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externals/openal-soft/alc/effects/fshifter.cpp

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+/**
+ * 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 <algorithm>
+#include <array>
+#include <cmath>
+#include <complex>
+#include <cstdlib>
+#include <iterator>
+
+#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<double>;
+
+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<double,HilSize> 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<double>(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<uint,2> mPhaseStep{};
+    std::array<uint,2> mPhase{};
+    std::array<double,2> mSign{};
+
+    /* Effects buffers */
+    std::array<double,HilSize> mInFIFO{};
+    std::array<complex_d,HilStep> mOutFIFO{};
+    std::array<complex_d,HilSize> mOutputAccum{};
+    std::array<complex_d,HilSize> mAnalytic{};
+    std::array<complex_d,BufferLineSize> 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 BufferStorage *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<const FloatBufferLine> samplesIn,
+        const al::span<FloatBufferLine> samplesOut) override;
+
+    DEF_NEWDEL(FshifterState)
+};
+
+void FshifterState::deviceUpdate(const DeviceBase*, const BufferStorage*)
+{
+    /* (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<float>(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<float>(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<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> 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<float>(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<EffectState> create() override
+    { return al::intrusive_ptr<EffectState>{new FshifterState{}}; }
+};
+
+} // namespace
+
+EffectStateFactory *FshifterStateFactory_getFactory()
+{
+    static FshifterStateFactory FshifterFactory{};
+    return &FshifterFactory;
+}