/**
* 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 <algorithm>
#include <functional>
#include "alcmain.h"
#include "alcontext.h"
#include "core/filters/biquad.h"
#include "effectslot.h"
#include "vecmat.h"
namespace {
/* The document "Effects Extension Guide.pdf" says that low and high *
* frequencies are cutoff frequencies. This is not fully correct, they *
* are corner frequencies for low and high shelf filters. If they were *
* just cutoff frequencies, there would be no need in cutoff frequency *
* gains, which are present. Documentation for "Creative Proteus X2" *
* software describes 4-band equalizer functionality in a much better *
* way. This equalizer seems to be a predecessor of OpenAL 4-band *
* equalizer. With low and high shelf filters we are able to cutoff *
* frequencies below and/or above corner frequencies using attenuation *
* gains (below 1.0) and amplify all low and/or high frequencies using *
* gains above 1.0. *
* *
* Low-shelf Low Mid Band High Mid Band High-shelf *
* corner center center corner *
* frequency frequency frequency frequency *
* 50Hz..800Hz 200Hz..3000Hz 1000Hz..8000Hz 4000Hz..16000Hz *
* *
* | | | | *
* | | | | *
* B -----+ /--+--\ /--+--\ +----- *
* O |\ | | | | | | /| *
* O | \ - | - - | - / | *
* S + | \ | | | | | | / | *
* T | | | | | | | | | | *
* ---------+---------------+------------------+---------------+-------- *
* C | | | | | | | | | | *
* U - | / | | | | | | \ | *
* T | / - | - - | - \ | *
* O |/ | | | | | | \| *
* F -----+ \--+--/ \--+--/ +----- *
* F | | | | *
* | | | | *
* *
* Gains vary from 0.126 up to 7.943, which means from -18dB attenuation *
* up to +18dB amplification. Band width varies from 0.01 up to 1.0 in *
* octaves for two mid bands. *
* *
* Implementation is based on the "Cookbook formulae for audio EQ biquad *
* filter coefficients" by Robert Bristow-Johnson *
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
struct EqualizerState final : public EffectState {
struct {
/* Effect parameters */
BiquadFilter filter[4];
/* Effect gains for each channel */
float CurrentGains[MAX_OUTPUT_CHANNELS]{};
float TargetGains[MAX_OUTPUT_CHANNELS]{};
} mChans[MaxAmbiChannels];
FloatBufferLine mSampleBuffer{};
void deviceUpdate(const ALCdevice *device) 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(EqualizerState)
};
void EqualizerState::deviceUpdate(const ALCdevice*)
{
for(auto &e : mChans)
{
std::for_each(std::begin(e.filter), std::end(e.filter), std::mem_fn(&BiquadFilter::clear));
std::fill(std::begin(e.CurrentGains), std::end(e.CurrentGains), 0.0f);
}
}
void EqualizerState::update(const ALCcontext *context, const EffectSlot *slot,
const EffectProps *props, const EffectTarget target)
{
const ALCdevice *device{context->mDevice.get()};
auto frequency = static_cast<float>(device->Frequency);
float gain, f0norm;
/* Calculate coefficients for the each type of filter. Note that the shelf
* and peaking filters' gain is for the centerpoint of the transition band,
* while the effect property gains are for the shelf/peak itself. So the
* property gains need their dB halved (sqrt of linear gain) for the
* shelf/peak to reach the provided gain.
*/
gain = std::sqrt(props->Equalizer.LowGain);
f0norm = props->Equalizer.LowCutoff / frequency;
mChans[0].filter[0].setParamsFromSlope(BiquadType::LowShelf, f0norm, gain, 0.75f);
gain = std::sqrt(props->Equalizer.Mid1Gain);
f0norm = props->Equalizer.Mid1Center / frequency;
mChans[0].filter[1].setParamsFromBandwidth(BiquadType::Peaking, f0norm, gain,
props->Equalizer.Mid1Width);
gain = std::sqrt(props->Equalizer.Mid2Gain);
f0norm = props->Equalizer.Mid2Center / frequency;
mChans[0].filter[2].setParamsFromBandwidth(BiquadType::Peaking, f0norm, gain,
props->Equalizer.Mid2Width);
gain = std::sqrt(props->Equalizer.HighGain);
f0norm = props->Equalizer.HighCutoff / frequency;
mChans[0].filter[3].setParamsFromSlope(BiquadType::HighShelf, f0norm, gain, 0.75f);
/* Copy the filter coefficients for the other input channels. */
for(size_t i{1u};i < slot->Wet.Buffer.size();++i)
{
mChans[i].filter[0].copyParamsFrom(mChans[0].filter[0]);
mChans[i].filter[1].copyParamsFrom(mChans[0].filter[1]);
mChans[i].filter[2].copyParamsFrom(mChans[0].filter[2]);
mChans[i].filter[3].copyParamsFrom(mChans[0].filter[3]);
}
mOutTarget = target.Main->Buffer;
auto set_gains = [slot,target](auto &chan, al::span<const float,MaxAmbiChannels> coeffs)
{ ComputePanGains(target.Main, coeffs.data(), slot->Gain, chan.TargetGains); };
SetAmbiPanIdentity(std::begin(mChans), slot->Wet.Buffer.size(), set_gains);
}
void EqualizerState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
const al::span<float> buffer{mSampleBuffer.data(), samplesToDo};
auto chan = std::addressof(mChans[0]);
for(const auto &input : samplesIn)
{
const al::span<const float> inbuf{input.data(), samplesToDo};
DualBiquad{chan->filter[0], chan->filter[1]}.process(inbuf, buffer.begin());
DualBiquad{chan->filter[2], chan->filter[3]}.process(buffer, buffer.begin());
MixSamples(buffer, samplesOut, chan->CurrentGains, chan->TargetGains, samplesToDo, 0u);
++chan;
}
}
struct EqualizerStateFactory final : public EffectStateFactory {
EffectState *create() override { return new EqualizerState{}; }
};
} // namespace
EffectStateFactory *EqualizerStateFactory_getFactory()
{
static EqualizerStateFactory EqualizerFactory{};
return &EqualizerFactory;
}