From ecc51c8c55f01e961bd2e82f8c408ce3c5a525d4 Mon Sep 17 00:00:00 2001 From: Chris Robinson Date: Wed, 21 Mar 2018 08:56:26 -0700 Subject: Clean up some code formatting in the pitch shifter source Clean up excessive newlines and extra-long comments, move static inline definitions to their declarations. --- Alc/effects/pshifter.c | 972 ++++++++++++++++++++++++------------------------- 1 file changed, 478 insertions(+), 494 deletions(-) (limited to 'Alc') diff --git a/Alc/effects/pshifter.c b/Alc/effects/pshifter.c index 2bf911f8..a9921028 100644 --- a/Alc/effects/pshifter.c +++ b/Alc/effects/pshifter.c @@ -1,494 +1,478 @@ -/** - * 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" - -#define MAX_SIZE 2048 - -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[MAX_SIZE]; - ALfloat OutFIFO[MAX_SIZE]; - ALfloat LastPhase[(MAX_SIZE>>1) +1]; - ALfloat SumPhase[(MAX_SIZE>>1) +1]; - ALfloat OutputAccum[MAX_SIZE<<1]; - ALfloat window[MAX_SIZE]; - - ALcomplex FFTbuffer[MAX_SIZE]; - - ALfrequencyDomain Analysis_buffer[MAX_SIZE]; - ALfrequencyDomain Syntesis_buffer[MAX_SIZE]; - - -} 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 = MAX_SIZE>>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, MAX_SIZE*sizeof(ALfloat)); - memset(state->OutFIFO, 0, MAX_SIZE*sizeof(ALfloat)); - memset(state->FFTbuffer, 0, MAX_SIZE*sizeof(ALcomplex)); - memset(state->LastPhase, 0, ((MAX_SIZE>>1) +1)*sizeof(ALfloat)); - memset(state->SumPhase, 0, ((MAX_SIZE>>1) +1)*sizeof(ALfloat)); - memset(state->OutputAccum, 0, (MAX_SIZE<<1)*sizeof(ALfloat)); - memset(state->Analysis_buffer, 0, MAX_SIZE*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); - } - } -} +/** + * 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" + +#define MAX_SIZE 2048 + +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[MAX_SIZE]; + ALfloat OutFIFO[MAX_SIZE]; + ALfloat LastPhase[(MAX_SIZE>>1) +1]; + ALfloat SumPhase[(MAX_SIZE>>1) +1]; + ALfloat OutputAccum[MAX_SIZE<<1]; + ALfloat window[MAX_SIZE]; + + ALcomplex FFTbuffer[MAX_SIZE]; + + ALfrequencyDomain Analysis_buffer[MAX_SIZE]; + ALfrequencyDomain Syntesis_buffer[MAX_SIZE]; +} 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); + + +/* 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); + } + } +} + + +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 = MAX_SIZE>>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, 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)); + + /* 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); -- cgit v1.2.3