Add Chorus and Flanger effects

Code provided by Mike Gorchak

1 files changed, 395 insertions, 0 deletions

diff --git a/Alc/alcChorus.c b/Alc/alcChorus.c
new file mode 100644
index 00000000..5a97ff17
--- /dev/null
+++ b/Alc/alcChorus.c
@@ -0,0 +1,395 @@
+/**
+ * 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 ALchorusState {
+    // Must be first in all effects!
+    ALeffectState state;
+
+    ALfloat *SampleBufferLeft;
+    ALfloat *SampleBufferRight;
+    ALuint BufferLength;
+    ALint offset;
+    ALfloat lfo_coeff;
+    ALint lfo_disp;
+
+    /* Gains for left and right sides */
+    ALfloat Gain[2][MaxChannels];
+
+    /* effect parameters */
+    ALint waveform;
+    ALint phase;
+    ALfloat rate;
+    ALfloat depth;
+    ALfloat feedback;
+    ALfloat delay;
+    ALfloat frequency;
+} ALchorusState;
+
+static ALvoid ChorusDestroy(ALeffectState *effect)
+{
+    ALchorusState *state = (ALchorusState*)effect;
+
+    if (state)
+    {
+        if (state->SampleBufferLeft != NULL)
+        {
+            free(state->SampleBufferLeft);
+            state->SampleBufferLeft = NULL;
+        }
+        if (state->SampleBufferRight != NULL)
+        {
+            free(state->SampleBufferRight);
+            state->SampleBufferRight = NULL;
+        }
+        free(state);
+    }
+}
+
+static ALboolean ChorusDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
+{
+    ALchorusState *state = (ALchorusState*)effect;
+    ALuint maxlen;
+    ALuint it;
+
+    maxlen = fastf2u(AL_CHORUS_MAX_DELAY * 3.0f * Device->Frequency) + 1;
+    maxlen = NextPowerOf2(maxlen);
+
+    if (maxlen != state->BufferLength)
+    {
+        void *temp;
+
+        temp = realloc(state->SampleBufferLeft, maxlen * sizeof(ALfloat));
+        if (!temp)
+        {
+            return AL_FALSE;
+        }
+        state->SampleBufferLeft = temp;
+
+        temp = realloc(state->SampleBufferRight, maxlen * sizeof(ALfloat));
+        if (!temp)
+        {
+            return AL_FALSE;
+        }
+        state->SampleBufferRight = temp;
+
+        state->BufferLength = maxlen;
+    }
+
+    for (it = 0; it < state->BufferLength; it++)
+    {
+        state->SampleBufferLeft[it] = 0.0f;
+        state->SampleBufferRight[it] = 0.0f;
+    }
+
+    state->frequency=(ALfloat)Device->Frequency;
+
+    return AL_TRUE;
+}
+
+static ALvoid ChorusUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
+{
+    ALchorusState *state = (ALchorusState*)effect;
+    ALuint it;
+
+    for (it = 0; it < MaxChannels; it++)
+    {
+        state->Gain[0][it] = 0.0f;
+        state->Gain[1][it] = 0.0f;
+    }
+
+    state->waveform = Slot->effect.Chorus.Waveform;
+    state->phase = Slot->effect.Chorus.Phase;
+    state->rate = Slot->effect.Chorus.Rate;
+    state->depth = Slot->effect.Chorus.Depth;
+    state->feedback = Slot->effect.Chorus.Feedback;
+    state->delay = Slot->effect.Chorus.Delay;
+    state->frequency=(ALfloat)Device->Frequency;
+
+    /* Gains for left and right sides */
+    ComputeAngleGains(Device, atan2f(-1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[0]);
+    ComputeAngleGains(Device, atan2f(+1.0f, 0.0f), 0.0f, Slot->Gain, state->Gain[1]);
+
+    /* Calculate LFO coefficient */
+    switch (state->waveform)
+    {
+        case AL_CHORUS_WAVEFORM_TRIANGLE:
+             if (state->rate == 0.0f)
+             {
+                 state->lfo_coeff = 0.0f;
+             }
+             else
+             {
+                 state->lfo_coeff = 1.0f / ((ALfloat)Device->Frequency / state->rate);
+             }
+             break;
+        case AL_CHORUS_WAVEFORM_SINUSOID:
+             if (state->rate == 0.0f)
+             {
+                 state->lfo_coeff = 0.0f;
+             }
+             else
+             {
+                 state->lfo_coeff = F_PI * 2.0f / ((ALfloat)Device->Frequency / state->rate);
