<commit_before>/**
 * Reverb for the OpenAL cross platform audio library
 * Copyright (C) 2008-2009 by <NAME>.
 * 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 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;

// Calculate the length of a delay line and store its mask and offset.
static ALuint CalcLineLength(ALfloat length, ALintptrEXT offset, ALuint frequency, DelayLine *Delay)
{
    ALuint samples;

    // All line lengths are powers of 2, calculated from their lengths, with
    // an additional sample in case of rounding errors.
    samples = NextPowerOf2((ALuint)(length * frequency) + 1);
    // All lines share a single sample buffer.
    Delay->Mask = samples - 1;
    Delay->Line = (ALfloat*)offset;
    // Return the sample count for accumulation.
    return samples;
}

// Given the allocated sample buffer, this function updates each delay line
// offset.
static __inline ALvoid RealizeLineOffset(ALfloat * sampleBuffer, DelayLine *Delay)
{
    Delay->Line = &sampleBuffer[(ALintptrEXT)Delay->Line];
}

/* Calculates the delay line metrics and allocates the shared sample buffer
 * for all lines given a flag indicating whether or not to allocate the EAX-
 * related delays (eaxFlag) and the sample rate (frequency).  If an
 * allocation failure occurs, it returns AL_FALSE.
 */
static ALboolean AllocLines(ALboolean eaxFlag, ALuint frequency, ALverbState *State)
{
    ALuint totalSamples, index;
    ALfloat length;
    ALfloat *newBuffer = NULL;

    // All delay line lengths are calculated to accomodate the full range of
    // lengths given their respective paramters.
    totalSamples = 0;
    if(eaxFlag)
    {
        /* The modulator's line length is calculated from the maximum
         * modulation time and depth coefficient, and halfed for the low-to-
         * high frequency swing.  An additional sample is added to keep it
         * stable when there is no modulation.
         */
        length = (AL_EAXREVERB_MAX_MODULATION_TIME * MODULATION_DEPTH_COEFF /
                  2.0f) + (1.0f / frequency);
        totalSamples += CalcLineLength(length, totalSamples, frequency,
                                       &State->Mod.Delay);
    }

    // The initial delay is the sum of the reflections and late reverb
    // delays.
    if(eaxFlag)
        length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY +
                 AL_EAXREVERB_MAX_LATE_REVERB_DELAY;
    else
        length = AL_REVERB_MAX_REFLECTIONS_DELAY +
                 AL_REVERB_MAX_LATE_REVERB_DELAY;
    totalSamples += CalcLineLength(length, totalSamples, frequency,
                                   &State->Delay);

    // The early reflection lines.
    for(index = 0;index < 4;index++)
        totalSamples += CalcLineLength(EARLY_LINE_LENGTH[index], totalSamples,
                                       frequency, &State->Early.Delay[index]);

    // The decorrelator line is calculated from the lowest reverb density (a
    // parameter value of 1).
    length = (DECO_FRACTION * DECO_MULTIPLIER * DECO_MULTIPLIER) *
             LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER);
    totalSamples += CalcLineLength(length, totalSamples, frequency,
                                   &State->Decorrelator);

    // The late all-pass lines.
    for(index = 0;index < 4;index++)
        totalSamples += CalcLineLength(ALLPASS_LINE_LENGTH[index], totalSamples,
                                       frequency, &State->Late.ApDelay[index]);

    // The late delay lines are calculated from the lowest reverb density.
    for(index = 0;index < 4;index++)
    {
        length = LATE_LINE_LENGTH[index] * (1.0f + LATE_LINE_MULTIPLIER);
        totalSamples += CalcLineLength(length, totalSamples, frequency,
                                       &State->Late.Delay[index]);
    }

    if(eaxFlag)
    {
        // The echo all-pass and delay lines.
        totalSamples += CalcLineLength(ECHO_ALLPASS_LENGTH, totalSamples,
                                       frequency, &State->Echo.ApDelay);
        totalSamples += CalcLineLength(AL_EAXREVERB_MAX_ECHO_TIME, totalSamples,
                                       frequency, &State->Echo.Delay);
    }

    if(totalSamples != State->TotalSamples)
    {
        newBuffer = realloc(State->SampleBuffer, sizeof(ALfloat) * totalSamples);
        if(newBuffer == NULL)
            return AL_FALSE;
        State->SampleBuffer = newBuffer;
        State->TotalSamples = totalSamples;
    }

    // Update all delays to reflect the new sample buffer.
    RealizeLineOffset(State->SampleBuffer, &State->Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Decorrelator);
    for(index = 0;index < 4;index++)
    {
        RealizeLineOffset(State->SampleBuffer, &State->Early.Delay[index]);
        RealizeLineOffset(State->SampleBuffer, &State->Late.ApDelay[index]);
        RealizeLineOffset(State->SampleBuffer, &State->Late.Delay[index]);
    }
    if(eaxFlag)
    {
        RealizeLineOffset(State->SampleBuffer, &State->Mod.Delay);
        RealizeLineOffset(State->SampleBuffer, &State->Echo.ApDelay);
        RealizeLineOffset(State->SampleBuffer, &State->Echo.Delay);
    }

