Make the DYNLOAD LoadFSynth function non-inline - openal-soft.git - Original: git://repo.or.cz/openal-soft.git

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author	Chris Robinson <[email protected]>	2014-08-05 19:38:33 -0700
committer	Chris Robinson <[email protected]>	2014-08-05 19:38:33 -0700
commit	23441be474d2e6c272913432ae25fbf509b551b8 (patch)
tree	1c2104b4eac854b7eb76210ac900d6da4555293f /Alc/midi
parent	951870b18978eab89391f43c3c5b85e0700eb198 (diff)

Make the DYNLOAD LoadFSynth function non-inline

Diffstat (limited to 'Alc/midi')

-rw-r--r--

Alc/midi/fluidsynth.c

1 files changed, 1 insertions, 1 deletions

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/**
 * Ambisonic reverb engine for the OpenAL cross platform audio library
 * Copyright (C) 2008-2017 by Chris Robinson and 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.,
 *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 * Or go to http://www.gnu.org/copyleft/lgpl.html
 */

#include "config.h"

#include <stdio.h>
#include <stdlib.h>
#include <math.h>

#include "alMain.h"
#include "alu.h"
#include "alAuxEffectSlot.h"
#include "alEffect.h"
#include "alFilter.h"
#include "alError.h"
#include "mixer_defs.h"

/* This is the maximum number of samples processed for each inner loop
 * iteration. */
#define MAX_UPDATE_SAMPLES  256

/* The number of samples used for cross-faded delay lines.  This can be used
 * to balance the compensation for abrupt line changes and attenuation due to
 * minimally lengthed recursive lines.  Try to keep this below the device
 * update size.
 */
#define FADE_SAMPLES  128

#ifdef __GNUC__
#define UNEXPECTED(x) __builtin_expect((bool)(x), 0)
#else
#define UNEXPECTED(x) (x)
#endif

static MixerFunc MixSamples = Mix_C;
static RowMixerFunc MixRowSamples = MixRow_C;

static alonce_flag mixfunc_inited = AL_ONCE_FLAG_INIT;
static void init_mixfunc(void)
{
    MixSamples = SelectMixer();
    MixRowSamples = SelectRowMixer();
}

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.
     */
    ALsizei  Mask;
    ALfloat *Line;
} DelayLine;

typedef struct VecAllpass {
    DelayLine Delay;
    ALsizei Offset[4][2];
} VecAllpass;

typedef struct ALreverbState {
    DERIVE_FROM_TYPE(ALeffectState);

    ALboolean IsEax;

    /* All delay lines are allocated as a single buffer to reduce memory
     * fragmentation and management code.
     */
    ALfloat *SampleBuffer;
    ALuint   TotalSamples;

    /* Master effect filters */
    struct {
        ALfilterState Lp;
        ALfilterState Hp; /* EAX only */
    } Filter[4];

    /* Core delay line (early reflections and late reverb tap from this). */
    DelayLine Delay;

    /* Tap points for early reflection delay. */
    ALsizei EarlyDelayTap[4][2];
    ALfloat EarlyDelayCoeff[4];

    /* Tap points for late reverb feed and delay. */
    ALsizei LateFeedTap;
    ALsizei LateDelayTap[4][2];

    /* The feed-back and feed-forward all-pass coefficient. */
    ALfloat ApFeedCoeff;

    /* Coefficients for the all-pass and line scattering matrices. */
    ALfloat MixX;
    ALfloat MixY;

    struct {
        /* A Gerzon vector all-pass filter is used to simulate initial
         * diffusion.  The spread from this filter also helps smooth out the
         * reverb tail.
         */
        VecAllpass VecAp;

        /* Echo lines are used to complete the second half of the early
         * reflections.
         */
        DelayLine Delay[4];
        ALsizei   Offset[4][2];
        ALfloat   Coeff[4];

        /* The gain for each output channel based on 3D panning. */
        ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS];
        ALfloat PanGain[4][MAX_OUTPUT_CHANNELS];
    } Early;

    struct {
        /* 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; /* EAX only */

    struct {
        /* Attenuation to compensate for the modal density and decay rate of
         * the late lines.
         */
        ALfloat DensityGain;

        /* Recursive delay lines are used fill in the reverb tail. */
        DelayLine Delay[4];
        ALsizei   Offset[4][2];

        /* T60 decay filters are used to simulate absorption. */
        struct {
            ALfloat LFCoeffs[3];
            ALfloat HFCoeffs[3];
            ALfloat MidCoeff;
            /* The LF and HF filters keep a state of the last input and last
             * output sample.
             */
            ALfloat States[2][2];
        } Filters[4];

        /* A Gerzon vector all-pass filter is used to simulate diffusion. */
        VecAllpass VecAp;

        /* The gain for each output channel based on 3D panning. */
        ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS];
        ALfloat PanGain[4][MAX_OUTPUT_CHANNELS];
    } Late;

    /* Indicates the cross-fade point for delay line reads [0,FADE_SAMPLES]. */
    ALsizei FadeCount;

    /* The current write offset for all delay lines. */
    ALsizei Offset;

    /* Temporary storage used when processing. */
    alignas(16) ALfloat AFormatSamples[4][MAX_UPDATE_SAMPLES];
    alignas(16) ALfloat ReverbSamples[4][MAX_UPDATE_SAMPLES];
    alignas(16) ALfloat EarlySamples[4][MAX_UPDATE_SAMPLES];
} ALreverbState;

static ALvoid ALreverbState_Destruct(ALreverbState *State);
static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device);
static ALvoid ALreverbState_update(ALreverbState *State, const ALCdevice *Device, const ALeffectslot *Slot, const ALeffectProps *props);
static ALvoid ALreverbState_process(ALreverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALreverbState)

DEFINE_ALEFFECTSTATE_VTABLE(ALreverbState);

static void ALreverbState_Construct(ALreverbState *state)
{
    ALsizei i, j;

    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
    SET_VTABLE2(ALreverbState, ALeffectState, state);

    state->IsEax = AL_FALSE;

    state->TotalSamples = 0;
    state->SampleBuffer = NULL;

    for(i = 0;i < 4;i++)
    {
        ALfilterState_clear(&state->Filter[i].Lp);
        ALfilterState_clear(&state->Filter[i].Hp);
    }

    state->Delay.Mask = 0;
    state->Delay.Line = NULL;

    for(i = 0;i < 4;i++)
    {
        state->EarlyDelayTap[i][0] = 0;
        state->EarlyDelayTap[i][1] = 0;
        state->EarlyDelayCoeff[i] = 0.0f;
    }

    state->LateFeedTap = 0;

    for(i = 0;i < 4;i++)
    {
        state->LateDelayTap[i][0] = 0;
        state->LateDelayTap[i][1] = 0;
    }

    state->ApFeedCoeff = 0.0f;
    state->MixX = 0.0f;
    state->MixY = 0.0f;

    state->Early.VecAp.Delay.Mask = 0;
    state->Early.VecAp.Delay.Line = NULL;
    for(i = 0;i < 4;i++)
    {
        state->Early.VecAp.Offset[i][0] = 0;
        state->Early.VecAp.Offset[i][1] = 0;
        state->Early.Delay[i].Mask = 0;
        state->Early.Delay[i].Line = NULL;
        state->Early.Offset[i][0] = 0;
        state->Early.Offset[i][1] = 0;
        state->Early.Coeff[i] = 0.0f;
    }

    state->Mod.Index = 0;
    state->Mod.Range = 1;
    state->Mod.Depth = 0.0f;
    state->Mod.Coeff = 0.0f;
    state->Mod.Filter = 0.0f;

    state->Late.DensityGain = 0.0f;

    state->Late.VecAp.Delay.Mask = 0;
    state->Late.VecAp.Delay.Line = NULL;
    for(i = 0;i < 4;i++)
    {
        state->Late.VecAp.Offset[i][0] = 0;
        state->Late.VecAp.Offset[i][1] = 0;

        state->Late.Delay[i].Mask = 0;
        state->Late.Delay[i].Line = NULL;
        state->Late.Offset[i][0] = 0;
        state->Late.Offset[i][1] = 0;

        for(j = 0;j < 3;j++)
        {
            state->Late.Filters[i].LFCoeffs[j] = 0.0f;
            state->Late.Filters[i].HFCoeffs[j] = 0.0f;
        }
        state->Late.Filters[i].MidCoeff = 0.0f;

        state->Late.Filters[i].States[0][0] = 0.0f;
        state->Late.Filters[i].States[0][1] = 0.0f;
        state->Late.Filters[i].States[1][0] = 0.0f;
        state->Late.Filters[i].States[1][1] = 0.0f;
    }

    for(i = 0;i < 4;i++)
    {
        for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
        {
            state->Early.CurrentGain[i][j] = 0.0f;
            state->Early.PanGain[i][j] = 0.0f;
            state->Late.CurrentGain[i][j] = 0.0f;
            state->Late.PanGain[i][j] = 0.0f;
        }
    }

    state->FadeCount = 0;
    state->Offset = 0;
}

static ALvoid ALreverbState_Destruct(ALreverbState *State)
{
    al_free(State->SampleBuffer);
    State->SampleBuffer = NULL;

