AuroraOpenALSoft/Alc/alcModulator.c
2011-05-20 09:36:36 -07:00

207 lines
6.3 KiB
C

/**
* OpenAL cross platform audio library
* Copyright (C) 2009 by Chris Robinson.
* 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 ALmodulatorState {
// Must be first in all effects!
ALeffectState state;
enum {
SINUSOID,
SAWTOOTH,
SQUARE
} Waveform;
ALuint index;
ALuint step;
ALfloat Gain[MAXCHANNELS];
FILTER iirFilter;
ALfloat history[1];
} ALmodulatorState;
#define WAVEFORM_FRACBITS 16
#define WAVEFORM_FRACMASK ((1<<WAVEFORM_FRACBITS)-1)
static __inline ALfloat sin_func(ALuint index)
{
return sin(index / (double)(1<<WAVEFORM_FRACBITS) * M_PI * 2.0f);
}
static __inline ALfloat saw_func(ALuint index)
{
return index*2.0f/(1<<WAVEFORM_FRACBITS) - 1.0f;
}
static __inline ALfloat square_func(ALuint index)
{
return ((index>>(WAVEFORM_FRACBITS-1))&1) ? -1.0f : 1.0f;
}
static __inline ALfloat hpFilter1P(FILTER *iir, ALuint offset, ALfloat input)
{
ALfloat *history = &iir->history[offset];
ALfloat a = iir->coeff;
ALfloat output = input;
output = output + (history[0]-output)*a;
history[0] = output;
return input - output;
}
static ALvoid ModulatorDestroy(ALeffectState *effect)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
free(state);
}
static ALboolean ModulatorDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
ALuint index;
for(index = 0;index < MAXCHANNELS;index++)
state->Gain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
{
Channel chan = Device->Speaker2Chan[index];
state->Gain[chan] = 1.0f;
}
return AL_TRUE;
}
static ALvoid ModulatorUpdate(ALeffectState *effect, ALCcontext *Context, const ALeffect *Effect)
{
ALmodulatorState *state = (ALmodulatorState*)effect;
ALfloat cw, a = 0.0f;
if(Effect->Params.Modulator.Waveform == AL_RING_MODULATOR_SINUSOID)
state->Waveform = SINUSOID;
else if(Effect->Params.Modulator.Waveform == AL_RING_MODULATOR_SAWTOOTH)
state->Waveform = SAWTOOTH;
else if(Effect->Params.Modulator.Waveform == AL_RING_MODULATOR_SQUARE)
state->Waveform = SQUARE;
state->step = Effect->Params.Modulator.Frequency*(1<<WAVEFORM_FRACBITS) /
Context->Device->Frequency;
if(!state->step)
state->step = 1;
cw = cos(2.0*M_PI * Effect->Params.Modulator.HighPassCutoff /
Context->Device->Frequency);
a = (2.0f-cw) - aluSqrt(aluPow(2.0f-cw, 2.0f) - 1.0f);
state->iirFilter.coeff = a;
}
static ALvoid ModulatorProcess(ALeffectState *effect, const ALeffectslot *Slot, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MAXCHANNELS])
{
ALmodulatorState *state = (ALmodulatorState*)effect;
const ALfloat gain = Slot->Gain;
const ALuint step = state->step;
ALuint index = state->index;
ALfloat samp;
ALuint i;
switch(state->Waveform)
{
case SINUSOID:
for(i = 0;i < SamplesToDo;i++)
{
#define FILTER_OUT(func) do { \
samp = SamplesIn[i]; \
\
index += step; \
index &= WAVEFORM_FRACMASK; \
samp *= func(index); \
\
samp = hpFilter1P(&state->iirFilter, 0, samp); \
\
/* Apply slot gain */ \
samp *= gain; \
\
SamplesOut[i][FRONT_LEFT] += state->Gain[FRONT_LEFT] * samp; \
SamplesOut[i][FRONT_RIGHT] += state->Gain[FRONT_RIGHT] * samp; \
SamplesOut[i][FRONT_CENTER] += state->Gain[FRONT_CENTER] * samp; \
SamplesOut[i][SIDE_LEFT] += state->Gain[SIDE_LEFT] * samp; \
SamplesOut[i][SIDE_RIGHT] += state->Gain[SIDE_RIGHT] * samp; \
SamplesOut[i][BACK_LEFT] += state->Gain[BACK_LEFT] * samp; \
SamplesOut[i][BACK_RIGHT] += state->Gain[BACK_RIGHT] * samp; \
SamplesOut[i][BACK_CENTER] += state->Gain[BACK_CENTER] * samp; \
} while(0)
FILTER_OUT(sin_func);
}
break;
case SAWTOOTH:
for(i = 0;i < SamplesToDo;i++)
{
FILTER_OUT(saw_func);
}
break;
case SQUARE:
for(i = 0;i < SamplesToDo;i++)
{
FILTER_OUT(square_func);
#undef FILTER_OUT
}
break;
}
state->index = index;
}
ALeffectState *ModulatorCreate(void)
{
ALmodulatorState *state;
state = malloc(sizeof(*state));
if(!state)
return NULL;
state->state.Destroy = ModulatorDestroy;
state->state.DeviceUpdate = ModulatorDeviceUpdate;
state->state.Update = ModulatorUpdate;
state->state.Process = ModulatorProcess;
state->index = 0.0f;
state->step = 1.0f;
state->iirFilter.coeff = 0.0f;
state->iirFilter.history[0] = 0.0f;
return &state->state;
}