    // Clear the sample buffer.
    for(index = 0;index < State->TotalSamples;index++)
        State->SampleBuffer[index] = 0.0f;

    return AL_TRUE;
}

// Calculate a decay coefficient given the length of each cycle and the time
// until the decay reaches -60 dB.
static __inline ALfloat CalcDecayCoeff(ALfloat length, ALfloat decayTime)
{
    return aluPow(10.0f, length / decayTime * -60.0f / 20.0f);
}

// Calculate a decay length from a coefficient and the time until the decay
// reaches -60 dB.
static __inline ALfloat CalcDecayLength(ALfloat coeff, ALfloat decayTime)
{
    return log10(coeff) / -60.0 * 20.0f * decayTime;
}

// Calculate the high frequency parameter for the I3DL2 coefficient
// calculation.
static __inline ALfloat CalcI3DL2HFreq(ALfloat hfRef, ALuint frequency)
{
    return cos(2.0f * M_PI * hfRef / frequency);
}

// Calculate an attenuation to be applied to the input of any echo models to
// compensate for modal density and decay time.
static __inline ALfloat CalcDensityGain(ALfloat a)
{
    /* The energy of a signal can be obtained by finding the area under the
     * squared signal.  This takes the form of Sum(x_n^2), where x is the
     * amplitude for the sample n.
     *
     * Decaying feedback matches exponential decay of the form Sum(a^n),
     * where a is the attenuation coefficient, and n is the sample.  The area
     * under this decay curve can be calculated as:  1 / (1 - a).
     *
     * Modifying the above equation to find the squared area under the curve
     * (for energy) yields:  1 / (1 - a^2).  Input attenuation can then be
     * calculated by inverting the square root of this approximation,
     * yielding:  1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2).
     */
    return aluSqrt(1.0f - (a * a));
}

// Calculate the mixing matrix coefficients given a diffusion factor.
static __inline ALvoid CalcMatrixCoeffs(ALfloat diffusion, ALfloat *x, ALfloat *y)
{
    ALfloat n, t;

    // The matrix is of order 4, so n is sqrt (4 - 1).
    n = aluSqrt(3.0f);
    t = diffusion * atan(n);

    // Calculate the first mixing matrix coefficient.
    *x = cos(t);
    // Calculate the second mixing matrix coefficient.
    *y = sin(t) / n;
}

// Calculate the limited HF ratio for use with the late reverb low-pass
// filters.
static ALfloat CalcLimitedHfRatio(ALfloat hfRatio, ALfloat airAbsorptionGainHF, ALfloat decayTime)
{
    ALfloat limitRatio;

    /* Find the attenuation due to air absorption in dB (converting delay
     * time to meters using the speed of sound).  Then reversing the decay
     * equation, solve for HF ratio.  The delay length is cancelled out of
     * the equation, so it can be calculated once for all lines.
     */
    limitRatio = 1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) *
                         SPEEDOFSOUNDMETRESPERSEC);
    // Need to limit the result to a minimum of 0.1, just like the HF ratio
    // parameter.
    limitRatio = __max(limitRatio, 0.1f);

    // Using the limit calculated above, apply the upper bound to the HF
    // ratio.
    return __min(hfRatio, limitRatio);
}

// Calculate the coefficient for a HF (and eventually LF) decay damping
// filter.
static __inline ALfloat CalcDampingCoeff(ALfloat hfRatio, ALfloat length, ALfloat decayTime, ALfloat decayCoeff, ALfloat cw)
{
    ALfloat coeff, g;

    // Eventually this should boost the high frequencies when the ratio
    // exceeds 1.
    coeff = 0.0f;
    if (hfRatio < 1.0f)
    {
        // Calculate the low-pass coefficient by dividing the HF decay
        // coefficient by the full decay coefficient.
        g = CalcDecayCoeff(length, decayTime * hfRatio) / decayCoeff;

        // Damping is done with a 1-pole filter, so g needs to be squared.
        g *= g;
        coeff = lpCoeffCalc(g, cw);

        // Very low decay times will produce minimal output, so apply an
        // upper bound to the coefficient.
        coeff = __min(coeff, 0.98f);
    }
    return coeff;
}

// Update the EAX modulation index, range, and depth.  Keep in mind that this
// kind of vibrato is additive and not multiplicative as one may expect.  The
// downswing will sound stronger than the upswing.
static ALvoid UpdateModulator(ALfloat modTime, ALfloat modDepth, ALuint frequency, ALverbState *State)
{
    ALfloat length;

    /* Modulation is calculated in two parts.
     *
     * The modulation time effects the sinus applied to the change in
     * frequency.  An index out of the current time range (both in samples)
     * is incremented each sample.  The range is bound to a reasonable
     * minimum (1 sample) and when the timing changes, the index is rescaled
     * to the new range (to keep the sinus consistent).
     */
    length = modTime * frequency;
    if (length >= 1.0f) {
       State->Mod.Index = (ALuint)(State->Mod.Index * length /
                                   State->Mod.Range);
       State->Mod.Range = (ALuint)length;
    } else {
       State->Mod.Index = 0;
       State->Mod.Range = 1;
    }

