/** * OpenAL cross platform audio library * Copyright (C) 2018 by Raul Herraiz. * 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 #include #include #include #include #include #include "alc/effects/base.h" #include "alcomplex.h" #include "almalloc.h" #include "alnumbers.h" #include "alnumeric.h" #include "alspan.h" #include "core/bufferline.h" #include "core/context.h" #include "core/devformat.h" #include "core/device.h" #include "core/effectslot.h" #include "core/mixer.h" #include "core/mixer/defs.h" #include "intrusive_ptr.h" namespace { using uint = unsigned int; using complex_d = std::complex; constexpr size_t HilSize{1024}; constexpr size_t HilHalfSize{HilSize >> 1}; constexpr size_t OversampleFactor{4}; static_assert(HilSize%OversampleFactor == 0, "Factor must be a clean divisor of the size"); constexpr size_t HilStep{HilSize / OversampleFactor}; /* Define a Hann window, used to filter the HIL input and output. */ struct Windower { alignas(16) std::array mData; Windower() { /* Create lookup table of the Hann window for the desired size. */ for(size_t i{0};i < HilHalfSize;i++) { constexpr double scale{al::numbers::pi / double{HilSize}}; const double val{std::sin((static_cast(i)+0.5) * scale)}; mData[i] = mData[HilSize-1-i] = val * val; } } }; const Windower gWindow{}; struct FshifterState final : public EffectState { /* Effect parameters */ size_t mCount{}; size_t mPos{}; std::array mPhaseStep{}; std::array mPhase{}; std::array mSign{}; /* Effects buffers */ std::array mInFIFO{}; std::array mOutFIFO{}; std::array mOutputAccum{}; std::array mAnalytic{}; std::array mOutdata{}; alignas(16) FloatBufferLine mBufferOut{}; /* Effect gains for each output channel */ struct { float Current[MaxAmbiChannels]{}; float Target[MaxAmbiChannels]{}; } mGains[2]; void deviceUpdate(const DeviceBase *device, const Buffer &buffer) override; void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props, const EffectTarget target) override; void process(const size_t samplesToDo, const al::span samplesIn, const al::span samplesOut) override; DEF_NEWDEL(FshifterState) }; void FshifterState::deviceUpdate(const DeviceBase*, const Buffer&) { /* (Re-)initializing parameters and clear the buffers. */ mCount = 0; mPos = HilSize - HilStep; mPhaseStep.fill(0u); mPhase.fill(0u); mSign.fill(1.0); mInFIFO.fill(0.0); mOutFIFO.fill(complex_d{}); mOutputAccum.fill(complex_d{}); mAnalytic.fill(complex_d{}); for(auto &gain : mGains) { std::fill(std::begin(gain.Current), std::end(gain.Current), 0.0f); std::fill(std::begin(gain.Target), std::end(gain.Target), 0.0f); } } void FshifterState::update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props, const EffectTarget target) { const DeviceBase *device{context->mDevice}; const float step{props->Fshifter.Frequency / static_cast(device->Frequency)}; mPhaseStep[0] = mPhaseStep[1] = fastf2u(minf(step, 1.0f) * MixerFracOne); switch(props->Fshifter.LeftDirection) { case FShifterDirection::Down: mSign[0] = -1.0; break; case FShifterDirection::Up: mSign[0] = 1.0; break; case FShifterDirection::Off: mPhase[0] = 0; mPhaseStep[0] = 0; break; } switch(props->Fshifter.RightDirection) { case FShifterDirection::Down: mSign[1] = -1.0; break; case FShifterDirection::Up: mSign[1] = 1.0; break; case