Rename Alc to alc

1 files changed, 0 insertions, 405 deletions

diff --git a/Alc/effects/pshifter.cpp b/Alc/effects/pshifter.cpp
deleted file mode 100644
index 39d3cf1a..00000000
--- a/Alc/effects/pshifter.cpp
+++ /dev/null
@@ -1,405 +0,0 @@
-/**
- * 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 "alcmain.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(const ALsizei samplesToDo, const FloatBufferLine *RESTRICT samplesIn, const ALsizei numInput, const al::span<FloatBufferLine> samplesOut) 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*, 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);
-
-    mOutTarget = target.Main->Buffer;
-    ComputePanGains(target.Main, coeffs, slot->Params.Gain, mTargetGains);
-}
-
-void PshifterState::process(const ALsizei samplesToDo, const FloatBufferLine *RESTRICT samplesIn, const ALsizei /*numInput*/, 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/
-     */
-
-    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, -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, 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, 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;
-}