From 6fd23f09842b81788298e1840b8626252fdf5e18 Mon Sep 17 00:00:00 2001 From: Raulshc <33253777+Raulshc@users.noreply.github.com> Date: Sun, 18 Mar 2018 17:47:17 +0100 Subject: EFX:Pitch Shifter implementation Add pitch shifter effect using standard phase vocoder, based on work of Stephan Bernsee. Only mono signal processing by now. --- Alc/effects/pshifter.c | 493 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 493 insertions(+) create mode 100644 Alc/effects/pshifter.c (limited to 'Alc/effects') diff --git a/Alc/effects/pshifter.c b/Alc/effects/pshifter.c new file mode 100644 index 00000000..1a2ddef7 --- /dev/null +++ b/Alc/effects/pshifter.c @@ -0,0 +1,493 @@ +/** + * 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 +#include + +#include "alMain.h" +#include "alFilter.h" +#include "alAuxEffectSlot.h" +#include "alError.h" +#include "alu.h" + + +typedef struct ALcomplex{ + + ALfloat Real; + ALfloat Imag; + +}ALcomplex; + +typedef struct ALphasor{ + + ALfloat Amplitude; + ALfloat Phase; + +}ALphasor; + +typedef struct ALFrequencyDomain{ + + ALfloat Amplitude; + ALfloat Frequency; + +}ALfrequencyDomain; + +typedef struct ALpshifterState { + DERIVE_FROM_TYPE(ALeffectState); + + /* Effect gains for each channel */ + ALfloat Gain[MAX_OUTPUT_CHANNELS]; + + /* Effect parameters */ + ALsizei count; + ALsizei STFT_size; + ALsizei step; + ALsizei FIFOLatency; + ALsizei oversamp; + ALfloat PitchShift; + ALfloat Frequency; + + /*Effects buffers*/ + ALfloat InFIFO[BUFFERSIZE]; + ALfloat OutFIFO[BUFFERSIZE]; + ALfloat LastPhase[(BUFFERSIZE>>1) +1]; + ALfloat SumPhase[(BUFFERSIZE>>1) +1]; + ALfloat OutputAccum[BUFFERSIZE<<1]; + ALfloat window[BUFFERSIZE]; + + ALcomplex FFTbuffer[BUFFERSIZE]; + + ALfrequencyDomain Analysis_buffer[BUFFERSIZE]; + ALfrequencyDomain Syntesis_buffer[BUFFERSIZE]; + + +} ALpshifterState; + +static inline ALphasor rect2polar( ALcomplex number ); +static inline ALcomplex polar2rect( ALphasor number ); +static inline ALvoid FFT(ALcomplex *FFTBuffer, ALsizei FFTSize, ALint Sign); + +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); + +static void ALpshifterState_Construct(ALpshifterState *state) +{ + ALsizei i; + + ALeffectState_Construct(STATIC_CAST(ALeffectState, state)); + SET_VTABLE2(ALpshifterState, ALeffectState, state); + + /*Initializing parameters and set to zero the buffers */ + state->STFT_size = BUFFERSIZE>>1; + state->oversamp = 1<<2; + + state->step = state->STFT_size / state->oversamp ; + state->FIFOLatency = state->step * ( state->oversamp-1 ); + state->count = state->FIFOLatency; + + memset(state->InFIFO, 0, BUFFERSIZE*sizeof(ALfloat)); + memset(state->OutFIFO, 0, BUFFERSIZE*sizeof(ALfloat)); + memset(state->FFTbuffer, 0, BUFFERSIZE*sizeof(ALcomplex)); + memset(state->LastPhase, 0, ((BUFFERSIZE>>1) +1)*sizeof(ALfloat)); + memset(state->SumPhase, 0, ((BUFFERSIZE>>1) +1)*sizeof(ALfloat)); + memset(state->OutputAccum, 0, (BUFFERSIZE<<1)*sizeof(ALfloat)); + memset(state->Analysis_buffer, 0, BUFFERSIZE*sizeof(ALfrequencyDomain)); + + /* Create lockup table of the Hann window for the desired size, i.e. STFT_size */ + for ( i = 0; i < state->STFT_size>>1 ; i++ ) + { + state->window[i] = state->window[state->STFT_size-(i+1)] \ + = 0.5f * ( 1 - cosf(F_TAU*(ALfloat)i/(ALfloat)(state->STFT_size-1))); + } +} + +static ALvoid ALpshifterState_Destruct(ALpshifterState *state) +{ + ALeffectState_Destruct(STATIC_CAST(ALeffectState,state)); +} + +static ALboolean ALpshifterState_deviceUpdate(ALpshifterState *UNUSED(state), ALCdevice *UNUSED(device)) +{ + 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]; + const ALfloat adjust = 0.707945784384f; /*-3dB adjust*/ + + state->Frequency = (ALfloat)device->Frequency; + state->PitchShift = powf(2.0f,((ALfloat)props->Pshifter.CoarseTune + props->Pshifter.FineTune/100.0f)/12.0f); + + CalcAngleCoeffs(0.0f, 0.0f, 0.0f, coeffs); + ComputeDryPanGains(&device->Dry, coeffs, slot->Params.Gain * adjust, state->Gain); +} + +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/ **/ + + ALsizei i, j, k, STFT_half_size; + ALfloat freq_bin, expected, tmp; + ALfloat bufferOut[BUFFERSIZE]; + ALphasor component; + + + STFT_half_size = state->STFT_size >> 1; + freq_bin = state->Frequency / (ALfloat)state->STFT_size; + expected = F_TAU / (ALfloat)state->oversamp; + + + for (i = 0; i < SamplesToDo; i++) + { + /* Fill FIFO buffer with samples data */ + state->InFIFO[state->count] = SamplesIn[0][i]; + bufferOut[i] = state->OutFIFO[state->count - state->FIFOLatency]; + + state->count++; + + /* Check whether FIFO buffer is filled */ + if ( state->count >= state->STFT_size ) + { + state->count = state->FIFOLatency; + + /* Real signal windowing and store in FFTbuffer */ + for ( k = 0; k < state->STFT_size; k++ ) + { + state->FFTbuffer[k].Real = state->InFIFO[k] * state->window[k]; + state->FFTbuffer[k].Imag = 0.0f; + } + + /* ANALYSIS */ + /* Apply FFT to FFTbuffer data */ + FFT( state->FFTbuffer, state->STFT_size, -1 ); + + /* 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; k++ ) + { + /* Compute amplitude and phase */ + component = rect2polar( state->FFTbuffer[k] ); + + /* Compute phase difference and subtract expected phase difference */ + tmp = ( component.Phase - state->LastPhase[k] ) - (ALfloat)k*expected; + + /* Map delta phase into +/- Pi interval */ + tmp -= F_PI*(ALfloat)( fastf2i(tmp/F_PI) + fastf2i(tmp/F_PI) % 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 = 2.0f * component.Amplitude; + state->Analysis_buffer[k].Frequency = ((ALfloat)k + tmp) * freq_bin; + + /* Store actual phase[k] for the calculations in the next frame*/ + state->LastPhase[k] = component.Phase; + + } + + /* PROCESSING */ + /* pitch shifting */ + memset(state->Syntesis_buffer, 0, state->STFT_size*sizeof(ALfrequencyDomain)); + + for (k = 0; k <= STFT_half_size; k++) + { + j = fastf2i( (ALfloat)k*state->PitchShift ); + + if ( j <= STFT_half_size ) + { + 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; k++ ) + { + /* Compute bin deviation from scaled freq */ + tmp = state->Syntesis_buffer[k].Frequency /freq_bin - (ALfloat)k; + + /* Calculate actual delta phase and accumulate it to get bin phase */ + state->SumPhase[k] += ((ALfloat)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 */ + memset( &state->FFTbuffer[STFT_half_size+1], 0, (STFT_half_size-1) * sizeof(ALcomplex) ); + + /* Apply iFFT to buffer data */ + FFT( state->FFTbuffer, state->STFT_size, 1 ); + + /* Windowing and add to output */ + for( k=0; k < state->STFT_size; k++ ) + { + state->OutputAccum[k] += 2.0f * state->window[k]*state->FFTbuffer[k].Real / (STFT_half_size * state->oversamp); + } + + /* Shift accumulator, input & output FIFO */ + memmove(state->OutFIFO , state->OutputAccum , state->step * sizeof(ALfloat)); + memmove(state->OutputAccum, state->OutputAccum + state->step, state->STFT_size * sizeof(ALfloat)); + memmove(state->InFIFO , state->InFIFO + state->step, state->FIFOLatency * sizeof(ALfloat)); + + } + } + + /* Now, mix the processed sound data to the output*/ + + for (j = 0; j < NumChannels; j++ ) + { + ALfloat gain = state->Gain[j]; + + if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) + continue; + + for(i = 0;i < SamplesToDo;i++) + SamplesOut[j][i] += gain * bufferOut[i]; + + } + + +} + +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); + + +/* Converts ALcomplex to ALphasor*/ +static inline ALphasor rect2polar( ALcomplex number ) +{ + ALphasor polar; + + polar.Amplitude = sqrtf ( number.Real*number.Real + number.Imag*number.Imag ); + polar.Phase = atan2f( number.Imag , number.Real ); + + return polar; +} + +/* Converts ALphasor to ALcomplex*/ +static inline ALcomplex polar2rect( ALphasor number ) +{ + ALcomplex cartesian; + + cartesian.Real = number.Amplitude * cosf( number.Phase ); + cartesian.Imag = number.Amplitude * sinf( number.Phase ); + + return cartesian; +} + +/* Addition of two complex numbers (ALcomplex format)*/ +static inline ALcomplex complex_add( ALcomplex a, ALcomplex b ) +{ + ALcomplex result; + + result.Real = ( a.Real + b.Real ); + result.Imag = ( a.Imag + b.Imag ); + + return result; +} + +/* Substraction of two complex numbers (ALcomplex format)*/ +static inline ALcomplex complex_subst( ALcomplex a, ALcomplex b ) +{ + ALcomplex result; + + result.Real = ( a.Real - b.Real ); + result.Imag = ( a.Imag - b.Imag ); + + return result; +} + +/* Multiplication of two complex numbers (ALcomplex format)*/ +static inline ALcomplex complex_mult( ALcomplex a, ALcomplex b ) +{ + ALcomplex result; + + result.Real = ( a.Real * b.Real - a.Imag * b.Imag ); + result.Imag = ( a.Imag * b.Real + a.Real * b.Imag ); + + return result; +} + +/* Iterative implementation of 2-radix FFT (In-place algorithm). Sign = -1 is FFT and 1 is + iFFT (inverse). Fills FFTBuffer[0...FFTSize-1] with the Discrete Fourier Transform (DFT) + of the time domain data stored in FFTBuffer[0...FFTSize-1]. FFTBuffer is an array of + complex numbers (ALcomplex), FFTSize MUST BE power of two.*/ + +static inline ALvoid FFT(ALcomplex *FFTBuffer, ALsizei FFTSize, ALint Sign) +{ + ALfloat arg; + ALsizei i, j, k, mask, step, step2; + ALcomplex temp, u, w; + + /*bit-reversal permutation applied to a sequence of FFTSize items*/ + for (i = 1; i < FFTSize-1; i++ ) + { + + for ( mask = 0x1, j = 0; mask < FFTSize; mask <<= 1 ) + { + if ( ( i & mask ) != 0 ) j++; + + j <<= 1; + } + + j >>= 1; + + if ( i < j ) + { + temp = FFTBuffer[i]; + FFTBuffer[i] = FFTBuffer[j]; + FFTBuffer[j] = temp; + } + } + + /* Iterative form of Danielson–Lanczos lemma */ + for ( i = 1, step = 2; i < FFTSize; i<<=1, step <<= 1 ) + { + + step2 = step >> 1; + arg = F_PI / step2; + + w.Real = cosf( arg ); + w.Imag = sinf( arg ) * Sign; + + u.Real = 1.0f; + u.Imag = 0.0f; + + for ( j = 0; j < step2; j++ ) + { + + for ( k = j; k < FFTSize; k += step ) + { + + temp = complex_mult( FFTBuffer[k+step2], u ); + FFTBuffer[k+step2] = complex_subst( FFTBuffer[k], temp ); + FFTBuffer[k] = complex_add( FFTBuffer[k], temp ); + } + + u = complex_mult(u,w); + } + } +} -- cgit v1.2.3