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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"
-
- #ifdef HAVE_SSE_INTRINSICS
- #include <emmintrin.h>
- #endif
-
- #include <cmath>
- #include <cstdlib>
- #include <array>
- #include <complex>
- #include <algorithm>
-
- #include "alMain.h"
- #include "alcontext.h"
- #include "alAuxEffectSlot.h"
- #include "alError.h"
- #include "alu.h"
-
- #include "alcomplex.h"
-
-
- namespace {
-
- 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))
-
- inline int double2int(double d)
- {
- #if defined(HAVE_SSE_INTRINSICS)
- return _mm_cvttsd_si32(_mm_set_sd(d));
-
- #elif ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
- !defined(__SSE2_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP < 2)
-
- int sign, shift;
- int64_t mant;
- union {
- double d;
- int64_t i64;
- } conv;
-
- conv.d = d;
- sign = (conv.i64>>63) | 1;
- shift = ((conv.i64>>52)&0x7ff) - (1023+52);
-
- /* Over/underflow */
- if(UNLIKELY(shift >= 63 || shift < -52))
- return 0;
-
- mant = (conv.i64&0xfffffffffffff_i64) | 0x10000000000000_i64;
- if(LIKELY(shift < 0))
- return (int)(mant >> -shift) * sign;
- return (int)(mant << shift) * sign;
-
- #else
-
- return static_cast<int>(d);
- #endif
- }
-
- /* Define a Hann window, used to filter the STFT input and output. */
- /* Making this constexpr seems to require C++14. */
- std::array<ALdouble,STFT_SIZE> InitHannWindow()
- {
- std::array<ALdouble,STFT_SIZE> ret;
- /* Create lookup table of the Hann window for the desired size, i.e. HIL_SIZE */
- for(ALsizei i{0};i < STFT_SIZE>>1;i++)
- {
- ALdouble val = std::sin(al::MathDefs<double>::Pi() * i / ALdouble{STFT_SIZE-1});
- ret[i] = ret[STFT_SIZE-1-i] = val * val;
- }
- return ret;
- }
- alignas(16) const std::array<ALdouble,STFT_SIZE> HannWindow = InitHannWindow();
-
-
- struct ALphasor {
- ALdouble Amplitude;
- ALdouble Phase;
- };
-
- struct ALfrequencyDomain {
- ALdouble Amplitude;
- ALdouble Frequency;
- };
-
-
- /* Converts complex to ALphasor */
- inline ALphasor rect2polar(const complex_d &number)
- {
- ALphasor polar;
- polar.Amplitude = std::abs(number);
- polar.Phase = std::arg(number);
- return polar;
- }
-
- /* Converts ALphasor to complex */
- inline complex_d polar2rect(const ALphasor &number)
- { return std::polar<double>(number.Amplitude, number.Phase); }
-
-
- struct PshifterState final : public EffectState {
- /* Effect parameters */
- ALsizei mCount;
- ALsizei mPitchShiftI;
- ALfloat mPitchShift;
- ALfloat mFreqPerBin;
-
- /* Effects buffers */
- ALfloat mInFIFO[STFT_SIZE];
- ALfloat mOutFIFO[STFT_STEP];
- ALdouble mLastPhase[STFT_HALF_SIZE+1];
- ALdouble mSumPhase[STFT_HALF_SIZE+1];
- ALdouble mOutputAccum[STFT_SIZE];
-
- complex_d mFFTbuffer[STFT_SIZE];
-
- ALfrequencyDomain mAnalysis_buffer[STFT_HALF_SIZE+1];
- ALfrequencyDomain mSyntesis_buffer[STFT_HALF_SIZE+1];
-
- alignas(16) ALfloat mBufferOut[BUFFERSIZE];
-
- /* Effect gains for each output channel */
- ALfloat mCurrentGains[MAX_OUTPUT_CHANNELS];
- ALfloat mTargetGains[MAX_OUTPUT_CHANNELS];
-
-
- ALboolean deviceUpdate(const ALCdevice *device) override;
- void update(const ALCcontext *context, const ALeffectslot *slot, const EffectProps *props, const EffectTarget target) override;
- void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
-
- DEF_NEWDEL(PshifterState)
- };
-
- ALboolean PshifterState::deviceUpdate(const ALCdevice *device)
- {