FShifterDirection::Off: mPhase[1] = 0; mPhaseStep[1] = 0; break; } static constexpr auto inv_sqrt2 = static_cast(1.0 / al::numbers::sqrt2); static constexpr auto lcoeffs_pw = CalcDirectionCoeffs({-1.0f, 0.0f, 0.0f}); static constexpr auto rcoeffs_pw = CalcDirectionCoeffs({ 1.0f, 0.0f, 0.0f}); static constexpr auto lcoeffs_nrml = CalcDirectionCoeffs({-inv_sqrt2, 0.0f, inv_sqrt2}); static constexpr auto rcoeffs_nrml = CalcDirectionCoeffs({ inv_sqrt2, 0.0f, inv_sqrt2}); auto &lcoeffs = (device->mRenderMode != RenderMode::Pairwise) ? lcoeffs_nrml : lcoeffs_pw; auto &rcoeffs = (device->mRenderMode != RenderMode::Pairwise) ? rcoeffs_nrml : rcoeffs_pw; mOutTarget = target.Main->Buffer; ComputePanGains(target.Main, lcoeffs.data(), slot->Gain, mGains[0].Target); ComputePanGains(target.Main, rcoeffs.data(), slot->Gain, mGains[1].Target); } void FshifterState::process(const size_t samplesToDo, const al::span samplesIn, const al::span samplesOut) { for(size_t base{0u};base < samplesToDo;) { size_t todo{minz(HilStep-mCount, samplesToDo-base)}; /* Fill FIFO buffer with samples data */ const size_t pos{mPos}; size_t count{mCount}; do { mInFIFO[pos+count] = samplesIn[0][base]; mOutdata[base] = mOutFIFO[count]; ++base; ++count; } while(--todo); mCount = count; /* Check whether FIFO buffer is filled */ if(mCount < HilStep) break; mCount = 0; mPos = (mPos+HilStep) & (HilSize-1); /* Real signal windowing and store in Analytic buffer */ for(size_t src{mPos}, k{0u};src < HilSize;++src,++k) mAnalytic[k] = mInFIFO[src]*gWindow.mData[k]; for(size_t src{0u}, k{HilSize-mPos};src < mPos;++src,++k) mAnalytic[k] = mInFIFO[src]*gWindow.mData[k]; /* Processing signal by Discrete Hilbert Transform (analytical signal). */ complex_hilbert(mAnalytic); /* Windowing and add to output accumulator */ for(size_t dst{mPos}, k{0u};dst < HilSize;++dst,++k) mOutputAccum[dst] += 2.0/OversampleFactor*gWindow.mData[k]*mAnalytic[k]; for(size_t dst{0u}, k{HilSize-mPos};dst < mPos;++dst,++k) mOutputAccum[dst] += 2.0/OversampleFactor*gWindow.mData[k]*mAnalytic[k]; /* Copy out the accumulated result, then clear for the next iteration. */ std::copy_n(mOutputAccum.cbegin() + mPos, HilStep, mOutFIFO.begin()); std::fill_n(mOutputAccum.begin() + mPos, HilStep, complex_d{}); } /* Process frequency shifter using the analytic signal obtained. */ float *RESTRICT BufferOut{al::assume_aligned<16>(mBufferOut.data())}; for(size_t c{0};c < 2;++c) { const uint phase_step{mPhaseStep[c]}; uint phase_idx{mPhase[c]}; for(size_t k{0};k < samplesToDo;++k) { const double phase{phase_idx * (al::numbers::pi*2.0 / MixerFracOne)}; BufferOut[k] = static_cast(mOutdata[k].real()*std::cos(phase) + mOutdata[k].imag()*std::sin(phase)*mSign[c]); phase_idx += phase_step; phase_idx &= MixerFracMask; } mPhase[c] = phase_idx; /* Now, mix the processed sound data to the output. */ MixSamples({BufferOut, samplesToDo}, samplesOut, mGains[c].Current, mGains[c].Target, maxz(samplesToDo, 512), 0); } } struct FshifterStateFactory final : public EffectStateFactory { al::intrusive_ptr create() override { return al::intrusive_ptr{new FshifterState{}}; } }; } // namespace EffectStateFactory *FshifterStateFactory_getFactory() { static FshifterStateFactory FshifterFactory{}; return &FshifterFactory; }