    /* The modulation depth effects the amount of frequency change over the
     * range of the sinus.  It needs to be scaled by the modulation time so
     * that a given depth produces a consistent change in frequency over all
     * ranges of time.  Since the depth is applied to a sinus value, it needs
     * to be halfed once for the sinus range and again for the sinus swing
     * in time (half of it is spent decreasing the frequency, half is spent
     * increasing it).
     */
    State->Mod.Depth = modDepth * MODULATION_DEPTH_COEFF * modTime / 2.0f /
                       2.0f * frequency;
}

// Update the offsets for the initial effect delay line.
static ALvoid UpdateDelayLine(ALfloat earlyDelay, ALfloat lateDelay, ALuint frequency, ALverbState *State)
{
    // Calculate the initial delay taps.
    State->DelayTap[0] = (ALuint)(earlyDelay * frequency);
    State->DelayTap[1] = (ALuint)((earlyDelay + lateDelay) * frequency);
}

// Update the early reflections gain and line coefficients.
static ALvoid UpdateEarlyLines(ALfloat reverbGain, ALfloat earlyGain, ALfloat lateDelay, ALverbState *State)
{
    ALuint index;

    // Calculate the early reflections gain (from the master effect gain, and
    // reflections gain parameters) with a constant attenuation of 0.5.
    State->Early.Gain = 0.5f * reverbGain * earlyGain;

    // Calculate the gain (coefficient) for each early delay line using the
    // late delay time.  This expands the early reflections to the start of
    // the late reverb.
    for(index = 0;index < 4;index++)
        State->Early.Coeff[index] = CalcDecayCoeff(EARLY_LINE_LENGTH[index],
                                                   lateDelay);
}

// Update the offsets for the decorrelator line.
static ALvoid UpdateDecorrelator(ALfloat density, ALuint frequency, ALverbState *State)
{
    ALuint index;
    ALfloat length;

    /* The late reverb inputs are decorrelated to smooth the reverb tail and
     * reduce harsh echos.  The first tap occurs immediately, while the
     * remaining taps are delayed by multiples of a fraction of the smallest
     * cyclical delay time.
     *
     * offset[index] = (FRACTION (MULTIPLIER^index)) smallest_delay
     */
    for(index = 0;index < 3;index++)
    {
        length = (DECO_FRACTION * aluPow(DECO_MULTIPLIER, (ALfloat)index)) *
                 LATE_LINE_LENGTH[0] * (1.0f + (density * LATE_LINE_MULTIPLIER));
        State->DecoTap[index] = (ALuint)(length * frequency);
    }
}

// Update the late reverb gains, line lengths, and line coefficients.
static ALvoid UpdateLateLines(ALfloat reverbGain, ALfloat lateGain, ALfloat xMix, ALfloat density, ALfloat decayTime, ALfloat diffusion, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALverbState *State)
{
    ALfloat length;
    ALuint index;

    /* Calculate the late reverb gain (from the master effect gain, and late
     * reverb gain parameters).  Since the output is tapped prior to the
     * application of the next delay line coefficients, this gain needs to be
     * attenuated by the 'x' mixing matrix coefficient as well.
     */
    State->Late.Gain = reverbGain * lateGain * xMix;

    /* To compensate for changes in modal density and decay time of the late
     * reverb signal, the input is attenuated based on the maximal energy of
     * the outgoing signal.  This approximation is used to keep the apparent
     * energy of the signal equal for all ranges of density and decay time.
     *
     * The average length of the cyclcical delay lines is used to calculate
     * the attenuation coefficient.
     */
    length = (LATE_LINE_LENGTH[0] + LATE_LINE_LENGTH[1] +
              LATE_LINE_LENGTH[2] + LATE_LINE_LENGTH[3]) / 4.0f;
    length *= 1.0f + (density * LATE_LINE_MULTIPLIER);
    State->Late.DensityGain = CalcDensityGain(CalcDecayCoeff(length,
                                                             decayTime));

    // Calculate the all-pass feed-back and feed-forward coefficient.
    State->Late.ApFeedCoeff = 0.5f * aluPow(diffusion, 2.0f);

    for(index = 0;index < 4;index++)
    {
        // Calculate the gain (coefficient) for each all-pass line.
        State->Late.ApCoeff[index] = CalcDecayCoeff(ALLPASS_LINE_LENGTH[index],
                                                    decayTime);

        // Calculate the length (in seconds) of each cyclical delay line.
        length = LATE_LINE_LENGTH[index] * (1.0f + (density *
                                                    LATE_LINE_MULTIPLIER));

        // Calculate the delay offset for each cyclical delay line.
        State->Late.Offset[index] = (ALuint)(length * frequency);

        // Calculate the gain (coefficient) for each cyclical line.
        State->Late.Coeff[index] = CalcDecayCoeff(length, decayTime);

        // Calculate the damping coefficient for each low-pass filter.
        State->Late.LpCoeff[index] =
            CalcDampingCoeff(hfRatio, length, decayTime,
                             State->Late.Coeff[index], cw);

        // Attenuate the cyclical line coefficients by the mixing coefficient
        // (x).
        State->Late.Coeff[index] *= xMix;
    }
}

// Update the echo gain, line offset, line coefficients, and mixing
// coefficients.
static ALvoid UpdateEchoLine(ALfloat reverbGain, ALfloat lateGain, ALfloat echoTime, ALfloat decayTime, ALfloat diffusion, ALfloat echoDepth, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALverbState *State)
{
    // Update the offset and coefficient for the echo delay line.
    State->Echo.Offset = (ALuint)(echoTime * frequency);

    // Calculate the decay coefficient for the echo line.
    State->Echo.Coeff = CalcDecayCoeff(echoTime, decayTime);

    // Calculate the energy-based attenuation coefficient for the echo delay
    // line.
    State->Echo.DensityGain = CalcDensityGain(State->Echo.Coeff);

    // Calculate the echo all-pass feed coefficient.
    State->Echo.ApFeedCoeff = 0.5f * aluPow(diffusion, 2.0f);

    // Calculate the echo all-pass attenuation coefficient.
    State->Echo.ApCoeff = CalcDecayCoeff(ECHO_ALLPASS_LENGTH, decayTime);

    // Calculate the damping coefficient for each low-pass filter.
    State->Echo.LpCoeff = CalcDampingCoeff(hfRatio, echoTime, decayTime,
                                           State->Echo.Coeff, cw);

    /* Calculate the echo mixing coefficients.  The first is applied to the
     * echo itself.  The second is used to attenuate the late reverb when
     * echo depth is high and diffusion is low, so the echo is slightly
     * stronger than the decorrelated echos in the reverb tail.
     */
    State->Echo.MixCoeff[0] = reverbGain * lateGain * echoDepth;
    State->Echo.MixCoeff[1] = 1.0f - (echoDepth * 0.5f * (1.0f - diffusion));
}

// Update the early and late 3D panning gains.
static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALverbState *State)
{
    ALfloat earlyPan[3] = { ReflectionsPan[0], ReflectionsPan[1],
                            ReflectionsPan[2] };
    ALfloat latePan[3] = { LateReverbPan[0], LateReverbPan[1],
                           LateReverbPan[2] };
    const ALfloat *speakerGain;
    ALfloat dirGain;
    ALfloat length;
    ALuint index;
    ALint pos;

    // Calculate the 3D-panning gains for the early reflections and late
    // reverb.
    length = earlyPan[0]*earlyPan[0] + earlyPan[1]*earlyPan[1] + earlyPan[2]*earlyPan[2];
    if(length > 1.0f)
    {
        length = 1.0f / aluSqrt(length);
        earlyPan[0] *= length;
        earlyPan[1] *= length;
        earlyPan[2] *= length;
    }
    length = latePan[0]*latePan[0] + latePan[1]*latePan[1] + latePan[2]*latePan[2];
    if(length > 1.0f)
    {
        length = 1.0f / aluSqrt(length);
        latePan[0] *= length;
        latePan[1] *= length;
        latePan[2] *= length;
    }

    /* This code applies directional reverb just like the mixer applies
     * directional sources.  It diffuses the sound toward all speakers as the
     * magnitude of the panning vector drops, which is only a rough
     * approximation of the expansion of sound across the speakers from the
     * panning direction.
     */
    pos = aluCart2LUTpos(earlyPan[2], earlyPan[0]);
    speakerGain = &Device->PanningLUT[MAXCHANNELS * pos];
    dirGain = aluSqrt((earlyPan[0] * earlyPan[0]) + (earlyPan[2] * earlyPan[2]));

    for(index = 0;index < MAXCHANNELS;index++)
        State->Early.PanGain[index] = 0.0f;
    for(index = 0;index < Device->NumChan;index++)
    {
        Channel chan = Device->Speaker2Chan[index];
        State->Early.PanGain[chan] = 1.0 + (speakerGain[chan]-1.0)*dirGain;
    }


    pos = aluCart2LUTpos(latePan[2], latePan[0]);
    speakerGain = &Device->PanningLUT[MAXCHANNELS * pos];
    dirGain = aluSqrt((latePan[0] * latePan[0]) + (latePan[2] * latePan[2]));

    for(index = 0;index < MAXCHANNELS;index++)
         State->Late.PanGain[index] = 0.0f;
    for(index = 0;index < Device->NumChan;index++)
    {
        Channel chan = Device->Speaker2Chan[index];
        State->Late.PanGain[chan] = 1.0 + (speakerGain[chan]-1.0)*dirGain;
    }
}

// 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;<commit_msg>Use lerp() in a couple more places<commit_after>/**
 * Reverb for the OpenAL cross platform audio library
 * Copyright (C) 2008-2009 by <NAME>.
 * 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 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;

// Calculate the length of a delay line and store its mask and offset.
static ALuint CalcLineLength(ALfloat length, ALintptrEXT offset, ALuint frequency, DelayLine *Delay)
{
    ALuint samples;

    // All line lengths are powers of 2, calculated from their lengths, with
    // an additional sample in case of rounding errors.
    samples = NextPowerOf2((ALuint)(length * frequency) + 1);
    // All lines share a single sample buffer.
    Delay->Mask = samples - 1;
    Delay->Line = (ALfloat*)offset;
    // Return the sample count for accumulation.
    return samples;
}

// Given the allocated sample buffer, this function updates each delay line
// offset.
static __inline ALvoid RealizeLineOffset(ALfloat * sampleBuffer, DelayLine *Delay)
{
    Delay->Line = &sampleBuffer[(ALintptrEXT)Delay->Line];
}

/* Calculates the delay line metrics and allocates the shared sample buffer
 * for all lines given a flag indicating whether or not to allocate the EAX-
 * related delays (eaxFlag) and the sample rate (frequency).  If an
 * allocation failure occurs, it returns AL_FALSE.
 */
static ALboolean AllocLines(ALboolean eaxFlag, ALuint frequency, ALverbState *State)
{
    ALuint totalSamples, index;
    ALfloat length;
    ALfloat *newBuffer = NULL;

    // All delay line lengths are calculated to accomodate the full range of
    // lengths given their respective paramters.
    totalSamples = 0;
    if(eaxFlag)
    {
        /* The modulator's line length is calculated from the maximum
         * modulation time and depth coefficient, and halfed for the low-to-
         * high frequency swing.  An additional sample is added to keep it
         * stable when there is no modulation.
         */
        length = (AL_EAXREVERB_MAX_MODULATION_TIME * MODULATION_DEPTH_COEFF /
                  2.0f) + (1.0f / frequency);
        totalSamples += CalcLineLength(length, totalSamples, frequency,
                                       &State->Mod.Delay);
    }

    // The initial delay is the sum of the reflections and late reverb
    // delays.
    if(eaxFlag)
        length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY +
                 AL_EAXREVERB_MAX_LATE_REVERB_DELAY;
    else
        length = AL_REVERB_MAX_REFLECTIONS_DELAY +
                 AL_REVERB_MAX_LATE_REVERB_DELAY;
    totalSamples += CalcLineLength(length, totalSamples, frequency,
                                   &State->Delay);

    // The early reflection lines.
    for(index = 0;index < 4;index++)
        totalSamples += CalcLineLength(EARLY_LINE_LENGTH[index], totalSamples,
                                       frequency, &State->Early.Delay[index]);

    // The decorrelator line is calculated from the lowest reverb density (a
    // parameter value of 1).
    length = (DECO_FRACTION * DECO_MULTIPLIER * DECO_MULTIPLIER) *
             LATE_LINE_LENGTH[0] * (1.0f + LATE_LINE_MULTIPLIER);
    totalSamples += CalcLineLength(length, totalSamples, frequency,
                                   &State->Decorrelator);

    // The late all-pass lines.
    for(index = 0;index < 4;index++)
        totalSamples += CalcLineLength(ALLPASS_LINE_LENGTH[index], totalSamples,
                                       frequency, &State->Late.ApDelay[index]);

    // The late delay lines are calculated from the lowest reverb density.
    for(index = 0;index < 4;index++)
    {
        length = LATE_LINE_LENGTH[index] * (1.0f + LATE_LINE_MULTIPLIER);
        totalSamples += CalcLineLength(length, totalSamples, frequency,
                                       &State->Late.Delay[index]);
    }

    if(eaxFlag)
    {
        // The echo all-pass and delay lines.
        totalSamples += CalcLineLength(ECHO_ALLPASS_LENGTH, totalSamples,
                                       frequency, &State->Echo.ApDelay);
        totalSamples += CalcLineLength(AL_EAXREVERB_MAX_ECHO_TIME, totalSamples,
                                       frequency, &State->Echo.Delay);
    }

    if(totalSamples != State->TotalSamples)
    {
        newBuffer = realloc(State->SampleBuffer, sizeof(ALfloat) * totalSamples);
        if(newBuffer == NULL)
            return AL_FALSE;
        State->SampleBuffer = newBuffer;
        State->TotalSamples = totalSamples;
    }

    // Update all delays to reflect the new sample buffer.
    RealizeLineOffset(State->SampleBuffer, &State->Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Decorrelator);
    for(index = 0;index < 4;index++)
    {
        RealizeLineOffset(State->SampleBuffer, &State->Early.Delay[index]);
        RealizeLineOffset(State->SampleBuffer, &State->Late.ApDelay[index]);
        RealizeLineOffset(State->SampleBuffer, &State->Late.Delay[index]);
    }
    if(eaxFlag)
    {
        RealizeLineOffset(State->SampleBuffer, &State->Mod.Delay);
        RealizeLineOffset(State->SampleBuffer, &State->Echo.ApDelay);
        RealizeLineOffset(State->SampleBuffer, &State->Echo.Delay);
    }

    // Clear the sample buffer.
    for(index = 0;index < State->TotalSamples;index++)
        State->SampleBuffer[index] = 0.0f;

    return AL_TRUE;
}

// Calculate a decay coefficient given the length of each cycle and the time
// until the decay reaches -60 dB.
static __inline ALfloat CalcDecayCoeff(ALfloat length, ALfloat decayTime)
{
    return aluPow(10.0f, length / decayTime * -60.0f / 20.0f);
}

