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-rw-r--r--Alc/effects/pshifter.c441
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diff --git a/Alc/effects/pshifter.c b/Alc/effects/pshifter.c
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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 <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-#include "filters/defs.h"
-
-#include "alcomplex.h"
-
-
-#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))
-
-
-typedef struct ALphasor {
- ALdouble Amplitude;
- ALdouble Phase;
-} ALphasor;
-
-typedef struct ALFrequencyDomain {
- ALdouble Amplitude;
- ALdouble Frequency;
-} ALfrequencyDomain;
-
-
-typedef struct ALpshifterState {
- DERIVE_FROM_TYPE(ALeffectState);
-
- /* Effect parameters */
- ALsizei count;
- ALsizei PitchShiftI;
- ALfloat PitchShift;
- ALfloat FreqPerBin;
-
- /*Effects buffers*/
- ALfloat InFIFO[STFT_SIZE];
- ALfloat OutFIFO[STFT_STEP];
- ALdouble LastPhase[STFT_HALF_SIZE+1];
- ALdouble SumPhase[STFT_HALF_SIZE+1];
- ALdouble OutputAccum[STFT_SIZE];
-
- ALcomplex FFTbuffer[STFT_SIZE];
-
- ALfrequencyDomain Analysis_buffer[STFT_HALF_SIZE+1];
- ALfrequencyDomain Syntesis_buffer[STFT_HALF_SIZE+1];
-
- alignas(16) ALfloat BufferOut[BUFFERSIZE];
-
- /* Effect gains for each output channel */
- ALfloat CurrentGains[MAX_OUTPUT_CHANNELS];
- ALfloat TargetGains[MAX_OUTPUT_CHANNELS];
-} ALpshifterState;
-
-static ALvoid ALpshifterState_Destruct(ALpshifterState *state);
-static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device);
-static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props);
-static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
-DECLARE_DEFAULT_ALLOCATORS(ALpshifterState)
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALpshifterState);
-
-
-/* Define a Hann window, used to filter the STFT input and output. */
-alignas(16) static ALdouble HannWindow[STFT_SIZE];
-
-static void InitHannWindow(void)
-{
- ALsizei i;
-
- /* Create lookup table of the Hann window for the desired size, i.e. STFT_SIZE */
- for(i = 0;i < STFT_SIZE>>1;i++)
- {
- ALdouble val = sin(M_PI * (ALdouble)i / (ALdouble)(STFT_SIZE-1));
- HannWindow[i] = HannWindow[STFT_SIZE-1-i] = val * val;
- }
-}
-static alonce_flag HannInitOnce = AL_ONCE_FLAG_INIT;
-
-
-static inline ALint double2int(ALdouble d)
-{
-#if ((defined(__GNUC__) || defined(__clang__)) && (defined(__i386__) || defined(__x86_64__)) && \
- !defined(__SSE2_MATH__)) || (defined(_MSC_VER) && defined(_M_IX86_FP) && _M_IX86_FP < 2)
- ALint sign, shift;
- ALint64 mant;
- union {
- ALdouble d;
- ALint64 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&I64(0xfffffffffffff)) | I64(0x10000000000000);
- if(LIKELY(shift < 0))
- return (ALint)(mant >> -shift) * sign;
- return (ALint)(mant << shift) * sign;
-
-#else
-
- return (ALint)d;
-#endif
-}
-
-
-/* Converts ALcomplex to ALphasor */
-static inline ALphasor rect2polar(ALcomplex number)
-{
- ALphasor polar;
-
- polar.Amplitude = sqrt(number.Real*number.Real + number.Imag*number.Imag);
- polar.Phase = atan2(number.Imag, number.Real);
-
- return polar;
-}
-
-/* Converts ALphasor to ALcomplex */
-static inline ALcomplex polar2rect(ALphasor number)
-{
- ALcomplex cartesian;
-
- cartesian.Real = number.Amplitude * cos(number.Phase);
- cartesian.Imag = number.Amplitude * sin(number.Phase);
-
- return cartesian;
-}
-
-