    ALeffectState_Destruct(STATIC_CAST(ALeffectState,State));
}

/* The B-Format to A-Format conversion matrix. The arrangement of rows is
 * deliberately chosen to align the resulting lines to their spatial opposites
 * (0:above front left <-> 3:above back right, 1:below front right <-> 2:below
 * back left). It's not quite opposite, since the A-Format results in a
 * tetrahedron, but it's close enough. Should the model be extended to 8-lines
 * in the future, true opposites can be used.
 */
static const aluMatrixf B2A = {{
    { 0.288675134595f,  0.288675134595f,  0.288675134595f,  0.288675134595f },
    { 0.288675134595f, -0.288675134595f, -0.288675134595f,  0.288675134595f },
    { 0.288675134595f,  0.288675134595f, -0.288675134595f, -0.288675134595f },
    { 0.288675134595f, -0.288675134595f,  0.288675134595f, -0.288675134595f }
}};

/* Converts A-Format to B-Format. */
static const aluMatrixf A2B = {{
    { 0.866025403785f,  0.866025403785f,  0.866025403785f,  0.866025403785f },
    { 0.866025403785f, -0.866025403785f,  0.866025403785f, -0.866025403785f },
    { 0.866025403785f, -0.866025403785f, -0.866025403785f,  0.866025403785f },
    { 0.866025403785f,  0.866025403785f, -0.866025403785f, -0.866025403785f }
}};

static const ALfloat FadeStep = 1.0f / FADE_SAMPLES;

/* 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 (no
 * high-pass, modulation, or echo).
 */
ALboolean EmulateEAXReverb = AL_FALSE;

/* This coefficient is used to define the sinus depth according to the
 * modulation depth property. This value must be below 1, which would cause the
 * sampler to stall on the downswing, and above 1 it will cause it to sample
 * backwards.
 */
static const ALfloat MODULATION_DEPTH_COEFF = 1.0f / 2048.0f;

/* 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;

/* The all-pass and delay lines have a variable length dependent on the
 * effect's density parameter.  The resulting density multiplier is:
 *
 *     multiplier = 1 + (density * LINE_MULTIPLIER)
 *
 * Thus the line multiplier below will result in a maximum density multiplier
 * of 10.
 */
static const ALfloat LINE_MULTIPLIER = 9.0f;

/* All delay line lengths are specified in seconds.
 *
 * To approximate early reflections, we break them up into primary (those
 * arriving from the same direction as the source) and secondary (those
 * arriving from the opposite direction).
 *
 * The early taps decorrelate the 4-channel signal to approximate an average
 * room response for the primary reflections after the initial early delay.
 *
 * Given an average room dimension (d_a) and the speed of sound (c) we can
 * calculate the average reflection delay (r_a) regardless of listener and
 * source positions as:
 *
 *     r_a = d_a / c
 *     c   = 343.3
 *
 * This can extended to finding the average difference (r_d) between the
 * maximum (r_1) and minimum (r_0) reflection delays:
 *
 *     r_0 = 2 / 3 r_a
 *         = r_a - r_d / 2
 *         = r_d
 *     r_1 = 4 / 3 r_a
 *         = r_a + r_d / 2
 *         = 2 r_d
 *     r_d = 2 / 3 r_a
 *         = r_1 - r_0
 *
 * As can be determined by integrating the 1D model with a source (s) and
 * listener (l) positioned across the dimension of length (d_a):
 *
 *     r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c
 *
 * The initial taps (T_(i=0)^N) are then specified by taking a power series
 * that ranges between r_0 and half of r_1 less r_0:
 *
 *     R_i = 2^(i / (2 N - 1)) r_d
 *         = r_0 + (2^(i / (2 N - 1)) - 1) r_d
 *         = r_0 + T_i
 *     T_i = R_i - r_0
 *         = (2^(i / (2 N - 1)) - 1) r_d
 *
 * Assuming an average of 5m (up to 50m with the density multiplier), we get
 * the following taps:
 */
static const ALfloat EARLY_TAP_LENGTHS[4] =
{
    0.000000e+0f, 1.010676e-3f, 2.126553e-3f, 3.358580e-3f
};

/* The early all-pass filter lengths are based on the early tap lengths:
 *
 *     A_i = R_i / a
 *
 * Where a is the approximate maximum all-pass cycle limit (20).
 */
static const ALfloat EARLY_ALLPASS_LENGTHS[4] =
{
    4.854840e-4f, 5.360178e-4f, 5.918117e-4f, 6.534130e-4f
};

/* The early delay lines are used to transform the primary reflections into
 * the secondary reflections.  The A-format is arranged in such a way that
 * the channels/lines are spatially opposite:
 *
 *     C_i is opposite C_(N-i-1)
 *
 * The delays of the two opposing reflections (R_i and O_i) from a source
 * anywhere along a particular dimension always sum to twice its full delay:
 *
 *     2 r_a = R_i + O_i
 *
 * With that in mind we can determine the delay between the two reflections
 * and thus specify our early line lengths (L_(i=0)^N) using:
 *
 *     O_i = 2 r_a - R_(N-i-1)
 *     L_i = O_i - R_(N-i-1)
 *         = 2 (r_a - R_(N-i-1))
 *         = 2 (r_a - T_(N-i-1) - r_0)
 *         = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1)))
 *
 * Using an average dimension of 5m, we get:
 */
static const ALfloat EARLY_LINE_LENGTHS[4] =
{
    2.992520e-3f, 5.456575e-3f, 7.688329e-3f, 9.709681e-3f
};

/* The late all-pass filter lengths are based on the late line lengths:
 *
 *     A_i = (5 / 3) L_i / r_1
 */
static const ALfloat LATE_ALLPASS_LENGTHS[4] =
{
    8.091400e-4f, 1.019453e-3f, 1.407968e-3f, 1.618280e-3f
};

/* The late lines are used to approximate the decaying cycle of recursive
 * late reflections.
 *
 * Splitting the lines in half, we start with the shortest reflection paths
 * (L_(i=0)^(N/2)):
 *
 *     L_i = 2^(i / (N - 1)) r_d
 *
 * Then for the opposite (longest) reflection paths (L_(i=N/2)^N):
 *
 *     L_i = 2 r_a - L_(i-N/2)
 *         = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d
 *
 * For our 5m average room, we get:
 */
static const ALfloat LATE_LINE_LENGTHS[4] =
{
    9.709681e-3f, 1.223343e-2f, 1.689561e-2f, 1.941936e-2f
};

/* HACK: Workaround for a modff bug in 32-bit Windows, which attempts to write
 * a 64-bit double to the 32-bit float parameter.
 */
#if defined(_WIN32) && !defined (_M_X64) && !defined(_M_ARM)
static inline float hack_modff(float x, float *y)
{
    double di;
    double df = modf((double)x, &di);
    *y = (float)di;
    return (float)df;
}
#define modff hack_modff
#endif

/**************************************
 *  Device Update                     *
 **************************************/

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

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

    /* All line lengths are powers of 2, calculated from their lengths in
     * seconds, rounded up.
     */
    samples = fastf2u(ceilf(length*frequency));
    samples = NextPowerOf2((samples+extra) * splmult);

    /* All lines share a single sample buffer. */
    Delay->Mask = samples - 1;
    Delay->Line = (ALfloat*)offset;

    /* Return the sample count for accumulation. */
    return samples;
}

/* Calculates the delay line metrics and allocates the shared sample buffer
 * for all lines given the sample rate (frequency).  If an allocation failure
 * occurs, it returns AL_FALSE.
 */
static ALboolean AllocLines(const ALuint frequency, ALreverbState *State)
{
    ALuint totalSamples, i;
    ALfloat multiplier, length;

    /* All delay line lengths are calculated to accomodate the full range of
     * lengths given their respective paramters.
     */
    totalSamples = 0;

    /* The main delay length includes the maximum early reflection delay, the
     * largest early tap width, the maximum late reverb delay, and the
     * largest late tap width.  Finally, it must also be extended by the
     * update size (MAX_UPDATE_SAMPLES*4) for block processing.
     */
    multiplier = 1.0f + LINE_MULTIPLIER;
    length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY +
             EARLY_TAP_LENGTHS[3]*multiplier +
             AL_EAXREVERB_MAX_LATE_REVERB_DELAY +
             (LATE_LINE_LENGTHS[3] - LATE_LINE_LENGTHS[0])*0.25f*multiplier;
    /* Multiply length by 4, since we're storing 4 interleaved channels in the
     * main delay line.
     */
    totalSamples += CalcLineLength(length, totalSamples, frequency, MAX_UPDATE_SAMPLES, 4,
                                   &State->Delay);

    /* The early all-pass line. Multiply by 4, for 4 interleaved channels. */
    length = (EARLY_ALLPASS_LENGTHS[0]+EARLY_ALLPASS_LENGTHS[1]+
              EARLY_ALLPASS_LENGTHS[2]+EARLY_ALLPASS_LENGTHS[3]) * multiplier;
    totalSamples += CalcLineLength(length, totalSamples, frequency, 0, 4,
                                   &State->Early.VecAp.Delay);

    /* The early reflection lines. */
    for(i = 0;i < 4;i++)
    {
        length = EARLY_LINE_LENGTHS[i] * multiplier;
        totalSamples += CalcLineLength(length, totalSamples, frequency, 0, 1,
                                       &State->Early.Delay[i]);
    }

    /* The late vector all-pass line. Multiply by 4, for 4 interleaved channels. */
    length = (LATE_ALLPASS_LENGTHS[0]+LATE_ALLPASS_LENGTHS[1]+
              LATE_ALLPASS_LENGTHS[2]+LATE_ALLPASS_LENGTHS[3]) * multiplier;
    totalSamples += CalcLineLength(length, totalSamples, frequency, 0, 4,
                                   &State->Late.VecAp.Delay);

    /* The late delay lines are calculated from the larger of the maximum
     * density line length or the maximum echo time, and includes the maximum
     * modulation-related delay. The modulator's delay is calculated from the
     * maximum modulation time and depth coefficient, and halved for the low-
     * to-high frequency swing.
     */
    for(i = 0;i < 4;i++)
    {
        length = maxf(AL_EAXREVERB_MAX_ECHO_TIME, LATE_LINE_LENGTHS[i]*multiplier) +
                 AL_EAXREVERB_MAX_MODULATION_TIME*MODULATION_DEPTH_COEFF/2.0f;
        totalSamples += CalcLineLength(length, totalSamples, frequency, 0, 1,
                                       &State->Late.Delay[i]);
    }

    if(totalSamples != State->TotalSamples)
    {
        ALfloat *newBuffer;

        TRACE("New reverb buffer length: %u samples\n", totalSamples);
        newBuffer = al_calloc(16, sizeof(ALfloat) * totalSamples);
        if(!newBuffer) return AL_FALSE;

        al_free(State->SampleBuffer);
        State->SampleBuffer = newBuffer;
        State->TotalSamples = totalSamples;
    }

    /* Update all delays to reflect the new sample buffer. */
    RealizeLineOffset(State->SampleBuffer, &State->Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Early.VecAp.Delay);
    for(i = 0;i < 4;i++)
        RealizeLineOffset(State->SampleBuffer, &State->Early.Delay[i]);
    RealizeLineOffset(State->SampleBuffer, &State->Late.VecAp.Delay);
    for(i = 0;i < 4;i++)
        RealizeLineOffset(State->SampleBuffer, &State->Late.Delay[i]);

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

    return AL_TRUE;
}

static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device)
{
    ALuint frequency = Device->Frequency, i;
    ALfloat multiplier;

    /* Allocate the delay lines. */
    if(!AllocLines(frequency, State))
        return AL_FALSE;

    /* Calculate the modulation filter coefficient.  Notice that the exponent
     * is calculated given the current sample rate.  This ensures that the
     * resulting filter response over time is consistent across all sample
     * rates.
     */
    State->Mod.Coeff = powf(MODULATION_FILTER_COEFF,
                            MODULATION_FILTER_CONST / frequency);

    multiplier = 1.0f + LINE_MULTIPLIER;

    /* The late feed taps are set a fixed position past the latest delay tap. */
    for(i = 0;i < 4;i++)
        State->LateFeedTap = fastf2u((AL_EAXREVERB_MAX_REFLECTIONS_DELAY +
                                      EARLY_TAP_LENGTHS[3]*multiplier) *
                                     frequency);

    return AL_TRUE;
}

/**************************************
 *  Effect Update                     *
 **************************************/

/* Calculate a decay coefficient given the length of each cycle and the time
 * until the decay reaches -60 dB.
 */
static inline ALfloat CalcDecayCoeff(const ALfloat length, const ALfloat decayTime)
{
    return powf(0.001f/*-60 dB*/, length/decayTime);
}

/* Calculate a decay length from a coefficient and the time until the decay
 * reaches -60 dB.
 */
static inline ALfloat CalcDecayLength(const ALfloat coeff, const ALfloat decayTime)
{
    return log10f(coeff) * decayTime / log10f(0.001f)/*-60 dB*/;
}

/* 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(const 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 area under the squared 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 sqrtf(1.0f - a*a);
}

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

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

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

/* Calculate the limited HF ratio for use with the late reverb low-pass
 * filters.
 */
static ALfloat CalcLimitedHfRatio(const ALfloat hfRatio, const ALfloat airAbsorptionGainHF,
                                  const 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);
    /* Using the limit calculated above, apply the upper bound to the HF
     * ratio. Also need to limit the result to a minimum of 0.1, just like
     * the HF ratio parameter.
     */
    return clampf(limitRatio, 0.1f, hfRatio);
}

