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/**
* OpenAL cross platform audio library
* Copyright (C) 2018 by Raul Herraiz.
* 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 "alcmain.h"
#include "alcontext.h"
#include "core/filters/biquad.h"
#include "effectslot.h"
#include "vecmat.h"
namespace {
constexpr float GainScale{31621.0f};
constexpr float MinFreq{20.0f};
constexpr float MaxFreq{2500.0f};
constexpr float QFactor{5.0f};
struct AutowahState final : public EffectState {
/* Effect parameters */
float mAttackRate;
float mReleaseRate;
float mResonanceGain;
float mPeakGain;
float mFreqMinNorm;
float mBandwidthNorm;
float mEnvDelay;
/* Filter components derived from the envelope. */
struct {
float cos_w0;
float alpha;
} mEnv[BufferLineSize];
struct {
/* Effect filters' history. */
struct {
float z1, z2;
} Filter;
/* Effect gains for each output channel */
float CurrentGains[MAX_OUTPUT_CHANNELS];
float TargetGains[MAX_OUTPUT_CHANNELS];
} mChans[MaxAmbiChannels];
/* Effects buffers */
alignas(16) float mBufferOut[BufferLineSize];
void deviceUpdate(const DeviceBase *device, const Buffer &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;
DEF_NEWDEL(AutowahState)
};
void AutowahState::deviceUpdate(const DeviceBase*, const Buffer&)
{
/* (Re-)initializing parameters and clear the buffers. */
mAttackRate = 1.0f;
mReleaseRate = 1.0f;
mResonanceGain = 10.0f;
mPeakGain = 4.5f;
mFreqMinNorm = 4.5e-4f;
mBandwidthNorm = 0.05f;
mEnvDelay = 0.0f;
for(auto &e : mEnv)
{
e.cos_w0 = 0.0f;
e.alpha = 0.0f;
}
for(auto &chan : mChans)
{
std::fill(std::begin(chan.CurrentGains), std::end(chan.CurrentGains), 0.0f);
chan.Filter.z1 = 0.0f;
chan.Filter.z2 = 0.0f;
}
}
void AutowahState::update(const ContextBase *context, const EffectSlot *slot,
const EffectProps *props, const EffectTarget target)
{
const DeviceBase *device{context->mDevice};
const auto frequency = static_cast<float>(device->Frequency);
const float ReleaseTime{clampf(props->Autowah.ReleaseTime, 0.001f, 1.0f)};
mAttackRate = std::exp(-1.0f / (props->Autowah.AttackTime*frequency));
mReleaseRate = std::exp(-1.0f / (ReleaseTime*frequency));
/* 0-20dB Resonance Peak gain */
mResonanceGain = std::sqrt(std::log10(props->Autowah.Resonance)*10.0f / 3.0f);
mPeakGain = 1.0f - std::log10(props->Autowah.PeakGain / GainScale);
mFreqMinNorm = MinFreq / frequency;
mBandwidthNorm = (MaxFreq-MinFreq) / frequency;
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 AutowahState::process(const size_t samplesToDo,
const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
const float attack_rate{mAttackRate};
const float release_rate{mReleaseRate};
const float res_gain{mResonanceGain};
const float peak_gain{mPeakGain};
const float freq_min{mFreqMinNorm};
const float bandwidth{mBandwidthNorm};
float env_delay{mEnvDelay};
for(size_t i{0u};i < samplesToDo;i++)
{
float w0, sample, a;
/* Envelope follower described on the book: Audio Effects, Theory,
* Implementation and Application.
*/
sample = peak_gain * std::fabs(samplesIn[0][i]);
a = (sample > env_delay) ? attack_rate : release_rate;
env_delay = lerp(sample, env_delay, a);
/* Calculate the cos and alpha components for this sample's filter. */
w0 = minf((bandwidth*env_delay + freq_min), 0.46f) * al::MathDefs<float>::Tau();
mEnv[i].cos_w0 = std::cos(w0);
mEnv[i].alpha = std::sin(w0)/(2.0f * QFactor);
}
mEnvDelay = env_delay;
auto chandata = std::addressof(mChans[0]);
for(const auto &insamples : samplesIn)
{
/* This effectively inlines BiquadFilter_setParams for a peaking
* filter and BiquadFilter_processC. The alpha and cosine components
* for the filter coefficients were previously calculated with the
* envelope. Because the filter changes for each sample, the
* coefficients are transient and don't need to be held.
*/
float z1{chandata->Filter.z1};
float z2{chandata->Filter.z2};
for(size_t i{0u};i < samplesToDo;i++)
{
const float alpha{mEnv[i].alpha};
const float cos_w0{mEnv[i].cos_w0};
float input, output;
float a[3], b[3];
b[0] = 1.0f + alpha*res_gain;
b[1] = -2.0f * cos_w0;
b[2] = 1.0f - alpha*res_gain;
a[0] = 1.0f + alpha/res_gain;
a[1] = -2.0f * cos_w0;
a[2] = 1.0f - alpha/res_gain;
input = insamples[i];
output = input*(b[0]/a[0]) + z1;
z1 = input*(b[1]/a[0]) - output*(a[1]/a[0]) + z2;
z2 = input*(b[2]/a[0]) - output*(a[2]/a[0]);
mBufferOut[i] = output;
}
chandata->Filter.z1 = z1;
chandata->Filter.z2 = z2;
/* Now, mix the processed sound data to the output. */
MixSamples({mBufferOut, samplesToDo}, samplesOut, chandata->CurrentGains,
chandata->TargetGains, samplesToDo, 0);
++chandata;
}
}
struct AutowahStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new AutowahState{}}; }
};
} // namespace
EffectStateFactory *AutowahStateFactory_getFactory()
{
static AutowahStateFactory AutowahFactory{};
return &AutowahFactory;
}
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