-static void ALpshifterState_Construct(ALpshifterState *state)
-{
- ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
- SET_VTABLE2(ALpshifterState, ALeffectState, state);
-
- alcall_once(&HannInitOnce, InitHannWindow);
-}
-
-static ALvoid ALpshifterState_Destruct(ALpshifterState *state)
-{
- ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
-}
-
-static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *state, ALCdevice *device)
-{
- /* (Re-)initializing parameters and clear the buffers. */
- state->count = FIFO_LATENCY;
- state->PitchShiftI = FRACTIONONE;
- state->PitchShift = 1.0f;
- state->FreqPerBin = device->Frequency / (ALfloat)STFT_SIZE;
-
- memset(state->InFIFO, 0, sizeof(state->InFIFO));
- memset(state->OutFIFO, 0, sizeof(state->OutFIFO));
- memset(state->FFTbuffer, 0, sizeof(state->FFTbuffer));
- memset(state->LastPhase, 0, sizeof(state->LastPhase));
- memset(state->SumPhase, 0, sizeof(state->SumPhase));
- memset(state->OutputAccum, 0, sizeof(state->OutputAccum));
- memset(state->Analysis_buffer, 0, sizeof(state->Analysis_buffer));
- memset(state->Syntesis_buffer, 0, sizeof(state->Syntesis_buffer));
-
- memset(state->CurrentGains, 0, sizeof(state->CurrentGains));
- memset(state->TargetGains, 0, sizeof(state->TargetGains));
-
- return AL_TRUE;
-}
-
-static ALvoid ALpshifterState_update(ALpshifterState *state, const ALCcontext *context, const ALeffectslot *slot, const ALeffectProps *props)
-{
- const ALCdevice *device = context->Device;
- ALfloat coeffs[MAX_AMBI_COEFFS];
- float pitch;
-
- pitch = powf(2.0f,
- (ALfloat)(props->Pshifter.CoarseTune*100 + props->Pshifter.FineTune) / 1200.0f
- );
- state->PitchShiftI = fastf2i(pitch*FRACTIONONE);
- state->PitchShift = state->PitchShiftI * (1.0f/FRACTIONONE);
-
- CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs);
- ComputePanGains(&device->Dry, coeffs, slot->Params.Gain, state->TargetGains);
-}
-
-static ALvoid ALpshifterState_process(ALpshifterState *state, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
-{
- /* Pitch shifter engine based on the work of Stephan Bernsee.
- * http://blogs.zynaptiq.com/bernsee/pitch-shifting-using-the-ft/
- */
-
- static const ALdouble expected = M_PI*2.0 / OVERSAMP;
- const ALdouble freq_per_bin = state->FreqPerBin;
- ALfloat *restrict bufferOut = state->BufferOut;
- ALsizei count = state->count;
- ALsizei i, j, k;
-
- for(i = 0;i < SamplesToDo;)
- {
- do {
- /* Fill FIFO buffer with samples data */
- state->InFIFO[count] = SamplesIn[0][i];
- bufferOut[i] = state->OutFIFO[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(k = 0;k < STFT_SIZE;k++)
- {
- state->FFTbuffer[k].Real = state->InFIFO[k] * HannWindow[k];
- state->FFTbuffer[k].Imag = 0.0;
- }
-
- /* ANALYSIS */
- /* Apply FFT to FFTbuffer data */
- complex_fft(state->FFTbuffer, STFT_SIZE, -1.0);
-
- /* Analyze the obtained data. Since the real FFT is symmetric, only
- * STFT_HALF_SIZE+1 samples are needed.
- */
- for(k = 0;k < STFT_HALF_SIZE+1;k++)
- {
- ALphasor component;
- ALdouble tmp;
- ALint qpd;
-
- /* Compute amplitude and phase */
- component = rect2polar(state->FFTbuffer[k]);
-
- /* Compute phase difference and subtract expected phase difference */
- tmp = (component.Phase - state->LastPhase[k]) - k*expected;
-
- /* Map delta phase into +/- Pi interval */
- qpd = double2int(tmp / M_PI);
- tmp -= M_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.
