500 lines
18 KiB
C
500 lines
18 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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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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typedef struct ALequalizerStateFactory {
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DERIVE_FROM_TYPE(ALeffectStateFactory);
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} ALequalizerStateFactory;
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static ALequalizerStateFactory EqualizerFactory;
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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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typedef enum ALEQFilterType {
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LOW_SHELF,
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HIGH_SHELF,
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PEAKING
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} ALEQFilterType;
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typedef struct ALEQFilter {
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ALEQFilterType type;
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ALfloat x[2]; /* History of two last input samples */
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ALfloat y[2]; /* History of two last output samples */
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ALfloat a[3]; /* Transfer function coefficients "a" */
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ALfloat b[3]; /* Transfer function coefficients "b" */
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} ALEQFilter;
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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[MaxChannels];
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/* Effect parameters */
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ALEQFilter bandfilter[4];
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} ALequalizerState;
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static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
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{
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(void)state;
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}
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static ALboolean ALequalizerState_DeviceUpdate(ALequalizerState *state, ALCdevice *device)
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{
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return AL_TRUE;
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(void)state;
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(void)device;
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}
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static ALvoid ALequalizerState_Update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot)
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{
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ALfloat frequency = (ALfloat)device->Frequency;
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ALfloat gain = sqrtf(1.0f / device->NumChan) * slot->Gain;
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ALuint it;
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for(it = 0;it < MaxChannels;it++)
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state->Gain[it] = 0.0f;
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for(it = 0; it < device->NumChan; it++)
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{
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enum Channel chan = device->Speaker2Chan[it];
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state->Gain[chan] = gain;
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}
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/* Calculate coefficients for the each type of filter */
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for(it = 0; it < 4; it++)
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{
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ALfloat gain;
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ALfloat filter_frequency;
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ALfloat bandwidth = 0.0f;
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ALfloat w0;
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ALfloat alpha = 0.0f;
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/* convert linear gains to filter gains */
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switch (it)
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{
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case 0: /* Low Shelf */
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gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.LowGain)) / 40.0f);
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filter_frequency = slot->effect.Equalizer.LowCutoff;
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break;
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case 1: /* Peaking */
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gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.Mid1Gain)) / 40.0f);
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filter_frequency = slot->effect.Equalizer.Mid1Center;
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bandwidth = slot->effect.Equalizer.Mid1Width;
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break;
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case 2: /* Peaking */
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gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.Mid2Gain)) / 40.0f);
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filter_frequency = slot->effect.Equalizer.Mid2Center;
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bandwidth = slot->effect.Equalizer.Mid2Width;
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break;
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case 3: /* High Shelf */
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gain = powf(10.0f, (20.0f * log10f(slot->effect.Equalizer.HighGain)) / 40.0f);
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filter_frequency = slot->effect.Equalizer.HighCutoff;
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break;
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}
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w0 = 2.0f*F_PI * filter_frequency / frequency;
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/* Calculate filter coefficients depending on filter type */
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switch(state->bandfilter[it].type)
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{
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case LOW_SHELF:
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alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
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(1.0f / 0.75f - 1.0f) + 2.0f);
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state->bandfilter[it].b[0] = gain * ((gain + 1.0f) -
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(gain - 1.0f) * cosf(w0) +
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2.0f * sqrtf(gain) * alpha);
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state->bandfilter[it].b[1] = 2.0f * gain * ((gain - 1.0f) -
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(gain + 1.0f) * cosf(w0));
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state->bandfilter[it].b[2] = gain * ((gain + 1.0f) -
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(gain - 1.0f) * cosf(w0) -
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2.0f * sqrtf(gain) * alpha);
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state->bandfilter[it].a[0] = (gain + 1.0f) +
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(gain - 1.0f) * cosf(w0) +
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2.0f * sqrtf(gain) * alpha;
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state->bandfilter[it].a[1] = -2.0f * ((gain - 1.0f) +
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(gain + 1.0f) * cosf(w0));
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state->bandfilter[it].a[2] = (gain + 1.0f) +
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(gain - 1.0f) * cosf(w0) -
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2.0f * sqrtf(gain) * alpha;
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break;
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case HIGH_SHELF:
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alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
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(1.0f / 0.75f - 1.0f) + 2.0f);
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state->bandfilter[it].b[0] = gain * ((gain + 1.0f) +
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(gain - 1.0f) * cosf(w0) +
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2.0f * sqrtf(gain) * alpha);
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state->bandfilter[it].b[1] = -2.0f * gain * ((gain - 1.0f) +
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(gain + 1.0f) *
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cosf(w0));
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state->bandfilter[it].b[2] = gain * ((gain + 1.0f) +
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(gain - 1.0f) * cosf(w0) -
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2.0f * sqrtf(gain) * alpha);
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state->bandfilter[it].a[0] = (gain + 1.0f) -
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(gain - 1.0f) * cosf(w0) +
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2.0f * sqrtf(gain) * alpha;
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state->bandfilter[it].a[1] = 2.0f * ((gain - 1.0f) -
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(gain + 1.0f) * cosf(w0));
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state->bandfilter[it].a[2] = (gain + 1.0f) -
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(gain - 1.0f) * cosf(w0) -
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2.0f * sqrtf(gain) * alpha;
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break;
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case PEAKING:
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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state->bandfilter[it].b[0] = 1.0f + alpha * gain;
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state->bandfilter[it].b[1] = -2.0f * cosf(w0);
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state->bandfilter[it].b[2] = 1.0f - alpha * gain;
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state->bandfilter[it].a[0] = 1.0f + alpha / gain;
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state->bandfilter[it].a[1] = -2.0f * cosf(w0);
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state->bandfilter[it].a[2] = 1.0f - alpha / gain;
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break;
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}
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}
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}
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static ALvoid ALequalizerState_Process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
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{
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ALuint base;
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ALuint it;
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ALuint kt;
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ALuint ft;
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for(base = 0;base < SamplesToDo;)
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{
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ALfloat temps[64];
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ALuint td = minu(SamplesToDo-base, 64);
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for(it = 0;it < td;it++)
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{
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ALfloat smp = SamplesIn[base+it];
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ALfloat tempsmp;
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for(ft = 0;ft < 4;ft++)
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{
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ALEQFilter *filter = &state->bandfilter[ft];
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tempsmp = filter->b[0] / filter->a[0] * smp +
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filter->b[1] / filter->a[0] * filter->x[0] +
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filter->b[2] / filter->a[0] * filter->x[1] -
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filter->a[1] / filter->a[0] * filter->y[0] -
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filter->a[2] / filter->a[0] * filter->y[1];
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filter->x[1] = filter->x[0];
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filter->x[0] = smp;
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filter->y[1] = filter->y[0];
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filter->y[0] = tempsmp;
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smp = tempsmp;
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}
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temps[it] = smp;
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}
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for(kt = 0;kt < MaxChannels;kt++)
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{
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ALfloat gain = state->Gain[kt];
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if(!(gain > 0.00001f))
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continue;
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for(it = 0;it < td;it++)
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SamplesOut[kt][base+it] += gain * temps[it];
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}
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base += td;
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}
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}
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static ALeffectStateFactory *ALequalizerState_getCreator(void)
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{
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return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
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}
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DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
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ALeffectState *ALequalizerStateFactory_create(void)
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{
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ALequalizerState *state;
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int it;
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state = malloc(sizeof(*state));
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if(!state) return NULL;
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SET_VTABLE2(ALequalizerState, ALeffectState, state);
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state->bandfilter[0].type = LOW_SHELF;
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state->bandfilter[1].type = PEAKING;
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state->bandfilter[2].type = PEAKING;
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state->bandfilter[3].type = HIGH_SHELF;
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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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state->bandfilter[it].x[0] = 0.0f;
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state->bandfilter[it].x[1] = 0.0f;
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state->bandfilter[it].y[0] = 0.0f;
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state->bandfilter[it].y[1] = 0.0f;
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}
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return STATIC_CAST(ALeffectState, state);
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}
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static ALvoid ALequalizerStateFactory_destroy(ALeffectState *effect)
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{
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ALequalizerState *state = STATIC_UPCAST(ALequalizerState, ALeffectState, effect);
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ALequalizerState_Destruct(state);
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free(state);
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}
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DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory);
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static void init_equalizer_factory(void)
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{
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SET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory, &EqualizerFactory);
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}
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ALeffectStateFactory *ALequalizerStateFactory_getFactory(void)
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{
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static pthread_once_t once = PTHREAD_ONCE_INIT;
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pthread_once(&once, init_equalizer_factory);
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return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
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}
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void equalizer_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
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{
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effect=effect;
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val=val;
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switch(param)
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{
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void equalizer_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
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{
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equalizer_SetParami(effect, context, param, vals[0]);
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}
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void equalizer_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
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{
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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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effect->Equalizer.LowGain = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.LowCutoff = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.Mid1Gain = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.Mid1Center = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.Mid1Width = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.Mid2Gain = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.Mid2Center = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.Mid2Width = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.HighGain = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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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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effect->Equalizer.HighCutoff = val;
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else
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alSetError(context, AL_INVALID_VALUE);
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break;
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void equalizer_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
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{
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equalizer_SetParamf(effect, context, param, vals[0]);
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}
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void equalizer_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
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{
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effect=effect;
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val=val;
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switch(param)
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{
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default:
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alSetError(context, AL_INVALID_ENUM);
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break;
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}
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}
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void equalizer_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
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{
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equalizer_GetParami(effect, context, param, vals);
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}
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void equalizer_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
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{
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switch(param)
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{
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case AL_EQUALIZER_LOW_GAIN:
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*val = effect->Equalizer.LowGain;
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break;
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case AL_EQUALIZER_LOW_CUTOFF:
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*val = effect->Equalizer.LowCutoff;
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break;
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case AL_EQUALIZER_MID1_GAIN:
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*val = effect->Equalizer.Mid1Gain;
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break;
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case AL_EQUALIZER_MID1_CENTER:
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*val = effect->Equalizer.Mid1Center;
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break;
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case AL_EQUALIZER_MID1_WIDTH:
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*val = effect->Equalizer.Mid1Width;
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break;
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case AL_EQUALIZER_MID2_GAIN:
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*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);
|
|
}
|