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337 lines
12 KiB
337 lines
12 KiB
/**
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* This file is part of the OpenAL Soft cross platform audio library
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*
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* Copyright (C) 2019 by Anis A. Hireche
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* * Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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*
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* * Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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*
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* * Neither the name of Spherical-Harmonic-Transform nor the names of its
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* contributors may be used to endorse or promote products derived from
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* this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
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* AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
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* IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
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* ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR CONTRIBUTORS BE
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* LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
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* CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
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* SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
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* INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
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* CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
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* ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
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* POSSIBILITY OF SUCH DAMAGE.
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*/
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#include "config.h"
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#include <algorithm>
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#include <array>
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#include <cstdlib>
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#include <functional>
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#include <iterator>
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#include "alc/effects/base.h"
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#include "almalloc.h"
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#include "alnumbers.h"
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#include "alnumeric.h"
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#include "alspan.h"
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#include "core/ambidefs.h"
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#include "core/bufferline.h"
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#include "core/context.h"
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#include "core/devformat.h"
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#include "core/device.h"
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#include "core/effectslot.h"
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#include "core/mixer.h"
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#include "intrusive_ptr.h"
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namespace {
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using uint = unsigned int;
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#define MAX_UPDATE_SAMPLES 256
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#define NUM_FORMANTS 4
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#define NUM_FILTERS 2
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#define Q_FACTOR 5.0f
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#define VOWEL_A_INDEX 0
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#define VOWEL_B_INDEX 1
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#define WAVEFORM_FRACBITS 24
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#define WAVEFORM_FRACONE (1<<WAVEFORM_FRACBITS)
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#define WAVEFORM_FRACMASK (WAVEFORM_FRACONE-1)
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inline float Sin(uint index)
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{
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constexpr float scale{al::numbers::pi_v<float>*2.0f / WAVEFORM_FRACONE};
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return std::sin(static_cast<float>(index) * scale)*0.5f + 0.5f;
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}
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inline float Saw(uint index)
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{ return static_cast<float>(index) / float{WAVEFORM_FRACONE}; }
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inline float Triangle(uint index)
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{ return std::fabs(static_cast<float>(index)*(2.0f/WAVEFORM_FRACONE) - 1.0f); }
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inline float Half(uint) { return 0.5f; }
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template<float (&func)(uint)>
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void Oscillate(float *RESTRICT dst, uint index, const uint step, size_t todo)
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{
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for(size_t i{0u};i < todo;i++)
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{
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index += step;
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index &= WAVEFORM_FRACMASK;
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dst[i] = func(index);
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}
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}
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struct FormantFilter
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{
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float mCoeff{0.0f};
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float mGain{1.0f};
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float mS1{0.0f};
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float mS2{0.0f};
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FormantFilter() = default;
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FormantFilter(float f0norm, float gain)
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: mCoeff{std::tan(al::numbers::pi_v<float> * f0norm)}, mGain{gain}
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{ }
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inline void process(const float *samplesIn, float *samplesOut, const size_t numInput)
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{
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/* A state variable filter from a topology-preserving transform.
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* Based on a talk given by Ivan Cohen: https://www.youtube.com/watch?v=esjHXGPyrhg
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*/
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const float g{mCoeff};
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const float gain{mGain};
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const float h{1.0f / (1.0f + (g/Q_FACTOR) + (g*g))};
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float s1{mS1};
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float s2{mS2};
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for(size_t i{0u};i < numInput;i++)
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{
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const float H{(samplesIn[i] - (1.0f/Q_FACTOR + g)*s1 - s2)*h};
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const float B{g*H + s1};
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const float L{g*B + s2};
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s1 = g*H + B;
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s2 = g*B + L;
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// Apply peak and accumulate samples.
