanceModel = props->mDistanceModel; AtomicReplaceHead(ctx->mFreeContextProps, props); return true; } bool CalcEffectSlotParams(EffectSlot *slot, EffectSlot **sorted_slots, ContextBase *context) { EffectSlotProps *props{slot->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->Target != props->Target) *sorted_slots = nullptr; slot->Gain = props->Gain; slot->AuxSendAuto = props->AuxSendAuto; slot->Target = props->Target; slot->EffectType = props->Type; slot->mEffectProps = props->Props; if(auto *reverbprops = std::get_if(&props->Props)) { slot->RoomRolloff = reverbprops->RoomRolloffFactor; slot->DecayTime = reverbprops->DecayTime; slot->DecayLFRatio = reverbprops->DecayLFRatio; slot->DecayHFRatio = reverbprops->DecayHFRatio; slot->DecayHFLimit = reverbprops->DecayHFLimit; slot->AirAbsorptionGainHF = reverbprops->AirAbsorptionGainHF; } else { slot->RoomRolloff = 0.0f; slot->DecayTime = 0.0f; slot->DecayLFRatio = 0.0f; slot->DecayHFRatio = 0.0f; slot->DecayHFLimit = false; slot->AirAbsorptionGainHF = 1.0f; } EffectState *state{props->State.release()}; EffectState *oldstate{slot->mEffectState.release()}; slot->mEffectState.reset(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(evt_vec.first.len > 0) LIKELY { auto &evt = InitAsyncEvent(evt_vec.first.buf); evt.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.reset(oldstate); } } AtomicReplaceHead(context->mFreeEffectslotProps, props); EffectTarget output; if(EffectSlot *target{slot->Target}) output = EffectTarget{&target->Wet, nullptr}; else { DeviceBase *device{context->mDevice}; output = EffectTarget{&device->Dry, &device->RealOut}; } state->update(context, slot, &slot->mEffectProps, output); return true; } /* Scales the azimuth of the given vector by 3 if it's in front. Effectively * scales +/-30 degrees to +/-90 degrees, leaving > +90 and < -90 alone. */ inline std::array ScaleAzimuthFront3(std::array pos) { if(pos[2] < 0.0f) { /* Normalize the length of the x,z components for a 2D vector of the * azimuth angle. Negate Z since {0,0,-1} is angle 0. */ const float len2d{std::sqrt(pos[0]*pos[0] + pos[2]*pos[2])}; float x{pos[0] / len2d}; float z{-pos[2] / len2d}; /* Z > cos(pi/6) = -30 < azimuth < 30 degrees. */ if(z > 0.866025403785f) { /* Triple the angle represented by x,z. */ x = x*3.0f - x*x*x*4.0f; z = z*z*z*4.0f - z*3.0f; /* Scale the vector back to fit in 3D. */ pos[0] = x * len2d; pos[2] = -z * len2d; } else { /* If azimuth >= 30 degrees, clamp to 90 degrees. */ pos[0] = std::copysign(len2d, pos[0]); pos[2] = 0.0f; } } return pos; } /* Scales the azimuth of the given vector by 1.5 (3/2) if it's in front. */ inline std::array ScaleAzimuthFront3_2(std::array pos) { if(pos[2] < 0.0f) { const float len2d{std::sqrt(pos[0]*pos[0] + pos[2]*pos[2])}; float x{pos[0] / len2d}; float z{-pos[2] / len2d}; /* Z > cos(pi/3) = -60 < azimuth < 60 degrees. */ if(z > 0.5f) { /* Halve the angle represented by x,z. */ x = std::copysign(std::sqrt((1.0f - z) * 0.5f), x); z = std::sqrt((1.0f + z) * 0.5f); /* Triple the angle represented by x,z. */ x = x*3.0f - x*x*x*4.0f; z = z*z*z*4.0f - z*3.0f; /* Scale the vector back to fit in 3D. */ pos[0] = x * len2d; pos[2] = -z * len2d; } else { /* If azimuth >= 60 degrees, clamp to 90 degrees. */ pos[0] = std::copysign(len2d, pos[0]); pos[2] = 0.0f; } } return pos; } /* 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. */ constexpr size_t CalcRotatorSize(size_t l) noexcept { if(l >= 2) return (l*2 + 1)*(l*2 + 1) + CalcRotatorSize(l-1); return 0; } struct RotatorCoeffs { struct CoeffValues { float u, v, w; }; std::array mCoeffs{}; RotatorCoeffs() { auto coeffs = mCoeffs.begin(); for(int l=2;l <= MaxAmbiOrder;++l) { for(int n{-l};n <= l;++n) { for(int m{-l};m <= l;++m) { /* compute u,v,w terms of Eq.8.1 (Table I) * * const bool d{m == 0}; // the delta function d_m0 * const double denom{(std::abs(n) == l) ? * (2*l) * (2*l - 1) : (l*l - n*n)}; * * const int abs_m{std::abs(m)}; * coeffs->u = std::sqrt((l*l - m*m) / denom); * coeffs->v = std::sqrt((l+abs_m-1) * (l+abs_m) / denom) * * (1.0+d) * (1.0 - 2.0*d) * 0.5; * coeffs->w = std::sqrt((l-abs_m-1) * (l-abs_m) / denom) * * (1.0-d) * -0.5; */ const double denom{static_cast((std::abs(n) == l) ? (2*l) * (2*l - 1) : (l*l - n*n))}; if(m == 0) { coeffs->u = static_cast(std::sqrt(l * l / denom)); coeffs->v = static_cast(std::sqrt((l-1) * l / denom) * -1.0); coeffs->w = 0.0f; } else { const int abs_m{std::abs(m)}; coeffs->u = static_cast(std::sqrt((l*l - m*m) / denom)); coeffs->v = static_cast(std::sqrt((l+abs_m-1) * (l+abs_m) / denom) * 0.5); coeffs->w = static_cast(std::sqrt((l-abs_m-1) * (l-abs_m) / denom) * -0.5); } ++coeffs; } } } } }; const RotatorCoeffs RotatorCoeffArray{}; /** * 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(AmbiRotateMatrix &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 AmbiRotateMatrix &R) { const float ri1{ R[ 1+2][static_cast(i+2_z)]}; const float rim1{R[-1+2][static_cast(i+2_z)]}; const float ri0{ R[ 0+2][static_cast(i+2_z)]}; const size_t y{last_band + static_cast(a+l-1)}; if(n == -l) return ri1*R[last_band][y] + rim1*R[last_band + static_cast(l-1_z)*2][y]; if(n == l) return ri1*R[last_band + static_cast(l-1_z)*2][y] - rim1*R[last_band][y]; return ri0*R[last_band + static_cast(l-1_z+n)][y]; }; auto U = [P](const int l, const int m, const int n, const size_t last_band, const AmbiRotateMatrix &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 AmbiRotateMatrix &R) { using namespace al::numbers; 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*sqrt2_v : (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*sqrt2_v : (p0 + p1); }; auto W = [P](const int l, const int m, const int n, const size_t last_band, const AmbiRotateMatrix &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.mCoeffs.cbegin(); size_t band_idx{4}, last_band{1}; for(int l{2};l <= order;++l) { size_t y{band_idx}; for(int n{-l};n <= l;++n,++y) { size_t x{band_idx}; for(int m{-l};m <= l;++m,++x) { float r{0.0f}; // computes Eq.8.1 if(const float u{coeffs->u}; u != 0.0f) r += u * U(l, m, n, last_band, matrix); if(const float v{coeffs->v}; v != 0.0f) r += v * V(l, m, n, last_band, matrix); if(const float w{coeffs->w}; 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)*2_uz + 1; } } /* End ambisonic rotation helpers. */ constexpr float sin30{0.5f}; constexpr float cos30{0.866025403785f}; constexpr float sin45{al::numbers::sqrt2_v*0.5f}; constexpr float cos45{al::numbers::sqrt2_v*0.5f}; constexpr float sin110{ 0.939692620786f}; constexpr float cos110{-0.342020143326f}; struct ChanPosMap { Channel channel; std::array pos; }; 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, const al::span SendSlots, const VoiceProps *props, const ContextParams &Context, DeviceBase *Device) { static constexpr std::array MonoMap{ ChanPosMap{FrontCenter, std::array{0.0f, 0.0f, -1.0f}} }; static constexpr std::array RearMap{ ChanPosMap{BackLeft, std::array{-sin30, 0.0f, cos30}}, ChanPosMap{BackRight, std::array{ sin30, 0.0f, cos30}}, }; static constexpr std::array QuadMap{ ChanPosMap{FrontLeft, std::array{-sin45, 0.0f, -cos45}}, ChanPosMap{FrontRight, std::array{ sin45, 0.0f, -cos45}}, ChanPosMap{BackLeft, std::array{-sin45, 0.0f, cos45}}, ChanPosMap{BackRight, std::array{ sin45, 0.0f, cos45}}, }; static constexpr std::array X51Map{ ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}}, ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}}, ChanPosMap{FrontCenter, std::array{ 0.0f, 0.0f, -1.0f}}, ChanPosMap{LFE, {}}, ChanPosMap{SideLeft, std::array{-sin110, 0.0f, -cos110}}, ChanPosMap{SideRight, std::array{ sin110, 0.0f, -cos110}}, }; static constexpr std::array X61Map{ ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}}, ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}}, ChanPosMap{FrontCenter, std::array{ 0.0f, 0.0f, -1.0f}}, ChanPosMap{LFE, {}}, ChanPosMap{BackCenter, std::array{ 0.0f, 0.0f, 1.0f}}, ChanPosMap{SideLeft, std::array{-1.0f, 0.0f, 0.0f}}, ChanPosMap{SideRight, std::array{ 1.0f, 0.0f, 0.0f}}, }; static constexpr std::array X71Map{ ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}}, ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}}, ChanPosMap{FrontCenter, std::array{ 0.0f, 0.0f, -1.0f}}, ChanPosMap{LFE, {}}, ChanPosMap{BackLeft, std::array{-sin30, 0.0f, cos30}}, ChanPosMap{BackRight, std::array{ sin30, 0.0f, cos30}}, ChanPosMap{SideLeft, std::array{ -1.0f, 0.0f, 0.0f}}, ChanPosMap{SideRight, std::array{ 1.0f, 0.0f, 0.0f}}, }; std::array StereoMap{ ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}}, ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}}, }; const auto Frequency = static_cast(Device->Frequency); const uint 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); }); } const auto getChans = [props,&StereoMap](FmtChannels chanfmt) noexcept -> std::pair> { switch(chanfmt) { case FmtMono: /* Mono buffers are never played direct. */ return {DirectMode::Off, al::span{MonoMap}}; case FmtStereo: if(props->DirectChannels == DirectMode::Off) { for(size_t i{0};i < 2;++i) { /* StereoPan is counter-clockwise in radians. */ const float a{props->StereoPan[i]}; StereoMap[i].pos[0] = -std::sin(a); StereoMap[i].pos[2] = -std::cos(a); } } return {props->DirectChannels, al::span{StereoMap}}; case FmtRear: return {props->DirectChannels, al::span{RearMap}}; case FmtQuad: return {props->DirectChannels, al::span{QuadMap}}; case FmtX51: return {props->DirectChannels, al::span{X51Map}}; case FmtX61: return {props->DirectChannels, al::span{X61Map}}; case FmtX71: return {props->DirectChannels, al::span{X71Map}}; case FmtBFormat2D: case FmtBFormat3D: case FmtUHJ2: case FmtUHJ3: case FmtUHJ4: case FmtSuperStereo: return {DirectMode::Off, {}}; } return {props->DirectChannels, {}}; }; const auto [DirectChannels,chans] = getChans(voice->mFmtChannels); voice->mFlags.reset(VoiceHasHrtf).reset(VoiceHasNfc); if(auto *decoder{voice->mDecoder.get()}) decoder->mWidthControl = minf(props->EnhWidth, 0.7f); if(IsAmbisonic(voice->mFmtChannels)) { /* Special handling for B-Format and UHJ sources. */ if(Device->AvgSpeakerDist > 0.0f && voice->mFmtChannels != FmtUHJ2 && voice->mFmtChannels != FmtSuperStereo) { if(!(Distance > std::numeric_limits::epsilon())) { /* 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); } else { /* Clamp the distance for really close sources, to prevent * excessive bass. */ const float mdist{maxf(Distance*NfcScale, Device->AvgSpeakerDist/4.0f)}; const float w0{SpeedOfSoundMetersPerSec / (mdist * Frequency)}; /* Only need to adjust the first channel of a B-Format source. */ voice->mChans[0].mDryParams.NFCtrlFilter.adjust(w0); } voice->mFlags.set(VoiceHasNfc); } /* Panning a B-Format sound toward some direction is easy. Just pan the * first (W) channel as a normal mono sound. The angular spread is used * as a directional scalar to blend between full coverage and full * panning. */ const float coverage{!