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Implement distortion and equalizer effects

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Code provided by Mike Gorchak
This commit is contained in:
Chris Robinson 2013-05-18 01:33:01 -07:00
parent a7ad6080f0
commit 78e7c1c27b
10 changed files with 919 additions and 15 deletions
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21
Alc/ALc.c
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@ -516,9 +516,7 @@ static const ALCenums enumeration[] = {
DECL(AL_EFFECT_REVERB),
DECL(AL_EFFECT_EAXREVERB),
DECL(AL_EFFECT_CHORUS),
#if 0
DECL(AL_EFFECT_DISTORTION),
#endif
DECL(AL_EFFECT_ECHO),
DECL(AL_EFFECT_FLANGER),
#if 0
@ -530,8 +528,8 @@ static const ALCenums enumeration[] = {
#if 0
DECL(AL_EFFECT_AUTOWAH),
DECL(AL_EFFECT_COMPRESSOR),
DECL(AL_EFFECT_EQUALIZER),
#endif
DECL(AL_EFFECT_EQUALIZER),
DECL(AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT),
DECL(AL_EFFECT_DEDICATED_DIALOGUE),
@ -593,6 +591,23 @@ static const ALCenums enumeration[] = {
DECL(AL_FLANGER_FEEDBACK),
DECL(AL_FLANGER_DELAY),
DECL(AL_EQUALIZER_LOW_GAIN),
DECL(AL_EQUALIZER_LOW_CUTOFF),
DECL(AL_EQUALIZER_MID1_GAIN),
DECL(AL_EQUALIZER_MID1_CENTER),
DECL(AL_EQUALIZER_MID1_WIDTH),
DECL(AL_EQUALIZER_MID2_GAIN),
DECL(AL_EQUALIZER_MID2_CENTER),
DECL(AL_EQUALIZER_MID2_WIDTH),
DECL(AL_EQUALIZER_HIGH_GAIN),
DECL(AL_EQUALIZER_HIGH_CUTOFF),
DECL(AL_DISTORTION_EDGE),
DECL(AL_DISTORTION_GAIN),
DECL(AL_DISTORTION_LOWPASS_CUTOFF),
DECL(AL_DISTORTION_EQCENTER),
DECL(AL_DISTORTION_EQBANDWIDTH),
DECL(AL_RING_MODULATOR_FREQUENCY),
DECL(AL_RING_MODULATOR_HIGHPASS_CUTOFF),
DECL(AL_RING_MODULATOR_WAVEFORM),

