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406 lines
14 KiB
406 lines
14 KiB
/**
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* OpenAL cross platform audio library
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* Copyright (C) 2018 by Raul Herraiz.
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#ifdef HAVE_SSE_INTRINSICS
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#include <emmintrin.h>
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#endif
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#include <cmath>
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#include <cstdlib>
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#include <array>
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#include <complex>
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#include <algorithm>
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#include "alMain.h"
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#include "alcontext.h"
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#include "alAuxEffectSlot.h"
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#include "alError.h"
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#include "alu.h"
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#include "alcomplex.h"
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namespace {
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using complex_d = std::complex<double>;
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#define STFT_SIZE 1024
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#define STFT_HALF_SIZE (STFT_SIZE>>1)
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#define OVERSAMP (1<<2)
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#define STFT_STEP (STFT_SIZE / OVERSAMP)
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#define FIFO_LATENCY (STFT_STEP * (OVERSAMP-1))
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inline int double2int(double d)
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{
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#if defined(HAVE_SSE_INTRINSICS)
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return _mm_cvttsd_si32(_mm_set_sd(d));
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#elif ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
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!defined(__SSE2_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP < 2)
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int sign, shift;
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int64_t mant;
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union {
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double d;
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int64_t i64;
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} conv;
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conv.d = d;
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sign = (conv.i64>>63) | 1;
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shift = ((conv.i64>>52)&0x7ff) - (1023+52);
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/* Over/underflow */
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if(UNLIKELY(shift >= 63 || shift < -52))
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return 0;
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mant = (conv.i64&0xfffffffffffff_i64) | 0x10000000000000_i64;
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if(LIKELY(shift < 0))
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return (int)(mant >> -shift) * sign;
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return (int)(mant << shift) * sign;
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#else
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return static_cast<int>(d);
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#endif
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}
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/* Define a Hann window, used to filter the STFT input and output. */
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/* Making this constexpr seems to require C++14. */
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std::array<ALdouble,STFT_SIZE> InitHannWindow()
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{
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std::array<ALdouble,STFT_SIZE> ret;
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/* Create lookup table of the Hann window for the desired size, i.e. HIL_SIZE */
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for(ALsizei i{0};i < STFT_SIZE>>1;i++)
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{
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ALdouble val = std::sin(al::MathDefs<double>::Pi() * i / ALdouble{STFT_SIZE-1});
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ret[i] = ret[STFT_SIZE-1-i] = val * val;
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}
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return ret;
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}
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alignas(16) const std::array<ALdouble,STFT_SIZE> HannWindow = InitHannWindow();
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struct ALphasor {
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ALdouble Amplitude;
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ALdouble Phase;
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};
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struct ALfrequencyDomain {
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ALdouble Amplitude;
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ALdouble Frequency;
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};
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/* Converts complex to ALphasor */
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inline ALphasor rect2polar(const complex_d &number)
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{
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ALphasor polar;
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polar.Amplitude = std::abs(number);
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polar.Phase = std::arg(number);
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return polar;
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}
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/* Converts ALphasor to complex */
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inline complex_d polar2rect(const ALphasor &number)
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{ return std::polar<double>(number.Amplitude, number.Phase); }
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struct PshifterState final : public EffectState {
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/* Effect parameters */
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ALsizei mCount;
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ALsizei mPitchShiftI;
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ALfloat mPitchShift;
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ALfloat mFreqPerBin;
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/* Effects buffers */
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ALfloat mInFIFO[STFT_SIZE];
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ALfloat mOutFIFO[STFT_STEP];
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ALdouble mLastPhase[STFT_HALF_SIZE+1];
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ALdouble mSumPhase[STFT_HALF_SIZE+1];
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ALdouble mOutputAccum[STFT_SIZE];
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complex_d mFFTbuffer[STFT_SIZE];
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ALfrequencyDomain mAnalysis_buffer[STFT_HALF_SIZE+1];
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ALfrequencyDomain mSyntesis_buffer[STFT_HALF_SIZE+1];
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alignas(16) ALfloat mBufferOut[BUFFERSIZE];
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/* Effect gains for each output channel */
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ALfloat mCurrentGains[MAX_OUTPUT_CHANNELS];