/* Calculates the first-order high-pass coefficients following the I3DL2
 * reference model.  This is the transfer function:
 *
 *                1 - z^-1
 *     H(z) = p ------------
 *               1 - p z^-1
 *
 * And this is the I3DL2 coefficient calculation given gain (g) and reference
 * angular frequency (w):
 *
 *                                    g
 *      p = ------------------------------------------------------
 *          g cos(w) + sqrt((cos(w) - 1) (g^2 cos(w) + g^2 - 2))
 *
 * The coefficient is applied to the partial differential filter equation as:
 *
 *     c_0 = p
 *     c_1 = -p
 *     c_2 = p
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcHighpassCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, g2, cw, p;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    g = maxf(0.001f, gain);
    g2 = g * g;
    cw = cosf(w);
    p = g / (g*cw + sqrt((cw - 1.0f) * (g2*cw + g2 - 2.0f)));

    coeffs[0] = p;
    coeffs[1] = -p;
    coeffs[2] = p;
}

/* Calculates the first-order low-pass coefficients following the I3DL2
 * reference model.  This is the transfer function:
 *
 *              (1 - a) z^0
 *     H(z) = ----------------
 *             1 z^0 - a z^-1
 *
 * And this is the I3DL2 coefficient calculation given gain (g) and reference
 * angular frequency (w):
 *
 *          1 - g^2 cos(w) - sqrt(2 g^2 (1 - cos(w)) - g^4 (1 - cos(w)^2))
 *     a = ----------------------------------------------------------------
 *                                    1 - g^2
 *
 * The coefficient is applied to the partial differential filter equation as:
 *
 *     c_0 = 1 - a
 *     c_1 = 0
 *     c_2 = a
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcLowpassCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, g2, cw, a;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    /* Be careful with gains < 0.001, as that causes the coefficient
     * to head towards 1, which will flatten the signal. */
    g = maxf(0.001f, gain);
    g2 = g * g;
    cw = cosf(w);
    a = (1.0f - g2*cw - sqrtf((2.0f*g2*(1.0f - cw)) - g2*g2*(1.0f - cw*cw))) /
        (1.0f - g2);

    coeffs[0] = 1.0f - a;
    coeffs[1] = 0.0f;
    coeffs[2] = a;
}

/* Calculates the first-order low-shelf coefficients.  The shelf filters are
 * used in place of low/high-pass filters to preserve the mid-band.  This is
 * the transfer function:
 *
 *             a_0 + a_1 z^-1
 *     H(z) = ----------------
 *              1 + b_1 z^-1
 *
 * And these are the coefficient calculations given cut gain (g) and a center
 * angular frequency (w):
 *
 *          sin(0.5 (pi - w) - 0.25 pi)
 *     p = -----------------------------
 *          sin(0.5 (pi - w) + 0.25 pi)
 *
 *          g + 1           g + 1
 *     a = ------- + sqrt((-------)^2 - 1)
 *          g - 1           g - 1
 *
 *            1 + g + (1 - g) a
 *     b_0 = -------------------
 *                    2
 *
 *            1 - g + (1 + g) a
 *     b_1 = -------------------
 *                    2
 *
 * The coefficients are applied to the partial differential filter equation
 * as:
 *
 *            b_0 + p b_1
 *     c_0 = -------------
 *              1 + p a
 *
 *            -(b_1 + p b_0)
 *     c_1 = ----------------
 *               1 + p a
 *
 *             p + a
 *     c_2 = ---------
 *            1 + p a
 *
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcLowShelfCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, rw, p, n;
    ALfloat alpha, beta0, beta1;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    g = maxf(0.001f, gain);
    rw = F_PI - w;
    p = sinf(0.5f*rw - 0.25f*F_PI) / sinf(0.5f*rw + 0.25f*F_PI);
    n = (g + 1.0f) / (g - 1.0f);
    alpha = n + sqrtf(n*n - 1.0f);
    beta0 = (1.0f + g + (1.0f - g)*alpha) / 2.0f;
    beta1 = (1.0f - g + (1.0f + g)*alpha) / 2.0f;

    coeffs[0] = (beta0 + p*beta1) / (1.0f + p*alpha);
    coeffs[1] = -(beta1 + p*beta0) / (1.0f + p*alpha);
    coeffs[2] = (p + alpha) / (1.0f + p*alpha);
}

