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- /**
- * 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 "config_simd.h"
- #include "alu.h"
- #include <algorithm>
- #include <array>
- #include <atomic>
- #include <cassert>
- #include <cmath>
- #include <cstdarg>
- #include <cstddef>
- #include <cstdint>
- #include <cstdio>
- #include <cstdlib>
- #include <iterator>
- #include <limits>
- #include <memory>
- #include <optional>
- #include <string>
- #include <string_view>
- #include <utility>
- #include <variant>
- #include "almalloc.h"
- #include "alnumbers.h"
- #include "alnumeric.h"
- #include "alsem.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/storage_formats.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"
- struct CTag;
- #if HAVE_SSE
- struct SSETag;
- #endif
- #if HAVE_SSE2
- struct SSE2Tag;
- #endif
- #if HAVE_SSE4_1
- struct SSE4Tag;
- #endif
- #if 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::string_view_literals;
- float InitConeScale()
- {
- float ret{1.0f};
- if(auto optval = al::getenv("__ALSOFT_HALF_ANGLE_CONES"))
- {
- if(al::case_compare(*optval, "true"sv) == 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<const FloatBufferLine> InSamples, const al::span<float2> AccumSamples,
- const al::span<float,BufferLineSize> TempBuf, const al::span<HrtfChannelState> ChanState,
- const size_t IrSize, const size_t SamplesToDo);
- HrtfDirectMixerFunc MixDirectHrtf{MixDirectHrtf_<CTag>};
- inline HrtfDirectMixerFunc SelectHrtfMixer()
- {
- #if HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return MixDirectHrtf_<NEONTag>;
- #endif
- #if HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return MixDirectHrtf_<SSETag>;
- #endif
- return MixDirectHrtf_<CTag>;
- }
- 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<float>(increment) - table->scaleBase;
- sf = std::max(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<float>(si)));
- }
- state->sf = sf;
- state->m = table->m[si];
- state->l = (state->m/2) - 1;
- state->filter = table->Tab.subspan(table->filterOffset[si]);
- }
- inline ResamplerFunc SelectResampler(Resampler resampler, uint increment)
- {
- switch(resampler)
- {
- case Resampler::Point:
- return Resample_<PointTag,CTag>;
- case Resampler::Linear:
- #if HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Resample_<LerpTag,NEONTag>;
- #endif
- #if HAVE_SSE4_1
- if((CPUCapFlags&CPU_CAP_SSE4_1))
- return Resample_<LerpTag,SSE4Tag>;
- #endif
- #if HAVE_SSE2
- if((CPUCapFlags&CPU_CAP_SSE2))
- return Resample_<LerpTag,SSE2Tag>;
- #endif
- return Resample_<LerpTag,CTag>;
- case Resampler::Spline:
- case Resampler::Gaussian:
- #if HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Resample_<CubicTag,NEONTag>;
- #endif
- #if HAVE_SSE4_1
- if((CPUCapFlags&CPU_CAP_SSE4_1))
- return Resample_<CubicTag,SSE4Tag>;
- #endif
- #if HAVE_SSE2
- if((CPUCapFlags&CPU_CAP_SSE2))
- return Resample_<CubicTag,SSE2Tag>;
- #endif
- #if HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return Resample_<CubicTag,SSETag>;
- #endif
- return Resample_<CubicTag,CTag>;
- case Resampler::BSinc12:
- case Resampler::BSinc24:
- case Resampler::BSinc48:
- if(increment > MixerFracOne)
- {
- #if HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Resample_<BSincTag,NEONTag>;
- #endif
- #if HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return Resample_<BSincTag,SSETag>;
- #endif
- return Resample_<BSincTag,CTag>;
- }
