/** * 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., * 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 "alFilter.h" #include "alAuxEffectSlot.h" #include "alError.h" #include "alu.h" /* 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 */ /* The maximum number of sample frames per update. */ #define MAX_UPDATE_SAMPLES 256 typedef struct ALequalizerState { DERIVE_FROM_TYPE(ALeffectState); /* Effect gains for each channel */ ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS]; /* Effect parameters */ ALfilterState filter[4][MAX_EFFECT_CHANNELS]; ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES]; } ALequalizerState; static ALvoid ALequalizerState_Destruct(ALequalizerState *UNUSED(state)) { } static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device)) { return AL_TRUE; } static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot) { ALfloat frequency = (ALfloat)device->Frequency; ALfloat gain, freq_mult; aluMatrixf matrix; ALuint i; gain = device->AmbiScale; aluMatrixfSet(&matrix, 1.0f, 0.0f, 0.0f, 0.0f, 0.0f, gain, 0.0f, 0.0f, 0.0f, 0.0f, gain, 0.0f, 0.0f, 0.0f, 0.0f, gain ); for(i = 0;i < MAX_EFFECT_CHANNELS;i++) ComputeFirstOrderGains(device->AmbiCoeffs, device->NumChannels, matrix.m[i], slot->Gain, state->Gain[i]); /* Calculate coefficients for the each type of filter. Note that the shelf * filters' gain is for the reference frequency, which is the centerpoint * of the transition band. */ gain = sqrtf(slot->EffectProps.Equalizer.LowGain); freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency; ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf, gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) ); /* Copy the filter coefficients for the other input channels. */ for(i = 1;i < MAX_EFFECT_CHANNELS;i++) { state->filter[0][i].a1 = state->filter[0][0].a1; state->filter[0][i].a2 = state->filter[0][0].a2; state->filter[0][i].b1 = state->filter[0][0].b1; state->filter[0][i].b2 = state->filter[0][0].b2; state->filter[0][i].input_gain = state->filter[0][0].input_gain; state->filter[0][i].process = state->filter[0][0].process; } gain = slot->EffectProps.Equalizer.Mid1Gain; freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency; ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking, gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width) ); for(i = 1;i < MAX_EFFECT_CHANNELS;i++) { state->filter[1][i].a1 = state->filter[1][0].a1; state->filter[1][i].a2 = state->filter[1][0].a2; state->filter[1][i].b1 = state->filter[1][0].b1; state->filter[1][i].b2 = state->filter[1][0].b2; state->filter[1][i].input_gain = state->filter[1][0].input_gain; state->filter[1][i].process = state->filter[1][0].process; } gain = slot->EffectProps.Equalizer.Mid2Gain; freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency; ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking, gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width) ); for(i = 1;i < MAX_EFFECT_CHANNELS;i++) { state->filter[2][i].a1 = state->filter[2][0].a1; state->filter[2][i].a2 = state->filter[2][0].a2; state->filter[2][i].b1 = state->filter[2][0].b1; state->filter[2][i].b2 = state->filter[2][0].b2; state->filter[2][i].input_gain = state->filter[2][0].input_gain; state->filter[2][i].process = state->filter[2][0].process; } gain = sqrtf(slot->EffectProps.Equalizer.HighGain); freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency; ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf, gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f) ); for(i = 1;i < MAX_EFFECT_CHANNELS;i++) { state->filter[3][i].a1 = state->filter[3][0].a1; state->filter[3][i].a2 = state->filter[3][0].a2; state->filter[3][i].b1 = state->filter[3][0].b1; state->filter[3][i].b2 = state->filter[3][0].b2; state->filter[3][i].input_gain = state->filter[3][0].input_gain; state->filter[3][i].process = state->filter[3][0].process; } } static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels) { ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer; ALuint it, kt, ft; ALuint base; for(base = 0;base < SamplesToDo;) { ALuint td = minu(MAX_UPDATE_SAMPLES, SamplesToDo-base); for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td); for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td); for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td); for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td); for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) { for(kt = 0;kt < NumChannels;kt++) { ALfloat gain = state->Gain[ft][kt]; if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD)) continue; for(it = 0;it < td;it++) SamplesOut[kt][base+it] += gain * Samples[3][ft][it]; } } base += td; } } DECLARE_DEFAULT_ALLOCATORS(ALequalizerState) DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState); typedef struct ALequalizerStateFactory { DERIVE_FROM_TYPE(ALeffectStateFactory); } ALequalizerStateFactory; ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory)) { ALequalizerState *state; int it, ft; state = ALequalizerState_New(sizeof(*state)); if(!state) return NULL; SET_VTABLE2(ALequalizerState, ALeffectState, state); /* Initialize sample history only on filter creation to avoid */ /* sound clicks if filter settings were changed in runtime. */ for(it = 0; it < 4; it++) { for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++) ALfilterState_clear(&state->filter[it][ft]); } return STATIC_CAST(ALeffectState, state); } DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory); ALeffectStateFactory *ALequalizerStateFactory_getFactory(void) { static ALequalizerStateFactory EqualizerFactory = { { GET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory) } }; return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory); } void ALequalizer_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val)) { SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } 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(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val)) { SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); } void ALequalizer_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals) { ALequalizer_getParami(effect, context, param, vals); } void ALequalizer_getParamf(const 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(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals) { ALequalizer_getParamf(effect, context, param, vals); } DEFINE_ALEFFECT_VTABLE(ALequalizer);