(Distance > std::numeric_limits::epsilon()) ? 1.0f : (al::numbers::inv_pi_v/2.0f * Spread)}; auto calc_coeffs = [xpos,ypos,zpos](RenderMode mode) { if(mode != RenderMode::Pairwise) return CalcDirectionCoeffs(std::array{xpos, ypos, zpos}, 0.0f); const auto pos = ScaleAzimuthFront3_2(std::array{xpos, ypos, zpos}); return CalcDirectionCoeffs(pos, 0.0f); }; const auto scales = GetAmbiScales(voice->mAmbiScaling); auto coeffs = calc_coeffs(Device->mRenderMode); if(!(coverage > 0.0f)) { ComputePanGains(&Device->Dry, coeffs, DryGain.Base*scales[0], voice->mChans[0].mDryParams.Gains.Target); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base*scales[0], voice->mChans[0].mWetParams[i].Gains.Target); } } else { /* 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 = Context.Matrix * N; V = Context.Matrix * V; } /* Build and normalize right-vector */ alu::Vector U{N.cross_product(V)}; U.normalize(); /* Build a rotation matrix. Manually fill the zeroth- and first- * order elements, then construct the rotation for the higher * orders. */ AmbiRotateMatrix &shrot = Device->mAmbiRotateMatrix; shrot.fill(AmbiRotateMatrix::value_type{}); shrot[0][0] = 1.0f; shrot[1][1] = U[0]; shrot[1][2] = -U[1]; shrot[1][3] = U[2]; shrot[2][1] = -V[0]; shrot[2][2] = V[1]; shrot[2][3] = -V[2]; shrot[3][1] = -N[0]; shrot[3][2] = N[1]; shrot[3][3] = -N[2]; AmbiRotator(shrot, static_cast(Device->mAmbiOrder)); /* If the device is higher order than the voice, "upsample" the * matrix. * * NOTE: Starting with second-order, a 2D upsample needs to be * applied with a 2D source and 3D output, even when they're the * same order. This is because higher orders have a height offset * on various channels (i.e. when elevation=0, those height-related * channels should be non-0). */ AmbiRotateMatrix &mixmatrix = Device->mAmbiRotateMatrix2; if(Device->mAmbiOrder > voice->mAmbiOrder || (Device->mAmbiOrder >= 2 && !Device->m2DMixing && Is2DAmbisonic(voice->mFmtChannels))) { if(voice->mAmbiOrder == 1) { const auto upsampler = Is2DAmbisonic(voice->mFmtChannels) ? al::span{AmbiScale::FirstOrder2DUp} : al::span{AmbiScale::FirstOrderUp}; UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder); } else if(voice->mAmbiOrder == 2) { const auto upsampler = Is2DAmbisonic(voice->mFmtChannels) ? al::span{AmbiScale::SecondOrder2DUp} : al::span{AmbiScale::SecondOrderUp}; UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder); } else if(voice->mAmbiOrder == 3) { const auto upsampler = Is2DAmbisonic(voice->mFmtChannels) ? al::span{AmbiScale::ThirdOrder2DUp} : al::span{AmbiScale::ThirdOrderUp}; UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder); } else if(voice->mAmbiOrder == 4) { const auto upsampler = al::span{AmbiScale::FourthOrder2DUp}; UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder); } else al::unreachable(); } else mixmatrix = shrot; /* Convert the rotation matrix for input ordering and scaling, and * whether input is 2D or 3D. */ const uint8_t *index_map{Is2DAmbisonic(voice->mFmtChannels) ? GetAmbi2DLayout(voice->mAmbiLayout).data() : GetAmbiLayout(voice->mAmbiLayout).data()}; /* Scale the panned W signal inversely to coverage (full coverage * means no panned signal), and according to the channel scaling. */ std::for_each(coeffs.begin(), coeffs.end(), [scale=(1.0f-coverage)*scales[0]](float &coeff) noexcept { coeff *= scale; }); for(size_t c{0};c < num_channels;c++) { const size_t acn{index_map[c]}; const float scale{scales[acn] * coverage}; /* For channel 0, combine the B-Format signal (scaled according * to the coverage amount) with the directional pan. For all * other channels, use just the (scaled) B-Format signal. */ for(size_t x{0};x < MaxAmbiChannels;++x) coeffs[x] += mixmatrix[acn][x] * scale; ComputePanGains(&Device->Dry, coeffs, DryGain.Base, voice->mChans[c].mDryParams.Gains.Target); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[c].mWetParams[i].Gains.Target); } coeffs = std::array{}; } } } else if(DirectChannels != DirectMode::Off && !Device->RealOut.RemixMap.empty()) { /* 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++) { uint idx{Device->channelIdxByName(chans[c].channel)}; if(idx != InvalidChannelIndex) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base; else if(DirectChannels == DirectMode::RemixMismatch) { auto match_channel = [channel=chans[c].channel](const InputRemixMap &map) noexcept { return 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 = Device->channelIdxByName(target.channel); if(idx != InvalidChannelIndex) 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++) { /* Skip LFE */ if(chans[c].channel == LFE) continue; const auto coeffs = CalcDirectionCoeffs(chans[c].pos, 0.0f); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[c].mWetParams[i].Gains.Target); } } } else if(Device->mRenderMode == RenderMode::Hrtf) { /* 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()) { if(voice->mFmtChannels == FmtMono) { const float src_ev{std::asin(clampf(ypos, -1.0f, 1.0f))}; const float