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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 <cmath>
#include <cstdlib>
#include <array>
#include <complex>
#include <algorithm>
#include "alcmain.h"
#include "alcomplex.h"
#include "alcontext.h"
#include "alu.h"
#include "effectslot.h"
#include "math_defs.h"
namespace {
using complex_d = std::complex<double>;
#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<double,HIL_SIZE> InitHannWindow()
{
std::array<double,HIL_SIZE> 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<double>::Pi() / double{HIL_SIZE}};
const double val{std::sin(static_cast<double>(i+1) * scale)};
ret[i] = ret[HIL_SIZE-1-i] = val * val;
}
return ret;
}
alignas(16) const std::array<double,HIL_SIZE> HannWindow = InitHannWindow();
struct FshifterState final : public EffectState {
/* Effect parameters */
size_t mCount{};
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 ALCdevice *device, const BufferStorage *buffer) override;
void update(const ALCcontext *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 ALCdevice*, const BufferStorage*)
{
/* (Re-)initializing parameters and clear the buffers. */
mCount = 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 ALCcontext *context, const EffectSlot *slot,
const EffectProps *props, const EffectTarget target)
{
const ALCdevice *device{context->mDevice.get()};
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;
}
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<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
for(size_t base{0u};base < samplesToDo;)
{
size_t todo{minz(HIL_SIZE-mCount, samplesToDo-base)};
/* Fill FIFO buffer with samples data */
size_t count{mCount};
do {
mInFIFO[count] = samplesIn[0][base];
mOutdata[base] = mOutFIFO[count-FIFO_LATENCY];
++base; ++count;
} while(--todo);
mCount = count;
/* Check whether FIFO buffer is filled */
if(mCount < HIL_SIZE) break;
mCount = FIFO_LATENCY;
/* Real signal windowing and store in Analytic buffer */
for(size_t k{0};k < HIL_SIZE;k++)
mAnalytic[k] = mInFIFO[k]*HannWindow[k];
/* Processing signal by Discrete Hilbert Transform (analytical signal). */
complex_hilbert(mAnalytic);
/* Windowing and add to output accumulator */
for(size_t k{0};k < HIL_SIZE;k++)
mOutputAccum[k] += 2.0/OVERSAMP*HannWindow[k]*mAnalytic[k];
/* Shift accumulator, input & output FIFO */
std::copy_n(mOutputAccum, HIL_STEP, mOutFIFO);
auto accum_iter = std::copy(std::begin(mOutputAccum)+HIL_STEP, std::end(mOutputAccum),
std::begin(mOutputAccum));
std::fill(accum_iter, std::end(mOutputAccum), complex_d{});
std::copy(std::begin(mInFIFO)+HIL_STEP, std::end(mInFIFO), std::begin(mInFIFO));
}
/* 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<double>::Tau())};
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 {
EffectState *create() override { return new FshifterState{}; }
};
} // namespace
EffectStateFactory *FshifterStateFactory_getFactory()
{
static FshifterStateFactory FshifterFactory{};
return &FshifterFactory;
}
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