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|
/**
* Reverb for the OpenAL cross platform audio library
* Copyright (C) 2008-2009 by Christopher Fitzgerald.
* 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 <stdio.h>
#include <stdlib.h>
#include <math.h>
#include "AL/al.h"
#include "AL/alc.h"
#include "alMain.h"
#include "alAuxEffectSlot.h"
#include "alEffect.h"
#include "alError.h"
#include "alu.h"
typedef struct DelayLine
{
// The delay lines use sample lengths that are powers of 2 to allow the
// use of bit-masking instead of a modulus for wrapping.
ALuint Mask;
ALfloat *Line;
} DelayLine;
typedef struct ALverbState {
// Must be first in all effects!
ALeffectState state;
// All delay lines are allocated as a single buffer to reduce memory
// fragmentation and management code.
ALfloat *SampleBuffer;
ALuint TotalSamples;
// Master effect low-pass filter (2 chained 1-pole filters).
FILTER LpFilter;
ALfloat LpHistory[2];
struct {
// Modulator delay line.
DelayLine Delay;
// The vibrato time is tracked with an index over a modulus-wrapped
// range (in samples).
ALuint Index;
ALuint Range;
// The depth of frequency change (also in samples) and its filter.
ALfloat Depth;
ALfloat Coeff;
ALfloat Filter;
} Mod;
// Initial effect delay.
DelayLine Delay;
// The tap points for the initial delay. First tap goes to early
// reflections, the last to late reverb.
ALuint DelayTap[2];
struct {
// Output gain for early reflections.
ALfloat Gain;
// Early reflections are done with 4 delay lines.
ALfloat Coeff[4];
DelayLine Delay[4];
ALuint Offset[4];
// The gain for each output channel based on 3D panning (only for the
// EAX path).
ALfloat PanGain[MAXCHANNELS];
} Early;
// Decorrelator delay line.
DelayLine Decorrelator;
// There are actually 4 decorrelator taps, but the first occurs at the
// initial sample.
ALuint DecoTap[3];
struct {
// Output gain for late reverb.
ALfloat Gain;
// Attenuation to compensate for the modal density and decay rate of
// the late lines.
ALfloat DensityGain;
// The feed-back and feed-forward all-pass coefficient.
ALfloat ApFeedCoeff;
// Mixing matrix coefficient.
ALfloat MixCoeff;
// Late reverb has 4 parallel all-pass filters.
ALfloat ApCoeff[4];
DelayLine ApDelay[4];
ALuint ApOffset[4];
// In addition to 4 cyclical delay lines.
ALfloat Coeff[4];
DelayLine Delay[4];
ALuint Offset[4];
// The cyclical delay lines are 1-pole low-pass filtered.
ALfloat LpCoeff[4];
ALfloat LpSample[4];
// The gain for each output channel based on 3D panning (only for the
// EAX path).
ALfloat PanGain[MAXCHANNELS];
} Late;
struct {
// Attenuation to compensate for the modal density and decay rate of
// the echo line.
ALfloat DensityGain;
// Echo delay and all-pass lines.
DelayLine Delay;
DelayLine ApDelay;
ALfloat Coeff;
ALfloat ApFeedCoeff;
ALfloat ApCoeff;
ALuint Offset;
ALuint ApOffset;
// The echo line is 1-pole low-pass filtered.
ALfloat LpCoeff;
ALfloat LpSample;
// Echo mixing coefficients.
ALfloat MixCoeff[2];
} Echo;
// The current read offset for all delay lines.
ALuint Offset;
// The gain for each output channel (non-EAX path only; aliased from
// Late.PanGain)
ALfloat *Gain;
} ALverbState;
/* This is a user config option for modifying the overall output of the reverb
* effect.
*/
ALfloat ReverbBoost = 1.0f;
/* Specifies whether to use a standard reverb effect in place of EAX reverb */
ALboolean EmulateEAXReverb = AL_FALSE;
/* This coefficient is used to define the maximum frequency range controlled
* by the modulation depth. The current value of 0.1 will allow it to swing
* from 0.9x to 1.1x. This value must be below 1. At 1 it will cause the
* sampler to stall on the downswing, and above 1 it will cause it to sample
* backwards.
*/
static const ALfloat MODULATION_DEPTH_COEFF = 0.1f;
/* A filter is used to avoid the terrible distortion caused by changing
* modulation time and/or depth. To be consistent across different sample
* rates, the coefficient must be raised to a constant divided by the sample
* rate: coeff^(constant / rate).
*/
static const ALfloat MODULATION_FILTER_COEFF = 0.048f;
static const ALfloat MODULATION_FILTER_CONST = 100000.0f;
// When diffusion is above 0, an all-pass filter is used to take the edge off
// the echo effect. It uses the following line length (in seconds).
static const ALfloat ECHO_ALLPASS_LENGTH = 0.0133f;
// Input into the late reverb is decorrelated between four channels. Their
// timings are dependent on a fraction and multiplier. See the
// UpdateDecorrelator() routine for the calculations involved.
static const ALfloat DECO_FRACTION = 0.15f;
static const ALfloat DECO_MULTIPLIER = 2.0f;
// All delay line lengths are specified in seconds.