src_az{std::atan2(xpos, -zpos)}; Device->mHrtf->getCoeffs(src_ev, src_az, Distance*NfcScale, Spread, voice->mChans[0].mDryParams.Hrtf.Target.Coeffs, voice->mChans[0].mDryParams.Hrtf.Target.Delay); voice->mChans[0].mDryParams.Hrtf.Target.Gain = DryGain.Base; const auto coeffs = CalcDirectionCoeffs(std::array{xpos, ypos, zpos}, Spread); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[0].mWetParams[i].Gains.Target); } } else for(size_t c{0};c < num_channels;c++) { using namespace al::numbers; /* Skip LFE */ if(chans[c].channel == LFE) continue; /* Warp the channel position toward the source position as the * source spread decreases. With no spread, all channels are at * the source position, at full spread (pi*2), each channel is * left unchanged. */ const float a{1.0f - (inv_pi_v/2.0f)*Spread}; std::array pos{ lerpf(chans[c].pos[0], xpos, a), lerpf(chans[c].pos[1], ypos, a), lerpf(chans[c].pos[2], zpos, a)}; const float len{std::sqrt(pos[0]*pos[0] + pos[1]*pos[1] + pos[2]*pos[2])}; if(len < 1.0f) { pos[0] /= len; pos[1] /= len; pos[2] /= len; } const float ev{std::asin(clampf(pos[1], -1.0f, 1.0f))}; const float az{std::atan2(pos[0], -pos[2])}; Device->mHrtf->getCoeffs(ev, az, Distance*NfcScale, 0.0f, voice->mChans[c].mDryParams.Hrtf.Target.Coeffs, voice->mChans[c].mDryParams.Hrtf.Target.Delay); voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain.Base; const auto coeffs = CalcDirectionCoeffs(pos, 0.0f); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[c].mWetParams[i].Gains.Target); } } } else { /* With no distance, spread is only meaningful for mono sources * where it can be 0 or full (non-mono sources are always full * spread here). */ const float spread{Spread * float(voice->mFmtChannels == FmtMono)}; /* 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. */ const float ev{std::asin(chans[c].pos[1])}; const float az{std::atan2(chans[c].pos[0], -chans[c].pos[2])}; Device->mHrtf->getCoeffs(ev, az, 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. */ const auto coeffs = CalcDirectionCoeffs(chans[c].pos, spread); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[c].mWetParams[i].Gains.Target); } } } voice->mFlags.set(VoiceHasHrtf); } 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*NfcScale, Device->AvgSpeakerDist/4.0f)}; const float w0{SpeedOfSoundMetersPerSec / (mdist * Frequency)}; /* Adjust NFC filters. */ for(size_t c{0};c < num_channels;c++) voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0); voice->mFlags.set(VoiceHasNfc); } if(voice->mFmtChannels == FmtMono) { auto calc_coeffs = [xpos,ypos,zpos,Spread](RenderMode mode) { if(mode != RenderMode::Pairwise) return CalcDirectionCoeffs(std::array{xpos, ypos, zpos}, Spread); const auto pos = ScaleAzimuthFront3_2(std::array{xpos, ypos, zpos}); return CalcDirectionCoeffs(pos, Spread); }; const auto coeffs = calc_coeffs(Device->mRenderMode); ComputePanGains(&Device->Dry, coeffs, DryGain.Base, voice->mChans[0].mDryParams.Gains.Target); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, voice->mChans[0].mWetParams[i].Gains.Target); } } else { using namespace al::numbers; 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 uint idx{Device->channelIdxByName(chans[c].channel)}; if(idx != InvalidChannelIndex) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base; } continue; } /* Warp the channel position toward the source position as * the spread decreases. With no spread, all channels are * at the source position, at full spread (pi*2), each * channel position is left unchanged. */ const float a{1.0f - (inv_pi_v/2.0f)*Spread}; std::array pos{ lerpf(chans[c].pos[0], xpos, a), lerpf(chans[c].pos[1], ypos, a), lerpf(chans[c].pos[2], zpos, a)}; const float len{std::sqrt(pos[0]*pos[0] + pos[1]*pos[1] + pos[2]*pos[2])}; if(len < 1.0f) { pos[0] /= len; pos[1] /= len; pos[2] /= len; } if(Device->mRenderMode == RenderMode::Pairwise) pos = ScaleAzimuthFront3(pos); const auto coeffs = CalcDirectionCoeffs(pos, 0.0f); ComputePanGains(&Device->Dry, coeffs, DryGain.Base, voice->mChans[c].mDryParams.Gains.Target); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *Slot{SendSlots[i]}) ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base, 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. */ static constexpr float w0{0.0f}; for(size_t c{0};c < num_channels;c++) voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0); voice->mFlags.set(VoiceHasNfc); } /* With no distance, spread is only meaningful for mono sources * where it can be 0 or full (non-mono sources are always full * spread here). */ const float spread{Spread * float(voice->mFmtChannels == FmtMono)}; 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 uint idx{Device->channelIdxByName(chans[c].channel)}; if(idx != InvalidChannelIndex) voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base; } continue; } const auto coeffs = CalcDirectionCoeffs((Device->mRenderMode==RenderMode::Pairwise) ? ScaleAzimuthFront3(chans[c].pos) : chans[c].pos, spread); ComputePanGains(&Device->Dry, coeffs, DryGain.Base, voice->mChans[c].mDryParams.Gains.Target); for(uint i{0};i < NumSends;i++) { if(const EffectSlot *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(uint 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 ContextBase *context) { DeviceBase *Device{context->mDevice}; std::array SendSlots; voice->mDirect.Buffer = Device->Dry.Buffer; for(uint i{0};i < Device->NumAuxSends;i++) { SendSlots[i] = props->Send[i].Slot; if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None) { 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{MaxPitch}) voice->mStep = MaxPitch<mStep = maxu(fastf2u(Pitch * MixerFracOne), 1); voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState); /* Calculate gains */ GainTriplet DryGain; DryGain.Base = minf(clampf(props->Gain, props->MinGain, props->MaxGain) * props->Direct.Gain * context->mParams.Gain, GainMixMax); DryGain.HF = props->Direct.GainHF; DryGain.LF = props->Direct.GainLF; std::array WetGain; for(uint i{0};i < Device->NumAuxSends;i++) { WetGain[i].Base = minf(clampf(props->Gain, props->MinGain, props->MaxGain) * props->Send[i].Gain * context->mParams.Gain, GainMixMax); 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, context->mParams, Device); } void CalcAttnSourceParams(Voice *voice, const VoiceProps *props, const ContextBase *context) { DeviceBase *Device{context->mDevice}; const uint NumSends{Device->NumAuxSends}; /* Set mixing buffers and get send parameters. */ voice->mDirect.Buffer = Device->Dry.Buffer; std::array SendSlots{}; std::array RoomRolloff{}; std::bitset UseDryAttnForRoom{0}; for(uint i{0};i < NumSends;i++) { SendSlots[i] = props->Send[i].Slot; if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None) SendSlots[i] = nullptr; else if(SendSlots[i]->AuxSendAuto) { /* NOTE: Contrary to the EFX docs, the effect's room rolloff factor * applies to the selected distance model along with the source's * room rolloff factor, not necessarily the inverse distance model. * * Generic Software also applies these rolloff factors regardless * of any setting. It doesn't seem to use the effect slot's send * auto for anything, though as far as I understand, it's supposed * to control whether the send gets the same gain/gainhf as the * direct path (excluding the filter). */ RoomRolloff[i] = props->RoomRolloffFactor + SendSlots[i]->RoomRolloff; } else UseDryAttnForRoom.set(i); 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) { /* Transform source vectors */ Position = context->mParams.Matrix * (Position - context->mParams.Position); Velocity = context->mParams.Matrix * Velocity; Direction = context->mParams.Matrix * Direction; } else { /* Offset the source velocity to be relative of the listener velocity */ Velocity += context->mParams.Velocity; } const bool directional{Direction.normalize() > 0.0f}; alu::Vector ToSource{Position[0], Position[1], Position[2], 0.0f}; const float Distance{ToSource.normalize()}; /* Calculate distance attenuation */ float ClampedDist{Distance}; float DryGainBase{props->Gain}; std::array WetGainBase{}; WetGainBase.fill(props->Gain); float DryAttnBase{1.0f}; switch(context->mParams.SourceDistanceModel ? props->mDistanceModel : context->mParams.mDistanceModel) { case DistanceModel::InverseClamped: if(props->MaxDistance < props->RefDistance) break; ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); /*fall-through*/ case DistanceModel::Inverse: if(props->RefDistance > 0.0f) { float dist{lerpf(props->RefDistance, ClampedDist, props->RolloffFactor)}; if(dist > 0.0f) { DryAttnBase = props->RefDistance / dist; DryGainBase *= DryAttnBase; } for(size_t i{0};i < NumSends;++i) { dist = lerpf(props->RefDistance, ClampedDist, RoomRolloff[i]); if(dist > 0.0f) WetGainBase[i] *= props->RefDistance / dist; } } break; case DistanceModel::LinearClamped: if(props->MaxDistance < props->RefDistance) break; ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); /*fall-through*/ case DistanceModel::Linear: if(props->MaxDistance != props->RefDistance) { float attn{(ClampedDist-props->RefDistance) / (props->MaxDistance-props->RefDistance) * props->RolloffFactor}; DryAttnBase = maxf(1.0f - attn, 0.0f); DryGainBase *= DryAttnBase; for(size_t i{0};i < NumSends;++i) { attn = (ClampedDist-props->RefDistance) / (props->MaxDistance-props->RefDistance) * RoomRolloff[i]; WetGainBase[i] *= maxf(1.0f - attn, 0.0f); } } break; case DistanceModel::ExponentClamped: if(props->MaxDistance < props->RefDistance) break; ClampedDist = clampf(ClampedDist, props->RefDistance, props->MaxDistance); /*fall-through*/ case DistanceModel::Exponent: if(ClampedDist > 0.0f && props->RefDistance > 0.0f) { const float dist_ratio{ClampedDist/props->RefDistance}; DryAttnBase = std::pow(dist_ratio, -props->RolloffFactor); DryGainBase *= DryAttnBase; for(size_t i{0};i < NumSends;++i) WetGainBase[i] *= std::pow(dist_ratio, -RoomRolloff[i]); } break; case DistanceModel::Disable: break; } /* Calculate directional soundcones */ float ConeHF{1.0f}, WetCone{1.0f}, WetConeHF{1.0f}; if(directional && props->InnerAngle < 360.0f) { static constexpr float Rad2Deg{static_cast(180.0 / al::numbers::pi)}; const float Angle{Rad2Deg*2.0f * std::acos(-Direction.dot_product(ToSource)) * ConeScale}; float ConeGain{1.0f}; if(Angle >= props->OuterAngle) { ConeGain = props->OuterGain; if(props->DryGainHFAuto) ConeHF = props->OuterGainHF; } else if(Angle >= props->InnerAngle) { const float scale{(Angle-props->InnerAngle) / (props->OuterAngle-props->InnerAngle)}; ConeGain = lerpf(1.0f, props->OuterGain, scale); if(props->DryGainHFAuto) ConeHF = lerpf(1.0f, props->OuterGainHF, scale); } DryGainBase *= ConeGain; if(props->WetGainAuto) WetCone = ConeGain; if(props->WetGainHFAuto) WetConeHF = ConeHF; } /* Apply gain and frequency filters */ GainTriplet DryGain{}; DryGainBase = clampf(DryGainBase, props->MinGain, props->MaxGain) * context->mParams.Gain; DryGain.Base = minf(DryGainBase * props->Direct.Gain, GainMixMax); DryGain.HF = ConeHF * props->Direct.GainHF; DryGain.LF = props->Direct.GainLF; std::array WetGain{}; for(uint i{0};i < NumSends;i++) { WetGainBase[i] = clampf(WetGainBase[i]*WetCone, props->MinGain, props->MaxGain) * context->mParams.Gain; /* If this effect slot's Auxiliary Send Auto is off, then use the dry * path distance and cone attenuation, otherwise use the wet (room) * path distance and cone attenuation. The send filter is used instead * of the direct filter, regardless. */ const bool use_room{!UseDryAttnForRoom.test(i)}; const float gain{use_room ? WetGainBase[i] : DryGainBase}; WetGain[i].Base = minf(gain * props->Send[i].Gain, GainMixMax); WetGain[i].HF = (use_room ? WetConeHF : ConeHF) * props->Send[i].GainHF; WetGain[i].LF = props->Send[i].GainLF; } /* Distance-based air absorption and initial send decay. */ if(Distance > props->RefDistance) LIKELY { const float distance_base{(Distance-props->RefDistance) * props->RolloffFactor}; const float distance_meters{distance_base * context->mParams.MetersPerUnit}; const float dryabsorb{distance_meters * props->AirAbsorptionFactor}; if(dryabsorb > std::numeric_limits::epsilon()) DryGain.HF *= std::pow(context->mParams.AirAbsorptionGainHF, dryabsorb); /* If the source's Auxiliary Send Filter Gain Auto is off, no extra * adjustment is applied to the send gains. */ for(uint i{props->WetGainAuto ? 0u : NumSends};i < NumSends;++i) { if(!SendSlots[i] || !(SendSlots[i]->DecayTime > 0.0f)) continue; if(distance_meters > std::numeric_limits::epsilon()) WetGain[i].HF *= std::pow(SendSlots[i]->AirAbsorptionGainHF, distance_meters); /* If this effect slot's Auxiliary Send Auto is off, don't apply * the automatic initial reverb decay. * * NOTE: Generic Software applies the initial decay regardless of * this setting. It doesn't seem to use it for anything, only the * source's send filter gain auto flag affects this. */ if(!SendSlots[i]->AuxSendAuto) continue; GainTriplet DecayDistance; /* Calculate the distances to where this effect's decay reaches * -60dB. */ DecayDistance.Base = SendSlots[i]->DecayTime * SpeedOfSoundMetersPerSec; DecayDistance.LF = DecayDistance.Base * SendSlots[i]->DecayLFRatio; DecayDistance.HF = SendSlots[i]->DecayHFRatio; if(SendSlots[i]->DecayHFLimit) { const float airAbsorption{SendSlots[i]->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). */ static constexpr float log10_decaygain{-3.0f/*std::log10(ReverbDecayGain)*/}; const float absorb_dist{log10_decaygain / std::log10(airAbsorption)}; DecayDistance.HF = minf(absorb_dist, DecayDistance.HF); } } DecayDistance.HF *= DecayDistance.Base; /* Apply a decay-time transformation to the wet path, based on the * source distance. The initial decay of the reverb effect is * calculated and applied to the wet path. * * FIXME: This is very likely not correct. It more likely should * work by calculating a rolloff dynamically based on the reverb * parameters (and source distance?) and add it to the room rolloff * with the reverb and source rolloff parameters. */ const float baseAttn{DryAttnBase}; const float fact{distance_base / DecayDistance.Base}; const float gain{std::pow(ReverbDecayGain, fact)*(1.0f-baseAttn) + baseAttn}; WetGain[i].Base *= gain; if(gain > 0.0f) { const float hffact{distance_base / DecayDistance.HF}; const float gainhf{std::pow(ReverbDecayGain, hffact)*(1.0f-baseAttn) + baseAttn}; WetGain[i].HF *= minf(gainhf/gain, 1.0f); const float lffact{distance_base / DecayDistance.LF}; const float gainlf{std::pow(ReverbDecayGain, lffact)*(1.0f-baseAttn) + baseAttn}; WetGain[i].LF *= minf(gainlf/gain, 1.0f); } } } /* Initial source pitch */ float Pitch{props->Pitch}; /* Calculate velocity-based doppler effect */ float DopplerFactor{props->DopplerFactor * context->mParams.DopplerFactor}; if(DopplerFactor > 0.0f) { const alu::Vector &lvelocity = context->mParams.Velocity; float vss{Velocity.dot_product(ToSource) * -DopplerFactor}; float vls{lvelocity.dot_product(ToSource) * -DopplerFactor}; const float SpeedOfSound{context->mParams.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{MaxPitch}) voice->mStep = MaxPitch<mStep = maxu(fastf2u(Pitch * MixerFracOne), 1); voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState); float spread{0.0f}; if(props->Radius > Distance) spread = al::numbers::pi_v*2.0f - Distance/props->Radius*al::numbers::pi_v; else if(Distance > 0.0f) spread = std::asin(props->Radius/Distance) * 2.0f; CalcPanningAndFilters(voice, ToSource[0]*XScale, ToSource[1]*YScale, ToSource[2]*ZScale, Distance, spread, DryGain, WetGain, SendSlots, props, context->mParams, Device); } void CalcSourceParams(Voice *voice, ContextBase *context, bool force) { VoicePropsItem *props{voice->mUpdate.exchange(nullptr, std::memory_order_acq_rel)}; if(!props && !force) return; if(props) { voice->mProps = static_cast(*props); AtomicReplaceHead(context->mFreeVoiceProps, props); } if((voice->mProps.DirectChannels != DirectMode::Off && voice->mFmtChannels != FmtMono && !IsAmbisonic(voice->mFmtChannels)) || voice->mProps.mSpatializeMode == SpatializeMode::Off || (voice->mProps.mSpatializeMode==SpatializeMode::Auto && voice->mFmtChannels != FmtMono)) CalcNonAttnSourceParams(voice, &voice->mProps, context); else CalcAttnSourceParams(voice, &voice->mProps, context); } void SendSourceStateEvent(ContextBase *context, uint id, VChangeState state) { RingBuffer *ring{context->mAsyncEvents.get()}; auto evt_vec = ring->getWriteVector(); if(evt_vec.first.len < 1) return; auto &evt = InitAsyncEvent(evt_vec.first.buf); evt.mId = id; switch(state) { case VChangeState::Reset: evt.mState = AsyncSrcState::Reset; break; case VChangeState::Stop: evt.mState = AsyncSrcState::Stop; break; case VChangeState::Play: evt.mState = AsyncSrcState::Play; break; case VChangeState::Pause: evt.mState = AsyncSrcState::Pause; break; /* Shouldn't happen. */ case VChangeState::Restart: al::unreachable(); } ring->writeAdvance(1); } void ProcessVoiceChanges(ContextBase *ctx) { VoiceChange *cur{ctx->mCurrentVoiceChange.load(std::memory_order_acquire)}; VoiceChange *next{cur->mNext.load(std::memory_order_acquire)}; if(!next) return; const auto enabledevt = ctx->mEnabledEvts.load(std::memory_order_acquire); do { cur = next; bool sendevt{false}; if(cur->mState == VChangeState::Reset || cur->mState == VChangeState::Stop) { if(Voice *voice{cur->mVoice}) { voice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed); voice->mLoopBuffer.store(nullptr, std::memory_order_relaxed); /* A source ID indicates the voice was playing or paused, which * gets a reset/stop event. */ sendevt = voice->mSourceID.exchange(0u, std::memory_order_relaxed) != 0u; Voice::State oldvstate{Voice::Playing}; voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping, std::memory_order_relaxed, std::memory_order_acquire); voice->mPendingChange.store(false, std::memory_order_release); } /* Reset state change events are always sent, even if the voice is * already stopped or even if there is no voice. */ sendevt |= (cur->mState == VChangeState::Reset); } else if(cur->mState == VChangeState::Pause) { 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 == VChangeState::Play) { /* 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 == VChangeState::Restart) { /* Restarting a voice 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.test(al::to_underlying(AsyncEnableBits::SourceState))) 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(ContextBase *ctx, const EffectSlotArray &slots, const al::span voices) { ProcessVoiceChanges(ctx); IncrementRef(ctx->mUpdateCount); if(!ctx->mHoldUpdates.load(std::memory_order_acquire)) LIKELY { bool force{CalcContextParams(ctx)}; auto sorted_slots = const_cast(slots.data() + slots.size()); for(EffectSlot *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(DeviceBase *device, const uint SamplesToDo) { ASSUME(SamplesToDo > 0); const nanoseconds curtime{device->mClockBase.load(std::memory_order_relaxed) + nanoseconds{seconds{device->mSamplesDone.load(std::memory_order_relaxed)}}/ device->Frequency}; for(ContextBase *ctx : *device->mContexts.load(std::memory_order_acquire)) { const EffectSlotArray &auxslots = *ctx->mActiveAuxSlots.load(std::memory_order_acquire); const al::span voices{ctx->getVoicesSpanAcquired()}; /* Process pending property updates for objects on the context. */ ProcessParamUpdates(ctx, auxslots, voices); /* Clear auxiliary effect slot mixing buffers. */ for(EffectSlot *slot : auxslots) { for(auto &buffer : slot->Wet.Buffer) buffer.fill(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, curtime, SamplesToDo); } /* Process effects. */ if(!auxslots.empty()) { /* Sort the slots into extra storage, so that effect slots come * before their effect slot target (or their targets' target). */ const al::span sorted_slots{const_cast(al::to_address(auxslots.end())), auxslots.size()}; /* Skip sorting if it has already been done. */ if(!sorted_slots[0]) { /* First, copy the slots to the sorted list, then partition the * sorted list so that all slots without a target slot go to * the end. */ std::copy(auxslots.begin(), auxslots.end(), sorted_slots.begin()); auto split_point = std::partition(sorted_slots.begin(), sorted_slots.end(), [](const EffectSlot *slot) noexcept -> bool { return slot->Target != nullptr; }); /* There must be at least one slot without a slot target. */ assert(split_point != sorted_slots.end()); /* Simple case: no more than 1 slot has a target slot. Either * all slots go right to the output, or the remaining one must * target an already-partitioned slot. */ if(split_point - sorted_slots.begin() > 1) { /* At least two slots target other slots. Starting from the * back of the sorted list, continue partitioning the front * of the list given each target until all targets are * accounted for. This ensures all slots without a target * go last, all slots directly targeting those last slots * go second-to-last, all slots directly targeting those * second-last slots go third-to-last, etc. */ auto next_target = sorted_slots.end(); do { /* This shouldn't happen, but if there's unsorted slots * left that don't target any sorted slots, they can't * contribute to the output, so leave them. */ if(next_target == split_point) UNLIKELY break; --next_target; split_point = std::partition(sorted_slots.begin(), split_point, [next_target](const