+             }
+             break;
+    }
+
+    /* Calculate lfo phase displacement */
+    if ((state->phase == 0) || (state->rate == 0.0f))
+    {
+        state->lfo_disp = 0;
+    }
+    else
+    {
+        state->lfo_disp = (ALint) ((ALfloat)Device->Frequency /
+                          state->rate / (360.0f / (ALfloat)state->phase));
+    }
+}
+
+static ALvoid ChorusProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
+{
+    ALchorusState *state = (ALchorusState*)effect;
+    const ALuint mask = state->BufferLength-1;
+    ALuint it;
+    ALuint kt;
+    ALint offset;
+    ALfloat lfo_value_left = 0.0f;
+    ALfloat lfo_value_right = 0.0f;
+    ALint delay_left = 0;
+    ALint delay_right = 0;
+    ALfloat smp;
+
+    offset=state->offset;
+
+    switch (state->waveform)
+    {
+        case AL_CHORUS_WAVEFORM_TRIANGLE:
+             for (it = 0; it < SamplesToDo; it++, offset++)
+             {
+                 lfo_value_left = 2.0f - fabsf(2.0f - fmodf(state->lfo_coeff *
+                                  offset * 4.0f, 4.0f));
+                 lfo_value_left *= state->depth * state->delay;
+                 lfo_value_left += state->delay;
+                 delay_left = (ALint)(lfo_value_left * state->frequency);
+                 lfo_value_right = 2.0f - fabsf(2.0f - fmodf(state->lfo_coeff *
+                                   (offset + state->lfo_disp) * 4.0f, 4.0f));
+                 lfo_value_right *= state->depth * state->delay;
+                 lfo_value_right += state->delay;
+                 delay_right = (ALint)(lfo_value_right * state->frequency);
+
+                 smp = state->SampleBufferLeft[(offset-delay_left) & mask];
+                 for (kt = 0; kt < MaxChannels; kt++)
+                 {
+                     SamplesOut[kt][it] += smp * state->Gain[0][kt];
+                 }
+                 state->SampleBufferLeft[offset & mask] = (smp + SamplesIn[it]) * state->feedback;
+                 smp = state->SampleBufferRight[(offset-delay_right) & mask];
+                 for (kt = 0; kt < MaxChannels; kt++)
+                 {
+                     SamplesOut[kt][it] += smp * state->Gain[1][kt];
+                 }
+                 state->SampleBufferRight[offset & mask] = (smp + SamplesIn[it]) * state->feedback;
+             }
+             break;
+        case AL_CHORUS_WAVEFORM_SINUSOID:
+             for (it = 0; it < SamplesToDo; it++, offset++)
+             {
+                 lfo_value_left = 1.0f + sinf(fmodf(state->lfo_coeff *
+                                  offset, 2 * F_PI));
+                 lfo_value_left *= state->depth * state->delay;
+                 lfo_value_left += state->delay;
+                 delay_left = (ALint)(lfo_value_left * state->frequency);
+                 lfo_value_right = 1.0f + sinf(fmodf(state->lfo_coeff *
+                                   (offset + state->lfo_disp), 2 * F_PI));
+                 lfo_value_right *= state->depth * state->delay;
+                 lfo_value_right += state->delay;
+                 delay_right = (ALint)(lfo_value_right * state->frequency);
+
+                 smp = state->SampleBufferLeft[(offset-delay_left) & mask];
+                 for (kt = 0; kt < MaxChannels; kt++)
+                 {
+                     SamplesOut[kt][it] += smp * state->Gain[0][kt];
+                 }
+                 state->SampleBufferLeft[offset & mask] = (smp + SamplesIn[it]) * state->feedback;
+                 smp = state->SampleBufferRight[(offset-delay_right) & mask];
+                 for (kt = 0; kt < MaxChannels; kt++)
+                 {
+                     SamplesOut[kt][it] += smp * state->Gain[1][kt];
+                 }
+                 state->SampleBufferRight[offset & mask] = (smp + SamplesIn[it]) * state->feedback;