// The lengths of the early delay lines.
static const ALfloat EARLY_LINE_LENGTH[4] =
{
0.0015f, 0.0045f, 0.0135f, 0.0405f
};
// The lengths of the late all-pass delay lines.
static const ALfloat ALLPASS_LINE_LENGTH[4] =
{
0.0151f, 0.0167f, 0.0183f, 0.0200f,
};
// The lengths of the late cyclical delay lines.
static const ALfloat LATE_LINE_LENGTH[4] =
{
0.0211f, 0.0311f, 0.0461f, 0.0680f
};
// The late cyclical delay lines have a variable length dependent on the
// effect's density parameter (inverted for some reason) and this multiplier.
static const ALfloat LATE_LINE_MULTIPLIER = 4.0f;
// Basic delay line input/output routines.
static __inline ALfloat DelayLineOut(DelayLine *Delay, ALuint offset)
{
return Delay->Line[offset&Delay->Mask];
}
static __inline ALvoid DelayLineIn(DelayLine *Delay, ALuint offset, ALfloat in)
{
Delay->Line[offset&Delay->Mask] = in;
}
// Attenuated delay line output routine.
static __inline ALfloat AttenuatedDelayLineOut(DelayLine *Delay, ALuint offset, ALfloat coeff)
{
return coeff * Delay->Line[offset&Delay->Mask];
}
// Basic attenuated all-pass input/output routine.
static __inline ALfloat AllpassInOut(DelayLine *Delay, ALuint outOffset, ALuint inOffset, ALfloat in, ALfloat feedCoeff, ALfloat coeff)
{
ALfloat out, feed;
out = DelayLineOut(Delay, outOffset);
feed = feedCoeff * in;
DelayLineIn(Delay, inOffset, (feedCoeff * (out - feed)) + in);
// The time-based attenuation is only applied to the delay output to
// keep it from affecting the feed-back path (which is already controlled
// by the all-pass feed coefficient).
return (coeff * out) - feed;
}
// Given an input sample, this function produces modulation for the late
// reverb.
static __inline ALfloat EAXModulation(ALverbState *State, ALfloat in)
{
ALfloat sinus, frac;
ALuint offset;
ALfloat out0, out1;
// Calculate the sinus rythm (dependent on modulation time and the
// sampling rate). The center of the sinus is moved to reduce the delay
// of the effect when the time or depth are low.
sinus = 1.0f - aluCos(F_PI*2.0f * State->Mod.Index / State->Mod.Range);
// The depth determines the range over which to read the input samples
// from, so it must be filtered to reduce the distortion caused by even
// small parameter changes.
State->Mod.Filter = lerp(State->Mod.Filter, State->Mod.Depth,
State->Mod.Coeff);
// Calculate the read offset and fraction between it and the next sample.
frac = (1.0f + (State->Mod.Filter * sinus));
offset = fastf2u(frac);
frac -= offset;
// Get the two samples crossed by the offset, and feed the delay line
// with the next input sample.
out0 = DelayLineOut(&State->Mod.Delay, State->Offset - offset);
out1 = DelayLineOut(&State->Mod.Delay, State->Offset - offset - 1);
DelayLineIn(&State->Mod.Delay, State->Offset, in);
// Step the modulation index forward, keeping it bound to its range.
State->Mod.Index = (State->Mod.Index + 1) % State->Mod.Range;
// The output is obtained by linearly interpolating the two samples that
// were acquired above.
return lerp(out0, out1, frac);
}
// Delay line output routine for early reflections.
static __inline ALfloat EarlyDelayLineOut(ALverbState *State, ALuint index)
{
return AttenuatedDelayLineOut(&State->Early.Delay[index],
State->Offset - State->Early.Offset[index],
State->Early.Coeff[index]);
}
// Given an input sample, this function produces four-channel output for the
// early reflections.
static __inline ALvoid EarlyReflection(ALverbState *State, ALfloat in, ALfloat *out)
{
ALfloat d[4], v, f[4];
// Obtain the decayed results of each early delay line.
d[0] = EarlyDelayLineOut(State, 0);
d[1] = EarlyDelayLineOut(State, 1);
d[2] = EarlyDelayLineOut(State, 2);
d[3] = EarlyDelayLineOut(State, 3);
/* The following uses a lossless scattering junction from waveguide
* theory. It actually amounts to a householder mixing matrix, which
* will produce a maximally diffuse response, and means this can probably
* be considered a simple feed-back delay network (FDN).
* N
* ---
* \
* v = 2/N / d_i
* ---
* i=1
*/
v = (d[0] + d[1] + d[2] + d[3]) * 0.5f;
|
