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- /**
- * 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;
- }
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