Implement distortion and equalizer effects

Code provided by Mike Gorchak

3 files changed, 820 insertions, 3 deletions

diff --git a/Alc/ALc.c b/Alc/ALc.c
index e859a6c2..24c2fbe5 100644
--- a/Alc/ALc.c
+++ b/Alc/ALc.c
@@ -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),
diff --git a/Alc/alcDistortion.c b/Alc/alcDistortion.c
new file mode 100644
index 00000000..717c7b3a
--- /dev/null
+++ b/Alc/alcDistortion.c
@@ -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);
+}
diff --git a/Alc/alcEqualizer.c b/Alc/alcEqualizer.c
new file mode 100644
index 00000000..2067c319
--- /dev/null
+++ b/Alc/alcEqualizer.c
@@ -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);
+}