/** * 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 #include #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->EffectProps.Equalizer.LowGain)) / 40.0f); filter_frequency = slot->EffectProps.Equalizer.LowCutoff; break; case 1: /* Peaking */ gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.Mid1Gain)) / 40.0f); filter_frequency = slot->EffectProps.Equalizer.Mid1Center; bandwidth = slot->EffectProps.Equalizer.Mid1Width; break; case 2: /* Peaking */ gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.Mid2Gain)) / 40.0f); filter_frequency = slot->EffectProps.Equalizer.Mid2Center; bandwidth = slot->EffectProps.Equalizer.Mid2Width; break; case 3: /* High Shelf */ gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.HighGain)) / 40.0f); filter_frequency = slot->EffectProps.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; } state->bandfilter[it].b[0] /= state->bandfilter[it].a[0]; state->bandfilter[it].b[1] /= state->bandfilter[it].a[0]; state->bandfilter[it].b[2] /= state->bandfilter[it].a[0]; state->bandfilter[it].a[0] /= state->bandfilter[it].a[0]; state->bandfilter[it].a[1] /= state->bandfilter[it].a[0]; state->bandfilter[it].a[2] /= state->bandfilter[it].a[0]; } } 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] * smp + filter->b[1] * filter->x[0] + filter->b[2] * filter->x[1] - filter->a[1] * filter->y[0] - filter->a[2] * 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 void ALequalizerState_Delete(ALequalizerState *state) { free(state); } 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); } 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 ALequalizer_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val) { SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); (void)effect;(void)param;(void)val; } void ALequalizer_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals) { ALequalizer_SetParami(effect, context, param, vals[0]); } void ALequalizer_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val) { ALeffectProps *props = &effect->Props; switch(param) { case AL_EQUALIZER_LOW_GAIN: if(!(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.LowGain = val; break; case AL_EQUALIZER_LOW_CUTOFF: if(!(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.LowCutoff = val; break; case AL_EQUALIZER_MID1_GAIN: if(!(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.Mid1Gain = val; break; case AL_EQUALIZER_MID1_CENTER: if(!(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.Mid1Center = val; break; case AL_EQUALIZER_MID1_WIDTH: if(!(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.Mid1Width = val; break; case AL_EQUALIZER_MID2_GAIN: if(!(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.Mid2Gain = val; break; case AL_EQUALIZER_MID2_CENTER: if(!(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.Mid2Center = val; break; case AL_EQUALIZER_MID2_WIDTH: if(!(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.Mid2Width = val; break; case AL_EQUALIZER_HIGH_GAIN: if(!(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.HighGain = val; break; case AL_EQUALIZER_HIGH_CUTOFF: if(!(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF)) SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE); props->Equalizer.HighCutoff = val; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALequalizer_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals) { ALequalizer_SetParamf(effect, context, param, vals[0]); } void ALequalizer_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val) { SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); (void)effect;(void)param;(void)val; } void ALequalizer_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALequalizer_GetParami(effect, context, param, vals); } void ALequalizer_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val) { const ALeffectProps *props = &effect->Props; switch(param) { case AL_EQUALIZER_LOW_GAIN: *val = props->Equalizer.LowGain; break; case AL_EQUALIZER_LOW_CUTOFF: *val = props->Equalizer.LowCutoff; break; case AL_EQUALIZER_MID1_GAIN: *val = props->Equalizer.Mid1Gain; break; case AL_EQUALIZER_MID1_CENTER: *val = props->Equalizer.Mid1Center; break; case AL_EQUALIZER_MID1_WIDTH: *val = props->Equalizer.Mid1Width; break; case AL_EQUALIZER_MID2_GAIN: *val = props->Equalizer.Mid2Gain; break; case AL_EQUALIZER_MID2_CENTER: *val = props->Equalizer.Mid2Center; break; case AL_EQUALIZER_MID2_WIDTH: *val = props->Equalizer.Mid2Width; break; case AL_EQUALIZER_HIGH_GAIN: *val = props->Equalizer.HighGain; break; case AL_EQUALIZER_HIGH_CUTOFF: *val = props->Equalizer.HighCutoff; break; default: SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } } void ALequalizer_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALequalizer_GetParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALequalizer);