TorqueGameEngines/Torque3D
1
0
Fork 0
mirror of https://github.com/TorqueGameEngines/Torque3D.git synced 2026-07-15 00:24:40 +00:00
Code Issues Projects Releases Packages Wiki Activity Actions

* BugFix: Correct convexDecomp compilation by setting the LINUX flag when necessary.

Browse source 
* BugFix: Update OpenAL to correct a compilation error on Linux.
This commit is contained in:
Robert MacGregor 2022-05-30 16:32:45 -04:00
parent e071f1d901
commit 7380161054
234 changed files with 30864 additions and 7523 deletions
Download patch file Download diff file Expand all files Collapse all files

274
Engine/lib/openal-soft/alc/effects/pshifter.cpp Normal file
View file

@ -0,0 +1,274 @@
/**
* OpenAL cross platform audio library
* Copyright (C) 2018 by Raul Herraiz.
* 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 <algorithm>
#include <array>
#include <cmath>
#include <complex>
#include <cstdlib>
#include <iterator>
#include "alc/effects/base.h"
#include "alcomplex.h"
#include "almalloc.h"
#include "alnumbers.h"
#include "alnumeric.h"
#include "alspan.h"
#include "core/bufferline.h"
#include "core/devformat.h"
#include "core/device.h"
#include "core/effectslot.h"
#include "core/mixer.h"
#include "core/mixer/defs.h"
#include "intrusive_ptr.h"
struct ContextBase;
namespace {
using uint = unsigned int;
using complex_d = std::complex<double>;
#define STFT_SIZE 1024
#define STFT_HALF_SIZE (STFT_SIZE>>1)
#define OVERSAMP (1<<2)
#define STFT_STEP (STFT_SIZE / OVERSAMP)
#define FIFO_LATENCY (STFT_STEP * (OVERSAMP-1))
/* Define a Hann window, used to filter the STFT input and output. */
std::array<double,STFT_SIZE> InitHannWindow()
{
std::array<double,STFT_SIZE> ret;
/* Create lookup table of the Hann window for the desired size, i.e. STFT_SIZE */
for(size_t i{0};i < STFT_SIZE>>1;i++)
{
constexpr double scale{al::numbers::pi / double{STFT_SIZE}};
const double val{std::sin(static_cast<double>(i+1) * scale)};
ret[i] = ret[STFT_SIZE-1-i] = val * val;
}
return ret;
}
alignas(16) const std::array<double,STFT_SIZE> HannWindow = InitHannWindow();
struct FrequencyBin {
double Amplitude;
double FreqBin;
};
struct PshifterState final : public EffectState {
/* Effect parameters */
size_t mCount;
size_t mPos;
uint mPitchShiftI;
double mPitchShift;
/* Effects buffers */
std::array<double,STFT_SIZE> mFIFO;
std::array<double,STFT_HALF_SIZE+1> mLastPhase;
std::array<double,STFT_HALF_SIZE+1> mSumPhase;
std::array<double,STFT_SIZE> mOutputAccum;
std::array<complex_d,STFT_SIZE> mFftBuffer;
std::array<FrequencyBin,STFT_HALF_SIZE+1> mAnalysisBuffer;
std::array<FrequencyBin,STFT_HALF_SIZE+1> mSynthesisBuffer;
alignas(16) FloatBufferLine mBufferOut;
/* Effect gains for each output channel */
float mCurrentGains[MAX_OUTPUT_CHANNELS];
float mTargetGains[MAX_OUTPUT_CHANNELS];
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;
DEF_NEWDEL(PshifterState)
};
void PshifterState::deviceUpdate(const DeviceBase*, const Buffer&)
{
/* (Re-)initializing parameters and clear the buffers. */
mCount = 0;
mPos = FIFO_LATENCY;
mPitchShiftI = MixerFracOne;
mPitchShift = 1.0;
std::fill(mFIFO.begin(), mFIFO.end(), 0.0);
std::fill(mLastPhase.begin(), mLastPhase.end(), 0.0);
std::fill(mSumPhase.begin(), mSumPhase.end(), 0.0);
std::fill(mOutputAccum.begin(), mOutputAccum.end(), 0.0);
std::fill(mFftBuffer.begin(), mFftBuffer.end(), complex_d{});
std::fill(mAnalysisBuffer.begin(), mAnalysisBuffer.end(), FrequencyBin{});
std::fill(mSynthesisBuffer.begin(), mSynthesisBuffer.end(), FrequencyBin{});
std::fill(std::begin(mCurrentGains), std::end(mCurrentGains), 0.0f);
std::fill(std::begin(mTargetGains), std::end(mTargetGains), 0.0f);
}
void PshifterState::update(const ContextBase*, const EffectSlot *slot,
const EffectProps *props, const EffectTarget target)
{
const int tune{props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune};
const float pitch{std::pow(2.0f, static_cast<float>(tune) / 1200.0f)};
mPitchShiftI = fastf2u(pitch*MixerFracOne);
mPitchShift = mPitchShiftI * double{1.0/MixerFracOne};
const auto coeffs = CalcDirectionCoeffs({0.0f, 0.0f, -1.0f}, 0.0f);
mOutTarget = target.Main->Buffer;
ComputePanGains(target.Main, coeffs.data(), slot->Gain, mTargetGains);
}
void PshifterState::process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
{
/* Pitch shifter engine based on the work of Stephan Bernsee.
* http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
*/
/* Cycle offset per update expected of each frequency bin (bin 0 is none,
* bin 1 is x1, bin 2 is x2, etc).
*/
constexpr double expected_cycles{al::numbers::pi*2.0 / OVERSAMP};
for(size_t base{0u};base < samplesToDo;)
{
const size_t todo{minz(STFT_STEP-mCount, samplesToDo-base)};
/* Retrieve the output samples from the FIFO and fill in the new input
* samples.
*/
auto fifo_iter = mFIFO.begin()+mPos + mCount;
std::transform(fifo_iter, fifo_iter+todo, mBufferOut.begin()+base,
[](double d) noexcept -> float { return static_cast<float>(d); });
std::copy_n(samplesIn[0].begin()+base, todo, fifo_iter);
mCount += todo;
base += todo;
/* Check whether FIFO buffer is filled with new samples. */
if(mCount < STFT_STEP) break;
mCount = 0;
mPos = (mPos+STFT_STEP) & (mFIFO.size()-1);
/* Time-domain signal windowing, store in FftBuffer, and apply a
* forward FFT to get the frequency-domain signal.
*/
for(size_t src{mPos}, k{0u};src < STFT_SIZE;++src,++k)
mFftBuffer[k] = mFIFO[src] * HannWindow[k];
for(size_t src{0u}, k{STFT_SIZE-mPos};src < mPos;++src,++k)
mFftBuffer[k] = mFIFO[src] * HannWindow[k];
forward_fft(mFftBuffer);
/* Analyze the obtained data. Since the real FFT is symmetric, only
* STFT_HALF_SIZE+1 samples are needed.
*/
for(size_t k{0u};k < STFT_HALF_SIZE+1;k++)
{
const double amplitude{std::abs(mFftBuffer[k])};
const double phase{std::arg(mFftBuffer[k])};
/* Compute phase difference and subtract expected phase difference */
double tmp{(phase - mLastPhase[k]) - static_cast<double>(k)*expected_cycles};
/* Map delta phase into +/- Pi interval */
int qpd{double2int(tmp / al::numbers::pi)};
tmp -= al::numbers::pi * (qpd + (qpd%2));
/* Get deviation from bin frequency from the +/- Pi interval */
tmp /= expected_cycles;
/* Compute the k-th partials' true frequency and store the
* amplitude and frequency bin in the analysis buffer.
*/
mAnalysisBuffer[k].Amplitude = amplitude;
mAnalysisBuffer[k].FreqBin = static_cast<double>(k) + tmp;
/* Store the actual phase[k] for the next frame. */
mLastPhase[k] = phase;
}
/* Shift the frequency bins according to the pitch adjustment,
* accumulating the amplitudes of overlapping frequency bins.
*/
std::fill(mSynthesisBuffer.begin(), mSynthesisBuffer.end(), FrequencyBin{});
const size_t bin_count{minz(STFT_HALF_SIZE+1,
(((STFT_HALF_SIZE+1)<<MixerFracBits) - (MixerFracOne>>1) - 1)/mPitchShiftI + 1)};
for(size_t k{0u};k < bin_count;k++)
{
const size_t j{(k*mPitchShiftI + (MixerFracOne>>1)) >> MixerFracBits};
mSynthesisBuffer[j].Amplitude += mAnalysisBuffer[k].Amplitude;
mSynthesisBuffer[j].FreqBin = mAnalysisBuffer[k].FreqBin * mPitchShift;
}
/* Reconstruct the frequency-domain signal from the adjusted frequency
* bins.
*/
for(size_t k{0u};k < STFT_HALF_SIZE+1;k++)
{
/* Calculate actual delta phase and accumulate it to get bin phase */
mSumPhase[k] += mSynthesisBuffer[k].FreqBin * expected_cycles;
mFftBuffer[k] = std::polar(mSynthesisBuffer[k].Amplitude, mSumPhase[k]);
}
for(size_t k{STFT_HALF_SIZE+1};k < STFT_SIZE;++k)
mFftBuffer[k] = std::conj(mFftBuffer[STFT_SIZE-k]);
/* Apply an inverse FFT to get the time-domain siganl, and accumulate
* for the output with windowing.
*/
inverse_fft(mFftBuffer);
for(size_t dst{mPos}, k{0u};dst < STFT_SIZE;++dst,++k)
mOutputAccum[dst] += HannWindow[k]*mFftBuffer[k].real() * (4.0/OVERSAMP/STFT_SIZE);
for(size_t dst{0u}, k{STFT_SIZE-mPos};dst < mPos;++dst,++k)
mOutputAccum[dst] += HannWindow[k]*mFftBuffer[k].real() * (4.0/OVERSAMP/STFT_SIZE);
/* Copy out the accumulated result, then clear for the next iteration. */
std::copy_n(mOutputAccum.begin() + mPos, STFT_STEP, mFIFO.begin() + mPos);
std::fill_n(mOutputAccum.begin() + mPos, STFT_STEP, 0.0);
}
/* Now, mix the processed sound data to the output. */
MixSamples({mBufferOut.data(), samplesToDo}, samplesOut, mCurrentGains, mTargetGains,
maxz(samplesToDo, 512), 0);
}
struct PshifterStateFactory final : public EffectStateFactory {
al::intrusive_ptr<EffectState> create() override
{ return al::intrusive_ptr<EffectState>{new PshifterState{}}; }
};
} // namespace
EffectStateFactory *PshifterStateFactory_getFactory()
{
static PshifterStateFactory PshifterFactory{};
return &PshifterFactory;
}