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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 - cos(2.0f * M_PI * 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 = (ALuint)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;
    // The junction is loaded with the input here.
    v += in;

    // Calculate the feed values for the delay lines.
    f[0] = v - d[0];
    f[1] = v - d[1];
    f[2] = v - d[2];
    f[3] = v - d[3];

    // Re-feed the delay lines.
    DelayLineIn(&State->Early.Delay[0], State->Offset, f[0]);
    DelayLineIn(&State->Early.Delay[1], State->Offset, f[1]);