356
Alc/alcDistortion.c Normal file
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@ -0,0 +1,356 @@
/**
* 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"
/* Filters 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 {
LOWPASS,
BANDPASS,
} 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 ALdistortionState {
/* Must be first in all effects! */
ALeffectState state;
/* Effect gains for each channel */
ALfloat Gain[MaxChannels];
/* Effect parameters */
ALEQFilter bandpass;
ALEQFilter lowpass;
ALfloat frequency;
ALfloat attenuation;
ALfloat edge;
/* Oversample data */
ALfloat oversample_buffer[BUFFERSIZE][4];
} ALdistortionState;
static ALvoid DistortionDestroy(ALeffectState *effect)
{
ALdistortionState *state = (ALdistortionState*)effect;
free(state);
}
static ALboolean DistortionDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
ALdistortionState *state = (ALdistortionState*)effect;
state->frequency = (ALfloat)Device->Frequency;
return AL_TRUE;
}
static ALvoid DistortionUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
{
ALdistortionState *state = (ALdistortionState*)effect;
ALfloat gain = sqrtf(1.0f / Device->NumChan) * Slot->Gain;
ALuint it;
ALfloat w0;
ALfloat alpha;
ALfloat bandwidth;
ALfloat cutoff;
for(it = 0; it < Device->NumChan; it++)
{
enum Channel chan = Device->Speaker2Chan[it];
state->Gain[chan] = gain;
}
/* Store distorted signal attenuation settings */
state->attenuation = Slot->effect.Distortion.Gain;
/* Store waveshaper edge settings */
state->edge = Slot->effect.Distortion.Edge;
/* Lowpass filter */
cutoff = Slot->effect.Distortion.LowpassCutoff;
/* Bandwidth value is constant in octaves */
bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
state->lowpass.b[0] = (1.0f - cosf(w0)) / 2.0f;
state->lowpass.b[1] = 1.0f - cosf(w0);
state->lowpass.b[2] = (1.0f - cosf(w0)) / 2.0f;
state->lowpass.a[0] = 1.0f + alpha;
state->lowpass.a[1] = -2.0f * cosf(w0);
state->lowpass.a[2] = 1.0f - alpha;
/* Bandpass filter */
cutoff = Slot->effect.Distortion.EQCenter;
/* Convert bandwidth in Hz to octaves */
bandwidth = Slot->effect.Distortion.EQBandwidth / (cutoff * 0.67f);
w0 = 2.0f * F_PI * cutoff / (state->frequency * 4.0f);
alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
state->bandpass.b[0] = alpha;
state->bandpass.b[1] = 0;
state->bandpass.b[2] = -alpha;
state->bandpass.a[0] = 1.0f + alpha;
state->bandpass.a[1] = -2.0f * cosf(w0);
state->bandpass.a[2] = 1.0f - alpha;
}
static ALvoid DistortionProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
{
ALdistortionState *state = (ALdistortionState*)effect;
float *RESTRICT oversample_buffer = &state->oversample_buffer[0][0];
ALfloat tempsmp;
ALuint it;
ALuint kt;
ALuint st;
/* Perform 4x oversampling to avoid aliasing. */
/* Oversampling greatly improves distortion */
/* quality and allows to implement lowpass and */
/* bandpass filters using high frequencies, at */
/* which classic IIR filters became unstable. */
/* Fill oversample buffer using zero stuffing */
for(it = 0; it < SamplesToDo; it++)
{
oversample_buffer[it*4 + 0] = SamplesIn[it];
oversample_buffer[it*4 + 1] = 0.0f;
oversample_buffer[it*4 + 2] = 0.0f;
oversample_buffer[it*4 + 3] = 0.0f;
}
/* First step, do lowpass filtering of original signal, */
/* additionally perform buffer interpolation and lowpass */
/* cutoff for oversampling (which is fortunately first */
/* step of distortion). So combine three operations into */
/* the one. */
for(it = 0; it < SamplesToDo * 4; it++)
{
tempsmp = state->lowpass.b[0] / state->lowpass.a[0] * oversample_buffer[it] +
state->lowpass.b[1] / state->lowpass.a[0] * state->lowpass.x[0] +
state->lowpass.b[2] / state->lowpass.a[0] * state->lowpass.x[1] -
state->lowpass.a[1] / state->lowpass.a[0] * state->lowpass.y[0] -
state->lowpass.a[2] / state->lowpass.a[0] * state->lowpass.y[1];
state->lowpass.x[1] = state->lowpass.x[0];
state->lowpass.x[0] = oversample_buffer[it];
state->lowpass.y[1] = state->lowpass.y[0];
state->lowpass.y[0] = tempsmp;
/* Restore signal power by multiplying sample by amount of oversampling */
oversample_buffer[it] = tempsmp * 4.0f;
}
for(it = 0; it < SamplesToDo * 4; it++)
{
ALfloat smp = oversample_buffer[it];
ALfloat edge = sinf(state->edge * (F_PI / 2.0f));
/* Second step, do distortion using waveshaper function */
/* to emulate signal processing during tube overdriving. */
/* Three steps of waveshaping are intended to modify */
/* waveform without boost/clipping/attenuation process. */
for(st = 0; st < 3; st++)
{
smp = (1.0f + 2.0f * edge / (1.0f - edge)) * smp / (1.0f + 2.0f * edge / (1.0f - edge) * fabsf(smp));
if((st & 0x00000001) == 0x00000001)
smp *= -1.0f;
}
/* Third step, do bandpass filtering of distorted signal */
tempsmp = state->bandpass.b[0] / state->bandpass.a[0] * smp +
state->bandpass.b[1] / state->bandpass.a[0] * state->bandpass.x[0] +
state->bandpass.b[2] / state->bandpass.a[0] * state->bandpass.x[1] -
state->bandpass.a[1] / state->bandpass.a[0] * state->bandpass.y[0] -
state->bandpass.a[2] / state->bandpass.a[0] * state->bandpass.y[1];
state->bandpass.x[1] = state->bandpass.x[0];
state->bandpass.x[0] = smp;
state->bandpass.y[1] = state->bandpass.y[0];
state->bandpass.y[0] = tempsmp;
smp = tempsmp;
/* Fourth step, final, do attenuation and perform decimation, */
/* store only one sample out of 4. */
if(!(it & 0x00000003))
{
smp *= state->attenuation;
for(kt = 0; kt < MaxChannels; kt++)
SamplesOut[kt][it>>2] += state->Gain[kt] * smp;
}
}
}
ALeffectState *DistortionCreate(void)
{
ALdistortionState *state;
state = malloc(sizeof(*state));
if(!state)
return NULL;
state->state.Destroy = DistortionDestroy;
state->state.DeviceUpdate = DistortionDeviceUpdate;
state->state.Update = DistortionUpdate;
state->state.Process = DistortionProcess;
state->bandpass.type = BANDPASS;
state->lowpass.type = LOWPASS;
/* Initialize sample history only on filter creation to avoid */
/* sound clicks if filter settings were changed in runtime. */
state->bandpass.x[0] = 0.0f;
state->bandpass.x[1] = 0.0f;
state->lowpass.y[0] = 0.0f;
state->lowpass.y[1] = 0.0f;
return &state->state;
}
void distortion_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
effect=effect;
val=val;
switch(param)
{
default:
alSetError(context, AL_INVALID_ENUM);
break;
}
}
void distortion_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
distortion_SetParami(effect, context, param, vals[0]);
}
void distortion_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
switch(param)
{
case AL_DISTORTION_EDGE:
if(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE)
effect->Distortion.Edge = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
case AL_DISTORTION_GAIN:
if(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN)
effect->Distortion.Gain = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
case AL_DISTORTION_LOWPASS_CUTOFF:
if(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF)
effect->Distortion.LowpassCutoff = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
case AL_DISTORTION_EQCENTER:
if(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER)
effect->Distortion.EQCenter = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
case AL_DISTORTION_EQBANDWIDTH:
if(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH)
effect->Distortion.EQBandwidth = val;
else
alSetError(context, AL_INVALID_VALUE);
break;
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
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@ -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);
}

2
CMakeLists.txt
View File

@ -456,7 +456,9 @@ SET(ALC_OBJS Alc/ALc.c
Alc/alcChorus.c
Alc/alcConfig.c
Alc/alcDedicated.c
Alc/alcDistortion.c
Alc/alcEcho.c
Alc/alcEqualizer.c
Alc/alcFlanger.c
Alc/alcModulator.c
Alc/alcReverb.c

2
OpenAL32/Include/alAuxEffectSlot.h
View File

@ -48,6 +48,8 @@ ALeffectState *ModulatorCreate(void);
ALeffectState *DedicatedCreate(void);
ALeffectState *ChorusCreate(void);
ALeffectState *FlangerCreate(void);
ALeffectState *EqualizerCreate(void);
ALeffectState *DistortionCreate(void);
#define ALeffectState_Destroy(a) ((a)->Destroy((a)))
#define ALeffectState_DeviceUpdate(a,b) ((a)->DeviceUpdate((a),(b)))

42
OpenAL32/Include/alEffect.h
View File

@ -15,6 +15,8 @@ enum {
DEDICATED,
CHORUS,
FLANGER,
EQUALIZER,
DISTORTION,
MAX_EFFECTS
};
@ -95,6 +97,28 @@ typedef struct ALeffect
ALfloat Delay;
} 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 (*SetParamiv)(struct ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
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_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_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
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_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_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals);
void flanger_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val);

