Move remaining effects to the effects subdir

1 files changed, 0 insertions, 499 deletions

diff --git a/Alc/alcEqualizer.c b/Alc/alcEqualizer.c
deleted file mode 100644
index d397a9ba..00000000
--- a/Alc/alcEqualizer.c
+++ /dev/null
@@ -1,499 +0,0 @@
-/**
- * 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"
-
-
-typedef struct ALequalizerStateFactory {
-    DERIVE_FROM_TYPE(ALeffectStateFactory);
-} ALequalizerStateFactory;
-
-static ALequalizerStateFactory EqualizerFactory;
-
-
-/*  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 {
-    DERIVE_FROM_TYPE(ALeffectState);
-
-    /* Effect gains for each channel */
-    ALfloat Gain[MaxChannels];
-
-    /* Effect parameters */
-    ALEQFilter bandfilter[4];
-} ALequalizerState;
-
-static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
-{
-    (void)state;
-}
-
-static ALboolean ALequalizerState_DeviceUpdate(ALequalizerState *state, ALCdevice *device)
-{
-    return AL_TRUE;
-    (void)state;
-    (void)device;
-}
-
-static ALvoid ALequalizerState_Update(ALequalizerState *state, ALCdevice *device, const ALeffectslot *slot)
-{
-    ALfloat frequency = (ALfloat)device->Frequency;
-    ALfloat gain = sqrtf(1.0f / device->NumChan) * slot->Gain;
-    ALuint it;
-
-    for(it = 0;it < MaxChannels;it++)
-        state->Gain[it] = 0.0f;
-    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 / 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 ALequalizerState_Process(ALequalizerState *state, ALuint SamplesToDo, const ALfloat *restrict SamplesIn, ALfloat (*restrict SamplesOut)[BUFFERSIZE])
-{
-    ALuint base;
-    ALuint it;
-    ALuint kt;
-    ALuint ft;
-
-    for(base = 0;base < SamplesToDo;)
-    {
-        ALfloat temps[64];
-        ALuint td = minu(SamplesToDo-base, 64);
-
-        for(it = 0;it < td;it++)
-        {
-            ALfloat smp = SamplesIn[base+it];
-            ALfloat tempsmp;
-
-            for(ft = 0;ft < 4;ft++)
-            {
-                ALEQFilter *filter = &state->bandfilter[ft];
-
-                tempsmp = filter->b[0] / filter->a[0] * smp +
-                          filter->b[1] / filter->a[0] * filter->x[0] +
-                          filter->b[2] / filter->a[0] * filter->x[1] -
-                          filter->a[1] / filter->a[0] * filter->y[0] -
-                          filter->a[2] / filter->a[0] * filter->y[1];
-
-                filter->x[1] = filter->x[0];
-                filter->x[0] = smp;
-                filter->y[1] = filter->y[0];
-                filter->y[0] = tempsmp;
-                smp = tempsmp;
-            }
-
-            temps[it] = smp;
-        }
-
-        for(kt = 0;kt < MaxChannels;kt++)
-        {
-            ALfloat gain = state->Gain[kt];
-            if(!(gain > 0.00001f))
-                continue;
-
-            for(it = 0;it < td;it++)
-                SamplesOut[kt][base+it] += gain * temps[it];
-        }
-
-        base += td;
-    }
-}
-
-static ALeffectStateFactory *ALequalizerState_getCreator(void)
-{
-    return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
-}
-
-DEFINE_ALEFFECTSTATE_VTABLE(ALequalizerState);
-
-
-ALeffectState *ALequalizerStateFactory_create(void)
-{
-    ALequalizerState *state;
-    int it;
-
-    state = malloc(sizeof(*state));
-    if(!state) return NULL;
-    SET_VTABLE2(ALequalizerState, ALeffectState, state);
-
-    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 STATIC_CAST(ALeffectState, state);
-}
-
-static ALvoid ALequalizerStateFactory_destroy(ALeffectState *effect)
-{
-    ALequalizerState *state = STATIC_UPCAST(ALequalizerState, ALeffectState, effect);
-    ALequalizerState_Destruct(state);
-    free(state);
-}
-
-DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALequalizerStateFactory);
-
-
-static void init_equalizer_factory(void)
-{
-    SET_VTABLE2(ALequalizerStateFactory, ALeffectStateFactory, &EqualizerFactory);
-}
-
-ALeffectStateFactory *ALequalizerStateFactory_getFactory(void)
-{
-    static pthread_once_t once = PTHREAD_ONCE_INIT;
-    pthread_once(&once, init_equalizer_factory);
-    return STATIC_CAST(ALeffectStateFactory, &EqualizerFactory);
-}
-
-
-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);
-}