/** * 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 "alc/effectslot.h" #include "alcomplex.h" #include "almalloc.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/mixer.h" #include "core/mixer/defs.h" #include "intrusive_ptr.h" #include "math_defs.h" namespace { using uint = unsigned int; using complex_d = std::complex; #define HIL_SIZE 1024 #define OVERSAMP (1<<2) #define HIL_STEP (HIL_SIZE / OVERSAMP) #define FIFO_LATENCY (HIL_STEP * (OVERSAMP-1)) /* Define a Hann window, used to filter the HIL input and output. */ std::array InitHannWindow() { std::array ret; /* Create lookup table of the Hann window for the desired size, i.e. HIL_SIZE */ for(size_t i{0};i < HIL_SIZE>>1;i++) { constexpr double scale{al::MathDefs::Pi() / double{HIL_SIZE}}; const double val{std::sin(static_cast(i+1) * scale)}; ret[i] = ret[HIL_SIZE-1-i] = val * val; } return ret; } alignas(16) const std::array HannWindow = InitHannWindow(); struct FshifterState final : public EffectState { /* Effect parameters */ size_t mCount{}; size_t mPos{}; uint mPhaseStep[2]{}; uint mPhase[2]{}; double mSign[2]{}; /* Effects buffers */ double mInFIFO[HIL_SIZE]{}; complex_d mOutFIFO[HIL_STEP]{}; complex_d mOutputAccum[HIL_SIZE]{}; complex_d mAnalytic[HIL_SIZE]{}; complex_d mOutdata[BufferLineSize]{}; alignas(16) float mBufferOut[BufferLineSize]{}; /* Effect gains for each output channel */ struct { float Current[MAX_OUTPUT_CHANNELS]{}; float Target[MAX_OUTPUT_CHANNELS]{}; } 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 = FIFO_LATENCY; std::fill(std::begin(mPhaseStep), std::end(mPhaseStep), 0u); std::fill(std::begin(mPhase), std::end(mPhase), 0u); std::fill(std::begin(mSign), std::end(mSign), 1.0); std::fill(std::begin(mInFIFO), std::end(mInFIFO), 0.0); std::fill(std::begin(mOutFIFO), std::end(mOutFIFO), complex_d{}); std::fill(std::begin(mOutputAccum), std::end(mOutputAccum), complex_d{}); std::fill(std::begin(mAnalytic), std::end(mAnalytic), 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; } const auto lcoeffs = CalcDirectionCoeffs({-1.0f, 0.0f, 0.0f}, 0.0f); const auto rcoeffs = CalcDirectionCoeffs({ 1.0f, 0.0f, 0.0f}, 0.0f); 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(HIL_STEP-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 < HIL_STEP) break; mCount = 0; mPos = (mPos+HIL_STEP) & (HIL_SIZE-1); /* Real signal windowing and store in Analytic buffer */ for(size_t src{mPos}, k{0u};src < HIL_SIZE;++src,++k) mAnalytic[k] = mInFIFO[src]*HannWindow[k]; for(size_t src{0u}, k{HIL_SIZE-mPos};src < mPos;++src,++k) mAnalytic[k] = mInFIFO[src]*HannWindow[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 < HIL_SIZE;++dst,++k) mOutputAccum[dst] += 2.0/OVERSAMP*HannWindow[k]*mAnalytic[k]; for(size_t dst{0u}, k{HIL_SIZE-mPos};dst < mPos;++dst,++k) mOutputAccum[dst] += 2.0/OVERSAMP*HannWindow[k]*mAnalytic[k]; /* Copy out the accumulated result, then clear for the next iteration. */ std::copy_n(mOutputAccum + mPos, HIL_STEP, mOutFIFO); std::fill_n(mOutputAccum + mPos, HIL_STEP, complex_d{}); } /* Process frequency shifter using the analytic signal obtained. */ float *RESTRICT BufferOut{mBufferOut}; for(int 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 * ((1.0/MixerFracOne) * al::MathDefs::Tau())}; 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; }