diff options
author | Chris Robinson <[email protected]> | 2019-07-28 18:56:04 -0700 |
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committer | Chris Robinson <[email protected]> | 2019-07-28 18:56:04 -0700 |
commit | cb3e96e75640730b9391f0d2d922eecd9ee2ce79 (patch) | |
tree | 23520551bddb2a80354e44da47f54201fdc084f0 /alc/effects/vmorpher.cpp | |
parent | 93e60919c8f387c36c267ca9faa1ac653254aea6 (diff) |
Rename Alc to alc
Diffstat (limited to 'alc/effects/vmorpher.cpp')
-rw-r--r-- | alc/effects/vmorpher.cpp | 430 |
1 files changed, 430 insertions, 0 deletions
diff --git a/alc/effects/vmorpher.cpp b/alc/effects/vmorpher.cpp new file mode 100644 index 00000000..eebba3f1 --- /dev/null +++ b/alc/effects/vmorpher.cpp @@ -0,0 +1,430 @@ +/** + * 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 "alAuxEffectSlot.h" +#include "alError.h" +#include "alu.h" + +namespace { + +#define MAX_UPDATE_SAMPLES 128 +#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 ALfloat Sin(ALsizei index) +{ + constexpr ALfloat scale{al::MathDefs<float>::Tau() / ALfloat{WAVEFORM_FRACONE}}; + return std::sin(static_cast<ALfloat>(index) * scale)*0.5f + 0.5f; +} + +inline ALfloat Saw(ALsizei index) +{ + return static_cast<ALfloat>(index) / ALfloat{WAVEFORM_FRACONE}; +} + +inline ALfloat Triangle(ALsizei index) +{ + return std::fabs(static_cast<ALfloat>(index)*(2.0f/WAVEFORM_FRACONE) - 1.0f); +} + +inline ALfloat Half(ALsizei) +{ + return 0.5f; +} + +template<ALfloat func(ALsizei)> +void Oscillate(ALfloat *RESTRICT dst, ALsizei index, const ALsizei step, ALsizei todo) +{ + for(ALsizei i{0};i < todo;i++) + { + index += step; + index &= WAVEFORM_FRACMASK; + dst[i] = func(index); + } +} + +struct FormantFilter +{ + ALfloat f0norm{0.0f}; + ALfloat fGain{1.0f}; + ALfloat s1{0.0f}; + ALfloat s2{0.0f}; + + FormantFilter() = default; + FormantFilter(ALfloat f0norm_, ALfloat gain) : f0norm{f0norm_}, fGain{gain} { } + + inline void process(const ALfloat* samplesIn, ALfloat* samplesOut, const ALsizei 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 ALfloat g = std::tan(al::MathDefs<float>::Pi() * f0norm); + const ALfloat h = 1.0f / (1 + (g / Q_FACTOR) + (g * g)); + + for (ALsizei i{0};i < numInput;i++) + { + const ALfloat H = h * (samplesIn[i] - (1.0f / Q_FACTOR + g) * s1 - s2); + const ALfloat B = g * H + s1; + const ALfloat L = g * B + s2; + + s1 = g * H + B; + s2 = g * B + L; + + // Apply peak and accumulate samples. + samplesOut[i] += B * fGain; + } + } + + inline void clear() + { + s1 = 0.0f; + s2 = 0.0f; + } +}; + + +struct VmorpherState final : public EffectState { + struct { + /* Effect parameters */ + FormantFilter Formants[NUM_FILTERS][NUM_FORMANTS]; + + /* Effect gains for each channel */ + ALfloat CurrentGains[MAX_OUTPUT_CHANNELS]{}; + ALfloat TargetGains[MAX_OUTPUT_CHANNELS]{}; + } mChans[MAX_AMBI_CHANNELS]; + + void (*mGetSamples)(ALfloat* RESTRICT, ALsizei, const ALsizei, ALsizei) {}; + + ALsizei mIndex{0}; + ALsizei mStep{1}; + + /* Effects buffers */ + ALfloat mSampleBufferA[MAX_UPDATE_SAMPLES]{}; + ALfloat mSampleBufferB[MAX_UPDATE_SAMPLES]{}; + + ALboolean deviceUpdate(const ALCdevice *device) override; + void update(const ALCcontext *context, const ALeffectslot *slot, const EffectProps *props, const EffectTarget target) override; + void process(const ALsizei samplesToDo, const FloatBufferLine *RESTRICT samplesIn, const ALsizei numInput, const al::span<FloatBufferLine> samplesOut) override; + + static std::array<FormantFilter,4> getFiltersByPhoneme(ALenum phoneme, ALfloat frequency, ALfloat pitch); + + DEF_NEWDEL(VmorpherState) +}; + +std::array<FormantFilter,4> VmorpherState::getFiltersByPhoneme(ALenum phoneme, ALfloat frequency, ALfloat 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 {}; +} + + +ALboolean 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); + } + + return AL_TRUE; +} + +void