// Calculate a decay length from a coefficient and the time until the decay
// reaches -60 dB.
static __inline ALfloat CalcDecayLength(ALfloat coeff, ALfloat decayTime)
{
    return log10(coeff) / -60.0 * 20.0f * decayTime;
}

// Calculate the high frequency parameter for the I3DL2 coefficient
// calculation.
static __inline ALfloat CalcI3DL2HFreq(ALfloat hfRef, ALuint frequency)
{
    return cos(2.0f * M_PI * hfRef / frequency);
}

// Calculate an attenuation to be applied to the input of any echo models to
// compensate for modal density and decay time.
static __inline ALfloat CalcDensityGain(ALfloat a)
{
    /* The energy of a signal can be obtained by finding the area under the
     * squared signal.  This takes the form of Sum(x_n^2), where x is the
     * amplitude for the sample n.
     *
     * Decaying feedback matches exponential decay of the form Sum(a^n),
     * where a is the attenuation coefficient, and n is the sample.  The area
     * under this decay curve can be calculated as:  1 / (1 - a).
     *
     * Modifying the above equation to find the squared area under the curve
     * (for energy) yields:  1 / (1 - a^2).  Input attenuation can then be
     * calculated by inverting the square root of this approximation,
     * yielding:  1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2).
     */
    return aluSqrt(1.0f - (a * a));
}

// Calculate the mixing matrix coefficients given a diffusion factor.
static __inline ALvoid CalcMatrixCoeffs(ALfloat diffusion, ALfloat *x, ALfloat *y)
{
    ALfloat n, t;

    // The matrix is of order 4, so n is sqrt (4 - 1).
    n = aluSqrt(3.0f);
    t = diffusion * atan(n);

    // Calculate the first mixing matrix coefficient.
    *x = cos(t);
    // Calculate the second mixing matrix coefficient.
    *y = sin(t) / n;
}

// Calculate the limited HF ratio for use with the late reverb low-pass
// filters.
static ALfloat CalcLimitedHfRatio(ALfloat hfRatio, ALfloat airAbsorptionGainHF, ALfloat decayTime)
{
    ALfloat limitRatio;

    /* Find the attenuation due to air absorption in dB (converting delay
     * time to meters using the speed of sound).  Then reversing the decay
     * equation, solve for HF ratio.  The delay length is cancelled out of
     * the equation, so it can be calculated once for all lines.
     */
    limitRatio = 1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) *
                         SPEEDOFSOUNDMETRESPERSEC);
    // Need to limit the result to a minimum of 0.1, just like the HF ratio
    // parameter.
    limitRatio = __max(limitRatio, 0.1f);

    // Using the limit calculated above, apply the upper bound to the HF
    // ratio.
    return __min(hfRatio, limitRatio);
}

// Calculate the coefficient for a HF (and eventually LF) decay damping
// filter.
static __inline ALfloat CalcDampingCoeff(ALfloat hfRatio, ALfloat length, ALfloat decayTime, ALfloat decayCoeff, ALfloat cw)
{
    ALfloat coeff, g;

    // Eventually this should boost the high frequencies when the ratio
    // exceeds 1.
    coeff = 0.0f;
    if (hfRatio < 1.0f)
    {
        // Calculate the low-pass coefficient by dividing the HF decay
        // coefficient by the full decay coefficient.
        g = CalcDecayCoeff(length, decayTime * hfRatio) / decayCoeff;

        // Damping is done with a 1-pole filter, so g needs to be squared.
        g *= g;
        coeff = lpCoeffCalc(g, cw);

        // Very low decay times will produce minimal output, so apply an
        // upper bound to the coefficient.
        coeff = __min(coeff, 0.98f);
    }
    return coeff;
}

// Update the EAX modulation index, range, and depth.  Keep in mind that this
// kind of vibrato is additive and not multiplicative as one may expect.  The
// downswing will sound stronger than the upswing.
static ALvoid UpdateModulator(ALfloat modTime, ALfloat modDepth, ALuint frequency, ALverbState *State)
{
    ALfloat length;

    /* Modulation is calculated in two parts.
     *
     * The modulation time effects the sinus applied to the change in
     * frequency.  An index out of the current time range (both in samples)
     * is incremented each sample.  The range is bound to a reasonable
     * minimum (1 sample) and when the timing changes, the index is rescaled
     * to the new range (to keep the sinus consistent).
     */
    length = modTime * frequency;
    if (length >= 1.0f) {
       State->Mod.Index = (ALuint)(State->Mod.Index * length /
                                   State->Mod.Range);
       State->Mod.Range = (ALuint)length;
    } else {
       State->Mod.Index = 0;
       State->Mod.Range = 1;
    }