EffectSlot *slot) noexcept -> bool { return slot->Target != *next_target; }); } while(split_point - sorted_slots.begin() > 1); } } for(const EffectSlot *slot : sorted_slots) { EffectState *state{slot->mEffectState.get()}; state->process(SamplesToDo, slot->Wet.Buffer, state->mOutTarget); } } /* Signal the event handler if there are any events to read. */ RingBuffer *ring{ctx->mAsyncEvents.get()}; if(ring->readSpace() > 0) ctx->mEventSem.post(); } } void ApplyDistanceComp(const al::span Samples, const size_t SamplesToDo, const DistanceComp::ChanData *distcomp) { ASSUME(SamplesToDo > 0); for(auto &chanbuffer : Samples) { const float gain{distcomp->Gain}; const size_t 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(SamplesToDo >= base) LIKELY { 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, [gain](float s) { return s * gain; }); } } void ApplyDither(const al::span Samples, uint *dither_seed, const float quant_scale, const size_t SamplesToDo) { static constexpr double invRNGRange{1.0 / std::numeric_limits::max()}; ASSUME(SamplesToDo > 0); /* 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 and before rounding. */ const float invscale{1.0f / quant_scale}; uint seed{*dither_seed}; auto dither_sample = [&seed,invscale,quant_scale](const float sample) noexcept -> float { float val{sample * quant_scale}; uint rng0{dither_rng(&seed)}; uint rng1{dither_rng(&seed)}; val += static_cast(rng0*invRNGRange - rng1*invRNGRange); return fast_roundf(val) * invscale; }; for(FloatBufferLine &inout : Samples) std::transform(inout.begin(), inout.begin()+SamplesToDo, inout.begin(), dither_sample); *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 size_t SamplesToDo, const size_t FrameStep) { ASSUME(FrameStep > 0); ASSUME(SamplesToDo > 0); DevFmtType_t *outbase{static_cast*>(OutBuffer) + Offset*FrameStep}; size_t c{0}; for(const FloatBufferLine &inbuf : InBuffer) { DevFmtType_t *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); ++c; } if(const size_t extra{FrameStep - c}) { const auto silence = SampleConv>(0.0f); for(size_t i{0};i < SamplesToDo;++i) { std::fill_n(outbase, extra, silence); outbase += FrameStep; } } } } // namespace uint DeviceBase::renderSamples(const uint numSamples) { const uint samplesToDo{minu(numSamples, BufferLineSize)}; /* Clear main mixing buffers. */ for(FloatBufferLine &buffer : MixBuffer) buffer.fill(0.0f); { const auto mixLock = getWriteMixLock(); /* Process and mix each context's sources and effects. */ ProcessContexts(this, samplesToDo); /* 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. */ auto samplesDone = mSamplesDone.load(std::memory_order_relaxed) + samplesToDo; auto clockBase = mClockBase.load(std::memory_order_relaxed) + std::chrono::seconds{samplesDone/Frequency}; mSamplesDone.store(samplesDone%Frequency, std::memory_order_relaxed); mClockBase.store(clockBase, std::memory_order_relaxed); } /* Apply any needed post-process for finalizing the Dry mix to the RealOut * (Ambisonic decode, UHJ encode, etc). */ postProcess(samplesToDo); /* Apply compression, limiting sample amplitude if needed or desired. */ if(Limiter) Limiter->process(samplesToDo, RealOut.Buffer.data()); /* Apply delays and attenuation for mismatched speaker distances. */ if(ChannelDelays) ApplyDistanceComp(RealOut.Buffer, samplesToDo, ChannelDelays->mChannels.data()); /* Apply dithering. The compressor should have left enough headroom for the * dither noise to not saturate. */ if(DitherDepth > 0.0f) ApplyDither(RealOut.Buffer, &DitherSeed, DitherDepth, samplesToDo); return samplesToDo; } void DeviceBase::renderSamples(const al::span outBuffers, const uint numSamples) { FPUCtl mixer_mode{}; uint total{0}; while(const uint todo{numSamples - total}) { const uint samplesToDo{renderSamples(todo)}; auto *srcbuf = RealOut.Buffer.data(); for(auto *dstbuf : outBuffers) { std::copy_n(srcbuf->data(), samplesToDo, dstbuf + total); ++srcbuf; } total += samplesToDo; } } void DeviceBase::renderSamples(void *outBuffer, const uint numSamples, const size_t frameStep) { FPUCtl mixer_mode{}; uint total{0}; while(const uint todo{numSamples - total}) { const uint samplesToDo{renderSamples(todo)}; if(outBuffer) LIKELY { /* Finally, interleave and convert samples, writing to the device's * output buffer. */ switch(FmtType) { #define HANDLE_WRITE(T) case T: \ Write(RealOut.Buffer, outBuffer, total, 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 } } total += samplesToDo; } } void DeviceBase::handleDisconnect(const char *msg, ...) { const auto mixLock = getWriteMixLock(); if(Connected.exchange(false, std::memory_order_acq_rel)) { AsyncEvent evt{std::in_place_type}; auto &disconnect = std::get(evt); va_list args; va_start(args, msg); int msglen{vsnprintf(disconnect.msg.data(), disconnect.msg.size(), msg, args)}; va_end(args); if(msglen < 0 || static_cast(msglen) >= disconnect.msg.size()) disconnect.msg[sizeof(disconnect.msg)-1] = 0; for(ContextBase *ctx : *mContexts.load()) { RingBuffer *ring{ctx->mAsyncEvents.get()}; auto evt_data = ring->getWriteVector().first; if(evt_data.len > 0) { al::construct_at(reinterpret_cast(evt_data.buf), evt); ring->writeAdvance(1); ctx->mEventSem.post(); } if(!ctx->mStopVoicesOnDisconnect) { ProcessVoiceChanges(ctx); continue; } 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); } } }