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axmol/external/openal/alc/effects/vmorpher.cpp

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/**
* OpenAL cross platform audio library
* Copyright (C) 2019 by Anis A. Hireche
* 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 "alu.h"
#include "effectslot.h"
#include "math_defs.h"
namespace {
#define MAX_UPDATE_SAMPLES 256
#define NUM_FORMANTS 4
#define NUM_FILTERS 2
#define Q_FACTOR 5.0f
#define VOWEL_A_INDEX 0
#define VOWEL_B_INDEX 1
#define WAVEFORM_FRACBITS 24
#define WAVEFORM_FRACONE (1<<WAVEFORM_FRACBITS)
#define WAVEFORM_FRACMASK (WAVEFORM_FRACONE-1)
inline float Sin(uint index)
{
constexpr float scale{al::MathDefs<float>::Tau() / WAVEFORM_FRACONE};
return std::sin(static_cast<float>(index) * scale)*0.5f + 0.5f;
}
inline float Saw(uint index)
{ return static_cast<float>(index) / float{WAVEFORM_FRACONE}; }
inline float Triangle(uint index)
{ return std::fabs(static_cast<float>(index)*(2.0f/WAVEFORM_FRACONE) - 1.0f); }
inline float Half(uint) { return 0.5f; }
template<float (&func)(uint)>
void Oscillate(float *RESTRICT dst, uint index, const uint step, size_t todo)
{
for(size_t i{0u};i < todo;i++)
{
index += step;
index &= WAVEFORM_FRACMASK;
dst[i] = func(index);
}
}
struct FormantFilter
{
float mCoeff{0.0f};
float mGain{1.0f};
float mS1{0.0f};
float mS2{0.0f};
FormantFilter() = default;
FormantFilter(float f0norm, float gain)
: mCoeff{std::tan(al::MathDefs<float>::Pi() * f0norm)}, mGain{gain}
{ }
inline void process(const float *samplesIn, float *samplesOut, const size_t numInput)
{
/* A state variable filter from a topology-preserving transform.
* Based on a talk given by Ivan Cohen: https://www.youtube.com/watch?v=esjHXGPyrhg
*/
const float g{mCoeff};
const float gain{mGain};
const float h{1.0f / (1.0f + (g/Q_FACTOR) + (g*g))};
float s1{mS1};
float s2{mS2};
for(size_t i{0u};i < numInput;i++)
{
const float H{(samplesIn[i] - (1.0f/Q_FACTOR + g)*s1 - s2)*h};
const float B{g*H + s1};
const float L{g*B + s2};
s1 = g*H + B;
s2 = g*B + L;
// Apply peak and accumulate samples.
samplesOut[i] += B * gain;
}
mS1 = s1;
mS2 = s2;
}
inline void clear()
{
mS1 = 0.0f;
mS2 = 0.0f;
}
};
struct VmorpherState final : public EffectState {
struct {
/* Effect parameters */
FormantFilter Formants[NUM_FILTERS][NUM_FORMANTS];
/* Effect gains for each channel */
float CurrentGains[MAX_OUTPUT_CHANNELS]{};
float TargetGains[MAX_OUTPUT_CHANNELS]{};
} mChans[MaxAmbiChannels];
void (*mGetSamples)(float*RESTRICT, uint, const uint, size_t){};
uint mIndex{0};
uint mStep{1};
/* Effects buffers */
alignas(16) float mSampleBufferA[MAX_UPDATE_SAMPLES]{};
alignas(16) float mSampleBufferB[MAX_UPDATE_SAMPLES]{};
alignas(16) float mLfo[MAX_UPDATE_SAMPLES]{};
void deviceUpdate(const ALCdevice *device, const Buffer &buffer) 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;
static std::array<FormantFilter,4> getFiltersByPhoneme(VMorpherPhenome phoneme,
float frequency, float pitch);
DEF_NEWDEL(VmorpherState)
};
std::array<FormantFilter,4> VmorpherState::getFiltersByPhoneme(VMorpherPhenome phoneme,
float frequency, float pitch)
{
/* Using soprano formant set of values to
* better match mid-range frequency space.
*
* See: https://www.classes.cs.uchicago.edu/archive/1999/spring/CS295/Computing_Resources/Csound/CsManual3.48b1.HTML/Appendices/table3.html
*/
switch(phoneme)
{
case VMorpherPhenome::A:
return {{
{( 800 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{(1150 * pitch) / frequency, 0.501187f}, /* std::pow(10.0f, -6 / 20.0f); */
{(2900 * pitch) / frequency, 0.025118f}, /* std::pow(10.0f, -32 / 20.0f); */
{(3900 * pitch) / frequency, 0.100000f} /* std::pow(10.0f, -20 / 20.0f); */
}};
case VMorpherPhenome::E:
return {{
{( 350 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{(2000 * pitch) / frequency, 0.100000f}, /* std::pow(10.0f, -20 / 20.0f); */
{(2800 * pitch) / frequency, 0.177827f}, /* std::pow(10.0f, -15 / 20.0f); */
{(3600 * pitch) / frequency, 0.009999f} /* std::pow(10.0f, -40 / 20.0f); */
}};
case VMorpherPhenome::I:
return {{
{( 270 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{(2140 * pitch) / frequency, 0.251188f}, /* std::pow(10.0f, -12 / 20.0f); */
{(2950 * pitch) / frequency, 0.050118f}, /* std::pow(10.0f, -26 / 20.0f); */
{(3900 * pitch) / frequency, 0.050118f} /* std::pow(10.0f, -26 / 20.0f); */
}};
case VMorpherPhenome::O:
return {{
{( 450 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{( 800 * pitch) / frequency, 0.281838f}, /* std::pow(10.0f, -11 / 20.0f); */
{(2830 * pitch) / frequency, 0.079432f}, /* std::pow(10.0f, -22 / 20.0f); */
{(3800 * pitch) / frequency, 0.079432f} /* std::pow(10.0f, -22 / 20.0f); */
}};
case VMorpherPhenome::U:
return {{
{( 325 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
{( 700 * pitch) / frequency, 0.158489f}, /* std::pow(10.0f, -16 / 20.0f); */
{(2700 * pitch) / frequency, 0.017782f}, /* std::pow(10.0f, -35 / 20.0f); */
{(3800 * pitch) / frequency, 0.009999f} /* std::pow(10.0f, -40 / 20.0f); */
}};
default:
break;
}
return {};
}
void VmorpherState::deviceUpdate(const ALCdevice*, const Buffer&)
{
for(auto &e : mChans)
{
std::for_each(std::begin(e.Formants[VOWEL_A_INDEX]), std::end(e.Formants[VOWEL_A_INDEX]),
std::mem_fn(&FormantFilter::clear));
std::for_each(std::begin(e.Formants[VOWEL_B_INDEX]), std::end(e.Formants[VOWEL_B_INDEX]),
std::mem_fn(&FormantFilter::clear));
std::fill(std::begin(e.CurrentGains), std::end(e.CurrentGains), 0.0f);
}
}
void VmorpherState::update(const ALCcontext *context, const EffectSlot *slot,
const EffectProps *props, const EffectTarget target)
{
const ALCdevice *device{context->mDevice.get()};
const float frequency{static_cast<float>(device->Frequency)};
const float step{props->Vmorpher.Rate / frequency};
mStep = fastf2u(clampf(step*WAVEFORM_FRACONE, 0.0f, float{WAVEFORM_FRACONE-1}));
if(mStep == 0)
mGetSamples = Oscillate<Half>;
else if(props->Vmorpher.Waveform == VMorpherWaveform::Sinusoid)
mGetSamples = Oscillate<Sin>;
else if(props->Vmorpher.Waveform == VMorpherWaveform::Triangle)
mGetSamples = Oscillate<Triangle>;
else /*if(props->Vmorpher.Waveform == VMorpherWaveform::Sawtooth)*/
mGetSamples = Oscillate<Saw>;
const float pitchA{std::pow(2.0f,
static_cast<float>(props->Vmorpher.PhonemeACoarseTuning) / 12.0f)};
const float pitchB{std::pow(2.0f,
static_cast<float>(props->Vmorpher.PhonemeBCoarseTuning) / 12.0f)};
auto vowelA = getFiltersByPhoneme(props->Vmorpher.PhonemeA, frequency, pitchA);
auto vowelB = getFiltersByPhoneme(props->Vmorpher.PhonemeB, frequency, pitchB);
/* Copy the filter coefficients to the input channels. */
for(size_t i{0u};i < slot->Wet.Buffer.size();++i)
{
std::copy(vowelA.begin(), vowelA.end(), std::begin(mChans[i].Formants[VOWEL_A_INDEX]));
std::copy(vowelB.begin(), vowelB.end(), std::begin(mChans[i].Formants[VOWEL_B_INDEX]));
}
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 VmorpherState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
/* Following the EFX specification for a conformant implementation which describes
* the effect as a pair of 4-band formant filters blended together using an LFO.
*/
for(size_t base{0u};base < samplesToDo;)
{
const size_t td{minz(MAX_UPDATE_SAMPLES, samplesToDo-base)};
mGetSamples(mLfo, mIndex, mStep, td);
mIndex += static_cast<uint>(mStep * td);
mIndex &= WAVEFORM_FRACMASK;
auto chandata = std::begin(mChans);
for(const auto &input : samplesIn)
{
auto& vowelA = chandata->Formants[VOWEL_A_INDEX];
auto& vowelB = chandata->Formants[VOWEL_B_INDEX];
/* Process first vowel. */
std::fill_n(std::begin(mSampleBufferA), td, 0.0f);
vowelA[0].process(&input[base], mSampleBufferA, td);
vowelA[1].process(&input[base], mSampleBufferA, td);
vowelA[2].process(&input[base], mSampleBufferA, td);
vowelA[3].process(&input[base], mSampleBufferA, td);
/* Process second vowel. */
std::fill_n(std::begin(mSampleBufferB), td, 0.0f);
vowelB[0].process(&input[base], mSampleBufferB, td);
vowelB[1].process(&input[base], mSampleBufferB, td);
vowelB[2].process(&input[base], mSampleBufferB, td);
vowelB[3].process(&input[base], mSampleBufferB, td);
alignas(16) float blended[MAX_UPDATE_SAMPLES];
for(size_t i{0u};i < td;i++)
blended[i] = lerp(mSampleBufferA[i], mSampleBufferB[i], mLfo[i]);
/* Now, mix the processed sound data to the output. */
MixSamples({blended, td}, samplesOut, chandata->CurrentGains, chandata->TargetGains,
samplesToDo-base, base);
++chandata;
}
base += td;
}
}
struct VmorpherStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new VmorpherState{}}; }
};
} // namespace
EffectStateFactory *VmorpherStateFactory_getFactory()
{
static VmorpherStateFactory VmorpherFactory{};
return &VmorpherFactory;
}
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