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samplesOut[i] += B * gain;
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}
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mS1 = s1;
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mS2 = s2;
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}
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inline void clear()
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{
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mS1 = 0.0f;
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mS2 = 0.0f;
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}
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};
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struct VmorpherState final : public EffectState {
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struct {
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/* Effect parameters */
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FormantFilter Formants[NUM_FILTERS][NUM_FORMANTS];
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/* Effect gains for each channel */
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float CurrentGains[MAX_OUTPUT_CHANNELS]{};
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float TargetGains[MAX_OUTPUT_CHANNELS]{};
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} mChans[MaxAmbiChannels];
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void (*mGetSamples)(float*RESTRICT, uint, const uint, size_t){};
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uint mIndex{0};
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uint mStep{1};
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/* Effects buffers */
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alignas(16) float mSampleBufferA[MAX_UPDATE_SAMPLES]{};
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alignas(16) float mSampleBufferB[MAX_UPDATE_SAMPLES]{};
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alignas(16) float mLfo[MAX_UPDATE_SAMPLES]{};
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void deviceUpdate(const DeviceBase *device, const Buffer &buffer) override;
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void update(const ContextBase *context, const EffectSlot *slot, const EffectProps *props,
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const EffectTarget target) override;
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void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
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const al::span<FloatBufferLine> samplesOut) override;
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static std::array<FormantFilter,4> getFiltersByPhoneme(VMorpherPhenome phoneme,
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float frequency, float pitch);
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DEF_NEWDEL(VmorpherState)
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};
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std::array<FormantFilter,4> VmorpherState::getFiltersByPhoneme(VMorpherPhenome phoneme,
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float frequency, float pitch)
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{
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/* Using soprano formant set of values to
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* better match mid-range frequency space.
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*
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* See: https://www.classes.cs.uchicago.edu/archive/1999/spring/CS295/Computing_Resources/Csound/CsManual3.48b1.HTML/Appendices/table3.html
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*/
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switch(phoneme)
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{
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case VMorpherPhenome::A:
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return {{
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{( 800 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
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{(1150 * pitch) / frequency, 0.501187f}, /* std::pow(10.0f, -6 / 20.0f); */
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{(2900 * pitch) / frequency, 0.025118f}, /* std::pow(10.0f, -32 / 20.0f); */
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{(3900 * pitch) / frequency, 0.100000f} /* std::pow(10.0f, -20 / 20.0f); */
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}};
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case VMorpherPhenome::E:
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return {{
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{( 350 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
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{(2000 * pitch) / frequency, 0.100000f}, /* std::pow(10.0f, -20 / 20.0f); */
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{(2800 * pitch) / frequency, 0.177827f}, /* std::pow(10.0f, -15 / 20.0f); */
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{(3600 * pitch) / frequency, 0.009999f} /* std::pow(10.0f, -40 / 20.0f); */
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}};
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case VMorpherPhenome::I:
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return {{
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{( 270 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
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{(2140 * pitch) / frequency, 0.251188f}, /* std::pow(10.0f, -12 / 20.0f); */
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{(2950 * pitch) / frequency, 0.050118f}, /* std::pow(10.0f, -26 / 20.0f); */
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{(3900 * pitch) / frequency, 0.050118f} /* std::pow(10.0f, -26 / 20.0f); */
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}};
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case VMorpherPhenome::O:
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return {{
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{( 450 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
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{( 800 * pitch) / frequency, 0.281838f}, /* std::pow(10.0f, -11 / 20.0f); */
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{(2830 * pitch) / frequency, 0.079432f}, /* std::pow(10.0f, -22 / 20.0f); */
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{(3800 * pitch) / frequency, 0.079432f} /* std::pow(10.0f, -22 / 20.0f); */
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}};
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case VMorpherPhenome::U:
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return {{
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{( 325 * pitch) / frequency, 1.000000f}, /* std::pow(10.0f, 0 / 20.0f); */
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{( 700 * pitch) / frequency, 0.158489f}, /* std::pow(10.0f, -16 / 20.0f); */
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{(2700 * pitch) / frequency, 0.017782f}, /* std::pow(10.0f, -35 / 20.0f); */
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{(3800 * pitch) / frequency, 0.009999f} /* std::pow(10.0f, -40 / 20.0f); */
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}};
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default:
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break;
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}
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return {};
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}
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void VmorpherState::deviceUpdate(const DeviceBase*, const Buffer&)
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{
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for(auto &e : mChans)
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{
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std::for_each(std::begin(e.Formants[VOWEL_A_INDEX]), std::end(e.Formants[VOWEL_A_INDEX]),
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std::mem_fn(&FormantFilter::clear));
