Implement a new reverb effect

Code created and graciously provided by Christopher Fitzgerald

2 files changed, 562 insertions, 59 deletions

diff --git a/Alc/ALu.c b/Alc/ALu.c
index fbf5a128..1f755621 100644
--- a/Alc/ALu.c
+++ b/Alc/ALu.c
@@ -33,6 +33,7 @@
 #include "alAuxEffectSlot.h"
 #include "alu.h"
 #include "bs2b.h"
+#include "alReverb.h"
 
 #if defined (HAVE_FLOAT_H)
 #include <float.h>
@@ -459,28 +460,24 @@ static ALvoid CalcSourceParams(ALCcontext *ALContext, ALsource *ALSource,
         if(ALSource->Send[0].Slot &&
            ALSource->Send[0].Slot->effect.type != AL_EFFECT_NULL)
         {
-            // If the slot's auxilliary send auto is off, the data sent to the
-            // effect slot is the same as the dry path, sans filter effects
-            if(!ALSource->Send[0].Slot->AuxSendAuto)
+            if(ALSource->Send[0].Slot->AuxSendAuto)
             {
+                // Apply minimal attenuation in place of missing statistical
+                // reverb model.
+                WetMix *= pow(DryMix, 1.0f / 2.0f);
+            }
+            else
+            {
+                // If the slot's auxilliary send auto is off, the data sent to the
+                // effect slot is the same as the dry path, sans filter effects
                 WetMix = DryMix;
                 WetGainHF = DryGainHF;
             }
 
-            // Note that these are really applied by the effect slot. However,
-            // it's easier to handle them here (particularly the lowpass
-            // filter). Applying the gain to the individual sources going to
-            // the effect slot should have the same effect as applying the gain
-            // to the accumulated sources in the effect slot.
-            // vol1*g + vol2*g + ... voln*g = (vol1+vol2+...voln)*g
-            WetMix *= ALSource->Send[0].Slot->Gain;
+            // Note that this is really applied by the effect slot. However,
+            // it's easier (more optimal) to handle it here.
             if(ALSource->Send[0].Slot->effect.type == AL_EFFECT_REVERB)
-            {
-                WetMix *= ALSource->Send[0].Slot->effect.Reverb.Gain;
                 WetGainHF *= ALSource->Send[0].Slot->effect.Reverb.GainHF;
-                WetGainHF *= pow(ALSource->Send[0].Slot->effect.Reverb.AirAbsorptionGainHF,
-                                 Distance * MetersPerUnit);
-            }
         }
         else
         {
@@ -969,50 +966,7 @@ ALvoid aluMixData(ALCcontext *ALContext,ALvoid *buffer,ALsizei size,ALenum forma
         while(ALEffectSlot)
         {
             if(ALEffectSlot->effect.type == AL_EFFECT_REVERB)
-            {
-                ALfloat *DelayBuffer = ALEffectSlot->ReverbBuffer;
-                ALuint Pos = ALEffectSlot->ReverbPos;
-                ALuint LatePos = ALEffectSlot->ReverbLatePos;
-                ALuint ReflectPos = ALEffectSlot->ReverbReflectPos;
-                ALuint Length = ALEffectSlot->ReverbLength;
-                ALfloat DecayGain = ALEffectSlot->ReverbDecayGain;
-                ALfloat DecayHFRatio = ALEffectSlot->effect.Reverb.DecayHFRatio;
-                ALfloat ReflectGain = ALEffectSlot->effect.Reverb.ReflectionsGain;
-                ALfloat LateReverbGain = ALEffectSlot->effect.Reverb.LateReverbGain;
-                ALfloat sample, lowsample;
-
-                WetFilter = &ALEffectSlot->iirFilter;
-                for(i = 0;i < SamplesToDo;i++)
-                {
-                    DelayBuffer[Pos] = WetBuffer[i];
-
-                    sample = DelayBuffer[ReflectPos] * ReflectGain;
-
-                    DelayBuffer[LatePos] *= LateReverbGain;
-
-                    Pos = (Pos+1) % Length;
-                    lowsample = lpFilter(WetFilter, DelayBuffer[Pos]);
-                    lowsample += (DelayBuffer[Pos]-lowsample) * DecayHFRatio;
-
-                    DelayBuffer[LatePos] += lowsample * DecayGain;
-
-                    sample += DelayBuffer[LatePos];
-
-                    DryBuffer[i][FRONT_LEFT]  += sample;
-                    DryBuffer[i][FRONT_RIGHT] += sample;
-                    DryBuffer[i][SIDE_LEFT]   += sample;
-                    DryBuffer[i][SIDE_RIGHT]  += sample;
-                    DryBuffer[i][BACK_LEFT]   += sample;
-                    DryBuffer[i][BACK_RIGHT]  += sample;
-
-                    LatePos = (LatePos+1) % Length;
-                    ReflectPos = (ReflectPos+1) % Length;
-                }
-
-                ALEffectSlot->ReverbPos = Pos;
-                ALEffectSlot->ReverbLatePos = LatePos;
-                ALEffectSlot->ReverbReflectPos = ReflectPos;
-            }
+                VerbProcess(ALEffectSlot->ReverbState, SamplesToDo, WetBuffer, DryBuffer);
 
             ALEffectSlot = ALEffectSlot->next;
         }
diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c
new file mode 100644
index 00000000..badbac34
--- /dev/null
+++ b/Alc/alcReverb.c
@@ -0,0 +1,549 @@
+// TODO:  Look into the distance attenuation done by the statistical model,
+//        and see if it matches 1 / sqrt (distance) or some variant thereof.
