openal-soft/Alc/effects/distortion.cpp

/**
 * 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 <cmath>
#include <cstdlib>

#include <cmath>

#include "alMain.h"
#include "alcontext.h"
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/biquad.h"


struct ALdistortionState final : public EffectState {
    /* Effect gains for each channel */
    ALfloat mGain[MAX_OUTPUT_CHANNELS]{};

    /* Effect parameters */
    BiquadFilter mLowpass;
    BiquadFilter mBandpass;
    ALfloat mAttenuation{};
    ALfloat mEdgeCoeff{};

    ALfloat mBuffer[2][BUFFERSIZE]{};


    ALboolean deviceUpdate(const ALCdevice *device) override;
    void update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target) override;
    void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], ALsizei numChannels) override;

    DEF_NEWDEL(ALdistortionState)
};

ALboolean ALdistortionState::deviceUpdate(const ALCdevice *UNUSED(device))
{
    mLowpass.clear();
    mBandpass.clear();
    return AL_TRUE;
}

void ALdistortionState::update(const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props, const EffectTarget target)
{
    const ALCdevice *device{context->Device};

    /* Store waveshaper edge settings. */
    const ALfloat edge{
        minf(std::sin(al::MathDefs<float>::Pi()*0.5f * props->Distortion.Edge), 0.99f)};
    mEdgeCoeff = 2.0f * edge / (1.0f-edge);

    ALfloat cutoff{props->Distortion.LowpassCutoff};
    /* Bandwidth value is constant in octaves. */
    ALfloat bandwidth{(cutoff / 2.0f) / (cutoff * 0.67f)};
    /* Multiply sampling frequency by the amount of oversampling done during
     * processing.
     */
    auto frequency = static_cast<ALfloat>(device->Frequency);
    mLowpass.setParams(BiquadType::LowPass, 1.0f, cutoff / (frequency*4.0f),
        calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
    );

    cutoff = props->Distortion.EQCenter;
    /* Convert bandwidth in Hz to octaves. */
    bandwidth = props->Distortion.EQBandwidth / (cutoff * 0.67f);
    mBandpass.setParams(BiquadType::BandPass, 1.0f, cutoff / (frequency*4.0f),
        calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
    );

    ALfloat coeffs[MAX_AMBI_CHANNELS];
    CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);

    mOutBuffer = target.Main->Buffer;
    mOutChannels = target.Main->NumChannels;
    ComputePanGains(target.Main, coeffs, slot->Params.Gain*props->Distortion.Gain, mGain);
}

void ALdistortionState::process(ALsizei SamplesToDo, const ALfloat (*RESTRICT SamplesIn)[BUFFERSIZE], ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
    ALfloat (*RESTRICT buffer)[BUFFERSIZE] = mBuffer;
    const ALfloat fc = mEdgeCoeff;
    ALsizei base;
    ALsizei i, k;

    for(base = 0;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.
         */
        ALsizei todo = mini(BUFFERSIZE, (SamplesToDo-base) * 4);

        /* Fill oversample buffer using zero stuffing. Multiply the sample by
         * the amount of oversampling to maintain the signal's power.
         */
        for(i = 0;i < todo;i++)
            buffer[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(buffer[1], buffer[0], todo);

        /* 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.
         */
        for(i = 0;i < todo;i++)
        {
            ALfloat smp = buffer[1][i];

            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
            smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));

            buffer[0][i] = smp;
        }

        /* Third step, do bandpass filtering of distorted signal. */
        mBandpass.process(buffer[1], buffer[0], todo);

        todo >>= 2;
        for(k = 0;k < NumChannels;k++)
        {
            /* Fourth step, final, do attenuation and perform decimation,
             * storing only one sample out of four.
             */
            ALfloat gain = mGain[k];
            if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
                continue;

            for(i = 0;i < todo;i++)
                SamplesOut[k][base+i] += gain * buffer[1][i*4];
        }

        base += todo;
    }
}


struct DistortionStateFactory final : public EffectStateFactory {
    EffectState *create() override;
};

EffectState *DistortionStateFactory::create()
{ return new ALdistortionState{}; }

EffectStateFactory *DistortionStateFactory_getFactory()
{
    static DistortionStateFactory DistortionFactory{};
    return &DistortionFactory;
}


void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); }
void ALdistortion_setParamiv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); }
void ALdistortion_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
    ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_DISTORTION_EDGE:
            if(!(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE))
                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion edge out of range");
            props->Distortion.Edge = val;
            break;

        case AL_DISTORTION_GAIN:
            if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN))
                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion gain out of range");
            props->Distortion.Gain = val;
            break;

        case AL_DISTORTION_LOWPASS_CUTOFF:
            if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF))
                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion low-pass cutoff out of range");
            props->Distortion.LowpassCutoff = val;
            break;

        case AL_DISTORTION_EQCENTER:
            if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER))
                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ center out of range");
            props->Distortion.EQCenter = val;
            break;

        case AL_DISTORTION_EQBANDWIDTH:
            if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH))
                SETERR_RETURN(context, AL_INVALID_VALUE,, "Distortion EQ bandwidth out of range");
            props->Distortion.EQBandwidth = val;
            break;

        default:
            alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x",
                       param);
    }
}
void ALdistortion_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{ ALdistortion_setParamf(effect, context, param, vals[0]); }

void ALdistortion_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(val))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer property 0x%04x", param); }
void ALdistortion_getParamiv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALint *UNUSED(vals))
{ alSetError(context, AL_INVALID_ENUM, "Invalid distortion integer-vector property 0x%04x", param); }
void ALdistortion_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
    const ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_DISTORTION_EDGE:
            *val = props->Distortion.Edge;
            break;

        case AL_DISTORTION_GAIN:
            *val = props->Distortion.Gain;
            break;

        case AL_DISTORTION_LOWPASS_CUTOFF:
            *val = props->Distortion.LowpassCutoff;
            break;

        case AL_DISTORTION_EQCENTER:
            *val = props->Distortion.EQCenter;
            break;

        case AL_DISTORTION_EQBANDWIDTH:
            *val = props->Distortion.EQBandwidth;
            break;

        default:
            alSetError(context, AL_INVALID_ENUM, "Invalid distortion float property 0x%04x",
                       param);
    }
}
void ALdistortion_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{ ALdistortion_getParamf(effect, context, param, vals); }

DEFINE_ALEFFECT_VTABLE(ALdistortion);