openal-soft/Alc/effects/distortion.c
Chris Robinson ef0d4f8210 Provide (mostly) lockless updates for effect slots
Similar to the listener, separate containers are provided atomically for the
mixer thread to apply updates without needing to block, and a free-list is used
to reuse container objects.

A couple things to note. First, the lock is still used when the effect state's
deviceUpdate method is called to prevent asynchronous calls to reset the device
from interfering. This can be fixed by using the list lock in ALc.c instead.

Secondly, old effect states aren't immediately deleted when the effect type
changes (the actual type, not just its properties). This is because the mixer
thread is intended to be real-time safe, and so can't be freeing anything. They
are cleared away when updates reuse the container they were kept in, and they
don't incur any extra processing cost, but there may be cases where the memory
is kept around until the effect slot is deleted.
2016-05-12 18:41:33 -07:00

297 lines
10 KiB
C

/**
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 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 ALdistortionState {
DERIVE_FROM_TYPE(ALeffectState);
/* Effect gains for each channel */
ALfloat Gain[MAX_OUTPUT_CHANNELS];
/* Effect parameters */
ALfilterState lowpass;
ALfilterState bandpass;
ALfloat attenuation;
ALfloat edge_coeff;
} ALdistortionState;
static ALvoid ALdistortionState_Destruct(ALdistortionState *state)
{
ALeffectState_Destruct(STATIC_CAST(ALeffectState,state));
}
static ALboolean ALdistortionState_deviceUpdate(ALdistortionState *UNUSED(state), ALCdevice *UNUSED(device))
{
return AL_TRUE;
}
static ALvoid ALdistortionState_update(ALdistortionState *state, const ALCdevice *Device, const ALeffectslot *Slot)
{
ALfloat frequency = (ALfloat)Device->Frequency;
ALfloat bandwidth;
ALfloat cutoff;
ALfloat edge;
/* Store distorted signal attenuation settings */
state->attenuation = Slot->Params.EffectProps.Distortion.Gain;
/* Store waveshaper edge settings */
edge = sinf(Slot->Params.EffectProps.Distortion.Edge * (F_PI_2));
edge = minf(edge, 0.99f);
state->edge_coeff = 2.0f * edge / (1.0f-edge);
/* Lowpass filter */
cutoff = Slot->Params.EffectProps.Distortion.LowpassCutoff;
/* Bandwidth value is constant in octaves */
bandwidth = (cutoff / 2.0f) / (cutoff * 0.67f);
ALfilterState_setParams(&state->lowpass, ALfilterType_LowPass, 1.0f,
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
/* Bandpass filter */
cutoff = Slot->Params.EffectProps.Distortion.EQCenter;
/* Convert bandwidth in Hz to octaves */
bandwidth = Slot->Params.EffectProps.Distortion.EQBandwidth / (cutoff * 0.67f);
ALfilterState_setParams(&state->bandpass, ALfilterType_BandPass, 1.0f,
cutoff / (frequency*4.0f), calc_rcpQ_from_bandwidth(cutoff / (frequency*4.0f), bandwidth)
);
ComputeAmbientGains(Device->Dry, Slot->Params.Gain, state->Gain);
}
static ALvoid ALdistortionState_process(ALdistortionState *state, ALuint SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALuint NumChannels)
{
const ALfloat fc = state->edge_coeff;
ALuint base;
ALuint it;
ALuint ot;
ALuint kt;
for(base = 0;base < SamplesToDo;)
{
float oversample_buffer[64][4];
ALuint td = minu(64, SamplesToDo-base);
/* 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 < td;it++)
{
oversample_buffer[it][0] = SamplesIn[0][it+base];
oversample_buffer[it][1] = 0.0f;
oversample_buffer[it][2] = 0.0f;
oversample_buffer[it][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 < td;it++)
{
for(ot = 0;ot < 4;ot++)
{
ALfloat smp;
smp = ALfilterState_processSingle(&state->lowpass, oversample_buffer[it][ot]);
/* Restore signal power by multiplying sample by amount of oversampling */
oversample_buffer[it][ot] = smp * 4.0f;
}
}
for(it = 0;it < td;it++)
{
/* 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(ot = 0;ot < 4;ot++)
{
ALfloat smp = oversample_buffer[it][ot];
smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp)) * -1.0f;
smp = (1.0f + fc) * smp/(1.0f + fc*fabsf(smp));