+//        If not, try to map it so it can replace the current dry-path model.
+//        Also see how it responds to the distance model.  Then if necessary
+//        update ALu.c to compensate.
+// TODO:  Finalize all updates and add necessary comments.  Merge changes
+//        with latest GIT snapshot and test all parameters thoroughly.  When
+//        ready produce some .diff files for the changes and include with
+//        alReverb.c for presentation.
+
+#include "config.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+#include "AL/al.h"
+#include "AL/alc.h"
+#include "alMain.h"
+#include "alAuxEffectSlot.h"
+#include "alEffect.h"
+#include "alReverb.h"
+
+#ifdef HAVE_SQRTF
+#define aluSqrt(x) ((ALfloat)sqrtf((float)(x)))
+#else
+#define aluSqrt(x) ((ALfloat)sqrt((double)(x)))
+#endif
+
+// fixes for mingw32.
+#if defined(max) && !defined(__max)
+#define __max max
+#endif
+#if defined(min) && !defined(__min)
+#define __min min
+#endif
+
+typedef struct DelayLine
+{
+    // The delay lines use lengths that are powers of 2 to allow bitmasking
+    // instead of modulus wrapping.
+    ALuint   Mask;
+    ALfloat *Line;
+} DelayLine;
+
+struct ALverbState
+{
+    // All delay lines are allocated as a single buffer to reduce memory
+    // fragmentation and teardown code.
+    ALfloat  *SampleBuffer;
+    // Master reverb gain.
+    ALfloat   Gain;
+    // Initial reverb delay.
+    DelayLine Delay;
+    // The tap points for the initial delay.  First tap goes to early
+    // reflections, the second to late reverb.
+    ALuint    Tap[2];
+    struct {
+        // Gain for early reflections.
+        ALfloat   Gain;
+        // Early reflections are done with 4 delay lines.
+        ALfloat   Coeff[4];
+        DelayLine Delay[4];
+        ALuint    Offset[4];
+    } Early;
+    struct {
+        // Gain for late reverb.
+        ALfloat   Gain;
+        // Diffusion of late reverb.
+        ALfloat   Diffusion;
+        // Late reverb is done with 8 delay lines.
+        ALfloat   Coeff[8];
+        DelayLine Delay[8];
+        ALuint    Offset[8];
+        // The input and last 4 delay lines are low-pass filtered.
+        ALfloat   LpCoeff[5];
+        ALfloat   LpSample[5];
+    } Late;
+    ALuint Offset;
+};
+
+// All delay line lengths are specified in seconds.
+
+// The length of the initial delay line (a sum of the maximum delay before
+// early reflections and late reverb; 0.3 + 0.1).
+static const ALfloat MASTER_LINE_LENGTH = 0.4000f;
+
+// 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 delay lines.
+static const ALfloat LATE_LINE_LENGTH[8] =
+{
+    0.0015f, 0.0037f, 0.0093f, 0.0234f,
+    0.0100f, 0.0150f, 0.0225f, 0.0337f
+};
+
+// The last 4 late delay lines have a variable length dependent on the effect
+// density parameter and this multiplier.
+static const ALfloat LATE_LINE_MULTIPLIER = 9.0f;
+
+static ALuint NextPowerOf2(ALuint value)
+{
+    ALuint powerOf2 = 1;
+
+    if(value)
+    {
+        value--;
+        while(value)
+        {
+            value >>= 1;
+            powerOf2 <<= 1;
+        }
+    }
+    return powerOf2;
+}
+
+// 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;
+}
+
+// Delay line output routine for early reflections.
+static __inline ALfloat EarlyDelayLineOut(ALverbState *State, ALuint index)
+{
+    return State->Early.Coeff[index] *
+           DelayLineOut(&State->Early.Delay[index],
+                        State->Offset - State->Early.Offset[index]);
+}
+
+// Given an input sample, this function produces a decorrelated stereo output
+// for early reflections.
+static __inline ALvoid EarlyReflection(ALverbState *State, ALfloat in, ALfloat *out)
+{
+    ALfloat d[4], v, f[4];
+
+    // Obtain the decayed results of each early delay line.
+    d[0] = EarlyDelayLineOut(State, 0);
+    d[1] = EarlyDelayLineOut(State, 1);
+    d[2] = EarlyDelayLineOut(State, 2);
+    d[3] = EarlyDelayLineOut(State, 3);
+
+    /* The following uses a lossless scattering junction from waveguide
+     * theory.  It actually amounts to a householder mixing matrix, which
+     * will produce a maximally diffuse response, and means this can probably
+     * be considered a simple FDN.
+     *          N
+     *         ---
+     *         \
+     * v = 2/N /   di
+     *         ---
+     *         i=1
+     */
+    v = (d[0] + d[1] + d[2] + d[3]) * 0.5f;
+    // The junction is loaded with the input here.
+    v += in;
+
+    // Calculate the feed values for the delay lines.
+    f[0] = v - d[0];
+    f[1] = v - d[1];
+    f[2] = v - d[2];
+    f[3] = v - d[3];
+
+    // To increase reflection complexity (and help reduce coloration) the
+    // delay lines cyclicly refeed themselves (0 -> 1 -> 3 -> 2 -> 0...).
+    DelayLineIn(&State->Early.Delay[0], State->Offset, f[2]);
+    DelayLineIn(&State->Early.Delay[1], State->Offset, f[0]);
+    DelayLineIn(&State->Early.Delay[2], State->Offset, f[3]);
+    DelayLineIn(&State->Early.Delay[3], State->Offset, f[1]);
+
+    // To decorrelate the output for stereo separation, the cyclical nature
+    // of the feed path is exploited.  The two outputs are obtained from the
+    // inner delay lines.
+    // Output is instant by using the inputs to them instead of taking the
+    // result of the two delay lines directly (f[0] and f[3] instead of d[1]
+    // and d[2]).
+    out[0] = State->Early.Gain * f[0];
+    out[1] = State->Early.Gain * f[3];
+}
+
+// Delay line output routine for late reverb.
+static __inline ALfloat LateDelayLineOut(ALverbState *State, ALuint index)
+{
+    return State->Late.Coeff[index] *
+           DelayLineOut(&State->Late.Delay[index],
+                        State->Offset - State->Late.Offset[index]);
+}
+
+// Low-pass filter input/output routine for late reverb.
+static __inline ALfloat LateLowPassInOut(ALverbState *State, ALuint index, ALfloat in)
+{
+    State->Late.LpSample[index] = in + ((State->Late.LpSample[index] - in) *
+                                        State->Late.LpCoeff[index]);
+    return State->Late.LpSample[index];
+}
+
+// Given an input sample, this function produces a decorrelated stereo output
+// for late reverb.
+static __inline ALvoid LateReverb(ALverbState *State, ALfloat in, ALfloat *out)
+{
+    ALfloat din, d[8], v, dv, f[8];
+
+    // Since the input will be sent directly to the output as in the early
+    // reflections function, it needs to take into account some immediate
+    // absorption.
+    in = LateLowPassInOut(State, 0, in);
+
+    // When diffusion is full, no input is directly passed to the variable-
+    // length delay lines (the last 4).
+    din = (1.0f - State->Late.Diffusion) * in;
+
+    // Obtain the decayed results of the fixed-length delay lines.
+    d[0] = LateDelayLineOut(State, 0);
+    d[1] = LateDelayLineOut(State, 1);
+    d[2] = LateDelayLineOut(State, 2);
+    d[3] = LateDelayLineOut(State, 3);
+    // Obtain the decayed and low-pass filtered results of the variable-
+    // length delay lines.
+    d[4] = LateLowPassInOut(State, 1, LateDelayLineOut(State, 4));
+    d[5] = LateLowPassInOut(State, 2, LateDelayLineOut(State, 5));
+    d[6] = LateLowPassInOut(State, 3, LateDelayLineOut(State, 6));
+    d[7] = LateLowPassInOut(State, 4, LateDelayLineOut(State, 7));
+
+    // The waveguide formula used in the early reflections function works
+    // great for high diffusion, but it is not obviously paramerized to allow
+    // a variable diffusion.  With only limited time and resources, what
+    // follows is the best variation of that formula I could come up with.