- [[fallthrough]];
- case Resampler::FastBSinc12:
- case Resampler::FastBSinc24:
- case Resampler::FastBSinc48:
- #if HAVE_NEON
- if((CPUCapFlags&CPU_CAP_NEON))
- return Resample_<FastBSincTag,NEONTag>;
- #endif
- #if HAVE_SSE
- if((CPUCapFlags&CPU_CAP_SSE))
- return Resample_<FastBSincTag,SSETag>;
- #endif
- return Resample_<FastBSincTag,CTag>;
- }
- return Resample_<PointTag,CTag>;
- }
- } // 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 = std::clamp(nfcscale, 0.0001f, 10000.0f);
- }
- ResamplerFunc PrepareResampler(Resampler resampler, uint increment, InterpState *state)
- {
- switch(resampler)
- {
- case Resampler::Point:
- case Resampler::Linear:
- break;
- case Resampler::Spline:
- state->emplace<CubicState>(al::span{gSplineFilter.mTable});
- break;
- case Resampler::Gaussian:
- state->emplace<CubicState>(al::span{gGaussianFilter.mTable});
- break;
- case Resampler::FastBSinc12:
- case Resampler::BSinc12:
- BsincPrepare(increment, &state->emplace<BsincState>(), &gBSinc12);
- break;
- case Resampler::FastBSinc24:
- case Resampler::BSinc24:
- BsincPrepare(increment, &state->emplace<BsincState>(), &gBSinc24);
- break;
- case Resampler::FastBSinc48:
- case Resampler::BSinc48:
- BsincPrepare(increment, &state->emplace<BsincState>(), &gBSinc48);
- 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,
- mHrtfState->mTemp, mHrtfState->mChannels, mHrtfState->mIrSize, SamplesToDo);
- }
- void DeviceBase::ProcessAmbiDec(const size_t SamplesToDo)
- {
- AmbiDecoder->process(RealOut.Buffer, Dry.Buffer, 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, 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, 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(al::span{RealOut.Buffer[lidx]}.first(SamplesToDo),
- al::span{RealOut.Buffer[ridx]}.first(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<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> output,
- const al::span<const std::array<float,MaxAmbiChannels>> upsampler,
- const al::span<const std::array<float,MaxAmbiChannels>,MaxAmbiChannels> rotator,
- size_t ambi_order)
- {
- const size_t num_chans{AmbiChannelsFromOrder(ambi_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)
- {
- const float a{upsampler[i][k]};
- /* Write the full number of channels. The compiler will have an
- * easier time optimizing if it has a fixed length.
- */
- std::transform(rotator[k].cbegin(), rotator[k].cend(), output[i].cbegin(),
- output[i].begin(), [a](float rot, float dst) noexcept { return rot*a + dst; });
- }
- }
- }
- 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
- #if ALSOFT_EAX
- * props->DistanceFactor
- #endif
- ;
- ctx->mParams.AirAbsorptionGainHF = props->AirAbsorptionGainHF;
- ctx->mParams.DopplerFactor = props->DopplerFactor;
- ctx->mParams.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity
- #if ALSOFT_EAX
- / props->DistanceFactor
- #endif
- ;
- 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;
- slot->RoomRolloff = 0.0f;
- slot->DecayTime = 0.0f;
- slot->DecayLFRatio = 0.0f;
- slot->DecayHFRatio = 0.0f;
- slot->DecayHFLimit = false;
- slot->AirAbsorptionGainHF = 1.0f;
- if(auto *reverbprops = std::get_if<ReverbProps>(&props->Props))
- {
- slot->RoomRolloff = reverbprops->RoomRolloffFactor;
- slot->AirAbsorptionGainHF = reverbprops->AirAbsorptionGainHF;
- /* If this effect slot's Auxiliary Send Auto is off, don't apply the
- * automatic send adjustments based on source distance.
- */
- if(slot->AuxSendAuto)
- {
- slot->DecayTime = reverbprops->DecayTime;
- slot->DecayLFRatio = reverbprops->DecayLFRatio;
- slot->DecayHFRatio = reverbprops->DecayHFRatio;
- slot->DecayHFLimit = reverbprops->DecayHFLimit;
- }
- }
- 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[0].len > 0) LIKELY
- {
- auto &evt = InitAsyncEvent<AsyncEffectReleaseEvent>(evt_vec[0].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);
- const auto output = [slot,context]() -> EffectTarget
- {
- if(EffectSlot *target{slot->Target})
- return EffectTarget{&target->Wet, nullptr};
- DeviceBase *device{context->mDevice};
- return 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<float,3> ScaleAzimuthFront3(std::array<float,3> 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<float,3> ScaleAzimuthFront3_2(std::array<float,3> 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
- * <https://github.com/polarch/Spherical-Harmonic-Transform/blob/master/getSHrotMtx.m>,
- * 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<CoeffValues,CalcRotatorSize(MaxAmbiOrder)> 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<double>((std::abs(n) == l) ?
- (2*l) * (2*l - 1) : (l*l - n*n))};
- if(m == 0)
- {
- coeffs->u = static_cast<float>(std::sqrt(l * l / denom));
- coeffs->v = static_cast<float>(std::sqrt((l-1) * l / denom) * -1.0);
- coeffs->w = 0.0f;
- }
- else
- {
- const int abs_m{std::abs(m)};
- coeffs->u = static_cast<float>(std::sqrt((l*l - m*m) / denom));
- coeffs->v = static_cast<float>(std::sqrt((l+abs_m-1) * (l+abs_m) / denom) *
- 0.5);
- coeffs->w = static_cast<float>(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;
- static constexpr 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<size_t>(i+2_z)]};
- const float rim1{R[-1+2][static_cast<size_t>(i+2_z)]};
- const float ri0{ R[ 0+2][static_cast<size_t>(i+2_z)]};
- const size_t y{last_band + static_cast<size_t>(a+l-1)};
- if(n == -l)
- return ri1*R[last_band][y] + rim1*R[last_band + static_cast<size_t>(l-1_z)*2][y];
- if(n == l)
- return ri1*R[last_band + static_cast<size_t>(l-1_z)*2][y] - rim1*R[last_band][y];
- return ri0*R[last_band + static_cast<size_t>(l-1_z+n)][y];
- };
- static constexpr auto U = [](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);
- };
- static constexpr auto V = [](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<float> : (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<float> : (p0 + p1);
- };
- static constexpr auto W = [](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<uint>(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<float>*0.5f};
- constexpr float cos45{al::numbers::sqrt2_v<float>*0.5f};
- constexpr float sin110{ 0.939692620786f};
- constexpr float cos110{-0.342020143326f};
- struct ChanPosMap {
- Channel channel;
- std::array<float,3> 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<const GainTriplet,MaxSendCount> WetGain,
- const al::span<EffectSlot*,MaxSendCount> 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<float>(Device->mSampleRate);
- 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<DirectMode,al::span<const ChanPosMap>>
- {
- switch(chanfmt)
- {
- case FmtMono:
- /* Mono buffers are never played direct. */
- return {DirectMode::Off, al::span{MonoMap}};
- case FmtStereo:
- case FmtMonoDup:
- 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 = std::min(props->EnhWidth, 0.7f);
- const float lgain{std::min(1.0f-props->Panning, 1.0f)};
- const float rgain{std::min(1.0f+props->Panning, 1.0f)};
- const float mingain{std::min(lgain, rgain)};
- auto SelectChannelGain = [lgain,rgain,mingain](const Channel chan) noexcept
- {
- switch(chan)
- {
- case FrontLeft: return lgain;
- case FrontRight: return rgain;
- case FrontCenter: break;
- case LFE: break;
- case BackLeft: return lgain;
- case BackRight: return rgain;
- case BackCenter: break;
- case SideLeft: return lgain;
- case SideRight: return rgain;
- case TopCenter: break;
- case TopFrontLeft: return lgain;
- case TopFrontCenter: break;
- case TopFrontRight: return rgain;
- case TopBackLeft: return lgain;
- case TopBackCenter: break;
- case TopBackRight: return rgain;
- case BottomFrontLeft: return lgain;
- case BottomFrontRight: return rgain;
- case BottomBackLeft: return lgain;
- case BottomBackRight: return rgain;
- case Aux0: case Aux1: case Aux2: case Aux3: case Aux4: case Aux5: case Aux6: case Aux7:
- case Aux8: case Aux9: case Aux10: case Aux11: case Aux12: case Aux13: case Aux14:
- case Aux15: case MaxChannels: break;
- }
- return mingain;
- };
- 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<float>::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{std::max(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<float>::epsilon()) ? 1.0f :
- (al::numbers::inv_pi_v<float>/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<int>(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 auto index_map = Is2DAmbisonic(voice->mFmtChannels) ?