/* Third step, do bandpass filtering of distorted signal */
smp = ALfilterState_processSingle(&state->bandpass, smp);
oversample_buffer[it][ot] = smp;
}
}
for(kt = 0;kt < NumChannels;kt++)
{
/* Fourth step, final, do attenuation and perform decimation,
* store only one sample out of 4.
*/
ALfloat gain = state->Gain[kt] * state->attenuation;
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(it = 0;it < td;it++)
SamplesOut[kt][base+it] += gain * oversample_buffer[it][0];
}
base += td;
}
}
DECLARE_DEFAULT_ALLOCATORS(ALdistortionState)
DEFINE_ALEFFECTSTATE_VTABLE(ALdistortionState);
typedef struct ALdistortionStateFactory {
DERIVE_FROM_TYPE(ALeffectStateFactory);
} ALdistortionStateFactory;
static ALeffectState *ALdistortionStateFactory_create(ALdistortionStateFactory *UNUSED(factory))
{
ALdistortionState *state;
state = ALdistortionState_New(sizeof(*state));
if(!state) return NULL;
SET_VTABLE2(ALdistortionState, ALeffectState, state);
ALfilterState_clear(&state->lowpass);
ALfilterState_clear(&state->bandpass);
return STATIC_CAST(ALeffectState, state);
}
DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALdistortionStateFactory);
ALeffectStateFactory *ALdistortionStateFactory_getFactory(void)
{
static ALdistortionStateFactory DistortionFactory = { { GET_VTABLE2(ALdistortionStateFactory, ALeffectStateFactory) } };
return STATIC_CAST(ALeffectStateFactory, &DistortionFactory);
}
void ALdistortion_setParami(ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint UNUSED(val))
{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
void ALdistortion_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
ALdistortion_setParami(effect, context, param, vals[0]);
}
void ALdistortion_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_DISTORTION_EDGE:
if(!(val >= AL_DISTORTION_MIN_EDGE && val <= AL_DISTORTION_MAX_EDGE))
SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
props->Distortion.Edge = val;
break;
case AL_DISTORTION_GAIN:
if(!(val >= AL_DISTORTION_MIN_GAIN && val <= AL_DISTORTION_MAX_GAIN))
SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
props->Distortion.Gain = val;
break;
case AL_DISTORTION_LOWPASS_CUTOFF:
if(!(val >= AL_DISTORTION_MIN_LOWPASS_CUTOFF && val <= AL_DISTORTION_MAX_LOWPASS_CUTOFF))
SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
props->Distortion.LowpassCutoff = val;
break;
case AL_DISTORTION_EQCENTER:
if(!(val >= AL_DISTORTION_MIN_EQCENTER && val <= AL_DISTORTION_MAX_EQCENTER))
SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
props->Distortion.EQCenter = val;
break;
case AL_DISTORTION_EQBANDWIDTH:
if(!(val >= AL_DISTORTION_MIN_EQBANDWIDTH && val <= AL_DISTORTION_MAX_EQBANDWIDTH))
SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
props->Distortion.EQBandwidth = val;
break;
default:
SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
}
}
void ALdistortion_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{
ALdistortion_setParamf(effect, context, param, vals[0]);
}
void ALdistortion_getParami(const ALeffect *UNUSED(effect), ALCcontext *context, ALenum UNUSED(param), ALint *UNUSED(val))
{ SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM); }
void ALdistortion_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{
ALdistortion_getParami(effect, context, param, vals);
}
void ALdistortion_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
const ALeffectProps *props = &effect->Props;
switch(param)
{
case AL_DISTORTION_EDGE:
*val = props->Distortion.Edge;
break;
case AL_DISTORTION_GAIN:
*val = props->Distortion.Gain;
break;
case AL_DISTORTION_LOWPASS_CUTOFF:
*val = props->Distortion.LowpassCutoff;
break;
case AL_DISTORTION_EQCENTER:
*val = props->Distortion.EQCenter;
break;
case AL_DISTORTION_EQBANDWIDTH:
*val = props->Distortion.EQBandwidth;
break;
default:
SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
}
}
void ALdistortion_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{
ALdistortion_getParamf(effect, context, param, vals);
}
DEFINE_ALEFFECT_VTABLE(ALdistortion);