399 lines
16 KiB
C
399 lines
16 KiB
C
/**
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* OpenAL cross platform audio library
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* Copyright (C) 2013 by Mike Gorchak
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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* Or go to http://www.gnu.org/copyleft/lgpl.html
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*/
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#include "config.h"
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#include <math.h>
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#include <stdlib.h>
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#include "alMain.h"
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#include "alFilter.h"
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#include "alAuxEffectSlot.h"
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#include "alError.h"
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#include "alu.h"
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/* The document "Effects Extension Guide.pdf" says that low and high *
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* frequencies are cutoff frequencies. This is not fully correct, they *
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* are corner frequencies for low and high shelf filters. If they were *
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* just cutoff frequencies, there would be no need in cutoff frequency *
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* gains, which are present. Documentation for "Creative Proteus X2" *
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* software describes 4-band equalizer functionality in a much better *
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* way. This equalizer seems to be a predecessor of OpenAL 4-band *
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* equalizer. With low and high shelf filters we are able to cutoff *
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* frequencies below and/or above corner frequencies using attenuation *
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* gains (below 1.0) and amplify all low and/or high frequencies using *
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* gains above 1.0. *
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* *
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* Low-shelf Low Mid Band High Mid Band High-shelf *
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* corner center center corner *
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* frequency frequency frequency frequency *
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* 50Hz..800Hz 200Hz..3000Hz 1000Hz..8000Hz 4000Hz..16000Hz *
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* *
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* | | | | *
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* | | | | *
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* B -----+ /--+--\ /--+--\ +----- *
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* O |\ | | | | | | /| *
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* O | \ - | - - | - / | *
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* S + | \ | | | | | | / | *
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* T | | | | | | | | | | *
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* ---------+---------------+------------------+---------------+-------- *
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* C | | | | | | | | | | *
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* U - | / | | | | | | \ | *
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* T | / - | - - | - \ | *
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* O |/ | | | | | | \| *
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* F -----+ \--+--/ \--+--/ +----- *
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* F | | | | *
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* | | | | *
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* *
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* Gains vary from 0.126 up to 7.943, which means from -18dB attenuation *
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* up to +18dB amplification. Band width varies from 0.01 up to 1.0 in *
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* octaves for two mid bands. *
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* *
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* Implementation is based on the "Cookbook formulae for audio EQ biquad *
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* filter coefficients" by Robert Bristow-Johnson *
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* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
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/* The maximum number of sample frames per update. */
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#define MAX_UPDATE_SAMPLES 256
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typedef struct ALequalizerState {
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DERIVE_FROM_TYPE(ALeffectState);
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/* Effect gains for each channel */
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ALfloat Gain[MAX_EFFECT_CHANNELS][MAX_OUTPUT_CHANNELS];
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/* Effect parameters */
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ALfilterState filter[4][MAX_EFFECT_CHANNELS];
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ALfloat SampleBuffer[4][MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES];
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} ALequalizerState;
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static ALvoid ALequalizerState_Destruct(ALequalizerState *UNUSED(state))
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{
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}
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static ALboolean ALequalizerState_deviceUpdate(ALequalizerState *UNUSED(state), ALCdevice *UNUSED(device))
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{
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return AL_TRUE;
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}
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static ALvoid ALequalizerState_update(ALequalizerState *state, const ALCdevice *device, const ALeffectslot *slot)
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{
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ALfloat frequency = (ALfloat)device->Frequency;
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ALfloat gain, freq_mult;
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aluMatrixf matrix;
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ALuint i;
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gain = device->AmbiScale;
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aluMatrixfSet(&matrix,
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1.0f, 0.0f, 0.0f, 0.0f,
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0.0f, gain, 0.0f, 0.0f,
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0.0f, 0.0f, gain, 0.0f,
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0.0f, 0.0f, 0.0f, gain
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);
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for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
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ComputeFirstOrderGains(device->AmbiCoeffs, device->NumChannels,
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matrix.m[i], slot->Gain, state->Gain[i]);
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/* Calculate coefficients for the each type of filter. Note that the shelf
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* filters' gain is for the reference frequency, which is the centerpoint
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* of the transition band.