- GetAmbi2DLayout(voice->mAmbiLayout).subspan(0) :
- GetAmbiLayout(voice->mAmbiLayout).subspan(0);
- /* 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.
- */
- std::transform(mixmatrix[acn].cbegin(), mixmatrix[acn].cend(), coeffs.begin(),
- coeffs.begin(), [scale](const float in, const float coeff) noexcept
- { return in*scale + coeff; });
- 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<float,MaxAmbiChannels>{};
- }
- }
- }
- 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++)
- {
- const float pangain{SelectChannelGain(chans[c].channel)};
- if(uint idx{Device->channelIdxByName(chans[c].channel)}; idx != InvalidChannelIndex)
- voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base * pangain;
- 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 * pangain
- * 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 float pangain{SelectChannelGain(chans[c].channel)};
- 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 * pangain,
- 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<float>::epsilon())
- {
- if(voice->mFmtChannels == FmtMono)
- {
- const float src_ev{std::asin(std::clamp(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++)
- {
- /* Skip LFE */
- if(chans[c].channel == LFE) continue;
- const float pangain{SelectChannelGain(chans[c].channel)};
- /* 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 - (al::numbers::inv_pi_v<float>/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(std::clamp(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 * pangain;
- 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 * pangain,
- 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;
- const float pangain{SelectChannelGain(chans[c].channel)};
- /* 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<float>::infinity(), spread,
- voice->mChans[c].mDryParams.Hrtf.Target.Coeffs,
- voice->mChans[c].mDryParams.Hrtf.Target.Delay);
- voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain.Base * pangain;
- /* 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 * pangain,
- 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<float>::epsilon())
- {
- /* Calculate NFC filter coefficient if needed. */
- if(Device->AvgSpeakerDist > 0.0f)
- {
- /* Clamp the distance for really close sources, to prevent
- * excessive bass.
- */
- const float mdist{std::max(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 for(size_t c{0};c < num_channels;c++)
- {
- const auto pangain = SelectChannelGain(chans[c].channel);
- /* Special-case LFE */
- if(chans[c].channel == LFE)
- {
- if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
- {
- const auto idx = uint{Device->channelIdxByName(chans[c].channel)};
- if(idx != InvalidChannelIndex)
- voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base * pangain;
- }
- 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 auto a = 1.0f - (al::numbers::inv_pi_v<float>/2.0f)*Spread;
- auto pos = std::array{
- lerpf(chans[c].pos[0], xpos, a),
- lerpf(chans[c].pos[1], ypos, a),
- lerpf(chans[c].pos[2], zpos, a)};
- const auto 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 * pangain,
- 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 * pangain,
- 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++)
- {
- const float pangain{SelectChannelGain(chans[c].channel)};
- /* 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 * pangain;
- }
- continue;
- }
- const auto coeffs = CalcDirectionCoeffs((Device->mRenderMode==RenderMode::Pairwise)
- ? ScaleAzimuthFront3(chans[c].pos) : chans[c].pos, spread);
- ComputePanGains(&Device->Dry, coeffs, DryGain.Base * pangain,
- 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 * pangain,
- 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<EffectSlot*,MaxSendCount> 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<float>(voice->mFrequency) /
- static_cast<float>(Device->mSampleRate) * props->Pitch;
- if(Pitch > float{MaxPitch})
- voice->mStep = MaxPitch<<MixerFracBits;
- else
- voice->mStep = std::max(fastf2u(Pitch * MixerFracOne), 1u);
- voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState);
- /* Calculate gains */
- GainTriplet DryGain{};
- DryGain.Base = std::min(std::clamp(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<GainTriplet,MaxSendCount> WetGain{};
- for(uint i{0};i < Device->NumAuxSends;i++)
- {
- WetGain[i].Base = std::min(std::clamp(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<EffectSlot*,MaxSendCount> SendSlots{};
- std::array<float,MaxSendCount> RoomRolloff{};
- for(uint i{0};i < NumSends;i++)
- {
- SendSlots[i] = props->Send[i].Slot;
- if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None)
- {
- SendSlots[i] = nullptr;
- voice->mSend[i].Buffer = {};
- }
- else
- {
- /* 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.
