Implement distortion and equalizer effects
Code provided by Mike Gorchak
This commit is contained in:
Chris Robinson
parent
a7ad6080f0
commit
78e7c1c27b
21
Alc/ALc.c
21
Alc/ALc.c
@ -516,9 +516,7 @@ static const ALCenums enumeration[] = {
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DECL(AL_EFFECT_REVERB),
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DECL(AL_EFFECT_REVERB),
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DECL(AL_EFFECT_EAXREVERB),
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DECL(AL_EFFECT_EAXREVERB),
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DECL(AL_EFFECT_CHORUS),
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DECL(AL_EFFECT_CHORUS),
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#if 0
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DECL(AL_EFFECT_DISTORTION),
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DECL(AL_EFFECT_DISTORTION),
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#endif
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DECL(AL_EFFECT_ECHO),
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DECL(AL_EFFECT_ECHO),
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DECL(AL_EFFECT_FLANGER),
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DECL(AL_EFFECT_FLANGER),
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#if 0
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#if 0
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@ -530,8 +528,8 @@ static const ALCenums enumeration[] = {
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#if 0
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#if 0
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DECL(AL_EFFECT_AUTOWAH),
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DECL(AL_EFFECT_AUTOWAH),
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DECL(AL_EFFECT_COMPRESSOR),
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DECL(AL_EFFECT_COMPRESSOR),
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DECL(AL_EFFECT_EQUALIZER),
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#endif
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#endif
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DECL(AL_EFFECT_EQUALIZER),
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DECL(AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT),
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DECL(AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT),
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DECL(AL_EFFECT_DEDICATED_DIALOGUE),
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DECL(AL_EFFECT_DEDICATED_DIALOGUE),
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@ -593,6 +591,23 @@ static const ALCenums enumeration[] = {
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DECL(AL_FLANGER_FEEDBACK),
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DECL(AL_FLANGER_FEEDBACK),
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DECL(AL_FLANGER_DELAY),
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DECL(AL_FLANGER_DELAY),
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DECL(AL_EQUALIZER_LOW_GAIN),
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DECL(AL_EQUALIZER_LOW_CUTOFF),
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DECL(AL_EQUALIZER_MID1_GAIN),
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DECL(AL_EQUALIZER_MID1_CENTER),
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DECL(AL_EQUALIZER_MID1_WIDTH),
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DECL(AL_EQUALIZER_MID2_GAIN),
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DECL(AL_EQUALIZER_MID2_CENTER),
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DECL(AL_EQUALIZER_MID2_WIDTH),
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DECL(AL_EQUALIZER_HIGH_GAIN),
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DECL(AL_EQUALIZER_HIGH_CUTOFF),
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DECL(AL_DISTORTION_EDGE),
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DECL(AL_DISTORTION_GAIN),
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DECL(AL_DISTORTION_LOWPASS_CUTOFF),
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DECL(AL_DISTORTION_EQCENTER),
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DECL(AL_DISTORTION_EQBANDWIDTH),
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DECL(AL_RING_MODULATOR_FREQUENCY),
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DECL(AL_RING_MODULATOR_FREQUENCY),
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DECL(AL_RING_MODULATOR_HIGHPASS_CUTOFF),
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DECL(AL_RING_MODULATOR_HIGHPASS_CUTOFF),
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DECL(AL_RING_MODULATOR_WAVEFORM),
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DECL(AL_RING_MODULATOR_WAVEFORM),
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356
Alc/alcDistortion.c
Normal file
356
Alc/alcDistortion.c
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@ -0,0 +1,356 @@
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/**
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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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/* Filters implementation is based on the "Cookbook formulae for audio *
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* EQ biquad 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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LOWPASS,
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BANDPASS,
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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 ALdistortionState {
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/* Must be first in all effects! */
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ALeffectState state;
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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 bandpass;
