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-rw-r--r--Alc/ALc.c18
-rw-r--r--Alc/alcChorus.c395
-rw-r--r--Alc/alcFlanger.c395
3 files changed, 806 insertions, 2 deletions
diff --git a/Alc/ALc.c b/Alc/ALc.c
index fd5b8894..094c974c 100644
--- a/Alc/ALc.c
+++ b/Alc/ALc.c
@@ -512,13 +512,13 @@ static const ALCenums enumeration[] = {
DECL(AL_EFFECT_NULL),
DECL(AL_EFFECT_REVERB),
DECL(AL_EFFECT_EAXREVERB),
-#if 0
DECL(AL_EFFECT_CHORUS),
+#if 0
DECL(AL_EFFECT_DISTORTION),
#endif
DECL(AL_EFFECT_ECHO),
-#if 0
DECL(AL_EFFECT_FLANGER),
+#if 0
DECL(AL_EFFECT_FREQUENCY_SHIFTER),
DECL(AL_EFFECT_VOCAL_MORPHER),
DECL(AL_EFFECT_PITCH_SHIFTER),
@@ -576,6 +576,20 @@ static const ALCenums enumeration[] = {
DECL(AL_ECHO_FEEDBACK),
DECL(AL_ECHO_SPREAD),
+ DECL(AL_CHORUS_WAVEFORM),
+ DECL(AL_CHORUS_PHASE),
+ DECL(AL_CHORUS_RATE),
+ DECL(AL_CHORUS_DEPTH),
+ DECL(AL_CHORUS_FEEDBACK),
+ DECL(AL_CHORUS_DELAY),
+
+ DECL(AL_FLANGER_WAVEFORM),
+ DECL(AL_FLANGER_PHASE),
+ DECL(AL_FLANGER_RATE),
+ DECL(AL_FLANGER_DEPTH),
+ DECL(AL_FLANGER_FEEDBACK),
+ DECL(AL_FLANGER_DELAY),
+
DECL(AL_RING_MODULATOR_FREQUENCY),
DECL(AL_RING_MODULATOR_HIGHPASS_CUTOFF),
DECL(AL_RING_MODULATOR_WAVEFORM),
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);
+}
diff --git a/Alc/alcFlanger.c b/Alc/alcFlanger.c
new file mode 100644
index 00000000..a1047c01
--- /dev/null
+++ b/Alc/alcFlanger.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 ALflangerState {
+ // 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;
+} ALflangerState;
+
+static ALvoid FlangerDestroy(ALeffectState *effect)
+{
+ ALflangerState *state = (ALflangerState*)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 FlangerDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
+{
+ ALflangerState *state = (ALflangerState*)effect;
+ ALuint maxlen;
+ ALuint it;
+
+ maxlen = fastf2u(AL_FLANGER_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 FlangerUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffectslot *Slot)
+{
+ ALflangerState *state = (ALflangerState*)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.Flanger.Waveform;
+ state->phase = Slot->effect.Flanger.Phase;
+ state->rate = Slot->effect.Flanger.Rate;
+ state->depth = Slot->effect.Flanger.Depth;
+ state->feedback = Slot->effect.Flanger.Feedback;
+ state->delay = Slot->effect.Flanger.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_FLANGER_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_FLANGER_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 FlangerProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *RESTRICT SamplesIn, ALfloat (*RESTRICT SamplesOut)[BUFFERSIZE])
+{
+ ALflangerState *state = (ALflangerState*)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_FLANGER_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_FLANGER_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 *FlangerCreate(void)
+{
+ ALflangerState *state;
+
+ state = malloc(sizeof(*state));
+ if(!state)
+ return NULL;
+
+ state->state.Destroy = FlangerDestroy;
+ state->state.DeviceUpdate = FlangerDeviceUpdate;
+ state->state.Update = FlangerUpdate;
+ state->state.Process = FlangerProcess;
+
+ state->BufferLength = 0;
+ state->SampleBufferLeft = NULL;
+ state->SampleBufferRight = NULL;
+ state->offset = 0;
+
+ return &state->state;
+}
+
+void flanger_SetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
+{
+ switch(param)
+ {
+ case AL_FLANGER_WAVEFORM:
+ if(val >= AL_FLANGER_MIN_WAVEFORM && val <= AL_FLANGER_MAX_WAVEFORM)
+ effect->Flanger.Waveform = val;
+ else
+ alSetError(context, AL_INVALID_VALUE);
+ break;
+
+ case AL_FLANGER_PHASE:
+ if(val >= AL_FLANGER_MIN_PHASE && val <= AL_FLANGER_MAX_PHASE)
+ effect->Flanger.Phase = val;
+ else
+ alSetError(context, AL_INVALID_VALUE);
+ break;
+
+ default:
+ alSetError(context, AL_INVALID_ENUM);
+ break;
+ }
+}
+void flanger_SetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
+{
+ flanger_SetParami(effect, context, param, vals[0]);
+}
+void flanger_SetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
+{
+ switch(param)
+ {
+ case AL_FLANGER_RATE:
+ if(val >= AL_FLANGER_MIN_RATE && val <= AL_FLANGER_MAX_RATE)
+ effect->Flanger.Rate = val;
+ else
+ alSetError(context, AL_INVALID_VALUE);
+ break;
+
+ case AL_FLANGER_DEPTH:
+ if(val >= AL_FLANGER_MIN_DEPTH && val <= AL_FLANGER_MAX_DEPTH)
+ effect->Flanger.Depth = val;
+ else
+ alSetError(context, AL_INVALID_VALUE);
+ break;
+
+ case AL_FLANGER_FEEDBACK:
+ if(val >= AL_FLANGER_MIN_FEEDBACK && val <= AL_FLANGER_MAX_FEEDBACK)
+ effect->Flanger.Feedback = val;
+ else
+ alSetError(context, AL_INVALID_VALUE);
+ break;
+
+ case AL_FLANGER_DELAY:
+ if(val >= AL_FLANGER_MIN_DELAY && val <= AL_FLANGER_MAX_DELAY)
+ effect->Flanger.Delay = val;
+ else
+ alSetError(context, AL_INVALID_VALUE);
+ break;
+
+ default:
+ alSetError(context, AL_INVALID_ENUM);
+ break;
+ }
+}
+void flanger_SetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
+{
+ flanger_SetParamf(effect, context, param, vals[0]);
+}
+
+void flanger_GetParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
+{
+ switch(param)
+ {
+ case AL_FLANGER_WAVEFORM:
+ *val = effect->Flanger.Waveform;
+ break;
+
+ case AL_FLANGER_PHASE:
+ *val = effect->Flanger.Phase;
+ break;
+
+ default:
+ alSetError(context, AL_INVALID_ENUM);
+ break;
+ }
+}
+void flanger_GetParamiv(ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
+{
+ flanger_GetParami(effect, context, param, vals);
+}
+void flanger_GetParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
+{
+ switch(param)
+ {
+ case AL_FLANGER_RATE:
+ *val = effect->Flanger.Rate;
+ break;
+
+ case AL_FLANGER_DEPTH:
+ *val = effect->Flanger.Depth;
+ break;
+
+ case AL_FLANGER_FEEDBACK:
+ *val = effect->Flanger.Feedback;
+ break;
+
+ case AL_FLANGER_DELAY:
+ *val = effect->Flanger.Delay;
+ break;
+
+ default:
+ alSetError(context, AL_INVALID_ENUM);
+ break;
+ }
+}
+void flanger_GetParamfv(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
+{
+ flanger_GetParamf(effect, context, param, vals);
+}
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-29 13:46:05 +0000