+    // First, there are 8 delay lines used.  The first 4 are fixed-length and
+    // generate the highest density of the diffuse response.  The last 4 are
+    // variable-length, and are used to smooth out the diffuse response.  The
+    // density effect parameter alters their length.  The inner two delay
+    // lines of each group have their signs reversed (more about this later).
+    v = (d[0] - d[1] - d[2] + d[3] +
+         d[4] - d[5] - d[6] + d[7]) * 0.25f;
+    // Diffusion is applied as a reduction of the junction pressure for all
+    // branches.  This presents two problems.  When the diffusion factor (0
+    // to 1) reaches 0.5, the average feed value is reduced (the junction
+    // becomes lossy).  Thus, at 0.5 the signal decays almost twice as fast
+    // as it should.  The second problem is the introduction of some
+    // resonant frequencies (coloration).  The reversed signs above are used
+    // to help combat some of the coloration by adding variations along the
+    // feed cycle.
+    v *= State->Late.Diffusion;
+    // Load the junction with the input.  To reduce the noticeable echo of
+    // the longer delay lines (the variable-length ones) the input is loaded
+    // with the inverse of the effect diffusion.  So at full diffusion, the
+    // input is not applied to the last 4 delay lines.  Input signs reversed
+    // to balance the equation.
+    dv = v + din;
+    v += in;
+
+    // As with the reversed signs above, to balance the equation the signs
+    // need to be reversed here, too.
+    f[0] = d[0] - v;
+    f[1] = d[1] + v;
+    f[2] = d[2] + v;
+    f[3] = d[3] - v;
+    f[4] = d[4] - dv;
+    f[5] = d[5] + dv;
+    f[6] = d[6] + dv;
+    f[7] = d[7] - dv;
+
+    // Feed the fixed-length delay lines with their own cycle (0 -> 1 -> 3 ->
+    // 2 -> 0...).
+    DelayLineIn(&State->Late.Delay[0], State->Offset, f[2]);
+    DelayLineIn(&State->Late.Delay[1], State->Offset, f[0]);
+    DelayLineIn(&State->Late.Delay[2], State->Offset, f[3]);
+    DelayLineIn(&State->Late.Delay[3], State->Offset, f[1]);
+    // Feed the variable-length delay lines with their cycle (4 -> 6 -> 7 ->
+    // 5 -> 4...).
+    DelayLineIn(&State->Late.Delay[4], State->Offset, f[5]);
+    DelayLineIn(&State->Late.Delay[5], State->Offset, f[7]);
+    DelayLineIn(&State->Late.Delay[6], State->Offset, f[4]);
+    DelayLineIn(&State->Late.Delay[7], State->Offset, f[6]);
+
+    // Output is derived from the values fed to the inner two variable-length
+    // delay lines (5 and 6).
+    out[0] = State->Late.Gain * f[7];
+    out[1] = State->Late.Gain * f[4];
+}
+
+// This creates the reverb state.  It should be called only when the reverb
+// effect is loaded into a slot that doesn't already have a reverb effect.
+ALverbState *VerbCreate(ALCcontext *Context)
+{
+    ALverbState *State = NULL;
+    ALuint length[13], totalLength, index;
+
+    State = malloc(sizeof(ALverbState));
+    if(!State)
+        return NULL;
+
+    // All line lengths are powers of 2, calculated from the line timings and
+    // the addition of an extra sample (for safety).
+    length[0] = NextPowerOf2((ALuint)(MASTER_LINE_LENGTH*Context->Frequency) + 1);
+    totalLength = length[0];
+    for(index = 0;index < 4;index++)
+    {
+        length[1+index] = NextPowerOf2((ALuint)(EARLY_LINE_LENGTH[index]*Context->Frequency) + 1);
+        totalLength += length[1+index];
+    }
+    for(index = 0;index < 4;index++)
+    {
+        length[5+index] = NextPowerOf2((ALuint)(LATE_LINE_LENGTH[index]*Context->Frequency) + 1);
+        totalLength += length[5+index];
+    }
+    for(index = 4;index < 8;index++)
+    {
+        length[5+index] = NextPowerOf2((ALuint)(LATE_LINE_LENGTH[index]*(1.0f + LATE_LINE_MULTIPLIER)*Context->Frequency) + 1);
+        totalLength += length[5+index];
+    }
+
+    // They all share a single sample buffer.