- /* (Re-)initializing parameters and clear the buffers. */
- mCount = FIFO_LATENCY;
- mPitchShiftI = FRACTIONONE;
- mPitchShift = 1.0f;
- mFreqPerBin = device->Frequency / static_cast<ALfloat>(STFT_SIZE);
-
- std::fill(std::begin(mInFIFO), std::end(mInFIFO), 0.0f);
- std::fill(std::begin(mOutFIFO), std::end(mOutFIFO), 0.0f);
- std::fill(std::begin(mLastPhase), std::end(mLastPhase), 0.0);
- std::fill(std::begin(mSumPhase), std::end(mSumPhase), 0.0);
- std::fill(std::begin(mOutputAccum), std::end(mOutputAccum), 0.0);
- std::fill(std::begin(mFFTbuffer), std::end(mFFTbuffer), complex_d{});
- std::fill(std::begin(mAnalysis_buffer), std::end(mAnalysis_buffer), ALfrequencyDomain{});
- std::fill(std::begin(mSyntesis_buffer), std::end(mSyntesis_buffer), ALfrequencyDomain{});
-
- std::fill(std::begin(mCurrentGains), std::end(mCurrentGains), 0.0f);
- std::fill(std::begin(mTargetGains), std::end(mTargetGains), 0.0f);
-
- return AL_TRUE;
- }
-
- void PshifterState::update(const ALCcontext* UNUSED(context), const ALeffectslot *slot, const EffectProps *props, const EffectTarget target)
- {
- const float pitch{std::pow(2.0f,
- static_cast<ALfloat>(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
- )};
- mPitchShiftI = fastf2i(pitch*FRACTIONONE);
- mPitchShift = mPitchShiftI * (1.0f/FRACTIONONE);
-
- ALfloat coeffs[MAX_AMBI_CHANNELS];
- CalcDirectionCoeffs({0.0f, 0.0f, -1.0f}, 0.0f, coeffs);
-
- mOutBuffer = target.Main->Buffer;
- mOutChannels = target.Main->NumChannels;
- ComputePanGains(target.Main, coeffs, slot->Params.Gain, mTargetGains);
- }
-
- void PshifterState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
- {
- /* Pitch shifter engine based on the work of Stephan Bernsee.
- * http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
- */
-
- static constexpr ALdouble expected{al::MathDefs<double>::Tau() / OVERSAMP};
- const ALdouble freq_per_bin{mFreqPerBin};
- ALfloat *RESTRICT bufferOut{mBufferOut};
- ALsizei count{mCount};
-
- for(ALsizei i{0};i < samplesToDo;)
- {
- do {
- /* Fill FIFO buffer with samples data */
- mInFIFO[count] = samplesIn[0][i];
- bufferOut[i] = mOutFIFO[count - FIFO_LATENCY];
-
- count++;
- } while(++i < samplesToDo && count < STFT_SIZE);
-
- /* Check whether FIFO buffer is filled */
- if(count < STFT_SIZE) break;
- count = FIFO_LATENCY;
-
- /* Real signal windowing and store in FFTbuffer */
- for(ALsizei k{0};k < STFT_SIZE;k++)
- {
- mFFTbuffer[k].real(mInFIFO[k] * HannWindow[k]);
- mFFTbuffer[k].imag(0.0);
- }
-
- /* ANALYSIS */
- /* Apply FFT to FFTbuffer data */
- complex_fft(mFFTbuffer, STFT_SIZE, -1.0);
-
- /* Analyze the obtained data. Since the real FFT is symmetric, only
- * STFT_HALF_SIZE+1 samples are needed.
- */
- for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
- {
- /* Compute amplitude and phase */
- ALphasor component{rect2polar(mFFTbuffer[k])};
-
- /* Compute phase difference and subtract expected phase difference */
- double tmp{(component.Phase - mLastPhase[k]) - k*expected};
-
- /* Map delta phase into +/- Pi interval */
- int qpd{double2int(tmp / al::MathDefs<double>::Pi())};
- tmp -= al::MathDefs<double>::Pi() * (qpd + (qpd%2));
-
- /* Get deviation from bin frequency from the +/- Pi interval */
- tmp /= expected;
-
- /* Compute the k-th partials' true frequency, twice the amplitude
- * for maintain the gain (because half of bins are used) and store
- * amplitude and true frequency in analysis buffer.