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ALEQFilter lowpass;
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ALfloat frequency;
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ALfloat attenuation;
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ALfloat edge;
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/* Oversample data */
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ALfloat oversample_buffer[BUFFERSIZE][4];
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} ALdistortionState;
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static ALvoid DistortionDestroy(ALeffectState *effect)
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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free(state);
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}
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static ALboolean DistortionDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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state->frequency = (ALfloat)Device->Frequency;
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return AL_TRUE;
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}
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static ALvoid DistortionUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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ALfloat gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
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ALuint it;
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ALfloat w0;
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ALfloat alpha;
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ALfloat bandwidth;
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ALfloat cutoff;
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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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/* Store distorted signal attenuation settings */
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state->attenuation = Slot->effect.Distortion.Gain;
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/* Store waveshaper edge settings */
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state->edge = Slot->effect.Distortion.Edge;
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/* Lowpass filter */
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cutoff = Slot->effect.Distortion.LowpassCutoff;
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/* Bandwidth value is constant in octaves */
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bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
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w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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state->lowpass.b[0] = (1.0f - cosf(w0)) / 2.0f;
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state->lowpass.b[1] = 1.0f - cosf(w0);
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state->lowpass.b[2] = (1.0f - cosf(w0)) / 2.0f;
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state->lowpass.a[0] = 1.0f + alpha;
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state->lowpass.a[1] = -2.0f * cosf(w0);
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state->lowpass.a[2] = 1.0f - alpha;
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/* Bandpass filter */
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cutoff = Slot->effect.Distortion.EQCenter;
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/* Convert bandwidth in Hz to octaves */
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bandwidth = Slot->effect.Distortion.EQBandwidth / (cutoff * 0.67f);
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w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
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alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
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state->bandpass.b[0] = alpha;
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state->bandpass.b[1] = 0;
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state->bandpass.b[2] = -alpha;
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state->bandpass.a[0] = 1.0f + alpha;
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state->bandpass.a[1] = -2.0f * cosf(w0);
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state->bandpass.a[2] = 1.0f - alpha;
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}
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static ALvoid DistortionProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
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{
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ALdistortionState *state = (ALdistortionState*)effect;
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float *RESTRICT oversample_buffer = &state->oversample_buffer[0][0];
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ALfloat tempsmp;
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ALuint it;
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ALuint kt;
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ALuint st;
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/* Perform 4x oversampling to avoid aliasing. */
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/* Oversampling greatly improves distortion */
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/* quality and allows to implement lowpass and */
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/* bandpass filters using high frequencies, at */
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/* which classic IIR filters became unstable. */
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/* Fill oversample buffer using zero stuffing */
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for(it = 0; it < SamplesToDo; it++)
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{
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oversample_buffer[it*4 + 0] = SamplesIn[it];
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oversample_buffer[it*4 + 1] = 0.0f;
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oversample_buffer[it*4 + 2] = 0.0f;
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oversample_buffer[it*4 + 3] = 0.0f;
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}
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/* First step, do lowpass filtering of original signal, */
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/* additionally perform buffer interpolation and lowpass */
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/* cutoff for oversampling (which is fortunately first */
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/* step of distortion). So combine three operations into */
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/* the one. */
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for(it = 0; it < SamplesToDo * 4; it++)
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{
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tempsmp = state->lowpass.b[0] / state->lowpass.a[0] * oversample_buffer[it] +
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state->lowpass.b[1] / state->lowpass.a[0] * state->lowpass.x[0] +
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state->lowpass.b[2] / state->lowpass.a[0] * state->lowpass.x[1] -
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state->lowpass.a[1] / state->lowpass.a[0] * state->lowpass.y[0] -
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state->lowpass.a[2] / state->lowpass.a[0] * state->lowpass.y[1];
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state->lowpass.x[1] = state->lowpass.x[0];
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state->lowpass.x[0] = oversample_buffer[it];
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state->lowpass.y[1] = state->lowpass.y[0];
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state->lowpass.y[0] = tempsmp;
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/* Restore signal power by multiplying sample by amount of oversampling */
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oversample_buffer[it] = tempsmp * 4.0f;
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}
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for(it = 0; it < SamplesToDo * 4; it++)
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{
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ALfloat smp = oversample_buffer[it];
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ALfloat edge = sinf(state->edge * (F_PI / 2.0f));
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/* Second step, do distortion using waveshaper function */
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/* to emulate signal processing during tube overdriving. */
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/* Three steps of waveshaping are intended to modify */
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/* waveform without boost/clipping/attenuation process. */
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for(st = 0; st < 3; st++)
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{
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smp = (1.0f + 2.0f * edge / (1.0f - edge)) * smp / (1.0f + 2.0f * edge / (1.0f - edge) * fabsf(smp));
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if((st & 0x00000001) == 0x00000001)
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smp *= -1.0f;
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}
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/* Third step, do bandpass filtering of distorted signal */
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tempsmp = state->bandpass.b[0] / state->bandpass.a[0] * smp +
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state->bandpass.b[1] / state->bandpass.a[0] * state->bandpass.x[0] +
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state->bandpass.b[2] / state->bandpass.a[0] * state->bandpass.x[1] -
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state->bandpass.a[1] / state->bandpass.a[0] * state->bandpass.y[0] -
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state->bandpass.a[2] / state->bandpass.a[0] * state->bandpass.y[1];
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state->bandpass.x[1] = state->bandpass.x[0];
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state->bandpass.x[0] = smp;
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state->bandpass.y[1] = state->bandpass.y[0];
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state->bandpass.y[0] = tempsmp;
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smp = tempsmp;
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/* Fourth step, final, do attenuation and perform decimation, */
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/* store only one sample out of 4. */
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if(!(it & 0x00000003))
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{
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smp *= state->attenuation;
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for(kt = 0; kt < MaxChannels; kt++)
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SamplesOut[kt][it>>2] += state->Gain[kt] * smp;
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}
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}
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}
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ALeffectState *DistortionCreate(void)
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{
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ALdistortionState *state;
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state = malloc(sizeof(*state));
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if(!state)
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return NULL;
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state->state.Destroy = DistortionDestroy;
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state->state.DeviceUpdate = DistortionDeviceUpdate;
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state->state.Update = DistortionUpdate;
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state->state.Process = DistortionProcess;
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state->bandpass.type = BANDPASS;
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state->lowpass.type = LOWPASS;
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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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state->bandpass.x[0] = 0.0f;
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state->bandpass.x[1] = 0.0f;
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state->lowpass.y[0] = 0.0f;
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state->lowpass.y[1] = 0.0f;
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return &state->state;
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}
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void distortion_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 distortion_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
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{
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distortion_SetParami(effect, context, param, vals[0]);
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}
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void distortion_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_DISTORTION_EDGE:
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if(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE)
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effect->Distortion.Edge = 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_DISTORTION_GAIN:
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if(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN)
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effect->Distortion.Gain = 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_DISTORTION_LOWPASS_CUTOFF:
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if(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF)
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effect->Distortion.LowpassCutoff = 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_DISTORTION_EQCENTER:
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if(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER)
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effect->Distortion.EQCenter = 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_DISTORTION_EQBANDWIDTH:
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if(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH)
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effect->Distortion.EQBandwidth = 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:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void distortion_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
|
||||||
|
{
|
||||||
|
distortion_SetParamf(effect, context, param, vals[0]);
|
||||||
|
}
|
||||||
|
|
||||||
|
void distortion_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
|
||||||
|
{
|
||||||
|
effect=effect;
|
||||||
|
val=val;
|
||||||
|
|
||||||
|
switch(param)
|
||||||
|
{
|
||||||
|
default:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void distortion_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
|
||||||
|
{
|
||||||
|
distortion_GetParami(effect, context, param, vals);
|
||||||
|
}
|
||||||
|
void distortion_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
|
||||||
|
{
|
||||||
|
switch(param)
|
||||||
|
{
|
||||||
|
case AL_DISTORTION_EDGE:
|
||||||
|
*val = effect->Distortion.Edge;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_DISTORTION_GAIN:
|
||||||
|
*val = effect->Distortion.Gain;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_DISTORTION_LOWPASS_CUTOFF:
|
||||||
|
*val = effect->Distortion.LowpassCutoff;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_DISTORTION_EQCENTER:
|
||||||
|
*val = effect->Distortion.EQCenter;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_DISTORTION_EQBANDWIDTH:
|
||||||
|
*val = effect->Distortion.EQBandwidth;
|
||||||
|
break;
|
||||||
|
|
||||||
|
default:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void distortion_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
|
||||||
|
{
|
||||||
|
distortion_GetParamf(effect, context, param, vals);
|
||||||
|
}
|
||||||
446
Alc/alcEqualizer.c
Normal file
446
Alc/alcEqualizer.c
Normal file
@ -0,0 +1,446 @@
|
|||||||
|
/**
|
||||||
|
* OpenAL cross platform audio library
|
||||||
|
* Copyright (C) 2013 by Mike Gorchak
|
||||||
|
* This library is free software; you can redistribute it and/or
|
||||||
|
* modify it under the terms of the GNU Library General Public
|
||||||
|
* License as published by the Free Software Foundation; either
|
||||||
|
* version 2 of the License, or (at your option) any later version.
|
||||||
|
*
|
||||||
|
* This library is distributed in the hope that it will be useful,
|
||||||
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
||||||
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
||||||
|
* Library General Public License for more details.
|
||||||
|
*
|
||||||
|
* You should have received a copy of the GNU Library General Public
|
||||||
|
* License along with this library; if not, write to the
|
||||||
|
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
|
||||||
|
* Boston, MA 02111-1307, USA.
|
||||||
|
* Or go to http://www.gnu.org/copyleft/lgpl.html
|
||||||
|
*/
|
||||||
|
|
||||||
|
#include "config.h"
|
||||||
|
|
||||||
|
#include <math.h>
|
||||||
|
#include <stdlib.h>
|
||||||
|
|
||||||
|
#include "alMain.h"
|
||||||
|
#include "alFilter.h"
|
||||||
|
#include "alAuxEffectSlot.h"
|
||||||
|
#include "alError.h"
|
||||||
|
#include "alu.h"
|
||||||
|
|
||||||
|
/* 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 {
|
||||||
|
/* Must be first in all effects! */
|
||||||
|
ALeffectState state;
|
||||||
|
|
||||||
|
/* Effect gains for each channel */
|
||||||
|
ALfloat Gain[MaxChannels];
|
||||||
|
|
||||||
|
/* Effect parameters */
|
||||||
|
ALEQFilter bandfilter[4];
|
||||||
|
ALfloat frequency;
|
||||||
|
} ALequalizerState;
|
||||||
|
|
||||||
|
static ALvoid EqualizerDestroy(ALeffectState *effect)
|
||||||
|
{
|
||||||
|
ALequalizerState *state = (ALequalizerState*)effect;
|
||||||
|
|
||||||
|
free(state);
|
||||||
|
}
|
||||||
|
|
||||||
|
static ALboolean EqualizerDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
|
||||||
|
{
|
||||||
|
ALequalizerState *state = (ALequalizerState*)effect;
|
||||||
|
|
||||||
|
state->frequency = (ALfloat)Device->Frequency;
|
||||||
|
|
||||||
|
return AL_TRUE;
|
||||||
|
}
|
||||||
|
|
||||||
|
static ALvoid EqualizerUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
|
||||||
|
{
|
||||||
|
ALequalizerState *state = (ALequalizerState*)effect;
|
||||||
|
ALfloat gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
|
||||||
|
ALuint it;
|
||||||
|
|
||||||
|
for(it = 0; it < Device->NumChan; it++)
|
||||||
|
{
|
||||||
|
enum Channel chan = Device->Speaker2Chan[it];
|
||||||
|
state->Gain[chan] = gain;
|
||||||
|
}
|
||||||
|
|
||||||
|
/* Calculate coefficients for the each type of filter */
|
||||||
|
for(it = 0; it < 4; it++)
|
||||||
|
{
|
||||||
|
ALfloat gain;
|
||||||
|
ALfloat filter_frequency;
|
||||||
|
ALfloat bandwidth = 0.0f;
|
||||||
|
ALfloat w0;
|
||||||
|
ALfloat alpha = 0.0f;
|
||||||
|
|
||||||
|
/* convert linear gains to filter gains */
|
||||||
|
switch (it)
|
||||||
|
{
|
||||||
|
case 0: /* Low Shelf */
|
||||||
|
gain = powf(10.0f, (20.0f * log10f(Slot->effect.Equalizer.LowGain)) / 40.0f);
|
||||||
|
filter_frequency = Slot->effect.Equalizer.LowCutoff;
|
||||||
|
break;
|
||||||
|
case 1: /* Peaking */
|
||||||
|
gain = powf(10.0f, (20.0f * log10f(Slot->effect.Equalizer.Mid1Gain)) / 40.0f);
|
||||||
|
filter_frequency = Slot->effect.Equalizer.Mid1Center;
|
||||||
|
bandwidth = Slot->effect.Equalizer.Mid1Width;
|
||||||
|
break;
|
||||||
|
case 2: /* Peaking */
|
||||||
|
gain = powf(10.0f, (20.0f * log10f(Slot->effect.Equalizer.Mid2Gain)) / 40.0f);
|
||||||
|
filter_frequency = Slot->effect.Equalizer.Mid2Center;
|
||||||
|
bandwidth = Slot->effect.Equalizer.Mid2Width;
|
||||||
|
break;
|
||||||
|
case 3: /* High Shelf */
|
||||||
|
gain = powf(10.0f, (20.0f * log10f(Slot->effect.Equalizer.HighGain)) / 40.0f);
|
||||||
|
filter_frequency = Slot->effect.Equalizer.HighCutoff;
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
w0 = 2.0f * F_PI * filter_frequency / state->frequency;
|
||||||
|
|
||||||
|
/* Calculate filter coefficients depending on filter type */
|
||||||
|
switch(state->bandfilter[it].type)
|
||||||
|
{
|
||||||
|
case LOW_SHELF:
|
||||||
|
alpha = sinf(w0) / 2.0f *
|
||||||
|
sqrtf((gain + 1.0f / gain) * (1.0f / 0.75f - 1.0f) + 2.0f);
|
||||||
|
state->bandfilter[it].b[0] = gain * ((gain + 1.0f) -
|
||||||
|
(gain - 1.0f) * cosf(w0) + 2.0f * sqrtf(gain) * alpha);
|
||||||
|
state->bandfilter[it].b[1] = 2.0f * gain * ((gain - 1.0f) -
|
||||||
|
(gain + 1.0f) * cosf(w0));
|
||||||
|
state->bandfilter[it].b[2] = gain * ((gain + 1.0f) -
|
||||||
|
(gain - 1.0f) * cosf(w0) - 2.0f * sqrtf(gain) * alpha);
|
||||||
|
state->bandfilter[it].a[0] = (gain + 1.0f) + (gain - 1.0f) *
|
||||||
|
cosf(w0) + 2.0f * sqrtf(gain) * alpha;
|
||||||
|
state->bandfilter[it].a[1] = -2.0f * ((gain - 1.0f) +
|
||||||
|
(gain + 1.0f) * cosf(w0));
|
||||||
|
state->bandfilter[it].a[2] = (gain + 1.0f) + (gain - 1.0f) *
|
||||||
|
cosf(w0) - 2.0f * sqrtf(gain) * alpha;
|
||||||
|
break;
|
||||||
|
case HIGH_SHELF:
|
||||||
|
alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
|
||||||
|
(1.0f / 0.75f - 1.0f) + 2.0f);
|
||||||
|
state->bandfilter[it].b[0] = gain * ((gain + 1.0f) +
|
||||||
|
(gain - 1.0f) * cosf(w0) +
|
||||||
|
2.0f * sqrtf(gain) * alpha);
|
||||||
|
state->bandfilter[it].b[1] = -2.0f * gain * ((gain - 1.0f) +
|
||||||
|
(gain + 1.0f) *
|
||||||
|
cosf(w0));
|
||||||
|
state->bandfilter[it].b[2] = gain * ((gain + 1.0f) +
|
||||||
|
(gain - 1.0f) * cosf(w0) -
|
||||||
|
2.0f * sqrtf(gain) * alpha);
|
||||||
|
state->bandfilter[it].a[0] = (gain + 1.0f) -
|
||||||
|
(gain - 1.0f) * cosf(w0) +
|
||||||
|
2.0f * sqrtf(gain) * alpha;
|
||||||
|
state->bandfilter[it].a[1] = 2.0f * ((gain - 1.0f) -
|
||||||
|
(gain + 1.0f) * cosf(w0));
|
||||||
|
state->bandfilter[it].a[2] = (gain + 1.0f) -
|
||||||
|
(gain - 1.0f) * cosf(w0) -
|
||||||
|
2.0f * sqrtf(gain) * alpha;
|
||||||
|
break;
|
||||||
|
case PEAKING:
|
||||||
|
alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
|
||||||
|
state->bandfilter[it].b[0] = 1.0f + alpha * gain;
|
||||||
|
state->bandfilter[it].b[1] = -2.0f * cosf(w0);
|
||||||
|
state->bandfilter[it].b[2] = 1.0f - alpha * gain;
|
||||||
|
state->bandfilter[it].a[0] = 1.0f + alpha / gain;
|
||||||
|
state->bandfilter[it].a[1] = -2.0f * cosf(w0);
|
||||||
|
state->bandfilter[it].a[2] = 1.0f - alpha / gain;
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
static ALvoid EqualizerProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
|
||||||
|
{
|
||||||
|
ALequalizerState *state = (ALequalizerState*)effect;
|
||||||
|
ALuint it;
|
||||||
|
ALuint kt;
|
||||||
|
ALuint ft;
|
||||||
|
|
||||||
|
for (it = 0; it < SamplesToDo; it++)
|
||||||
|
{
|
||||||
|
ALfloat tempsmp;
|
||||||
|
ALfloat smp = SamplesIn[it];
|
||||||
|
|
||||||
|
for(ft = 0;ft < 4;ft++)
|
||||||
|
{
|
||||||
|
tempsmp = state->bandfilter[ft].b[0] / state->bandfilter[ft].a[0] * smp +
|
||||||
|
state->bandfilter[ft].b[1] / state->bandfilter[ft].a[0] * state->bandfilter[ft].x[0] +
|
||||||
|
state->bandfilter[ft].b[2] / state->bandfilter[ft].a[0] * state->bandfilter[ft].x[1] -
|
||||||
|
state->bandfilter[ft].a[1] / state->bandfilter[ft].a[0] * state->bandfilter[ft].y[0] -
|
||||||
|
state->bandfilter[ft].a[2] / state->bandfilter[ft].a[0] * state->bandfilter[ft].y[1];
|
||||||
|
|
||||||
|
state->bandfilter[ft].x[1] = state->bandfilter[ft].x[0];
|
||||||
|
state->bandfilter[ft].x[0] = smp;
|
||||||
|
state->bandfilter[ft].y[1] = state->bandfilter[ft].y[0];
|
||||||
|
state->bandfilter[ft].y[0] = tempsmp;
|
||||||
|
smp = tempsmp;
|
||||||
|
}
|
||||||
|
|
||||||
|
for(kt = 0;kt < MaxChannels;kt++)
|
||||||
|
SamplesOut[kt][it] += state->Gain[kt] * smp;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
ALeffectState *EqualizerCreate(void)
|
||||||
|
{
|
||||||
|
ALequalizerState *state;
|
||||||
|
int it;
|
||||||
|
|
||||||
|
state = malloc(sizeof(*state));
|
||||||
|
if(!state)
|
||||||
|
return NULL;
|
||||||
|
|
||||||
|
state->state.Destroy = EqualizerDestroy;
|
||||||
|
state->state.DeviceUpdate = EqualizerDeviceUpdate;
|
||||||
|
state->state.Update = EqualizerUpdate;
|
||||||
|
state->state.Process = EqualizerProcess;
|
||||||
|
|
||||||
|
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 &state->state;
|
||||||
|
}
|
||||||
|
|
||||||
|
void equalizer_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
|
||||||
|
{
|
||||||
|
effect=effect;
|
||||||
|
val=val;
|
||||||
|
|
||||||
|
switch(param)
|
||||||
|
{
|
||||||
|
default:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void equalizer_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
|
||||||
|
{
|
||||||
|
equalizer_SetParami(effect, context, param, vals[0]);
|
||||||
|
}
|
||||||
|
void equalizer_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
|
||||||
|
{
|
||||||
|
switch(param)
|
||||||
|
{
|
||||||
|
case AL_EQUALIZER_LOW_GAIN:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_LOW_GAIN && val <= AL_EQUALIZER_MAX_LOW_GAIN)
|
||||||
|
effect->Equalizer.LowGain = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_LOW_CUTOFF:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_LOW_CUTOFF && val <= AL_EQUALIZER_MAX_LOW_CUTOFF)
|
||||||
|
effect->Equalizer.LowCutoff = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID1_GAIN:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_MID1_GAIN && val <= AL_EQUALIZER_MAX_MID1_GAIN)
|
||||||
|
effect->Equalizer.Mid1Gain = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID1_CENTER:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_MID1_CENTER && val <= AL_EQUALIZER_MAX_MID1_CENTER)
|
||||||
|
effect->Equalizer.Mid1Center = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID1_WIDTH:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_MID1_WIDTH && val <= AL_EQUALIZER_MAX_MID1_WIDTH)
|
||||||
|
effect->Equalizer.Mid1Width = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID2_GAIN:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_MID2_GAIN && val <= AL_EQUALIZER_MAX_MID2_GAIN)
|
||||||
|
effect->Equalizer.Mid2Gain = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID2_CENTER:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_MID2_CENTER && val <= AL_EQUALIZER_MAX_MID2_CENTER)
|
||||||
|
effect->Equalizer.Mid2Center = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID2_WIDTH:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_MID2_WIDTH && val <= AL_EQUALIZER_MAX_MID2_WIDTH)
|
||||||
|
effect->Equalizer.Mid2Width = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_HIGH_GAIN:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_HIGH_GAIN && val <= AL_EQUALIZER_MAX_HIGH_GAIN)
|
||||||
|
effect->Equalizer.HighGain = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_HIGH_CUTOFF:
|
||||||
|
if(val >= AL_EQUALIZER_MIN_HIGH_CUTOFF && val <= AL_EQUALIZER_MAX_HIGH_CUTOFF)
|
||||||
|
effect->Equalizer.HighCutoff = val;
|
||||||
|
else
|
||||||
|
alSetError(context, AL_INVALID_VALUE);
|
||||||
|
break;
|
||||||
|
|
||||||
|
default:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void equalizer_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
|
||||||
|
{
|
||||||
|
equalizer_SetParamf(effect, context, param, vals[0]);
|
||||||
|
}
|
||||||
|
|
||||||
|
void equalizer_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
|
||||||
|
{
|
||||||
|
effect=effect;
|
||||||
|
val=val;
|
||||||
|
|
||||||
|
switch(param)
|
||||||
|
{
|
||||||
|
default:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void equalizer_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
|
||||||
|
{
|
||||||
|
equalizer_GetParami(effect, context, param, vals);
|
||||||
|
}
|
||||||
|
void equalizer_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
|
||||||
|
{
|
||||||
|
switch(param)
|
||||||
|
{
|
||||||
|
case AL_EQUALIZER_LOW_GAIN:
|
||||||
|
*val = effect->Equalizer.LowGain;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_LOW_CUTOFF:
|
||||||
|
*val = effect->Equalizer.LowCutoff;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID1_GAIN:
|
||||||
|
*val = effect->Equalizer.Mid1Gain;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID1_CENTER:
|
||||||
|
*val = effect->Equalizer.Mid1Center;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID1_WIDTH:
|
||||||
|
*val = effect->Equalizer.Mid1Width;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID2_GAIN:
|
||||||
|
*val = effect->Equalizer.Mid2Gain;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID2_CENTER:
|
||||||
|
*val = effect->Equalizer.Mid2Center;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_MID2_WIDTH:
|
||||||
|
*val = effect->Equalizer.Mid2Width;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_HIGH_GAIN:
|
||||||
|
*val = effect->Equalizer.HighGain;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case AL_EQUALIZER_HIGH_CUTOFF:
|
||||||
|
*val = effect->Equalizer.HighCutoff;
|
||||||
|
break;
|
||||||
|
|
||||||
|
default:
|
||||||
|
alSetError(context, AL_INVALID_ENUM);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
void equalizer_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
|
||||||
|
{
|
||||||
|
equalizer_GetParamf(effect, context, param, vals);
|
||||||
|
}
|
||||||
@ -456,7 +456,9 @@ SET(ALC_OBJS Alc/ALc.c
|
|||||||
Alc/alcChorus.c
|
Alc/alcChorus.c
|
||||||
Alc/alcConfig.c
|
Alc/alcConfig.c
|
||||||
Alc/alcDedicated.c
|
Alc/alcDedicated.c
|
||||||
|
Alc/alcDistortion.c
|
||||||
Alc/alcEcho.c
|
Alc/alcEcho.c
|
||||||
|
Alc/alcEqualizer.c
|
||||||
Alc/alcFlanger.c
|
Alc/alcFlanger.c
|
||||||
Alc/alcModulator.c
|
Alc/alcModulator.c
|
||||||
Alc/alcReverb.c
|
Alc/alcReverb.c
|
||||||
|
|||||||
@ -48,6 +48,8 @@ ALeffectState *ModulatorCreate(void);
|
|||||||
ALeffectState *DedicatedCreate(void);
|
ALeffectState *DedicatedCreate(void);
|
||||||
ALeffectState *ChorusCreate(void);
|
ALeffectState *ChorusCreate(void);
|
||||||
ALeffectState *FlangerCreate(void);
|
ALeffectState *FlangerCreate(void);
|
||||||
|
ALeffectState *EqualizerCreate(void);
|
||||||
|
ALeffectState *DistortionCreate(void);
|
||||||
|
|
||||||
#define ALeffectState_Destroy(a) ((a)->Destroy((a)))
|
#define ALeffectState_Destroy(a) ((a)->Destroy((a)))
|
||||||
#define ALeffectState_DeviceUpdate(a,b) ((a)->DeviceUpdate((a),(b)))
|
#define ALeffectState_DeviceUpdate(a,b) ((a)->DeviceUpdate((a),(b)))
|
||||||
|
|||||||
@ -15,6 +15,8 @@ enum {
|
|||||||
DEDICATED,
|
DEDICATED,
|
||||||
CHORUS,
|
CHORUS,
|
||||||
FLANGER,
|
FLANGER,
|
||||||
|
EQUALIZER,
|
||||||
|
DISTORTION,
|
||||||
|
|
||||||
MAX_EFFECTS
|
MAX_EFFECTS
|
||||||
};
|
};
|
||||||
@ -95,6 +97,28 @@ typedef struct ALeffect
|
|||||||
ALfloat Delay;
|
ALfloat Delay;
|
||||||
} Flanger;
|
} Flanger;
|
||||||
|
|
||||||
|
struct {
|
||||||
|
ALfloat Delay;
|
||||||
|
ALfloat LowCutoff;
|
||||||
|
ALfloat LowGain;
|
||||||
|
ALfloat Mid1Center;
|
||||||
|
ALfloat Mid1Gain;
|
||||||
|
ALfloat Mid1Width;
|
||||||
|
ALfloat Mid2Center;
|
||||||
|
ALfloat Mid2Gain;
|
||||||
|
ALfloat Mid2Width;
|
||||||
|
ALfloat HighCutoff;
|
||||||
|
ALfloat HighGain;
|
||||||
|
} Equalizer;
|
||||||
|
|
||||||
|
struct {
|
||||||
|
ALfloat Edge;
|
||||||
|
ALfloat Gain;
|
||||||
|
ALfloat LowpassCutoff;
|
||||||
|
ALfloat EQCenter;
|
||||||
|
ALfloat EQBandwidth;
|
||||||
|
} Distortion;
|
||||||
|
|
||||||
void (*SetParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
void (*SetParami)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
||||||
void (*SetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
void (*SetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
||||||
void (*SetParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
void (*SetParamf)(struct ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
||||||
@ -149,6 +173,15 @@ void chorus_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALin
|
|||||||
void chorus_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
|
void chorus_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
|
||||||
void chorus_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
|
void chorus_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
|
||||||
|
|
||||||
|
void distortion_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
||||||
|
void distortion_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
||||||
|
void distortion_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
||||||
|
void distortion_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
|
||||||
|
void distortion_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
|
||||||
|
void distortion_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
|
||||||
|
void distortion_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
|
||||||
|
void distortion_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
|
||||||
|
|
||||||
void echo_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
void echo_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
||||||
void echo_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
void echo_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
||||||
void echo_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
void echo_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
||||||
@ -158,6 +191,15 @@ void echo_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint
|
|||||||
void echo_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
|
void echo_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
|
||||||
void echo_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
|
void echo_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
|
||||||
|
|
||||||
|
void equalizer_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
||||||
|
void equalizer_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
||||||
|
void equalizer_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
||||||
|
void equalizer_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals);
|
||||||
|
void equalizer_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val);
|
||||||
|
void equalizer_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals);
|
||||||
|
void equalizer_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val);
|
||||||
|
void equalizer_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals);
|
||||||
|
|
||||||
void flanger_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
void flanger_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val);
|
||||||
void flanger_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
void flanger_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
|
||||||
void flanger_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
void flanger_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);
|
||||||
|
|||||||
@ -513,8 +513,12 @@ static ALeffectState *CreateStateByType(ALenum type)
|
|||||||
return ReverbCreate();
|
return ReverbCreate();
|
||||||
case AL_EFFECT_CHORUS:
|
case AL_EFFECT_CHORUS:
|
||||||
return ChorusCreate();
|
return ChorusCreate();
|
||||||
|
case AL_EFFECT_DISTORTION:
|
||||||
|
return DistortionCreate();
|
||||||
case AL_EFFECT_ECHO:
|
case AL_EFFECT_ECHO:
|
||||||
return EchoCreate();
|
return EchoCreate();
|
||||||
|
case AL_EFFECT_EQUALIZER:
|
||||||
|
return EqualizerCreate();
|
||||||
case AL_EFFECT_FLANGER:
|
case AL_EFFECT_FLANGER:
|
||||||
return FlangerCreate();
|
return FlangerCreate();
|
||||||
case AL_EFFECT_RING_MODULATOR:
|
case AL_EFFECT_RING_MODULATOR:
|
||||||
|
|||||||