+             }
+             break;
+    }
+
+    state->offset=offset;
+}
+
+ALeffectState *ChorusCreate(void)
+{
+    ALchorusState *state;
+
+    state = malloc(sizeof(*state));
+    if(!state)
+        return NULL;
+
+    state->state.Destroy = ChorusDestroy;
+    state->state.DeviceUpdate = ChorusDeviceUpdate;
+    state->state.Update = ChorusUpdate;
+    state->state.Process = ChorusProcess;
+
+    state->BufferLength = 0;
+    state->SampleBufferLeft = NULL;
+    state->SampleBufferRight = NULL;
+    state->offset = 0;
+
+    return &state->state;
+}
+
+void chorus_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+    switch(param)
+    {
+        case AL_CHORUS_WAVEFORM:
+            if(val >= AL_CHORUS_MIN_WAVEFORM && val <= AL_CHORUS_MAX_WAVEFORM)
+                effect->Chorus.Waveform = val;
+            else
+                alSetError(context, AL_INVALID_VALUE);
+            break;
+
+        case AL_CHORUS_PHASE:
+            if(val >= AL_CHORUS_MIN_PHASE && val <= AL_CHORUS_MAX_PHASE)
+                effect->Chorus.Phase = val;
+            else
+                alSetError(context, AL_INVALID_VALUE);
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM);
+            break;
+    }
+}
+void chorus_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+    chorus_SetParami(effect, context, param, vals[0]);
+}
+void chorus_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+    switch(param)
+    {
+        case AL_CHORUS_RATE:
+            if(val >= AL_CHORUS_MIN_RATE && val <= AL_CHORUS_MAX_RATE)
+                effect->Chorus.Rate = val;
+            else
+                alSetError(context, AL_INVALID_VALUE);
+            break;
+
+        case AL_CHORUS_DEPTH:
+            if(val >= AL_CHORUS_MIN_DEPTH && val <= AL_CHORUS_MAX_DEPTH)
+                effect->Chorus.Depth = val;
+            else
+                alSetError(context, AL_INVALID_VALUE);
+            break;
+
+        case AL_CHORUS_FEEDBACK:
+            if(val >= AL_CHORUS_MIN_FEEDBACK && val <= AL_CHORUS_MAX_FEEDBACK)
+                effect->Chorus.Feedback = val;
+            else
+                alSetError(context, AL_INVALID_VALUE);
+            break;
+
+        case AL_CHORUS_DELAY:
+            if(val >= AL_CHORUS_MIN_DELAY && val <= AL_CHORUS_MAX_DELAY)
+                effect->Chorus.Delay = val;
+            else
+                alSetError(context, AL_INVALID_VALUE);
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM);
+            break;
+    }
+}
+void chorus_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+    chorus_SetParamf(effect, context, param, vals[0]);
+}
+
+void chorus_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+    switch(param)
+    {
+        case AL_CHORUS_WAVEFORM:
+            *val = effect->Chorus.Waveform;
+            break;
+
+        case AL_CHORUS_PHASE:
+            *val = effect->Chorus.Phase;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM);
+            break;
+    }
+}
+void chorus_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+    chorus_GetParami(effect, context, param, vals);
+}
+void chorus_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+    switch(param)
+    {
+        case AL_CHORUS_RATE:
+            *val = effect->Chorus.Rate;
+            break;
+
+        case AL_CHORUS_DEPTH:
+            *val = effect->Chorus.Depth;
+            break;
+
+        case AL_CHORUS_FEEDBACK:
+            *val = effect->Chorus.Feedback;
+            break;
+
+        case AL_CHORUS_DELAY:
+            *val = effect->Chorus.Delay;
+            break;
+
+        default:
+            alSetError(context, AL_INVALID_ENUM);
+            break;
+    }
+}
+void chorus_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+    chorus_GetParamf(effect, context, param, vals);
+}