    /* The modulation depth effects the amount of frequency change over the
     * range of the sinus.  It needs to be scaled by the modulation time so
     * that a given depth produces a consistent change in frequency over all
     * ranges of time.  Since the depth is applied to a sinus value, it needs
     * to be halfed once for the sinus range and again for the sinus swing
     * in time (half of it is spent decreasing the frequency, half is spent
     * increasing it).
     */
    State->Mod.Depth = modDepth * MODULATION_DEPTH_COEFF * modTime / 2.0f /
                       2.0f * frequency;
}

// Update the offsets for the initial effect delay line.
static ALvoid UpdateDelayLine(ALfloat earlyDelay, ALfloat lateDelay, ALuint frequency, ALverbState *State)
{
    // Calculate the initial delay taps.
    State->DelayTap[0] = (ALuint)(earlyDelay * frequency);
    State->DelayTap[1] = (ALuint)((earlyDelay + lateDelay) * frequency);
}

// Update the early reflections gain and line coefficients.
static ALvoid UpdateEarlyLines(ALfloat reverbGain, ALfloat earlyGain, ALfloat lateDelay, ALverbState *State)
{
    ALuint index;

    // Calculate the early reflections gain (from the master effect gain, and
    // reflections gain parameters) with a constant attenuation of 0.5.
    State->Early.Gain = 0.5f * reverbGain * earlyGain;

    // Calculate the gain (coefficient) for each early delay line using the
    // late delay time.  This expands the early reflections to the start of
    // the late reverb.
    for(index = 0;index < 4;index++)
        State->Early.Coeff[index] = CalcDecayCoeff(EARLY_LINE_LENGTH[index],
                                                   lateDelay);
}

// Update the offsets for the decorrelator line.
static ALvoid UpdateDecorrelator(ALfloat density, ALuint frequency, ALverbState *State)
{
    ALuint index;
    ALfloat length;

    /* The late reverb inputs are decorrelated to smooth the reverb tail and
     * reduce harsh echos.  The first tap occurs immediately, while the
     * remaining taps are delayed by multiples of a fraction of the smallest
     * cyclical delay time.
     *
     * offset[index] = (FRACTION (MULTIPLIER^index)) smallest_delay
     */
    for(index = 0;index < 3;index++)
    {
        length = (DECO_FRACTION * aluPow(DECO_MULTIPLIER, (ALfloat)index)) *
                 LATE_LINE_LENGTH[0] * (1.0f + (density * LATE_LINE_MULTIPLIER));
        State->DecoTap[index] = (ALuint)(length * frequency);
    }
}

// Update the late reverb gains, line lengths, and line coefficients.
static ALvoid UpdateLateLines(ALfloat reverbGain, ALfloat lateGain, ALfloat xMix, ALfloat density, ALfloat decayTime, ALfloat diffusion, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALverbState *State)
{
    ALfloat length;
    ALuint index;

    /* Calculate the late reverb gain (from the master effect gain, and late
     * reverb gain parameters).  Since the output is tapped prior to the
     * application of the next delay line coefficients, this gain needs to be
     * attenuated by the 'x' mixing matrix coefficient as well.
     */
    State->Late.Gain = reverbGain * lateGain * xMix;

    /* To compensate for changes in modal density and decay time of the late
     * reverb signal, the input is attenuated based on the maximal energy of
     * the outgoing signal.  This approximation is used to keep the apparent
     * energy of the signal equal for all ranges of density and decay time.
     *
     * The average length of the cyclcical delay lines is used to calculate
     * the attenuation coefficient.
     */
    length = (LATE_LINE_LENGTH[0] + LATE_LINE_LENGTH[1] +
              LATE_LINE_LENGTH[2] + LATE_LINE_LENGTH[3]) / 4.0f;
    length *= 1.0f + (density * LATE_LINE_MULTIPLIER);
    State->Late.DensityGain = CalcDensityGain(CalcDecayCoeff(length,
                                                             decayTime));

    // Calculate the all-pass feed-back and feed-forward coefficient.
    State->Late.ApFeedCoeff = 0.5f * aluPow(diffusion, 2.0f);

    for(index = 0;index < 4;index++)
    {
        // Calculate the gain (coefficient) for each all-pass line.
        State->Late.ApCoeff[index] = CalcDecayCoeff(ALLPASS_LINE_LENGTH[index],
                                                    decayTime);

        // Calculate the length (in seconds) of each cyclical delay line.
        length = LATE_LINE_LENGTH[index] * (1.0f + (density *
                                                    LATE_LINE_MULTIPLIER));

        // Calculate the delay offset for each cyclical delay line.
        State->Late.Offset[index] = (ALuint)(length * frequency);

        // Calculate the gain (coefficient) for each cyclical line.
        State->Late.Coeff[index] = CalcDecayCoeff(length, decayTime);

        // Calculate the damping coefficient for each low-pass filter.
        State->Late.LpCoeff[index] =
            CalcDampingCoeff(hfRatio, length, decayTime,
                             State->Late.Coeff[index], cw);

        // Attenuate the cyclical line coefficients by the mixing coefficient
        // (x).
        State->Late.Coeff[index] *= xMix;
    }
}