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ALfloat mTargetGains[MAX_OUTPUT_CHANNELS];
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ALboolean deviceUpdate(const ALCdevice *device) override;
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void update(const ALCcontext *context, const ALeffectslot *slot, const EffectProps *props, const EffectTarget target) override;
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void process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei numInput, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput) override;
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DEF_NEWDEL(PshifterState)
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};
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ALboolean PshifterState::deviceUpdate(const ALCdevice *device)
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{
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/* (Re-)initializing parameters and clear the buffers. */
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mCount = FIFO_LATENCY;
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mPitchShiftI = FRACTIONONE;
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mPitchShift = 1.0f;
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mFreqPerBin = device->Frequency / static_cast<ALfloat>(STFT_SIZE);
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std::fill(std::begin(mInFIFO), std::end(mInFIFO), 0.0f);
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std::fill(std::begin(mOutFIFO), std::end(mOutFIFO), 0.0f);
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std::fill(std::begin(mLastPhase), std::end(mLastPhase), 0.0);
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std::fill(std::begin(mSumPhase), std::end(mSumPhase), 0.0);
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std::fill(std::begin(mOutputAccum), std::end(mOutputAccum), 0.0);
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std::fill(std::begin(mFFTbuffer), std::end(mFFTbuffer), complex_d{});
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std::fill(std::begin(mAnalysis_buffer), std::end(mAnalysis_buffer), ALfrequencyDomain{});
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std::fill(std::begin(mSyntesis_buffer), std::end(mSyntesis_buffer), ALfrequencyDomain{});
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std::fill(std::begin(mCurrentGains), std::end(mCurrentGains), 0.0f);
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std::fill(std::begin(mTargetGains), std::end(mTargetGains), 0.0f);
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return AL_TRUE;
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}
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void PshifterState::update(const ALCcontext* UNUSED(context), const ALeffectslot *slot, const EffectProps *props, const EffectTarget target)
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{
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const float pitch{std::pow(2.0f,
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static_cast<ALfloat>(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
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)};
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mPitchShiftI = fastf2i(pitch*FRACTIONONE);
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mPitchShift = mPitchShiftI * (1.0f/FRACTIONONE);
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ALfloat coeffs[MAX_AMBI_CHANNELS];
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CalcDirectionCoeffs({0.0f, 0.0f, -1.0f}, 0.0f, coeffs);
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mOutBuffer = target.Main->Buffer;
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mOutChannels = target.Main->NumChannels;
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ComputePanGains(target.Main, coeffs, slot->Params.Gain, mTargetGains);
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}
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void PshifterState::process(ALsizei samplesToDo, const ALfloat (*RESTRICT samplesIn)[BUFFERSIZE], const ALsizei /*numInput*/, ALfloat (*RESTRICT samplesOut)[BUFFERSIZE], const ALsizei numOutput)
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{
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/* Pitch shifter engine based on the work of Stephan Bernsee.
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* http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
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*/
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static constexpr ALdouble expected{al::MathDefs<double>::Tau() / OVERSAMP};
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const ALdouble freq_per_bin{mFreqPerBin};
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ALfloat *RESTRICT bufferOut{mBufferOut};
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ALsizei count{mCount};
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for(ALsizei i{0};i < samplesToDo;)
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{
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do {
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/* Fill FIFO buffer with samples data */
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mInFIFO[count] = samplesIn[0][i];
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bufferOut[i] = mOutFIFO[count - FIFO_LATENCY];
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count++;
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} while(++i < samplesToDo && count < STFT_SIZE);
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/* Check whether FIFO buffer is filled */
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if(count < STFT_SIZE) break;
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count = FIFO_LATENCY;
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/* Real signal windowing and store in FFTbuffer */
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for(ALsizei k{0};k < STFT_SIZE;k++)
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{
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mFFTbuffer[k].real(mInFIFO[k] * HannWindow[k]);
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mFFTbuffer[k].imag(0.0);
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}
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/* ANALYSIS */
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/* Apply FFT to FFTbuffer data */
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complex_fft(mFFTbuffer, STFT_SIZE, -1.0);
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/* Analyze the obtained data. Since the real FFT is symmetric, only
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* STFT_HALF_SIZE+1 samples are needed.
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*/
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for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
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{
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/* Compute amplitude and phase */
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ALphasor component{rect2polar(mFFTbuffer[k])};
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/* Compute phase difference and subtract expected phase difference */
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double tmp{(component.Phase - mLastPhase[k]) - k*expected};
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/* Map delta phase into +/- Pi interval */
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int qpd{double2int(tmp / al::MathDefs<double>::Pi())};
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tmp -= al::MathDefs<double>::Pi() * (qpd + (qpd%2));
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/* Get deviation from bin frequency from the +/- Pi interval */
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tmp /= expected;
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/* Compute the k-th partials' true frequency, twice the amplitude
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* for maintain the gain (because half of bins are used) and store
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* amplitude and true frequency in analysis buffer.