- */
- RoomRolloff[i] = props->RoomRolloffFactor + SendSlots[i]->RoomRolloff;
- 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<float,MaxSendCount> 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 = std::clamp(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 = std::clamp(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 = std::max(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] *= std::max(1.0f - attn, 0.0f);
- }
- }
- break;
- case DistanceModel::ExponentClamped:
- if(props->MaxDistance < props->RefDistance) break;
- ClampedDist = std::clamp(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<float>(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 = std::clamp(DryGainBase, props->MinGain, props->MaxGain) * context->mParams.Gain;
- DryGain.Base = std::min(DryGainBase * props->Direct.Gain, GainMixMax);
- DryGain.HF = ConeHF * props->Direct.GainHF;
- DryGain.LF = props->Direct.GainLF;
- std::array<GainTriplet,MaxSendCount> WetGain{};
- for(uint i{0};i < NumSends;i++)
- {
- const auto gain = std::clamp(WetGainBase[i]*WetCone, props->MinGain, props->MaxGain) *
- context->mParams.Gain;
- WetGain[i].Base = std::min(gain * props->Send[i].Gain, GainMixMax);
- WetGain[i].HF = WetConeHF * props->Send[i].GainHF;
- WetGain[i].LF = props->Send[i].GainLF;
- }
- /* Distance-based air absorption and initial send decay. */
- if(Distance > props->RefDistance) LIKELY
- {
- /* FIXME: In keeping with EAX, the base air absorption gain should be
- * taken from the reverb property in the "primary fx slot" when it has
- * a reverb effect and the environment flag set, and be applied to the
- * direct path and all environment sends, rather than each path using
- * the air absorption gain associated with the given slot's effect. At
- * this point in the mixer, and even in EFX itself, there's no concept
- * of a "primary fx slot" so it's unclear which effect slot should be
- * checked.
- *
- * The HF reference is also intended to be handled the same way, but
- * again, there's no concept of a "primary fx slot" here and no way to
- * know which effect slot to look at for the reference frequency.
- */
- const auto distance_units = float{(Distance-props->RefDistance) * props->RolloffFactor};
- const auto distance_meters = float{distance_units * context->mParams.MetersPerUnit};
- const auto absorb = float{distance_meters * props->AirAbsorptionFactor};
- if(absorb > std::numeric_limits<float>::epsilon())
- DryGain.HF *= std::pow(context->mParams.AirAbsorptionGainHF, absorb);
- /* 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(SendSlots[i]->AirAbsorptionGainHF < 1.0f
- && absorb > std::numeric_limits<float>::epsilon())
- WetGain[i].HF *= std::pow(SendSlots[i]->AirAbsorptionGainHF, absorb);
- const float DecayDistance{SendSlots[i]->DecayTime * SpeedOfSoundMetersPerSec};
- /* 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_meters / DecayDistance};
- const float gain{std::pow(ReverbDecayGain, fact)*(1.0f-baseAttn) + baseAttn};
- WetGain[i].Base *= gain;
- }
- }
- /* 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<float>::infinity();
- }
- else
- {
- /* Source and listener movement is nominal. Calculate the proper
- * doppler shift.
- */
- Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss);
- }
- }
- /* Adjust pitch based on the buffer and output frequencies, and calculate
- * fixed-point stepping value.
- */
- Pitch *= static_cast<float>(voice->mFrequency) / static_cast<float>(Device->mSampleRate);
- if(Pitch > float{MaxPitch})
- voice->mStep = MaxPitch<<MixerFracBits;
- else
- voice->mStep = std::max(fastf2u(Pitch * MixerFracOne), 1u);
- voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState);
- float spread{0.0f};
- if(props->Radius > Distance)
- spread = al::numbers::pi_v<float>*2.0f - Distance/props->Radius*al::numbers::pi_v<float>;
- 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<VoiceProps&>(*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[0].len < 1) return;
- auto &evt = InitAsyncEvent<AsyncSourceStateEvent>(evt_vec[0].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 al::span<EffectSlot*> slots,
- const al::span<EffectSlot*> sorted_slots, const al::span<Voice*> voices)
- {
- ProcessVoiceChanges(ctx);
- IncrementRef(ctx->mUpdateCount);
- if(!ctx->mHoldUpdates.load(std::memory_order_acquire)) LIKELY
- {
- bool force{CalcContextParams(ctx)};
- auto sorted_slot_base = al::to_address(sorted_slots.begin());
- for(EffectSlot *slot : slots)
- force |= CalcEffectSlotParams(slot, sorted_slot_base, 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 auto curtime = device->getClockTime();
- auto proc_context = [SamplesToDo,curtime](ContextBase *ctx)
- {
- const auto auxslotspan = al::span{*ctx->mActiveAuxSlots.load(std::memory_order_acquire)};
- const auto auxslots = auxslotspan.first(auxslotspan.size()>>1);
- const auto sorted_slots = auxslotspan.last(auxslotspan.size()>>1);
- const auto voices = ctx->getVoicesSpanAcquired();
- /* Process pending property updates for objects on the context. */
- ProcessParamUpdates(ctx, auxslots, sorted_slots, voices);
- /* Clear auxiliary effect slot mixing buffers. */
- auto clear_wetbuffers = [](EffectSlot *slot)
- {
- auto clear_buffer = [](const FloatBufferSpan buffer)
- { std::fill(buffer.begin(), buffer.end(), 0.0f); };
- std::for_each(slot->Wet.Buffer.begin(), slot->Wet.Buffer.end(), clear_buffer);
- };
- std::for_each(auxslots.begin(), auxslots.end(), clear_wetbuffers);
- /* Process voices that have a playing source. */
- auto proc_voice = [ctx,curtime,SamplesToDo](Voice *voice)
- {
- const Voice::State vstate{voice->mPlayState.load(std::memory_order_acquire)};
- if(vstate != Voice::Stopped && vstate != Voice::Pending)
- voice->mix(vstate, ctx, curtime, SamplesToDo);
- };
- std::for_each(voices.begin(), voices.end(), proc_voice);
- /* 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). Skip
- * sorting if it has already been done.
- */
- if(!sorted_slots[0])
- {
- /* First, copy the slots to the sorted list and partition them,
- * so that all slots without a target slot go to the end.
- */
- auto has_target = [](const EffectSlot *slot) noexcept -> bool
- { return slot->Target != nullptr; };
- auto split_point = std::partition_copy(auxslots.rbegin(), auxslots.rend(),
- sorted_slots.begin(), sorted_slots.rbegin(), has_target).first;
- /* There must be at least one slot without a slot target. */
- assert(split_point != sorted_slots.end());
- /* 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();
- while(std::distance(sorted_slots.begin(), split_point) > 1)
- {
- /* 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;
- auto not_next = [next_target](const EffectSlot *slot) noexcept -> bool
- { return slot->Target != *next_target; };
- split_point = std::partition(sorted_slots.begin(), split_point, not_next);
- }
- }
- auto proc_slot = [SamplesToDo](const EffectSlot *slot)
- {
- EffectState *state{slot->mEffectState.get()};
- state->process(SamplesToDo, slot->Wet.Buffer, state->mOutTarget);
- };
- std::for_each(sorted_slots.begin(), sorted_slots.end(), proc_slot);
- }
- /* Signal the event handler if there are any events to read. */
- if(RingBuffer *ring{ctx->mAsyncEvents.get()}; ring->readSpace() > 0)
- ctx->mEventSem.post();
- };
- const auto contexts = al::span{*device->mContexts.load(std::memory_order_acquire)};
- std::for_each(contexts.begin(), contexts.end(), proc_context);
- }
- void ApplyDistanceComp(const al::span<FloatBufferLine> Samples, const size_t SamplesToDo,
- const al::span<const DistanceComp::ChanData,MaxOutputChannels> chandata)
- {
- ASSUME(SamplesToDo > 0);
- auto distcomp = chandata.begin();
- for(auto &chanbuffer : Samples)
- {
- const float gain{distcomp->Gain};
- auto distbuf = al::span{al::assume_aligned<16>(distcomp->Buffer.data()),
- distcomp->Buffer.size()};
- ++distcomp;
- const size_t base{distbuf.size()};
- if(base < 1) continue;
- const auto inout = al::span{al::assume_aligned<16>(chanbuffer.data()), SamplesToDo};
- if(SamplesToDo >= base) LIKELY
- {
- auto delay_end = std::rotate(inout.begin(), inout.end()-ptrdiff_t(base), inout.end());
- std::swap_ranges(inout.begin(), delay_end, distbuf.begin());
- }
- else
- {
- auto delay_start = std::swap_ranges(inout.begin(), inout.end(), distbuf.begin());
- std::rotate(distbuf.begin(), delay_start, distbuf.begin()+ptrdiff_t(base));
- }
- std::transform(inout.begin(), inout.end(), inout.begin(),
- [gain](float s) { return s*gain; });
- }
- }
- void ApplyDither(const al::span<FloatBufferLine> Samples, uint *dither_seed,
- const float quant_scale, const size_t SamplesToDo)
- {
- static constexpr double invRNGRange{1.0 / std::numeric_limits<uint>::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<float>(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<typename T>
- 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(std::clamp(val*2147483648.0f, -2147483648.0f, 2147483520.0f));
- }
- template<> inline int16_t SampleConv(float val) noexcept
- { return static_cast<int16_t>(fastf2i(std::clamp(val*32768.0f, -32768.0f, 32767.0f))); }
- template<> inline int8_t SampleConv(float val) noexcept
- { return static_cast<int8_t>(fastf2i(std::clamp(val*128.0f, -128.0f, 127.0f))); }
- /* Define unsigned output variations. */
- template<> inline uint32_t SampleConv(float val) noexcept
- { return static_cast<uint32_t>(SampleConv<int32_t>(val)) + 2147483648u; }
- template<> inline uint16_t SampleConv(float val) noexcept
- { return static_cast<uint16_t>(SampleConv<int16_t>(val) + 32768); }
- template<> inline uint8_t SampleConv(float val) noexcept
- { return static_cast<uint8_t>(SampleConv<int8_t>(val) + 128); }
- template<typename T>
- void Write(const al::span<const FloatBufferLine> InBuffer, void *OutBuffer, const size_t Offset,
- const size_t SamplesToDo, const size_t FrameStep)
- {
- ASSUME(FrameStep > 0);
- ASSUME(SamplesToDo > 0);
- /* Some Clang versions don't like calling subspan on an rvalue here. */
- const auto output_ = al::span{static_cast<T*>(OutBuffer), (Offset+SamplesToDo)*FrameStep};
- const auto output = output_.subspan(Offset*FrameStep);
- /* If there's extra channels in the interleaved output buffer to skip,
- * clear the whole output buffer. This is simpler to ensure the extra
- * channels are silent than trying to clear just the extra channels.
- */
- if(FrameStep > InBuffer.size())
- std::fill(output.begin(), output.end(), SampleConv<T>(0.0f));
- auto outbase = output.begin();
- for(const auto &srcbuf : InBuffer)
- {
- const auto src = al::span{srcbuf}.first(SamplesToDo);
- auto out = outbase++;
- *out = SampleConv<T>(src.front());
- std::for_each(src.begin()+1, src.end(), [FrameStep,&out](const float s) noexcept
- {
- out += ptrdiff_t(FrameStep);
- *out = SampleConv<T>(s);
- });
- }
- }
- template<typename T>
- void Write(const al::span<const FloatBufferLine> InBuffer, al::span<void*> OutBuffers,
- const size_t Offset, const size_t SamplesToDo)
- {
- ASSUME(SamplesToDo > 0);
- auto srcbuf = InBuffer.cbegin();
- for(auto *dstbuf : OutBuffers)
- {
- const auto src = al::span{*srcbuf}.first(SamplesToDo);
- /* Some Clang versions don't like calling subspan on an rvalue here. */
- const auto dst_ = al::span{static_cast<T*>(dstbuf), Offset+SamplesToDo};
- const auto dst = dst_.subspan(Offset);
- std::transform(src.begin(), src.end(), dst.begin(), SampleConv<T>);
- ++srcbuf;
- }
- }
- } // namespace
- uint DeviceBase::renderSamples(const uint numSamples)
- {
- const uint samplesToDo{std::min(numSamples, uint{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 clockBaseSec = mClockBaseSec.load(std::memory_order_relaxed) +
- seconds32{samplesDone/mSampleRate};
- mSamplesDone.store(samplesDone%mSampleRate, std::memory_order_relaxed);
- mClockBaseSec.store(clockBaseSec, 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);
- /* Apply delays and attenuation for mismatched speaker distances. */
- if(ChannelDelays)
- ApplyDistanceComp(RealOut.Buffer, samplesToDo, ChannelDelays->mChannels);
- /* 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<void*> outBuffers, const uint numSamples)
- {
- FPUCtl mixer_mode{};
- uint total{0};
- while(const uint todo{numSamples - total})
- {
- const uint samplesToDo{renderSamples(todo)};
- switch(FmtType)
- {
- #define HANDLE_WRITE(T) case T: \
- Write<DevFmtType_t<T>>(RealOut.Buffer, outBuffers, total, samplesToDo); break;
- HANDLE_WRITE(DevFmtByte)
- HANDLE_WRITE(DevFmtUByte)
- HANDLE_WRITE(DevFmtShort)
- HANDLE_WRITE(DevFmtUShort)
- HANDLE_WRITE(DevFmtInt)
- HANDLE_WRITE(DevFmtUInt)
- HANDLE_WRITE(DevFmtFloat)
- }
- #undef HANDLE_WRITE
- 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<DevFmtType_t<T>>(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::doDisconnect(std::string msg)
- {
- const auto mixLock = getWriteMixLock();
- if(Connected.exchange(false, std::memory_order_acq_rel))
- {
- AsyncEvent evt{std::in_place_type<AsyncDisconnectEvent>};
- auto &disconnect = std::get<AsyncDisconnectEvent>(evt);
- disconnect.msg = std::move(msg);
- for(ContextBase *ctx : *mContexts.load())
- {
- RingBuffer *ring{ctx->mAsyncEvents.get()};
- auto evt_data = ring->getWriteVector()[0];
- if(evt_data.len > 0)
- {
- al::construct_at(reinterpret_cast<AsyncEvent*>(evt_data.buf), evt);
- ring->writeAdvance(1);
- ctx->mEventSem.post();
- }
- if(!ctx->mStopVoicesOnDisconnect.load())
- {
- 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);
- }
- }
- }
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