@ -430,7 +430,7 @@ static void InitEffectParams(ALeffect *effect, ALenum type)
|
|||||||
case AL_EFFECT_CHORUS:
|
case AL_EFFECT_CHORUS:
|
||||||
effect->Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM;
|
effect->Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM;
|
||||||
effect->Chorus.Phase = AL_CHORUS_DEFAULT_PHASE;
|
effect->Chorus.Phase = AL_CHORUS_DEFAULT_PHASE;
|
||||||
effect->Chorus.Rate = AL_CHORUS_MAX_RATE;
|
effect->Chorus.Rate = AL_CHORUS_DEFAULT_RATE;
|
||||||
effect->Chorus.Depth = AL_CHORUS_DEFAULT_DEPTH;
|
effect->Chorus.Depth = AL_CHORUS_DEFAULT_DEPTH;
|
||||||
effect->Chorus.Feedback = AL_CHORUS_DEFAULT_FEEDBACK;
|
effect->Chorus.Feedback = AL_CHORUS_DEFAULT_FEEDBACK;
|
||||||
effect->Chorus.Delay = AL_CHORUS_DEFAULT_DELAY;
|
effect->Chorus.Delay = AL_CHORUS_DEFAULT_DELAY;
|
||||||
@ -443,6 +443,21 @@ static void InitEffectParams(ALeffect *effect, ALenum type)
|
|||||||
effect->GetParamf = chorus_GetParamf;
|
effect->GetParamf = chorus_GetParamf;
|
||||||
effect->GetParamfv = chorus_GetParamfv;
|
effect->GetParamfv = chorus_GetParamfv;
|
||||||
break;
|
break;
|
||||||
|
case AL_EFFECT_DISTORTION:
|
||||||
|
effect->Distortion.Edge = AL_DISTORTION_DEFAULT_EDGE;
|
||||||
|
effect->Distortion.Gain = AL_DISTORTION_DEFAULT_GAIN;
|
||||||
|
effect->Distortion.LowpassCutoff = AL_DISTORTION_DEFAULT_LOWPASS_CUTOFF;
|
||||||
|
effect->Distortion.EQCenter = AL_DISTORTION_DEFAULT_EQCENTER;
|
||||||
|
effect->Distortion.EQBandwidth = AL_DISTORTION_DEFAULT_EQBANDWIDTH;
|
||||||
|
effect->SetParami = distortion_SetParami;
|
||||||
|
effect->SetParamiv = distortion_SetParamiv;
|
||||||
|
effect->SetParamf = distortion_SetParamf;
|
||||||
|
effect->SetParamfv = distortion_SetParamfv;
|
||||||
|
effect->GetParami = distortion_GetParami;
|
||||||
|
effect->GetParamiv = distortion_GetParamiv;
|
||||||
|
effect->GetParamf = distortion_GetParamf;
|
||||||
|
effect->GetParamfv = distortion_GetParamfv;
|
||||||
|
break;
|
||||||
case AL_EFFECT_ECHO:
|
case AL_EFFECT_ECHO:
|
||||||
effect->Echo.Delay = AL_ECHO_DEFAULT_DELAY;
|
effect->Echo.Delay = AL_ECHO_DEFAULT_DELAY;
|
||||||
effect->Echo.LRDelay = AL_ECHO_DEFAULT_LRDELAY;
|
effect->Echo.LRDelay = AL_ECHO_DEFAULT_LRDELAY;
|
||||||
@ -458,10 +473,30 @@ static void InitEffectParams(ALeffect *effect, ALenum type)
|
|||||||
effect->GetParamf = echo_GetParamf;
|
effect->GetParamf = echo_GetParamf;
|
||||||
effect->GetParamfv = echo_GetParamfv;
|
effect->GetParamfv = echo_GetParamfv;
|
||||||
break;
|
break;
|
||||||
|
case AL_EFFECT_EQUALIZER:
|
||||||
|
effect->Equalizer.LowCutoff = AL_EQUALIZER_DEFAULT_LOW_CUTOFF;
|
||||||
|
effect->Equalizer.LowGain = AL_EQUALIZER_DEFAULT_LOW_GAIN;
|
||||||
|
effect->Equalizer.Mid1Center = AL_EQUALIZER_DEFAULT_MID1_CENTER;
|
||||||
|
effect->Equalizer.Mid1Gain = AL_EQUALIZER_DEFAULT_MID1_GAIN;
|
||||||
|
effect->Equalizer.Mid1Width = AL_EQUALIZER_DEFAULT_MID1_WIDTH;
|
||||||
|
effect->Equalizer.Mid2Center = AL_EQUALIZER_DEFAULT_MID2_CENTER;
|
||||||
|
effect->Equalizer.Mid2Gain = AL_EQUALIZER_DEFAULT_MID2_GAIN;
|
||||||
|
effect->Equalizer.Mid2Width = AL_EQUALIZER_DEFAULT_MID2_WIDTH;
|
||||||
|
effect->Equalizer.HighCutoff = AL_EQUALIZER_DEFAULT_HIGH_CUTOFF;
|
||||||
|
effect->Equalizer.HighGain = AL_EQUALIZER_DEFAULT_HIGH_GAIN;
|
||||||
|
effect->SetParami = equalizer_SetParami;
|
||||||
|
effect->SetParamiv = equalizer_SetParamiv;
|
||||||
|
effect->SetParamf = equalizer_SetParamf;
|
||||||
|
effect->SetParamfv = equalizer_SetParamfv;
|
||||||
|
effect->GetParami = equalizer_GetParami;
|
||||||
|
effect->GetParamiv = equalizer_GetParamiv;
|
||||||
|
effect->GetParamf = equalizer_GetParamf;
|
||||||
|
effect->GetParamfv = equalizer_GetParamfv;
|
||||||
|
break;
|
||||||
case AL_EFFECT_FLANGER:
|
case AL_EFFECT_FLANGER:
|
||||||
effect->Flanger.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
|
effect->Flanger.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
|
||||||
effect->Flanger.Phase = AL_FLANGER_DEFAULT_PHASE;
|
effect->Flanger.Phase = AL_FLANGER_DEFAULT_PHASE;
|
||||||
effect->Flanger.Rate = AL_FLANGER_MAX_RATE;
|
effect->Flanger.Rate = AL_FLANGER_DEFAULT_RATE;
|
||||||
effect->Flanger.Depth = AL_FLANGER_DEFAULT_DEPTH;
|
effect->Flanger.Depth = AL_FLANGER_DEFAULT_DEPTH;
|
||||||
effect->Flanger.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
|
effect->Flanger.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
|
||||||
effect->Flanger.Delay = AL_FLANGER_DEFAULT_DELAY;
|
effect->Flanger.Delay = AL_FLANGER_DEFAULT_DELAY;
|
||||||
|
|||||||
@ -44,6 +44,8 @@ const struct EffectList EffectList[] = {
|
|||||||
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
|
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
|
||||||
{ "chorus", CHORUS, "AL_EFFECT_CHORUS", AL_EFFECT_CHORUS },
|
{ "chorus", CHORUS, "AL_EFFECT_CHORUS", AL_EFFECT_CHORUS },
|
||||||
{ "flanger", FLANGER, "AL_EFFECT_FLANGER", AL_EFFECT_FLANGER },
|
{ "flanger", FLANGER, "AL_EFFECT_FLANGER", AL_EFFECT_FLANGER },
|
||||||
|
{ "equalizer", EQUALIZER, "AL_EFFECT_EQUALIZER", AL_EFFECT_EQUALIZER },
|
||||||
|
{ "distortion", DISTORTION, "AL_EFFECT_DISTORTION", AL_EFFECT_DISTORTION },
|
||||||
{ NULL, 0, NULL, (ALenum)0 }
|
{ NULL, 0, NULL, (ALenum)0 }
|
||||||
};
|
};
|
||||||
|
|
||||||
|
|||||||
@ -126,8 +126,8 @@
|
|||||||
## excludefx:
|
## excludefx:
|
||||||
# Sets which effects to exclude, preventing apps from using them. This can
|
# Sets which effects to exclude, preventing apps from using them. This can
|
||||||
# help for apps that try to use effects which are too CPU intensive for the
|
# help for apps that try to use effects which are too CPU intensive for the
|
||||||
# system to handle. Available effects are: eaxreverb,reverb,chorus,echo,
|
# system to handle. Available effects are: eaxreverb,reverb,chorus,distortion,
|
||||||
# flanger,modulator,dedicated
|
# echo,equalizer,flanger,modulator,dedicated
|
||||||
#excludefx =
|
#excludefx =
|
||||||
|
|
||||||
## slots:
|
## slots:
|
||||||
|
|||||||
Loading…
Reference in New Issue
Block a user