/** * 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 #include #include #include #include "al/auxeffectslot.h" #include "alcmain.h" #include "alcontext.h" #include "alu.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<::Tau() / WAVEFORM_FRACONE}; return std::sin(static_cast(index) * scale)*0.5f + 0.5f; } inline float Saw(ALuint index) { return static_cast(index) / float{WAVEFORM_FRACONE}; } inline float Triangle(ALuint index) { return std::fabs(static_cast(index)*(2.0f/WAVEFORM_FRACONE) - 1.0f); } inline float Half(ALuint) { return 0.5f; } template void Oscillate(float *RESTRICT dst, ALuint index, const ALuint 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::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, ALuint, const ALuint, size_t){}; ALuint mIndex{0}; ALuint 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) 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 samplesIn, const al::span samplesOut) override; static std::array getFiltersByPhoneme(ALenum phoneme, float frequency, float pitch); DEF_NEWDEL(VmorpherState) }; std::array VmorpherState::getFiltersByPhoneme(ALenum 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 AL_VOCAL_MORPHER_PHONEME_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 AL_VOCAL_MORPHER_PHONEME_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 AL_VOCAL_MORPHER_PHONEME_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 AL_VOCAL_MORPHER_PHONEME_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 AL_VOCAL_MORPHER_PHONEME_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); */ }}; } return {}; } void VmorpherState::deviceUpdate(const ALCdevice* /*device*/) { 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(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; else if(props->Vmorpher.Waveform == AL_VOCAL_MORPHER_WAVEFORM_SINUSOID) mGetSamples = Oscillate; else if(props->Vmorpher.Waveform == AL_VOCAL_MORPHER_WAVEFORM_SAWTOOTH) mGetSamples = Oscillate; else /*if(props->Vmorpher.Waveform == AL_VOCAL_MORPHER_WAVEFORM_TRIANGLE)*/ mGetSamples = Oscillate; const float pitchA{std::pow(2.0f, static_cast(props->Vmorpher.PhonemeACoarseTuning) / 12.0f)}; const float pitchB{std::pow(2.0f, static_cast(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 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 samplesIn, const al::span 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(mStep * td); mIndex &= WAVEFORM_FRACMASK; auto chandata = std::addressof(mChans[0]); 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 { EffectState *create() override { return new VmorpherState{}; } }; } // namespace EffectStateFactory *VmorpherStateFactory_getFactory() { static VmorpherStateFactory VmorpherFactory{}; return &VmorpherFactory; }