// Update the echo gain, line offset, line coefficients, and mixing
// coefficients.
static ALvoid UpdateEchoLine(ALfloat reverbGain, ALfloat lateGain, ALfloat echoTime, ALfloat decayTime, ALfloat diffusion, ALfloat echoDepth, ALfloat hfRatio, ALfloat cw, ALuint frequency, ALverbState *State)
{
    // Update the offset and coefficient for the echo delay line.
    State->Echo.Offset = (ALuint)(echoTime * frequency);

    // Calculate the decay coefficient for the echo line.
    State->Echo.Coeff = CalcDecayCoeff(echoTime, decayTime);

    // Calculate the energy-based attenuation coefficient for the echo delay
    // line.
    State->Echo.DensityGain = CalcDensityGain(State->Echo.Coeff);

    // Calculate the echo all-pass feed coefficient.
    State->Echo.ApFeedCoeff = 0.5f * aluPow(diffusion, 2.0f);

    // Calculate the echo all-pass attenuation coefficient.
    State->Echo.ApCoeff = CalcDecayCoeff(ECHO_ALLPASS_LENGTH, decayTime);

    // Calculate the damping coefficient for each low-pass filter.
    State->Echo.LpCoeff = CalcDampingCoeff(hfRatio, echoTime, decayTime,
                                           State->Echo.Coeff, cw);

    /* Calculate the echo mixing coefficients.  The first is applied to the
     * echo itself.  The second is used to attenuate the late reverb when
     * echo depth is high and diffusion is low, so the echo is slightly
     * stronger than the decorrelated echos in the reverb tail.
     */
    State->Echo.MixCoeff[0] = reverbGain * lateGain * echoDepth;
    State->Echo.MixCoeff[1] = 1.0f - (echoDepth * 0.5f * (1.0f - diffusion));
}

// Update the early and late 3D panning gains.
static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, ALverbState *State)
{
    ALfloat earlyPan[3] = { ReflectionsPan[0], ReflectionsPan[1],
                            ReflectionsPan[2] };
    ALfloat latePan[3] = { LateReverbPan[0], LateReverbPan[1],
                           LateReverbPan[2] };
    const ALfloat *speakerGain;
    ALfloat dirGain;
    ALfloat length;
    ALuint index;
    ALint pos;

    // Calculate the 3D-panning gains for the early reflections and late
    // reverb.
    length = earlyPan[0]*earlyPan[0] + earlyPan[1]*earlyPan[1] + earlyPan[2]*earlyPan[2];
    if(length > 1.0f)
    {
        length = 1.0f / aluSqrt(length);
        earlyPan[0] *= length;
        earlyPan[1] *= length;
        earlyPan[2] *= length;
    }
    length = latePan[0]*latePan[0] + latePan[1]*latePan[1] + latePan[2]*latePan[2];
    if(length > 1.0f)
    {
        length = 1.0f / aluSqrt(length);
        latePan[0] *= length;
        latePan[1] *= length;
        latePan[2] *= length;
    }

    /* This code applies directional reverb just like the mixer applies
     * directional sources.  It diffuses the sound toward all speakers as the
     * magnitude of the panning vector drops, which is only a rough
     * approximation of the expansion of sound across the speakers from the
     * panning direction.
     */
    pos = aluCart2LUTpos(earlyPan[2], earlyPan[0]);
    speakerGain = &Device->PanningLUT[MAXCHANNELS * pos];
    dirGain = aluSqrt((earlyPan[0] * earlyPan[0]) + (earlyPan[2] * earlyPan[2]));

    for(index = 0;index < MAXCHANNELS;index++)
        State->Early.PanGain[index] = 0.0f;
    for(index = 0;index < Device->NumChan;index++)
    {
        Channel chan = Device->Speaker2Chan[index];
        State->Early.PanGain[chan] = lerp(1.0, speakerGain[chan], dirGain);
    }


    pos = aluCart2LUTpos(latePan[2], latePan[0]);
    speakerGain = &Device->PanningLUT[MAXCHANNELS * pos];
    dirGain = aluSqrt((latePan[0] * latePan[0]) + (latePan[2] * latePan[2]));

    for(index = 0;index < MAXCHANNELS;index++)
         State->Late.PanGain[index] = 0.0f;
    for(index = 0;index < Device->NumChan;index++)
    {
        Channel chan = Device->Speaker2Chan[index];
        State->Late.PanGain[chan] = lerp(1.0, speakerGain[chan], dirGain);
    }
}

// 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;
{'fraction_non_alphanumeric': 0.08822678064877174, 'fraction_numerical': 0.013546631674417707, 'mean_word_length': 2.6180967674369895, 'pattern_counts': {'":': 0, '<': 37, '<?xml version=': 0, '>': 175, 'https://': 0, 'lorem ipsum': 0, 'www.': 2, 'xml': 0}, 'pii_count': 0, 'substrings_counts': 0, 'word_list_counts': {'cursed_substrings.json': 38, 'profanity_word_list.json': 0, 'sexual_word_list.json': 0, 'zh_pornsignals.json': 0}}
{'dir': 'git-commits-cleaned', 'id': '2043739', 'n_tokens_mistral': 16046, 'n_tokens_neox': 15107, 'n_words': 5327}