VmorpherState::update(const ALCcontext *context, const ALeffectslot *slot, const EffectProps *props, const EffectTarget target) +{ + const ALCdevice *device{context->Device}; + const ALfloat frequency{static_cast<ALfloat>(device->Frequency)}; + const ALfloat step{props->Vmorpher.Rate / static_cast<ALfloat>(device->Frequency)}; + mStep = fastf2i(clampf(step*WAVEFORM_FRACONE, 0.0f, ALfloat{WAVEFORM_FRACONE-1})); + + if(mStep == 0) + mGetSamples = Oscillate<Half>; + else if(props->Vmorpher.Waveform == AL_VOCAL_MORPHER_WAVEFORM_SINUSOID) + mGetSamples = Oscillate<Sin>; + else if(props->Vmorpher.Waveform == AL_VOCAL_MORPHER_WAVEFORM_SAWTOOTH) + mGetSamples = Oscillate<Saw>; + else /*if(props->Vmorpher.Waveform == AL_VOCAL_MORPHER_WAVEFORM_TRIANGLE)*/ + mGetSamples = Oscillate<Triangle>; + + const ALfloat pitchA{fastf2i(std::pow(2.0f, props->Vmorpher.PhonemeACoarseTuning*100.0f / 2400.0f)*FRACTIONONE) * (1.0f/FRACTIONONE)}; + const ALfloat pitchB{fastf2i(std::pow(2.0f, props->Vmorpher.PhonemeBCoarseTuning*100.0f / 2400.0f)*FRACTIONONE) * (1.0f/FRACTIONONE)}; + + 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; + for(size_t i{0u};i < slot->Wet.Buffer.size();++i) + { + auto coeffs = GetAmbiIdentityRow(i); + ComputePanGains(target.Main, coeffs.data(), slot->Params.Gain, mChans[i].TargetGains); + } +} + +void VmorpherState::process(const ALsizei samplesToDo, const FloatBufferLine *RESTRICT samplesIn, const ALsizei numInput, 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(ALsizei base{0};base < samplesToDo;) + { + alignas(16) ALfloat lfo[MAX_UPDATE_SAMPLES]; + const ALsizei td = mini(MAX_UPDATE_SAMPLES, samplesToDo-base); + + mGetSamples(lfo, mIndex, mStep, td); + mIndex += (mStep * td) & WAVEFORM_FRACMASK; + mIndex &= WAVEFORM_FRACMASK; + + ASSUME(numInput > 0); + for(ALsizei c{0};c < numInput;c++) + { + for (ALsizei i{0};i < td;i++) + { + mSampleBufferA[i] = 0.0f; + mSampleBufferB[i] = 0.0f; + } + + auto& vowelA = mChans[c].Formants[VOWEL_A_INDEX]; + auto& vowelB = mChans[c].Formants[VOWEL_B_INDEX]; + + /* Process first vowel. */ + vowelA[0].process(&samplesIn[c][base], mSampleBufferA, td); + vowelA[1].process(&samplesIn[c][base], mSampleBufferA, td); + vowelA[2].process(&samplesIn[c][base], mSampleBufferA, td); + vowelA[3].process(&samplesIn[c][base], mSampleBufferA, td); + + /* Process second vowel. */ + vowelB[0].process(&samplesIn[c][base], mSampleBufferB, td); + vowelB[1].process(&samplesIn[c][base], mSampleBufferB, td); + vowelB[2].process(&samplesIn[c][base], mSampleBufferB, td); + vowelB[3].process(&samplesIn[c][base], mSampleBufferB, td); + + alignas(16) ALfloat samplesBlended[MAX_UPDATE_SAMPLES]; + + for (ALsizei i{0};i < td;i++) + samplesBlended[i] = lerp(mSampleBufferA[i], mSampleBufferB[i], lfo[i]); + + /* Now, mix the processed sound data to the output. */ + MixSamples(samplesBlended, samplesOut, mChans[c].CurrentGains, mChans[c].TargetGains, + samplesToDo-base, base, td); + } + + base += td; + } +} + + +void Vmorpher_setParami(EffectProps* props, ALCcontext *context, ALenum param, ALint val) +{ + switch(param) + { + case AL_VOCAL_MORPHER_WAVEFORM: + if(!(val >= AL_VOCAL_MORPHER_MIN_WAVEFORM && val <= AL_VOCAL_MORPHER_MAX_WAVEFORM)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Vocal morpher waveform out of range"); + props->Vmorpher.Waveform = val; + break; + + case AL_VOCAL_MORPHER_PHONEMEA: + if(!(val >= AL_VOCAL_MORPHER_MIN_PHONEMEA && val <= AL_VOCAL_MORPHER_MAX_PHONEMEA)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Vocal morpher phoneme-a out of range"); + props->Vmorpher.PhonemeA = val; + break; + + case AL_VOCAL_MORPHER_PHONEMEB: + if(!(val >= AL_VOCAL_MORPHER_MIN_PHONEMEB && val <= AL_VOCAL_MORPHER_MAX_PHONEMEB)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Vocal morpher phoneme-b out of range"); + props->Vmorpher.PhonemeB = val; + break; + + case AL_VOCAL_MORPHER_PHONEMEA_COARSE_TUNING: + if(!