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*/
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gain = sqrtf(slot->EffectProps.Equalizer.LowGain);
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freq_mult = slot->EffectProps.Equalizer.LowCutoff/frequency;
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ALfilterState_setParams(&state->filter[0][0], ALfilterType_LowShelf,
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gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
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);
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/* Copy the filter coefficients for the other input channels. */
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[0][i].a1 = state->filter[0][0].a1;
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state->filter[0][i].a2 = state->filter[0][0].a2;
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state->filter[0][i].b1 = state->filter[0][0].b1;
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state->filter[0][i].b2 = state->filter[0][0].b2;
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state->filter[0][i].input_gain = state->filter[0][0].input_gain;
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state->filter[0][i].process = state->filter[0][0].process;
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}
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gain = slot->EffectProps.Equalizer.Mid1Gain;
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freq_mult = slot->EffectProps.Equalizer.Mid1Center/frequency;
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ALfilterState_setParams(&state->filter[1][0], ALfilterType_Peaking,
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gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid1Width)
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);
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[1][i].a1 = state->filter[1][0].a1;
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state->filter[1][i].a2 = state->filter[1][0].a2;
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state->filter[1][i].b1 = state->filter[1][0].b1;
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state->filter[1][i].b2 = state->filter[1][0].b2;
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state->filter[1][i].input_gain = state->filter[1][0].input_gain;
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state->filter[1][i].process = state->filter[1][0].process;
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}
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gain = slot->EffectProps.Equalizer.Mid2Gain;
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freq_mult = slot->EffectProps.Equalizer.Mid2Center/frequency;
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ALfilterState_setParams(&state->filter[2][0], ALfilterType_Peaking,
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gain, freq_mult, calc_rcpQ_from_bandwidth(freq_mult, slot->EffectProps.Equalizer.Mid2Width)
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);
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[2][i].a1 = state->filter[2][0].a1;
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state->filter[2][i].a2 = state->filter[2][0].a2;
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state->filter[2][i].b1 = state->filter[2][0].b1;
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state->filter[2][i].b2 = state->filter[2][0].b2;
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state->filter[2][i].input_gain = state->filter[2][0].input_gain;
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state->filter[2][i].process = state->filter[2][0].process;
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}
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gain = sqrtf(slot->EffectProps.Equalizer.HighGain);
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freq_mult = slot->EffectProps.Equalizer.HighCutoff/frequency;
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ALfilterState_setParams(&state->filter[3][0], ALfilterType_HighShelf,
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gain, freq_mult, calc_rcpQ_from_slope(gain, 0.75f)
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);
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for(i = 1;i < MAX_EFFECT_CHANNELS;i++)
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{
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state->filter[3][i].a1 = state->filter[3][0].a1;
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state->filter[3][i].a2 = state->filter[3][0].a2;
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state->filter[3][i].b1 = state->filter[3][0].b1;
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state->filter[3][i].b2 = state->filter[3][0].b2;
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state->filter[3][i].input_gain = state->filter[3][0].input_gain;
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state->filter[3][i].process = state->filter[3][0].process;
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}
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}
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static ALvoid ALequalizerState_process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
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{
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ALfloat (*Samples)[MAX_EFFECT_CHANNELS][MAX_UPDATE_SAMPLES] = state->SampleBuffer;
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ALuint it, kt, ft;
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ALuint base;
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for(base = 0;base < SamplesToDo;)
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{
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ALuint td = minu(MAX_UPDATE_SAMPLES, SamplesToDo-base);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[0][ft], Samples[0][ft], &SamplesIn[ft][base], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[1][ft], Samples[1][ft], Samples[0][ft], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[2][ft], Samples[2][ft], Samples[1][ft], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_process(&state->filter[3][ft], Samples[3][ft], Samples[2][ft], td);
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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{
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for(kt = 0;kt < NumChannels;kt++)
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{
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ALfloat gain = state->Gain[ft][kt];
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if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
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continue;
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for(it = 0;it < td;it++)
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SamplesOut[kt][base+it] += gain * Samples[3][ft][it];
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}
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}
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base += td;
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}
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}
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DECLARE_DEFAULT_ALLOCATORS(ALequalizerState)
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DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
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typedef struct ALequalizerStateFactory {
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DERIVE_FROM_TYPE(ALeffectStateFactory);
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} ALequalizerStateFactory;
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ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *UNUSED(factory))
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{
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ALequalizerState *state;
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int it, ft;
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state = ALequalizerState_New(sizeof(*state));
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if(!state) return NULL;
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SET_VTABLE2(ALequalizerState, ALeffectState, state);
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/* Initialize sample history only on filter creation to avoid */
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/* sound clicks if filter settings were changed in runtime. */
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for(it = 0; it < 4; it++)
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{
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for(ft = 0;ft < MAX_EFFECT_CHANNELS;ft++)
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ALfilterState_clear(&state->filter[it][ft]);
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}
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return STATIC_CAST(ALeffectState, state);
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}