- */
- mAnalysis_buffer[k].Amplitude = 2.0 * component.Amplitude;
- mAnalysis_buffer[k].Frequency = (k + tmp) * freq_per_bin;
-
- /* Store actual phase[k] for the calculations in the next frame*/
- mLastPhase[k] = component.Phase;
- }
-
- /* PROCESSING */
- /* pitch shifting */
- for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
- {
- mSyntesis_buffer[k].Amplitude = 0.0;
- mSyntesis_buffer[k].Frequency = 0.0;
- }
-
- for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
- {
- ALsizei j{(k*mPitchShiftI) >> FRACTIONBITS};
- if(j >= STFT_HALF_SIZE+1) break;
-
- mSyntesis_buffer[j].Amplitude += mAnalysis_buffer[k].Amplitude;
- mSyntesis_buffer[j].Frequency = mAnalysis_buffer[k].Frequency * mPitchShift;
- }
-
- /* SYNTHESIS */
- /* Synthesis the processing data */
- for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
- {
- ALphasor component;
- ALdouble tmp;
-
- /* Compute bin deviation from scaled freq */
- tmp = mSyntesis_buffer[k].Frequency/freq_per_bin - k;
-
- /* Calculate actual delta phase and accumulate it to get bin phase */
- mSumPhase[k] += (k + tmp) * expected;
-
- component.Amplitude = mSyntesis_buffer[k].Amplitude;
- component.Phase = mSumPhase[k];
-
- /* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/
- mFFTbuffer[k] = polar2rect(component);
- }
- /* zero negative frequencies for recontruct a real signal */
- for(ALsizei k{STFT_HALF_SIZE+1};k < STFT_SIZE;k++)
- mFFTbuffer[k] = complex_d{};
-
- /* Apply iFFT to buffer data */
- complex_fft(mFFTbuffer, STFT_SIZE, 1.0);
-
- /* Windowing and add to output */
- for(ALsizei k{0};k < STFT_SIZE;k++)
- mOutputAccum[k] += HannWindow[k] * mFFTbuffer[k].real() /
- (0.5 * STFT_HALF_SIZE * OVERSAMP);
-
- /* Shift accumulator, input & output FIFO */
- ALsizei j, k;
- for(k = 0;k < STFT_STEP;k++) mOutFIFO[k] = static_cast<ALfloat>(mOutputAccum[k]);
- for(j = 0;k < STFT_SIZE;k++,j++) mOutputAccum[j] = mOutputAccum[k];
- for(;j < STFT_SIZE;j++) mOutputAccum[j] = 0.0;
- for(k = 0;k < FIFO_LATENCY;k++)
- mInFIFO[k] = mInFIFO[k+STFT_STEP];
- }
- mCount = count;
-
- /* Now, mix the processed sound data to the output. */
- MixSamples(bufferOut, numOutput, samplesOut, mCurrentGains, mTargetGains,
- maxi(samplesToDo, 512), 0, samplesToDo);
- }
-
-
- void Pshifter_setParamf(EffectProps*, ALCcontext *context, ALenum param, ALfloat)
- { alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param); }
- void Pshifter_setParamfv(EffectProps*, ALCcontext *context, ALenum param, const ALfloat*)
- { alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float-vector property 0x%04x", param); }
-
- void Pshifter_setParami(EffectProps *props, ALCcontext *context, ALenum param, ALint val)
- {
- switch(param)
- {
- case AL_PITCH_SHIFTER_COARSE_TUNE:
- if(!(val >= AL_PITCH_SHIFTER_MIN_COARSE_TUNE && val <= AL_PITCH_SHIFTER_MAX_COARSE_TUNE))
- SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter coarse tune out of range");
- props->Pshifter.CoarseTune = val;
- break;
-
- case AL_PITCH_SHIFTER_FINE_TUNE:
- if(!(val >= AL_PITCH_SHIFTER_MIN_FINE_TUNE && val <= AL_PITCH_SHIFTER_MAX_FINE_TUNE))
- SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter fine tune out of range");
- props->Pshifter.FineTune = val;
- break;
-
- default:
- alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
- }
- }
- void Pshifter_setParamiv(EffectProps *props, ALCcontext *context, ALenum param, const ALint *vals)
- { Pshifter_setParami(props, context, param, vals[0]); }
-
- void Pshifter_getParami(const EffectProps *props, ALCcontext *context, ALenum param, ALint *val)
- {
- switch(param)
- {
- case AL_PITCH_SHIFTER_COARSE_TUNE:
- *val = props->Pshifter.CoarseTune;
- break;
- case AL_PITCH_SHIFTER_FINE_TUNE:
- *val = props->Pshifter.FineTune;
- break;
-
- default:
- alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
- }
- }
- void Pshifter_getParamiv(const EffectProps *props, ALCcontext *context, ALenum param, ALint *vals)
- { Pshifter_getParami(props, context, param, vals); }
-
- void Pshifter_getParamf(const EffectProps*, ALCcontext *context, ALenum param, ALfloat*)
- { alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param); }
- void Pshifter_getParamfv(const EffectProps*, ALCcontext *context, ALenum param, ALfloat*)
- { alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float vector-property 0x%04x", param); }
-
- DEFINE_ALEFFECT_VTABLE(Pshifter);
-
-
- struct PshifterStateFactory final : public EffectStateFactory {
- EffectState *create() override;
- EffectProps getDefaultProps() const noexcept override;
- const EffectVtable *getEffectVtable() const noexcept override { return &Pshifter_vtable; }
- };
-
- EffectState *PshifterStateFactory::create()
- { return new PshifterState{}; }
-
- EffectProps PshifterStateFactory::getDefaultProps() const noexcept
- {
- EffectProps props{};
- props.Pshifter.CoarseTune = AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE;
- props.Pshifter.FineTune = AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE;
- return props;
- }
-
- } // namespace
-
- EffectStateFactory *PshifterStateFactory_getFactory()
- {
- static PshifterStateFactory PshifterFactory{};
- return &PshifterFactory;
- }
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