(val >= AL_VOCAL_MORPHER_MIN_PHONEMEA_COARSE_TUNING && val <= AL_VOCAL_MORPHER_MAX_PHONEMEA_COARSE_TUNING)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Vocal morpher phoneme-a coarse tuning out of range"); + props->Vmorpher.PhonemeACoarseTuning = val; + break; + + case AL_VOCAL_MORPHER_PHONEMEB_COARSE_TUNING: + if(!(val >= AL_VOCAL_MORPHER_MIN_PHONEMEB_COARSE_TUNING && val <= AL_VOCAL_MORPHER_MAX_PHONEMEB_COARSE_TUNING)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Vocal morpher phoneme-b coarse tuning out of range"); + props->Vmorpher.PhonemeBCoarseTuning = val; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid vocal morpher integer property 0x%04x", param); + } +} +void Vmorpher_setParamiv(EffectProps*, ALCcontext *context, ALenum param, const ALint*) +{ alSetError(context, AL_INVALID_ENUM, "Invalid vocal morpher integer-vector property 0x%04x", param); } +void Vmorpher_setParamf(EffectProps *props, ALCcontext *context, ALenum param, ALfloat val) +{ + switch(param) + { + case AL_VOCAL_MORPHER_RATE: + if(!(val >= AL_VOCAL_MORPHER_MIN_RATE && val <= AL_VOCAL_MORPHER_MAX_RATE)) + SETERR_RETURN(context, AL_INVALID_VALUE,, "Vocal morpher rate out of range"); + props->Vmorpher.Rate = val; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid vocal morpher float property 0x%04x", param); + } +} +void Vmorpher_setParamfv(EffectProps *props, ALCcontext *context, ALenum param, const ALfloat *vals) +{ Vmorpher_setParamf(props, context, param, vals[0]); } + +void Vmorpher_getParami(const EffectProps* props, ALCcontext *context, ALenum param, ALint* val) +{ + switch(param) + { + case AL_VOCAL_MORPHER_PHONEMEA: + *val = props->Vmorpher.PhonemeA; + break; + + case AL_VOCAL_MORPHER_PHONEMEB: + *val = props->Vmorpher.PhonemeB; + break; + + case AL_VOCAL_MORPHER_PHONEMEA_COARSE_TUNING: + *val = props->Vmorpher.PhonemeACoarseTuning; + break; + + case AL_VOCAL_MORPHER_PHONEMEB_COARSE_TUNING: + *val = props->Vmorpher.PhonemeBCoarseTuning; + break; + + case AL_VOCAL_MORPHER_WAVEFORM: + *val = props->Vmorpher.Waveform; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid vocal morpher integer property 0x%04x", param); + } +} +void Vmorpher_getParamiv(const EffectProps*, ALCcontext *context, ALenum param, ALint*) +{ alSetError(context, AL_INVALID_ENUM, "Invalid vocal morpher integer-vector property 0x%04x", param); } +void Vmorpher_getParamf(const EffectProps *props, ALCcontext *context, ALenum param, ALfloat *val) +{ + switch(param) + { + case AL_VOCAL_MORPHER_RATE: + *val = props->Vmorpher.Rate; + break; + + default: + alSetError(context, AL_INVALID_ENUM, "Invalid vocal morpher float property 0x%04x", param); + } +} +void Vmorpher_getParamfv(const EffectProps *props, ALCcontext *context, ALenum param, ALfloat *vals) +{ Vmorpher_getParamf(props, context, param, vals); } + +DEFINE_ALEFFECT_VTABLE(Vmorpher); + + +struct VmorpherStateFactory final : public EffectStateFactory { + EffectState *create() override { return new VmorpherState{}; } + EffectProps getDefaultProps() const noexcept override; + const EffectVtable *getEffectVtable() const noexcept override { return &Vmorpher_vtable; } +}; + +EffectProps VmorpherStateFactory::getDefaultProps() const noexcept +{ + EffectProps props{}; + props.Vmorpher.Rate = AL_VOCAL_MORPHER_DEFAULT_RATE; + props.Vmorpher.PhonemeA = AL_VOCAL_MORPHER_DEFAULT_PHONEMEA; + props.Vmorpher.PhonemeB = AL_VOCAL_MORPHER_DEFAULT_PHONEMEB; + props.Vmorpher.PhonemeACoarseTuning = AL_VOCAL_MORPHER_DEFAULT_PHONEMEA_COARSE_TUNING; + props.Vmorpher.PhonemeBCoarseTuning = AL_VOCAL_MORPHER_DEFAULT_PHONEMEB_COARSE_TUNING; + props.Vmorpher.Waveform = AL_VOCAL_MORPHER_DEFAULT_WAVEFORM; + return props; +} + +} // namespace + +EffectStateFactory *VmorpherStateFactory_getFactory() +{ + static VmorpherStateFactory VmorpherFactory{}; + return &VmorpherFactory; +} |