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DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory);
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ALeffectStateFactory *ALequalizerStateFactory_getFactory(void)
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{
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static ALequalizerStateFactory EqualizerFactory = { { GET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory) } };
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return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
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}
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void ALequalizer_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
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{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
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void ALequalizer_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
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{
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ALequalizer_setParami(effect, context, param, vals[0]);
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}
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void ALequalizer_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
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{
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ALeffectProps *props = &effect->Props;
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switch(param)
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{
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case AL_EQUALIZER_LOW_GAIN:
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if(!(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.LowGain = val;
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break;
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case AL_EQUALIZER_LOW_CUTOFF:
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if(!(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.LowCutoff = val;
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break;
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case AL_EQUALIZER_MID1_GAIN:
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if(!(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.Mid1Gain = val;
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break;
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case AL_EQUALIZER_MID1_CENTER:
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if(!(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.Mid1Center = val;
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break;
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case AL_EQUALIZER_MID1_WIDTH:
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if(!(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.Mid1Width = val;
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break;
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case AL_EQUALIZER_MID2_GAIN:
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if(!(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.Mid2Gain = val;
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break;
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case AL_EQUALIZER_MID2_CENTER:
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if(!(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.Mid2Center = val;
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break;
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case AL_EQUALIZER_MID2_WIDTH:
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if(!(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.Mid2Width = val;
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break;
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case AL_EQUALIZER_HIGH_GAIN:
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if(!(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.HighGain = val;
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break;
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case AL_EQUALIZER_HIGH_CUTOFF:
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if(!(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF))
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SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
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props->Equalizer.HighCutoff = val;
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break;
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default:
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SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
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}
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}
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void ALequalizer_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
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{
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ALequalizer_setParamf(effect, context, param, vals[0]);
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}
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void ALequalizer_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
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{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
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void ALequalizer_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
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{
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ALequalizer_getParami(effect, context, param, vals);
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}
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void ALequalizer_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
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{
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const ALeffectProps *props = &effect->Props;
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switch(param)
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{
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case AL_EQUALIZER_LOW_GAIN:
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*val = props->Equalizer.LowGain;
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break;
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case AL_EQUALIZER_LOW_CUTOFF:
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*val = props->Equalizer.LowCutoff;
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break;
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case AL_EQUALIZER_MID1_GAIN:
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*val = props->Equalizer.Mid1Gain;
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break;
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case AL_EQUALIZER_MID1_CENTER:
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*val = props->Equalizer.Mid1Center;
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break;
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case AL_EQUALIZER_MID1_WIDTH:
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*val = props->Equalizer.Mid1Width;
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break;
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case AL_EQUALIZER_MID2_GAIN:
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*val = props->Equalizer.Mid2Gain;
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break;
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case AL_EQUALIZER_MID2_CENTER:
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*val = props->Equalizer.Mid2Center;
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break;
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case AL_EQUALIZER_MID2_WIDTH:
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*val = props->Equalizer.Mid2Width;
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break;
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case AL_EQUALIZER_HIGH_GAIN:
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*val = props->Equalizer.HighGain;
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break;
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case AL_EQUALIZER_HIGH_CUTOFF:
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*val = props->Equalizer.HighCutoff;
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break;
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default:
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SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
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}
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}
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void ALequalizer_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
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{
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ALequalizer_getParamf(effect, context, param, vals);
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}
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DEFINE_ALEFFECT_VTABLE(ALequalizer);
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