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-/**
- * OpenAL cross platform audio library
- * Copyright (C) 2013 by Mike Gorchak
- * 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., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- * Or go to http://www.gnu.org/copyleft/lgpl.html
- */
-
-#include "config.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "alMain.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
-#include "alError.h"
-#include "alu.h"
-
-
-typedef struct ALequalizerStateFactory {
- DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALequalizerStateFactory;
-
-static ALequalizerStateFactory EqualizerFactory;
-
-
-/* The document "Effects Extension Guide.pdf" says that low and high *
- * frequencies are cutoff frequencies. This is not fully correct, they *
- * are corner frequencies for low and high shelf filters. If they were *
- * just cutoff frequencies, there would be no need in cutoff frequency *
- * gains, which are present. Documentation for "Creative Proteus X2" *
- * software describes 4-band equalizer functionality in a much better *
- * way. This equalizer seems to be a predecessor of OpenAL 4-band *
- * equalizer. With low and high shelf filters we are able to cutoff *
- * frequencies below and/or above corner frequencies using attenuation *
- * gains (below 1.0) and amplify all low and/or high frequencies using *
- * gains above 1.0. *
- * *
- * Low-shelf Low Mid Band High Mid Band High-shelf *
- * corner center center corner *
- * frequency frequency frequency frequency *
- * 50Hz..800Hz 200Hz..3000Hz 1000Hz..8000Hz 4000Hz..16000Hz *
- * *
- * | | | | *
- * | | | | *
- * B -----+ /--+--\ /--+--\ +----- *
- * O |\ | | | | | | /| *
- * O | \ - | - - | - / | *
- * S + | \ | | | | | | / | *
- * T | | | | | | | | | | *
- * ---------+---------------+------------------+---------------+-------- *
- * C | | | | | | | | | | *
- * U - | / | | | | | | \ | *
- * T | / - | - - | - \ | *
- * O |/ | | | | | | \| *
- * F -----+ \--+--/ \--+--/ +----- *
- * F | | | | *
- * | | | | *
- * *
- * Gains vary from 0.126 up to 7.943, which means from -18dB attenuation *
- * up to +18dB amplification. Band width varies from 0.01 up to 1.0 in *
- * octaves for two mid bands. *
- * *
- * Implementation is based on the "Cookbook formulae for audio EQ biquad *
- * filter coefficients" by Robert Bristow-Johnson *
- * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
-
-typedef enum ALEQFilterType {
- LOW_SHELF,
- HIGH_SHELF,
- PEAKING
-} ALEQFilterType;
-
-typedef struct ALEQFilter {
- ALEQFilterType type;
- ALfloat x[2]; /* History of two last input samples */
- ALfloat y[2]; /* History of two last output samples */
- ALfloat a[3]; /* Transfer function coefficients "a" */
- ALfloat b[3]; /* Transfer function coefficients "b" */
-} ALEQFilter;
-
-typedef struct ALequalizerState {
- DERIVE_FROM_TYPE(ALeffectState);
-
- /* Effect gains for each channel */
- ALfloat Gain[MaxChannels];
-
- /* Effect parameters */
- ALEQFilter bandfilter[4];
-} ALequalizerState;
-
-static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
-{
- (void)state;
-}
-
-static ALboolean ALequalizerState_DeviceUpdate(ALequalizerState *state, ALCdevice *device)
-{
- return AL_TRUE;
- (void)state;
- (void)device;
-}
-
-static ALvoid ALequalizerState_Update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot)
-{
- ALfloat frequency = (ALfloat)device->Frequency;
- ALfloat gain = sqrtf(1.0f / device->NumChan) * slot->Gain;
- ALuint it;
-
- for(it = 0;it < MaxChannels;it++)
- state->Gain[it] = 0.0f;
- for(it = 0; it < device->NumChan; it++)
- {
- enum Channel chan = device->Speaker2Chan[it];
- state->Gain[chan] = gain;
- }
-
- /* Calculate coefficients for the each type of filter */
- for(it = 0; it < 4; it++)
- {
- ALfloat gain;
- ALfloat filter_frequency;
- ALfloat bandwidth = 0.0f;
- ALfloat w0;
- ALfloat alpha = 0.0f;
-
- /* convert linear gains to filter gains */
- switch (it)
- {
- case 0: /* Low Shelf */
- gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.LowGain)) / 40.0f);
- filter_frequency = slot->effect.Equalizer.LowCutoff;
- break;
- case 1: /* Peaking */
- gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.Mid1Gain)) / 40.0f);
- filter_frequency = slot->effect.Equalizer.Mid1Center;