- */
- state->Analysis_buffer[k].Amplitude = 2.0 * component.Amplitude;
- state->Analysis_buffer[k].Frequency = (k + tmp) * freq_per_bin;
-
- /* Store actual phase[k] for the calculations in the next frame*/
- state->LastPhase[k] = component.Phase;
- }
-
- /* PROCESSING */
- /* pitch shifting */
- for(k = 0;k < STFT_HALF_SIZE+1;k++)
- {
- state->Syntesis_buffer[k].Amplitude = 0.0;
- state->Syntesis_buffer[k].Frequency = 0.0;
- }
-
- for(k = 0;k < STFT_HALF_SIZE+1;k++)
- {
- j = (k*state->PitchShiftI) >> FRACTIONBITS;
- if(j >= STFT_HALF_SIZE+1) break;
-
- state->Syntesis_buffer[j].Amplitude += state->Analysis_buffer[k].Amplitude;
- state->Syntesis_buffer[j].Frequency = state->Analysis_buffer[k].Frequency *
- state->PitchShift;
- }
-
- /* SYNTHESIS */
- /* Synthesis the processing data */
- for(k = 0;k < STFT_HALF_SIZE+1;k++)
- {
- ALphasor component;
- ALdouble tmp;
-
- /* Compute bin deviation from scaled freq */
- tmp = state->Syntesis_buffer[k].Frequency/freq_per_bin - k;
-
- /* Calculate actual delta phase and accumulate it to get bin phase */
- state->SumPhase[k] += (k + tmp) * expected;
-
- component.Amplitude = state->Syntesis_buffer[k].Amplitude;
- component.Phase = state->SumPhase[k];
-
- /* Compute phasor component to cartesian complex number and storage it into FFTbuffer*/
- state->FFTbuffer[k] = polar2rect(component);
- }
- /* zero negative frequencies for recontruct a real signal */
- for(k = STFT_HALF_SIZE+1;k < STFT_SIZE;k++)
- {
- state->FFTbuffer[k].Real = 0.0;
- state->FFTbuffer[k].Imag = 0.0;
- }
-
- /* Apply iFFT to buffer data */
- complex_fft(state->FFTbuffer, STFT_SIZE, 1.0);
-
- /* Windowing and add to output */
- for(k = 0;k < STFT_SIZE;k++)
- state->OutputAccum[k] += HannWindow[k] * state->FFTbuffer[k].Real /
- (0.5 * STFT_HALF_SIZE * OVERSAMP);
-
- /* Shift accumulator, input & output FIFO */
- for(k = 0;k < STFT_STEP;k++) state->OutFIFO[k] = (ALfloat)state->OutputAccum[k];
- for(j = 0;k < STFT_SIZE;k++,j++) state->OutputAccum[j] = state->OutputAccum[k];
- for(;j < STFT_SIZE;j++) state->OutputAccum[j] = 0.0;
- for(k = 0;k < FIFO_LATENCY;k++)
- state->InFIFO[k] = state->InFIFO[k+STFT_STEP];
- }
- state->count = count;
-
- /* Now, mix the processed sound data to the output. */
- MixSamples(bufferOut, NumChannels, SamplesOut, state->CurrentGains, state->TargetGains,
- maxi(SamplesToDo, 512), 0, SamplesToDo);
-}
-
-typedef struct PshifterStateFactory {
- DERIVE_FROM_TYPE(EffectStateFactory);
-} PshifterStateFactory;
-
-static ALeffectState *PshifterStateFactory_create(PshifterStateFactory *UNUSED(factory))
-{
- ALpshifterState *state;
-
- NEW_OBJ0(state, ALpshifterState)();
- if(!state) return NULL;
-
- return STATIC_CAST(ALeffectState, state);
-}
-
-DEFINE_EFFECTSTATEFACTORY_VTABLE(PshifterStateFactory);
-
-EffectStateFactory *PshifterStateFactory_getFactory(void)
-{
- static PshifterStateFactory PshifterFactory = { { GET_VTABLE2(PshifterStateFactory, EffectStateFactory) } };
-
- return STATIC_CAST(EffectStateFactory, &PshifterFactory);
-}
-
-
-void ALpshifter_setParamf(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat UNUSED(val))
-{
- alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param );
-}
-
-void ALpshifter_setParamfv(ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, const ALfloat *UNUSED(vals))
-{
- alSetError( context, AL_INVALID_ENUM, "Invalid pitch shifter float-vector property 0x%04x", param );
-}
-
-void ALpshifter_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
-{
- ALeffectProps *props = &effect->Props;
- 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 ALpshifter_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
- ALpshifter_setParami(effect, context, param, vals[0]);
-}
-
-void ALpshifter_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
-{
- const ALeffectProps *props = &effect->Props;
- switch(param)
- {
- case AL_PITCH_SHIFTER_COARSE_TUNE:
- *val = (ALint)props->Pshifter.CoarseTune;
- break;
- case AL_PITCH_SHIFTER_FINE_TUNE:
- *val = (ALint)props->Pshifter.FineTune;
- break;
-
- default:
- alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter integer property 0x%04x", param);
- }
-}
-void ALpshifter_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
- ALpshifter_getParami(effect, context, param, vals);
-}
-
-void ALpshifter_getParamf(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(val))
-{
- alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float property 0x%04x", param);
-}
-
-void ALpshifter_getParamfv(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum param, ALfloat *UNUSED(vals))
-{
- alSetError(context, AL_INVALID_ENUM, "Invalid pitch shifter float vector-property 0x%04x", param);
-}
-
-DEFINE_ALEFFECT_VTABLE(ALpshifter);