4
OpenAL32/alAuxEffectSlot.c
View File

@ -513,8 +513,12 @@ static ALeffectState *CreateStateByType(ALenum type)
return ReverbCreate();
case AL_EFFECT_CHORUS:
return ChorusCreate();
case AL_EFFECT_DISTORTION:
return DistortionCreate();
case AL_EFFECT_ECHO:
return EchoCreate();
case AL_EFFECT_EQUALIZER:
return EqualizerCreate();
case AL_EFFECT_FLANGER:
return FlangerCreate();
case AL_EFFECT_RING_MODULATOR:

39
OpenAL32/alEffect.c
View File

@ -430,7 +430,7 @@ static void InitEffectParams(ALeffect *effect, ALenum type)
case AL_EFFECT_CHORUS:
effect->Chorus.Waveform = AL_CHORUS_DEFAULT_WAVEFORM;
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.Feedback = AL_CHORUS_DEFAULT_FEEDBACK;
effect->Chorus.Delay = AL_CHORUS_DEFAULT_DELAY;
@ -443,6 +443,21 @@ static void InitEffectParams(ALeffect *effect, ALenum type)
effect->GetParamf = chorus_GetParamf;
effect->GetParamfv = chorus_GetParamfv;
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:
effect->Echo.Delay = AL_ECHO_DEFAULT_DELAY;
effect->Echo.LRDelay = AL_ECHO_DEFAULT_LRDELAY;
@ -458,10 +473,30 @@ static void InitEffectParams(ALeffect *effect, ALenum type)
effect->GetParamf = echo_GetParamf;
effect->GetParamfv = echo_GetParamfv;
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:
effect->Flanger.Waveform = AL_FLANGER_DEFAULT_WAVEFORM;
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.Feedback = AL_FLANGER_DEFAULT_FEEDBACK;
effect->Flanger.Delay = AL_FLANGER_DEFAULT_DELAY;

18
OpenAL32/alExtension.c
View File

@ -36,14 +36,16 @@
const struct EffectList EffectList[] = {
{ "eaxreverb", EAXREVERB, "AL_EFFECT_EAXREVERB", AL_EFFECT_EAXREVERB },
{ "reverb", REVERB, "AL_EFFECT_REVERB", AL_EFFECT_REVERB },
{ "echo", ECHO, "AL_EFFECT_ECHO", AL_EFFECT_ECHO },
{ "modulator", MODULATOR, "AL_EFFECT_RING_MODULATOR", AL_EFFECT_RING_MODULATOR },
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT", AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
{ "chorus", CHORUS, "AL_EFFECT_CHORUS", AL_EFFECT_CHORUS },
{ "flanger", FLANGER, "AL_EFFECT_FLANGER", AL_EFFECT_FLANGER },
{ "eaxreverb", EAXREVERB, "AL_EFFECT_EAXREVERB", AL_EFFECT_EAXREVERB },
{ "reverb", REVERB, "AL_EFFECT_REVERB", AL_EFFECT_REVERB },
{ "echo", ECHO, "AL_EFFECT_ECHO", AL_EFFECT_ECHO },
{ "modulator", MODULATOR, "AL_EFFECT_RING_MODULATOR", AL_EFFECT_RING_MODULATOR },
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT", AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT },
{ "dedicated", DEDICATED, "AL_EFFECT_DEDICATED_DIALOGUE", AL_EFFECT_DEDICATED_DIALOGUE },
{ "chorus", CHORUS, "AL_EFFECT_CHORUS", AL_EFFECT_CHORUS },
{ "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 }
};

4
alsoftrc.sample
View File

@ -126,8 +126,8 @@
## excludefx:
# 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
# system to handle. Available effects are: eaxreverb,reverb,chorus,echo,
# flanger,modulator,dedicated
# system to handle. Available effects are: eaxreverb,reverb,chorus,distortion,
# echo,equalizer,flanger,modulator,dedicated
#excludefx =
## slots: