alc/effects/distortion.cpp

   1 /**
   2  * OpenAL cross platform audio library
   3  * Copyright (C) 2013 by Mike Gorchak
   4  * This library is free software; you can redistribute it and/or
   5  *  modify it under the terms of the GNU Library General Public
   6  *  License as published by the Free Software Foundation; either
   7  *  version 2 of the License, or (at your option) any later version.
   8  *
   9  * This library is distributed in the hope that it will be useful,
  10  *  but WITHOUT ANY WARRANTY; without even the implied warranty of
  11  *  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  12  *  Library General Public License for more details.
  13  *
  14  * You should have received a copy of the GNU Library General Public
  15  *  License along with this library; if not, write to the
  16  *  Free Software Foundation, Inc.,
  17  *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
  18  * Or go to http://www.gnu.org/copyleft/lgpl.html
  19  */
  20
  21 #include "config.h"
  22
  23 #include <algorithm>
  24 #include <array>
  25 #include <cmath>
  26 #include <cstdlib>
  27 #include <variant>
  28
  29 #include "alc/effects/base.h"
  30 #include "alnumbers.h"
  31 #include "alnumeric.h"
  32 #include "alspan.h"
  33 #include "core/ambidefs.h"
  34 #include "core/bufferline.h"
  35 #include "core/context.h"
  36 #include "core/device.h"
  37 #include "core/effects/base.h"
  38 #include "core/effectslot.h"
  39 #include "core/filters/biquad.h"
  40 #include "core/mixer.h"
  41 #include "core/mixer/defs.h"
  42 #include "intrusive_ptr.h"
  43
  44 struct BufferStorage;
  45
  46 namespace {
  47
  48 struct DistortionState final : public EffectState {
  49     /* Effect gains for each channel */
  50     std::array<float,MaxAmbiChannels> mGain{};
  51
  52     /* Effect parameters */
  53     BiquadFilter mLowpass;
  54     BiquadFilter mBandpass;
  55     float mAttenuation{};
  56     float mEdgeCoeff{};
  57
  58     alignas(16) std::array<FloatBufferLine,2> mBuffer{};
  59
  60
  61     void deviceUpdate(const DeviceBase *device, const BufferStorage *buffer) override;
  62     void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
  63         const EffectTarget target) override;
  64     void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
  65         const al::span<FloatBufferLine> samplesOut) override;
  66 };
  67
  68 void DistortionState::deviceUpdate(const DeviceBase*, const BufferStorage*)
  69 {
  70     mLowpass.clear();
  71     mBandpass.clear();
  72 }
  73
  74 void DistortionState::update(const ContextBase *context, const EffectSlot *slot,
  75     const EffectProps *props_, const EffectTarget target)
  76 {
  77     auto &props = std::get<DistortionProps>(*props_);
  78     const DeviceBase *device{context->mDevice};
  79
  80     /* Store waveshaper edge settings. */
  81     const float edge{std::min(std::sin(al::numbers::pi_v<float>*0.5f * props.Edge), 0.99f)};
  82     mEdgeCoeff = 2.0f * edge / (1.0f-edge);
  83
  84     float cutoff{props.LowpassCutoff};
  85     /* Bandwidth value is constant in octaves. */
  86     float bandwidth{(cutoff / 2.0f) / (cutoff * 0.67f)};
  87     /* Divide normalized frequency by the amount of oversampling done during
  88      * processing.
  89      */
  90     auto frequency = static_cast<float>(device->Frequency);
  91     mLowpass.setParamsFromBandwidth(BiquadType::LowPass, cutoff/frequency/4.0f, 1.0f, bandwidth);
  92
  93     cutoff = props.EQCenter;
  94     /* Convert bandwidth in Hz to octaves. */
  95     bandwidth = props.EQBandwidth / (cutoff * 0.67f);
  96     mBandpass.setParamsFromBandwidth(BiquadType::BandPass, cutoff/frequency/4.0f, 1.0f, bandwidth);
  97
  98     static constexpr auto coeffs = CalcDirectionCoeffs(std::array{0.0f, 0.0f, -1.0f});
  99
 100     mOutTarget = target.Main->Buffer;
 101     ComputePanGains(target.Main, coeffs, slot->Gain*props.Gain, mGain);
 102 }
 103
 104 void DistortionState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
 105 {
 106     const float fc{mEdgeCoeff};
 107     for(size_t base{0u};base < samplesToDo;)
 108     {
 109         /* Perform 4x oversampling to avoid aliasing. Oversampling greatly
 110          * improves distortion quality and allows to implement lowpass and
 111          * bandpass filters using high frequencies, at which classic IIR
 112          * filters became unstable.
 113          */
 114         size_t todo{std::min(BufferLineSize, (samplesToDo-base) * 4_uz)};
 115
 116         /* Fill oversample buffer using zero stuffing. Multiply the sample by
 117          * the amount of oversampling to maintain the signal's power.
 118          */
 119         for(size_t i{0u};i < todo;i++)
 120             mBuffer[0][i] = !(i&3) ? samplesIn[0][(i>>2)+base] * 4.0f : 0.0f;
 121
 122         /* First step, do lowpass filtering of original signal. Additionally
 123          * perform buffer interpolation and lowpass cutoff for oversampling
 124          * (which is fortunately first step of distortion). So combine three
 125          * operations into the one.
 126          */
 127         mLowpass.process({mBuffer[0].data(), todo}, mBuffer[1]);
 128
 129         /* Second step, do distortion using waveshaper function to emulate
 130          * signal processing during tube overdriving. Three steps of
 131          * waveshaping are intended to modify waveform without boost/clipping/
 132          * attenuation process.
 133          */
 134         auto proc_sample = [fc](float smp) -> float
 135         {
 136             smp = (1.0f + fc) * smp/(1.0f + fc*std::fabs(smp));
 137             smp = (1.0f + fc) * smp/(1.0f + fc*std::fabs(smp)) * -1.0f;
 138             smp = (1.0f + fc) * smp/(1.0f + fc*std::fabs(smp));
 139             return smp;
 140         };
 141         std::transform(mBuffer[1].begin(), mBuffer[1].begin()+todo, mBuffer[0].begin(),
 142             proc_sample);
 143
 144         /* Third step, do bandpass filtering of distorted signal. */
 145         mBandpass.process({mBuffer[0].data(), todo}, mBuffer[1]);
 146
 147         todo >>= 2;
 148         auto outgains = mGain.cbegin();
 149         for(FloatBufferLine &RESTRICT output : samplesOut)
 150         {
 151             /* Fourth step, final, do attenuation and perform decimation,
 152              * storing only one sample out of four.
 153              */
 154             const float gain{*(outgains++)};
 155             if(!(std::fabs(gain) > GainSilenceThreshold))
 156                 continue;
 157
 158             for(size_t i{0u};i < todo;i++)
 159                 output[base+i] += gain * mBuffer[1][i*4];
 160         }
 161
 162         base += todo;
 163     }
 164 }
 165
 166
 167 struct DistortionStateFactory final : public EffectStateFactory {
 168     al::intrusive_ptr<EffectState> create() override
 169     { return al::intrusive_ptr<EffectState>{new DistortionState{}}; }
 170 };
 171
 172 } // namespace
 173
 174 EffectStateFactory *DistortionStateFactory_getFactory()
 175 {
 176     static DistortionStateFactory DistortionFactory{};
 177     return &DistortionFactory;
 178 }