- bandwidth = slot->effect.Equalizer.Mid1Width;
- break;
- case 2: /* Peaking */
- gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.Mid2Gain)) / 40.0f);
- filter_frequency = slot->effect.Equalizer.Mid2Center;
- bandwidth = slot->effect.Equalizer.Mid2Width;
- break;
- case 3: /* High Shelf */
- gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.HighGain)) / 40.0f);
- filter_frequency = slot->effect.Equalizer.HighCutoff;
- break;
- }
-
- w0 = 2.0f*F_PI * filter_frequency / frequency;
-
- /* Calculate filter coefficients depending on filter type */
- switch(state->bandfilter[it].type)
- {
- case LOW_SHELF:
- alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
- (1.0f / 0.75f - 1.0f) + 2.0f);
- state->bandfilter[it].b[0] = gain * ((gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].b[1] = 2.0f * gain * ((gain - 1.0f) -
- (gain + 1.0f) * cosf(w0));
- state->bandfilter[it].b[2] = gain * ((gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].a[0] = (gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha;
- state->bandfilter[it].a[1] = -2.0f * ((gain - 1.0f) +
- (gain + 1.0f) * cosf(w0));
- state->bandfilter[it].a[2] = (gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha;
- break;
- case HIGH_SHELF:
- alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
- (1.0f / 0.75f - 1.0f) + 2.0f);
- state->bandfilter[it].b[0] = gain * ((gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].b[1] = -2.0f * gain * ((gain - 1.0f) +
- (gain + 1.0f) *
- cosf(w0));
- state->bandfilter[it].b[2] = gain * ((gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].a[0] = (gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha;
- state->bandfilter[it].a[1] = 2.0f * ((gain - 1.0f) -
- (gain + 1.0f) * cosf(w0));
- state->bandfilter[it].a[2] = (gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha;
- break;
- case PEAKING:
- alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
- state->bandfilter[it].b[0] = 1.0f + alpha * gain;
- state->bandfilter[it].b[1] = -2.0f * cosf(w0);
- state->bandfilter[it].b[2] = 1.0f - alpha * gain;
- state->bandfilter[it].a[0] = 1.0f + alpha / gain;
- state->bandfilter[it].a[1] = -2.0f * cosf(w0);
- state->bandfilter[it].a[2] = 1.0f - alpha / gain;
- break;
- }
- }
-}
-
-static ALvoid ALequalizerState_Process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
-{
- ALuint base;
- ALuint it;
- ALuint kt;
- ALuint ft;
-
- for(base = 0;base < SamplesToDo;)
- {
- ALfloat temps[64];
- ALuint td = minu(SamplesToDo-base, 64);
-
- for(it = 0;it < td;it++)
- {
- ALfloat smp = SamplesIn[base+it];
- ALfloat tempsmp;
-
- for(ft = 0;ft < 4;ft++)
- {
- ALEQFilter *filter = &state->bandfilter[ft];
-
- tempsmp = filter->b[0] / filter->a[0] * smp +
- filter->b[1] / filter->a[0] * filter->x[0] +
- filter->b[2] / filter->a[0] * filter->x[1] -
- filter->a[1] / filter->a[0] * filter->y[0] -
- filter->a[2] / filter->a[0] * filter->y[1];
-
- filter->x[1] = filter->x[0];
- filter->x[0] = smp;
- filter->y[1] = filter->y[0];
- filter->y[0] = tempsmp;
- smp = tempsmp;
- }
-
- temps[it] = smp;
- }
-
- for(kt = 0;kt < MaxChannels;kt++)
- {
- ALfloat gain = state->Gain[kt];
- if(!(gain > 0.00001f))
- continue;
-
- for(it = 0;it < td;it++)
- SamplesOut[kt][base+it] += gain * temps[it];
- }
-
- base += td;
- }
-}
-
-static ALeffectStateFactory *ALequalizerState_getCreator(void)
-{
- return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
-}
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
-
-
-ALeffectState *ALequalizerStateFactory_create(void)
-{
- ALequalizerState *state;
- int it;
-
- state = malloc(sizeof(*state));
- if(!state) return NULL;
- SET_VTABLE2(ALequalizerState, ALeffectState, state);
-
- state->bandfilter[0].type = LOW_SHELF;
- state->bandfilter[1].type = PEAKING;
- state->bandfilter[2].type = PEAKING;
- state->bandfilter[3].type = HIGH_SHELF;
-
- /* Initialize sample history only on filter creation to avoid */
- /* sound clicks if filter settings were changed in runtime. */
- for(it = 0; it < 4; it++)
- {
- state->bandfilter[it].x[0] = 0.0f;
- state->bandfilter[it].x[1] = 0.0f;
- state->bandfilter[it].y[0] = 0.0f;
- state->bandfilter[it].y[1] = 0.0f;
- }
-
- return STATIC_CAST(ALeffectState, state);
-}
-
-static ALvoid ALequalizerStateFactory_destroy(ALeffectState *effect)
-{
- ALequalizerState *state = STATIC_UPCAST(ALequalizerState, ALeffectState, effect);
- ALequalizerState_Destruct(state);
- free(state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory);
-
-
-static void init_equalizer_factory(void)
-{
- SET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory, &EqualizerFactory);
-}
-
-ALeffectStateFactory *ALequalizerStateFactory_getFactory(void)
-{
- static pthread_once_t once = PTHREAD_ONCE_INIT;
- pthread_once(&once, init_equalizer_factory);
- return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
-}
-
-
-void equalizer_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
-{
- effect=effect;
- val=val;
-
- switch(param)
- {
- default:
- alSetError(context, AL_INVALID_ENUM);
- break;
- }
-}
-void equalizer_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
-{
- equalizer_SetParami(effect, context, param, vals[0]);
-}
-void equalizer_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
-{
- switch(param)
- {
- case AL_EQUALIZER_LOW_GAIN:
- if(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN)
- effect->Equalizer.LowGain = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_LOW_CUTOFF:
- if(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF)
- effect->Equalizer.LowCutoff = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_MID1_GAIN:
- if(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN)
- effect->Equalizer.Mid1Gain = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_MID1_CENTER:
- if(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER)
- effect->Equalizer.Mid1Center = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_MID1_WIDTH:
- if(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH)
- effect->Equalizer.Mid1Width = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_MID2_GAIN:
- if(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN)
- effect->Equalizer.Mid2Gain = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_MID2_CENTER:
- if(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER)
- effect->Equalizer.Mid2Center = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_MID2_WIDTH:
- if(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH)
- effect->Equalizer.Mid2Width = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_HIGH_GAIN:
- if(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN)
- effect->Equalizer.HighGain = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- case AL_EQUALIZER_HIGH_CUTOFF:
- if(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF)
- effect->Equalizer.HighCutoff = val;
- else
- alSetError(context, AL_INVALID_VALUE);
- break;
-
- default:
- alSetError(context, AL_INVALID_ENUM);
- break;
- }
-}
-void equalizer_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
-{
- equalizer_SetParamf(effect, context, param, vals[0]);
-}
-
-void equalizer_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
-{
- effect=effect;
- val=val;
-
- switch(param)
- {
- default:
- alSetError(context, AL_INVALID_ENUM);
- break;
- }
-}
-void equalizer_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
-{
- equalizer_GetParami(effect, context, param, vals);
-}
-void equalizer_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
-{
- switch(param)
- {
- case AL_EQUALIZER_LOW_GAIN:
- *val = effect->Equalizer.LowGain;
- break;
-
- case AL_EQUALIZER_LOW_CUTOFF:
- *val = effect->Equalizer.LowCutoff;
- break;
-
- case AL_EQUALIZER_MID1_GAIN:
- *val = effect->Equalizer.Mid1Gain;
- break;
-
- case AL_EQUALIZER_MID1_CENTER:
- *val = effect->Equalizer.Mid1Center;
- break;
-
- case AL_EQUALIZER_MID1_WIDTH:
- *val = effect->Equalizer.Mid1Width;
- break;
-
- case AL_EQUALIZER_MID2_GAIN:
- *val = effect->Equalizer.Mid2Gain;
- break;
-
- case AL_EQUALIZER_MID2_CENTER:
- *val = effect->Equalizer.Mid2Center;
- break;
-
- case AL_EQUALIZER_MID2_WIDTH:
- *val = effect->Equalizer.Mid2Width;
- break;
-
- case AL_EQUALIZER_HIGH_GAIN:
- *val = effect->Equalizer.HighGain;
- break;
-
- case AL_EQUALIZER_HIGH_CUTOFF:
- *val = effect->Equalizer.HighCutoff;
- break;
-
- default:
- alSetError(context, AL_INVALID_ENUM);
- break;
- }
-}
-void equalizer_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
-{
- equalizer_GetParamf(effect, context, param, vals);
-}