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178 lines
6.2 KiB
178 lines
6.2 KiB
/**
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* OpenAL cross platform audio library
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* Copyright (C) 2013 by Mike Gorchak
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <algorithm>
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#include <array>
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#include <cstdlib>
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#include <iterator>
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#include "alc/effects/base.h"
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#include "almalloc.h"
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#include "alnumbers.h"
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#include "alnumeric.h"
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#include "alspan.h"
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#include "core/bufferline.h"
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#include "core/context.h"
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#include "core/devformat.h"
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#include "core/device.h"
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#include "core/effectslot.h"
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#include "core/filters/biquad.h"
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#include "core/mixer.h"
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#include "core/mixer/defs.h"
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#include "intrusive_ptr.h"
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namespace {
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struct DistortionState final : public EffectState {
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/* Effect gains for each channel */
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float mGain[MAX_OUTPUT_CHANNELS]{};
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/* Effect parameters */
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BiquadFilter mLowpass;
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BiquadFilter mBandpass;
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float mAttenuation{};
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float mEdgeCoeff{};
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float mBuffer[2][BufferLineSize]{};
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void deviceUpdate(const DeviceBase *device, const Buffer &buffer) override;
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void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
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const EffectTarget target) override;
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void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
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const al::span<FloatBufferLine> samplesOut) override;
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DEF_NEWDEL(DistortionState)
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};
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void DistortionState::deviceUpdate(const DeviceBase*, const Buffer&)
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{
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mLowpass.clear();
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mBandpass.clear();
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}
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void DistortionState::update(const ContextBase *context, const EffectSlot *slot,
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const EffectProps *props, const EffectTarget target)
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{
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const DeviceBase *device{context->mDevice};
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/* Store waveshaper edge settings. */
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const float edge{minf(std::sin(al::numbers::pi_v<float>*0.5f * props->Distortion.Edge),
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0.99f)};
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mEdgeCoeff = 2.0f * edge / (1.0f-edge);
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float cutoff{props->Distortion.LowpassCutoff};
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/* Bandwidth value is constant in octaves. */
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float bandwidth{(cutoff / 2.0f) / (cutoff * 0.67f)};
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/* Divide normalized frequency by the amount of oversampling done during
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* processing.
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*/
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auto frequency = static_cast<float>(device->Frequency);
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mLowpass.setParamsFromBandwidth(BiquadType::LowPass, cutoff/frequency/4.0f, 1.0f, bandwidth);
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cutoff = props->Distortion.EQCenter;
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/* Convert bandwidth in Hz to octaves. */
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bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f);
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mBandpass.setParamsFromBandwidth(BiquadType::BandPass, cutoff/frequency/4.0f, 1.0f, bandwidth);
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const auto coeffs = CalcDirectionCoeffs({0.0f, 0.0f, -1.0f}, 0.0f);
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mOutTarget = target.Main->Buffer;
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ComputePanGains(target.Main, coeffs.data(), slot->Gain*props->Distortion.Gain, mGain);
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}
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void DistortionState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
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{
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const float fc{mEdgeCoeff};
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for(size_t base{0u};base < samplesToDo;)
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{
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/* Perform 4x oversampling to avoid aliasing. Oversampling greatly
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* improves distortion quality and allows to implement lowpass and
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* bandpass filters using high frequencies, at which classic IIR
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* filters became unstable.
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*/
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size_t todo{minz(BufferLineSize, (samplesToDo-base) * 4)};
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/* Fill oversample buffer using zero stuffing. Multiply the sample by
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* the amount of oversampling to maintain the signal's power.
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*/
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for(size_t i{0u};i < todo;i++)
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mBuffer[0][i] = !(i&3) ? samplesIn[0][(i>>2)+base] * 4.0f : 0.0f;
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/* First step, do lowpass filtering of original signal. Additionally
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* perform buffer interpolation and lowpass cutoff for oversampling
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* (which is fortunately first step of distortion). So combine three
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* operations into the one.
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*/
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mLowpass.process({mBuffer[0], todo}, mBuffer[1]);
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/* Second step, do distortion using waveshaper function to emulate
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* signal processing during tube overdriving. Three steps of
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* waveshaping are intended to modify waveform without boost/clipping/
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* attenuation process.
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*/
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auto proc_sample = [fc](float smp) -> float
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{
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smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp));
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smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp)) * -1.0f;
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smp = (1.0f + fc) * smp/(1.0f + fc*std::abs(smp));
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return smp;
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};
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std::transform(std::begin(mBuffer[1]), std::begin(mBuffer[1])+todo, std::begin(mBuffer[0]),
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proc_sample);
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/* Third step, do bandpass filtering of distorted signal. */
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mBandpass.process({mBuffer[0], todo}, mBuffer[1]);
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todo >>= 2;
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const float *outgains{mGain};
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for(FloatBufferLine &output : samplesOut)
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{
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/* Fourth step, final, do attenuation and perform decimation,
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* storing only one sample out of four.
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*/
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const float gain{*(outgains++)};
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if(!(std::fabs(gain) > GainSilenceThreshold))
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continue;
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for(size_t i{0u};i < todo;i++)
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output[base+i] += gain * mBuffer[1][i*4];
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}
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base += todo;
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}
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}
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struct DistortionStateFactory final : public EffectStateFactory {
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al::intrusive_ptr<EffectState> create() override
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{ return al::intrusive_ptr<EffectState>{new DistortionState{}}; }
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};
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} // namespace
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EffectStateFactory *DistortionStateFactory_getFactory()
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{
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static DistortionStateFactory DistortionFactory{};
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return &DistortionFactory;
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}
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