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std::for_each(std::begin(e.Formants[VOWEL_B_INDEX]), std::end(e.Formants[VOWEL_B_INDEX]),
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std::mem_fn(&FormantFilter::clear));
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std::fill(std::begin(e.CurrentGains), std::end(e.CurrentGains), 0.0f);
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}
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}
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void VmorpherState::update(const ContextBase *context, const EffectSlot *slot,
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const EffectProps *props, const EffectTarget target)
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{
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const DeviceBase *device{context->mDevice};
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const float frequency{static_cast<float>(device->Frequency)};
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const float step{props->Vmorpher.Rate / frequency};
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mStep = fastf2u(clampf(step*WAVEFORM_FRACONE, 0.0f, float{WAVEFORM_FRACONE-1}));
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if(mStep == 0)
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mGetSamples = Oscillate<Half>;
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else if(props->Vmorpher.Waveform == VMorpherWaveform::Sinusoid)
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mGetSamples = Oscillate<Sin>;
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else if(props->Vmorpher.Waveform == VMorpherWaveform::Triangle)
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mGetSamples = Oscillate<Triangle>;
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else /*if(props->Vmorpher.Waveform == VMorpherWaveform::Sawtooth)*/
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mGetSamples = Oscillate<Saw>;
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const float pitchA{std::pow(2.0f,
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static_cast<float>(props->Vmorpher.PhonemeACoarseTuning) / 12.0f)};
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const float pitchB{std::pow(2.0f,
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static_cast<float>(props->Vmorpher.PhonemeBCoarseTuning) / 12.0f)};
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auto vowelA = getFiltersByPhoneme(props->Vmorpher.PhonemeA, frequency, pitchA);
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auto vowelB = getFiltersByPhoneme(props->Vmorpher.PhonemeB, frequency, pitchB);
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/* Copy the filter coefficients to the input channels. */
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for(size_t i{0u};i < slot->Wet.Buffer.size();++i)
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{
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std::copy(vowelA.begin(), vowelA.end(), std::begin(mChans[i].Formants[VOWEL_A_INDEX]));
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std::copy(vowelB.begin(), vowelB.end(), std::begin(mChans[i].Formants[VOWEL_B_INDEX]));
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}
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mOutTarget = target.Main->Buffer;
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auto set_gains = [slot,target](auto &chan, al::span<const float,MaxAmbiChannels> coeffs)
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{ ComputePanGains(target.Main, coeffs.data(), slot->Gain, chan.TargetGains); };
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SetAmbiPanIdentity(std::begin(mChans), slot->Wet.Buffer.size(), set_gains);
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}
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void VmorpherState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
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{
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/* Following the EFX specification for a conformant implementation which describes
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* the effect as a pair of 4-band formant filters blended together using an LFO.
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*/
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for(size_t base{0u};base < samplesToDo;)
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{
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const size_t td{minz(MAX_UPDATE_SAMPLES, samplesToDo-base)};
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mGetSamples(mLfo, mIndex, mStep, td);
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mIndex += static_cast<uint>(mStep * td);
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mIndex &= WAVEFORM_FRACMASK;
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auto chandata = std::begin(mChans);
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for(const auto &input : samplesIn)
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{
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auto& vowelA = chandata->Formants[VOWEL_A_INDEX];
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auto& vowelB = chandata->Formants[VOWEL_B_INDEX];
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/* Process first vowel. */
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std::fill_n(std::begin(mSampleBufferA), td, 0.0f);
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vowelA[0].process(&input[base], mSampleBufferA, td);
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vowelA[1].process(&input[base], mSampleBufferA, td);
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vowelA[2].process(&input[base], mSampleBufferA, td);
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vowelA[3].process(&input[base], mSampleBufferA, td);
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/* Process second vowel. */
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std::fill_n(std::begin(mSampleBufferB), td, 0.0f);
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vowelB[0].process(&input[base], mSampleBufferB, td);
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vowelB[1].process(&input[base], mSampleBufferB, td);
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vowelB[2].process(&input[base], mSampleBufferB, td);
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vowelB[3].process(&input[base], mSampleBufferB, td);
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alignas(16) float blended[MAX_UPDATE_SAMPLES];
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for(size_t i{0u};i < td;i++)
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blended[i] = lerpf(mSampleBufferA[i], mSampleBufferB[i], mLfo[i]);
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/* Now, mix the processed sound data to the output. */
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MixSamples({blended, td}, samplesOut, chandata->CurrentGains, chandata->TargetGains,
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samplesToDo-base, base);
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++chandata;
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}
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base += td;
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}
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}
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struct VmorpherStateFactory final : public EffectStateFactory {
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al::intrusive_ptr<EffectState> create() override
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{ return al::intrusive_ptr<EffectState>{new VmorpherState{}}; }
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};
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} // namespace
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EffectStateFactory *VmorpherStateFactory_getFactory()
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{
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static VmorpherStateFactory VmorpherFactory{};
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return &VmorpherFactory;
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}
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