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/**
* OpenAL cross platform audio library
* Copyright (C) 2013 by Mike Gorchak
* 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 <cstdlib>
#include <iterator>
#include "alc/effects/base.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/filters/biquad.h"
#include "core/mixer.h"
#include "core/mixer/defs.h"
#include "intrusive_ptr.h"
namespace {
struct DistortionState final : public EffectState {
/* Effect gains for each channel */
std::array<float,MaxAmbiChannels> mGain{};
/* Effect parameters */
BiquadFilter mLowpass;
BiquadFilter mBandpass;
float mAttenuation{};
float mEdgeCoeff{};
alignas(16) std::array<FloatBufferLine,2> mBuffer{};
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;
};
void DistortionState::deviceUpdate(const DeviceBase*, const BufferStorage*)
{
mLowpass.clear();
mBandpass.clear();
}
void DistortionState::update(const ContextBase *context, const EffectSlot *slot,
const EffectProps *props_, const EffectTarget target)
{
auto &props = std::get<DistortionProps>(*props_);
const DeviceBase *device{context->mDevice};
/* Store waveshaper edge settings. */
const float edge{minf(std::sin(al::numbers::pi_v<float>*0.5f * props.Edge),
0.99f)};
mEdgeCoeff = 2.0f * edge / (1.0f-edge);
float cutoff{props.LowpassCutoff};
/* Bandwidth value is constant in octaves. */
float bandwidth{(cutoff / 2.0f) / (cutoff * 0.67f)};
/* Divide normalized frequency by the amount of oversampling done during
* processing.
*/
auto frequency = static_cast<float>(device->Frequency);
mLowpass.setParamsFromBandwidth(BiquadType::LowPass, cutoff/frequency/4.0f, 1.0f, bandwidth);
cutoff = props.EQCenter;
/* Convert bandwidth in Hz to octaves. */
bandwidth = props.EQBandwidth / (cutoff * 0.67f);
mBandpass.setParamsFromBandwidth(BiquadType::BandPass, cutoff/frequency/4.0f, 1.0f, bandwidth);
static constexpr auto coeffs = CalcDirectionCoeffs(std::array{0.0f, 0.0f, -1.0f});
mOutTarget = target.Main->Buffer;
ComputePanGains(target.Main, coeffs, slot->Gain*props.Gain, mGain);
}
void DistortionState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
const float fc{mEdgeCoeff};
for(size_t base{0u};base < samplesToDo;)
{
/* Perform 4x oversampling to avoid aliasing. Oversampling greatly
* improves distortion quality and allows to implement lowpass and
* bandpass filters using high frequencies, at which classic IIR
* filters became unstable.
*/
size_t todo{minz(BufferLineSize, (samplesToDo-base) * 4)};
/* Fill oversample buffer using zero stuffing. Multiply the sample by
* the amount of oversampling to maintain the signal's power.
*/
for(size_t i{0u};i < todo;i++)
mBuffer[0][i] = !(i&3) ? samplesIn[0][(i>>2)+base] * 4.0f : 0.0f;
/* First step, do lowpass filtering of original signal. Additionally
* perform buffer interpolation and lowpass cutoff for oversampling
* (which is fortunately first step of distortion). So combine three
* operations into the one.
*/
mLowpass.process({mBuffer[0].data(), todo}, mBuffer[1].data());
/* Second step, do distortion using waveshaper function to emulate
* signal processing during tube overdriving. Three steps of
* waveshaping are intended to modify waveform without boost/clipping/
* attenuation process.
*/
auto proc_sample = [fc](float smp) -> float
{
smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp));
smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp)) * -1.0f;
smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp));
return smp;
};
std::transform(mBuffer[1].begin(), mBuffer[1].begin()+todo, mBuffer[0].begin(),
proc_sample);
/* Third step, do bandpass filtering of distorted signal. */
mBandpass.process({mBuffer[0].data(), todo}, mBuffer[1].data());
todo >>= 2;
const float *outgains{mGain.data()};
for(FloatBufferLine &RESTRICT output : samplesOut)
{
/* Fourth step, final, do attenuation and perform decimation,
* storing only one sample out of four.
*/
const float gain{*(outgains++)};
if(!(std::fabs(gain) > GainSilenceThreshold))
continue;
for(size_t i{0u};i < todo;i++)
output[base+i] += gain * mBuffer[1][i*4];
}
base += todo;
}
}
struct DistortionStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new DistortionState{}}; }
};
} // namespace
EffectStateFactory *DistortionStateFactory_getFactory()
{
static DistortionStateFactory DistortionFactory{};
return &DistortionFactory;
}
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