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*/
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mAnalysis_buffer[k].Amplitude = 2.0 * component.Amplitude;
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mAnalysis_buffer[k].Frequency = (k + tmp) * freq_per_bin;
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/* Store actual phase[k] for the calculations in the next frame*/
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mLastPhase[k] = component.Phase;
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}
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/* PROCESSING */
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/* pitch shifting */
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for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
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{
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mSyntesis_buffer[k].Amplitude = 0.0;
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mSyntesis_buffer[k].Frequency = 0.0;
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}
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for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
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{
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ALsizei j{(k*mPitchShiftI) >> FRACTIONBITS};
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if(j >= STFT_HALF_SIZE+1) break;
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mSyntesis_buffer[j].Amplitude += mAnalysis_buffer[k].Amplitude;
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mSyntesis_buffer[j].Frequency = mAnalysis_buffer[k].Frequency * mPitchShift;
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}
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/* SYNTHESIS */
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/* Synthesis the processing data */
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for(ALsizei k{0};k < STFT_HALF_SIZE+1;k++)
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{
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ALphasor component;
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ALdouble tmp;
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/* Compute bin deviation from scaled freq */
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tmp = mSyntesis_buffer[k].Frequency/freq_per_bin - k;
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/* Calculate actual delta phase and accumulate it to get bin phase */
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mSumPhase[k] += (k + tmp) * expected;
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component.Amplitude = mSyntesis_buffer[k].Amplitude;
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component.Phase = mSumPhase[k];
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/* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/
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mFFTbuffer[k] = polar2rect(component);
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}
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/* zero negative frequencies for recontruct a real signal */
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for(ALsizei k{STFT_HALF_SIZE+1};k < STFT_SIZE;k++)
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mFFTbuffer[k] = complex_d{};
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/* Apply iFFT to buffer data */
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complex_fft(mFFTbuffer, STFT_SIZE, 1.0);
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/* Windowing and add to output */
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for(ALsizei k{0};k < STFT_SIZE;k++)
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mOutputAccum[k] += HannWindow[k] * mFFTbuffer[k].real() /
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(0.5 * STFT_HALF_SIZE * OVERSAMP);
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/* Shift accumulator, input & output FIFO */
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ALsizei j, k;
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for(k = 0;k < STFT_STEP;k++) mOutFIFO[k] = static_cast<ALfloat>(mOutputAccum[k]);
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for(j = 0;k < STFT_SIZE;k++,j++) mOutputAccum[j] = mOutputAccum[k];
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for(;j < STFT_SIZE;j++) mOutputAccum[j] = 0.0;
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for(k = 0;k < FIFO_LATENCY;k++)
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mInFIFO[k] = mInFIFO[k+STFT_STEP];
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}
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mCount = count;
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/* Now, mix the processed sound data to the output. */
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MixSamples(bufferOut, numOutput, samplesOut, mCurrentGains, mTargetGains,
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maxi(samplesToDo, 512), 0, samplesToDo);
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}
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void Pshifter_setParamf(EffectProps*, ALCcontext *context, ALenum param, ALfloat)
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{ alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param); }
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void Pshifter_setParamfv(EffectProps*, ALCcontext *context, ALenum param, const ALfloat*)
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{ alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float-vector property 0x%04x", param); }
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void Pshifter_setParami(EffectProps *props, ALCcontext *context, ALenum param, ALint val)
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{
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switch(param)
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{
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case AL_PITCH_SHIFTER_COARSE_TUNE:
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if(!(val >= AL_PITCH_SHIFTER_MIN_COARSE_TUNE && val <= AL_PITCH_SHIFTER_MAX_COARSE_TUNE))
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SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter coarse tune out of range");
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props->Pshifter.CoarseTune = val;
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break;
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case AL_PITCH_SHIFTER_FINE_TUNE:
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if(!(val >= AL_PITCH_SHIFTER_MIN_FINE_TUNE && val <= AL_PITCH_SHIFTER_MAX_FINE_TUNE))
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SETERR_RETURN(context, AL_INVALID_VALUE,,"Pitch shifter fine tune out of range");
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props->Pshifter.FineTune = val;
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break;
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default:
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alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
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}
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}
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void Pshifter_setParamiv(EffectProps *props, ALCcontext *context, ALenum param, const ALint *vals)
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{ Pshifter_setParami(props, context, param, vals[0]); }
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void Pshifter_getParami(const EffectProps *props, ALCcontext *context, ALenum param, ALint *val)
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{
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switch(param)
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{
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case AL_PITCH_SHIFTER_COARSE_TUNE:
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*val = props->Pshifter.CoarseTune;
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break;
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case AL_PITCH_SHIFTER_FINE_TUNE:
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*val = props->Pshifter.FineTune;
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break;
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default:
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alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
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}
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}
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void Pshifter_getParamiv(const EffectProps *props, ALCcontext *context, ALenum param, ALint *vals)
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{ Pshifter_getParami(props, context, param, vals); }
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void Pshifter_getParamf(const EffectProps*, ALCcontext *context, ALenum param, ALfloat*)
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{ alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param); }
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void Pshifter_getParamfv(const EffectProps*, ALCcontext *context, ALenum param, ALfloat*)
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{ alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float vector-property 0x%04x", param); }
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DEFINE_ALEFFECT_VTABLE(Pshifter);
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struct PshifterStateFactory final : public EffectStateFactory {
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EffectState *create() override;
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EffectProps getDefaultProps() const noexcept override;
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const EffectVtable *getEffectVtable() const noexcept override { return &Pshifter_vtable; }
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};
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EffectState *PshifterStateFactory::create()
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{ return new PshifterState{}; }
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EffectProps PshifterStateFactory::getDefaultProps() const noexcept
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{
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EffectProps props{};
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props.Pshifter.CoarseTune = AL_PITCH_SHIFTER_DEFAULT_COARSE_TUNE;
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props.Pshifter.FineTune = AL_PITCH_SHIFTER_DEFAULT_FINE_TUNE;
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return props;
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}
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} // namespace
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EffectStateFactory *PshifterStateFactory_getFactory()
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{
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static PshifterStateFactory PshifterFactory{};
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return &PshifterFactory;
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}
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