mirror of
https://github.com/TorqueGameEngines/Torque3D.git
synced 2026-03-04 13:00:33 +00:00
added libraries: opus flac libsndfile updated: libvorbis libogg openal - Everything works as expected for now. Bare in mind libsndfile needed the check for whether or not it could find the xiph libraries removed in order for this to work.
350 lines
12 KiB
C++
350 lines
12 KiB
C++
/**
|
|
* 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 {
|
|
uint mTargetChannel{InvalidChannelIndex};
|
|
|
|
/* Effect parameters */
|
|
FormantFilter mFormants[NUM_FILTERS][NUM_FORMANTS];
|
|
|
|
/* Effect gains for each channel */
|
|
float mCurrentGain{};
|
|
float mTargetGain{};
|
|
} 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 BufferStorage *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 BufferStorage*)
|
|
{
|
|
for(auto &e : mChans)
|
|
{
|
|
e.mTargetChannel = InvalidChannelIndex;
|
|
std::for_each(std::begin(e.mFormants[VOWEL_A_INDEX]), std::end(e.mFormants[VOWEL_A_INDEX]),
|
|
std::mem_fn(&FormantFilter::clear));
|
|
std::for_each(std::begin(e.mFormants[VOWEL_B_INDEX]), std::end(e.mFormants[VOWEL_B_INDEX]),
|
|
std::mem_fn(&FormantFilter::clear));
|
|
e.mCurrentGain = 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].mFormants[VOWEL_A_INDEX]));
|
|
std::copy(vowelB.begin(), vowelB.end(), std::begin(mChans[i].mFormants[VOWEL_B_INDEX]));
|
|
}
|
|
|
|
mOutTarget = target.Main->Buffer;
|
|
auto set_channel = [this](size_t idx, uint outchan, float outgain)
|
|
{
|
|
mChans[idx].mTargetChannel = outchan;
|
|
mChans[idx].mTargetGain = outgain;
|
|
};
|
|
target.Main->setAmbiMixParams(slot->Wet, slot->Gain, set_channel);
|
|
}
|
|
|
|
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)
|
|
{
|
|
const size_t outidx{chandata->mTargetChannel};
|
|
if(outidx == InvalidChannelIndex)
|
|
{
|
|
++chandata;
|
|
continue;
|
|
}
|
|
|
|
auto& vowelA = chandata->mFormants[VOWEL_A_INDEX];
|
|
auto& vowelB = chandata->mFormants[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[outidx].data()+base, chandata->mCurrentGain,
|
|
chandata->mTargetGain, samplesToDo-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;
|
|
}
|