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- /**
- * This file is part of the OpenAL Soft cross platform audio library
- *
- * Copyright (C) 2019 by Anis A. Hireche
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * * Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- *
- * * Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- *
- * * Neither the name of Spherical-Harmonic-Transform nor the names of its
- * contributors may be used to endorse or promote products derived from
- * this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
- * AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
- * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
- * ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE
- * LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
- * CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
- * SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
- * INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
- * CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
- * ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
- * POSSIBILITY OF SUCH DAMAGE.
- */
-
- #include "config.h"
-
- #include <algorithm>
- #include <array>
- #include <cstdlib>
- #include <functional>
- #include <iterator>
-
- #include "alc/effects/base.h"
- #include "almalloc.h"
- #include "alnumbers.h"
- #include "alnumeric.h"
- #include "alspan.h"
- #include "core/ambidefs.h"
- #include "core/bufferline.h"
- #include "core/context.h"
- #include "core/devformat.h"
- #include "core/device.h"
- #include "core/effectslot.h"
- #include "core/mixer.h"
- #include "intrusive_ptr.h"
-
-
- namespace {
-
- using uint = unsigned int;
-
- #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::numbers::pi_v<float>*2.0f / 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::numbers::pi_v<float> * 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 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;
-
- 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 DeviceBase*, 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 ContextBase *context, const EffectSlot *slot,
- const EffectProps *props, const EffectTarget target)
- {
- const DeviceBase *device{context->mDevice};
- 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] = lerpf(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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