/* Calculates the first-order high-shelf coefficients.  The shelf filters are
 * used in place of low/high-pass filters to preserve the mid-band.  This is
 * the transfer function:
 *
 *             a_0 + a_1 z^-1
 *     H(z) = ----------------
 *              1 + b_1 z^-1
 *
 * And these are the coefficient calculations given cut gain (g) and a center
 * angular frequency (w):
 *
 *          sin(0.5 w - 0.25 pi)
 *     p = ----------------------
 *          sin(0.5 w + 0.25 pi)
 *
 *          g + 1           g + 1
 *     a = ------- + sqrt((-------)^2 - 1)
 *          g - 1           g - 1
 *
 *            1 + g + (1 - g) a
 *     b_0 = -------------------
 *                    2
 *
 *            1 - g + (1 + g) a
 *     b_1 = -------------------
 *                    2
 *
 * The coefficients are applied to the partial differential filter equation
 * as:
 *
 *            b_0 + p b_1
 *     c_0 = -------------
 *              1 + p a
 *
 *            b_1 + p b_0
 *     c_1 = -------------
 *              1 + p a
 *
 *            -(p + a)
 *     c_2 = ----------
 *            1 + p a
 *
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcHighShelfCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, p, n;
    ALfloat alpha, beta0, beta1;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    g = maxf(0.001f, gain);
    p = sinf(0.5f*w - 0.25f*F_PI) / sinf(0.5f*w + 0.25f*F_PI);
    n = (g + 1.0f) / (g - 1.0f);
    alpha = n + sqrtf(n*n - 1.0f);
    beta0 = (1.0f + g + (1.0f - g)*alpha) / 2.0f;
    beta1 = (1.0f - g + (1.0f + g)*alpha) / 2.0f;

    coeffs[0] = (beta0 + p*beta1) / (1.0f + p*alpha);
    coeffs[1] = (beta1 + p*beta0) / (1.0f + p*alpha);
    coeffs[2] = -(p + alpha) / (1.0f + p*alpha);
}

/* Calculates the 3-band T60 damping coefficients for a particular delay line
 * of specified length using a combination of two low/high-pass/shelf or
 * pass-through filter sections (producing 3 coefficients each) and a general
 * gain (7th coefficient) given decay times for each band split at two (LF/
 * HF) reference frequencies (w).
 */
static void CalcT60DampingCoeffs(const ALfloat length, const ALfloat lfDecayTime,
                                 const ALfloat mfDecayTime, const ALfloat hfDecayTime,
                                 const ALfloat lfW, const ALfloat hfW, ALfloat lfcoeffs[3],
                                 ALfloat hfcoeffs[3], ALfloat *midcoeff)
{
    ALfloat lfGain = CalcDecayCoeff(length, lfDecayTime);
    ALfloat mfGain = CalcDecayCoeff(length, mfDecayTime);
    ALfloat hfGain = CalcDecayCoeff(length, hfDecayTime);

    if(lfGain < mfGain)
    {
        if(mfGain < hfGain)
        {
            CalcLowShelfCoeffs(mfGain / hfGain, hfW, lfcoeffs);
            CalcHighpassCoeffs(lfGain / mfGain, lfW, hfcoeffs);
            *midcoeff = hfGain;
        }
        else if(mfGain > hfGain)
        {
            CalcHighpassCoeffs(lfGain / mfGain, lfW, lfcoeffs);
            CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
            *midcoeff = mfGain;
        }
        else
        {
            lfcoeffs[0] = 1.0f;
            lfcoeffs[1] = 0.0f;
            lfcoeffs[2] = 0.0f;
            CalcHighpassCoeffs(lfGain / mfGain, lfW, hfcoeffs);
            *midcoeff = mfGain;
        }
    }
    else if(lfGain > mfGain)
    {
        if(mfGain < hfGain)
        {
            double hg = mfGain / lfGain;
            double lg = mfGain / hfGain;

            CalcHighShelfCoeffs(hg, lfW, lfcoeffs);
            CalcLowShelfCoeffs(lg, hfW, hfcoeffs);
            *midcoeff = maxf(lfGain, hfGain) / maxf(hg, lg);
        }
        else if(mfGain > hfGain)
        {
            CalcHighShelfCoeffs(mfGain / lfGain, lfW, lfcoeffs);
            CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
            *midcoeff = lfGain;
        }
        else
        {
            lfcoeffs[0] = 1.0f;
            lfcoeffs[1] = 0.0f;
            lfcoeffs[2] = 0.0f;
            CalcHighShelfCoeffs(mfGain / lfGain, lfW, hfcoeffs);
            *midcoeff = lfGain;
        }
    }
    else
    {
        lfcoeffs[0] = 1.0f;
        lfcoeffs[1] = 0.0f;
        lfcoeffs[2] = 0.0f;

        if(mfGain < hfGain)
        {
            CalcLowShelfCoeffs(mfGain / hfGain, hfW, hfcoeffs);
            *midcoeff = hfGain;
        }
        else if(mfGain > hfGain)
        {
            CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
            *midcoeff = mfGain;
        }
        else
        {
            hfcoeffs[3] = 1.0f;
            hfcoeffs[4] = 0.0f;
            hfcoeffs[5] = 0.0f;
            *midcoeff = mfGain;
        }
    }
}

/* 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(const ALfloat modTime, const ALfloat modDepth,
                              const ALuint frequency, ALreverbState *State)
{
    ALuint range;

    /* Modulation is calculated in two parts.
     *
     * The modulation time effects the speed of the sinus. An index out of the
     * current range (both in samples) is incremented each sample, so a longer
     * time implies a larger range. 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.
     */
    range = maxu(fastf2u(modTime*frequency), 1);
    State->Mod.Index = (ALuint)(State->Mod.Index * (ALuint64)range /
                                State->Mod.Range);
    State->Mod.Range = range;

    /* The modulation depth effects the scale of the sinus, which changes how
     * much extra delay is added to the delay line. This delay changing over
     * time changes the pitch, creating the modulation effect. The scale needs
     * to be multiplied by the modulation time so that a given depth produces a
     * consistent shift in frequency over all ranges of time. Since the depth
     * is applied to a sinus value, it needs to be halved once for the sinus
     * range (-1...+1 to 0...1) 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 main effect delay line. */
static ALvoid UpdateDelayLine(const ALfloat earlyDelay, const ALfloat lateDelay, const ALfloat density, const ALfloat decayTime, const ALuint frequency, ALreverbState *State)
{
    ALfloat multiplier, length;
    ALuint i;

    multiplier = 1.0f + density*LINE_MULTIPLIER;

    /* Early reflection taps are decorrelated by means of an average room
     * reflection approximation described above the definition of the taps.
     * This approximation is linear and so the above density multiplier can
     * be applied to adjust the width of the taps.  A single-band decay
     * coefficient is applied to simulate initial attenuation and absorption.
     *
     * Late reverb taps are based on the late line lengths to allow a zero-
     * delay path and offsets that would continue the propagation naturally
     * into the late lines.
     */
    for(i = 0;i < 4;i++)
    {
        length = earlyDelay + EARLY_TAP_LENGTHS[i]*multiplier;
        State->EarlyDelayTap[i][1] = fastf2u(length * frequency);

        length = EARLY_TAP_LENGTHS[i]*multiplier;
        State->EarlyDelayCoeff[i] = CalcDecayCoeff(length, decayTime);

        length = lateDelay + (LATE_LINE_LENGTHS[i] - LATE_LINE_LENGTHS[0])*0.25f*multiplier;
        State->LateDelayTap[i][1] = State->LateFeedTap + fastf2u(length * frequency);
    }
}

/* Update the early reflection line lengths and gain coefficients. */
static ALvoid UpdateEarlyLines(const ALfloat density, const ALfloat decayTime, const ALuint frequency, ALreverbState *State)
{
    ALfloat multiplier, length;
    ALsizei i;

    multiplier = 1.0f + density*LINE_MULTIPLIER;

    for(i = 0;i < 4;i++)
    {
        /* Calculate the length (in seconds) of each all-pass line. */
        length = EARLY_ALLPASS_LENGTHS[i] * multiplier;

        /* Calculate the delay offset for each all-pass line. */
        State->Early.VecAp.Offset[i][1] = fastf2i(length * frequency);

        /* Calculate the length (in seconds) of each delay line. */
        length = EARLY_LINE_LENGTHS[i] * multiplier;

        /* Calculate the delay offset for each delay line. */
        State->Early.Offset[i][1] = fastf2u(length * frequency);

        /* Calculate the gain (coefficient) for each line. */
        State->Early.Coeff[i] = CalcDecayCoeff(length, decayTime);
    }
}

/* Update the late reverb line lengths and T60 coefficients. */
static ALvoid UpdateLateLines(const ALfloat density, const ALfloat diffusion, const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lfW, const ALfloat hfW, const ALfloat echoTime, const ALfloat echoDepth, const ALuint frequency, ALreverbState *State)
{
    ALfloat multiplier, length, bandWeights[3];
    ALsizei i;

    /* 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 delay lines is used to calculate the
     * attenuation coefficient.
     */
    multiplier = 1.0f + density*LINE_MULTIPLIER;
    length = (LATE_LINE_LENGTHS[0] + LATE_LINE_LENGTHS[1] +
              LATE_LINE_LENGTHS[2] + LATE_LINE_LENGTHS[3]) / 4.0f * multiplier;
    /* Include the echo transformation (see below). */
    length = lerp(length, echoTime, echoDepth);
    length += (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] +
               LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f * multiplier;
    /* The density gain calculation uses an average decay time weighted by
     * approximate bandwidth.  This attempts to compensate for losses of
     * energy that reduce decay time due to scattering into highly attenuated
     * bands.
     */
    bandWeights[0] = lfW;
    bandWeights[1] = hfW - lfW;


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