+    State->SampleBuffer = malloc(totalLength * sizeof(ALfloat));
+    if(!State->SampleBuffer)
+    {
+        free(State);
+        return NULL;
+    }
+    for(index = 0; index < totalLength;index++)
+        State->SampleBuffer[index] = 0.0f;
+
+    // Each one has its mask and start address calculated one time.
+    State->Gain = 0.0f;
+    State->Delay.Mask = length[0] - 1;
+    State->Delay.Line = &State->SampleBuffer[0];
+    totalLength = length[0];
+
+    State->Tap[0] = 0;
+    State->Tap[1] = 0;
+
+    State->Early.Gain = 0.0f;
+    // All fixed-length delay lines have their read-write offsets calculated
+    // one time.
+    for(index = 0;index < 4;index++)
+    {
+        State->Early.Coeff[index] = 0.0f;
+        State->Early.Delay[index].Mask = length[1 + index] - 1;
+        State->Early.Delay[index].Line = &State->SampleBuffer[totalLength];
+        totalLength += length[1 + index];
+
+        State->Early.Offset[index] = (ALuint)(EARLY_LINE_LENGTH[index] * Context->Frequency);
+    }
+
+    State->Late.Gain = 0.0f;
+    State->Late.Diffusion = 0.0f;
+    for(index = 0;index < 8;index++)
+    {
+        State->Late.Coeff[index] = 0.0f;
+        State->Late.Delay[index].Mask = length[5 + index] - 1;
+        State->Late.Delay[index].Line = &State->SampleBuffer[totalLength];
+        totalLength += length[5 + index];
+
+        State->Late.Offset[index] = 0;
+        if(index < 4)
+        {
+            State->Late.Offset[index] = (ALuint)(LATE_LINE_LENGTH[index] * Context->Frequency);
+            State->Late.LpCoeff[index] = 0.0f;
+            State->Late.LpSample[index] = 0.0f;
+        }
+        else if(index == 4)
+        {
+            State->Late.LpCoeff[index] = 0.0f;
+            State->Late.LpSample[index] = 0.0f;
+        }
+    }
+
+    State->Offset = 0;
+    return State;
+}
+
+// This destroys the reverb state.  It should be called only when the effect
+// slot has a different (or no) effect loaded over the reverb effect.
+ALvoid VerbDestroy(ALverbState *State)
+{
+    if(State)
+    {
+        free(State->SampleBuffer);
+        State->SampleBuffer = NULL;
+        free(State);
+    }
+}
+
+// This updates the reverb state.  This is called any time the reverb effect
+// is loaded into a slot.
+ALvoid VerbUpdate(ALCcontext *Context, ALeffectslot *Slot, ALeffect *Effect)
+{
+    ALverbState *State = Slot->ReverbState;
+    ALuint index, index2;
+    ALfloat length, lpcoeff, cw, g;
+    ALfloat hfRatio = Effect->Reverb.DecayHFRatio;
+
+    // Calculate the master gain (from the slot and master reverb gain).
+    State->Gain = Slot->Gain * Effect->Reverb.Gain;
+
+    // Calculate the initial delay taps.
+    length = Effect->Reverb.ReflectionsDelay;
+    State->Tap[0] = (ALuint)(length * Context->Frequency);
+    length += Effect->Reverb.LateReverbDelay;
+    State->Tap[1] = (ALuint)(length * Context->Frequency);
+
+    // Calculate the early reflections gain.  Right now this uses a gain of
+    // 0.75 to compensate for the increase in density.  It should probably
+    // use a power (RMS) based measurement from the resulting distribution of
+    // early delay lines.
+    State->Early.Gain = Effect->Reverb.ReflectionsGain * 0.75f;
+
+    // Calculate the gain (coefficient) for each early delay line.
+    for(index = 0;index < 4;index++)
+        State->Early.Coeff[index] = pow(10.0f, EARLY_LINE_LENGTH[index] /
+                                               Effect->Reverb.LateReverbDelay *
+                                               -60.0f / 20.0f);
+
+    // Calculate the late reverb gain, adjusted by density, diffusion, and
+    // decay time.  To be accurate, the adjustments should probably use power
+    // measurements for each contribution, but they are not too bad as they
+    // are.
+    State->Late.Gain = Effect->Reverb.LateReverbGain *
+                       (0.45f + (0.55f * Effect->Reverb.Density)) *
+                       (1.0f - (0.25f * Effect->Reverb.Diffusion)) *
+                       (1.0f - (0.025f * Effect->Reverb.DecayTime));
+    State->Late.Diffusion = Effect->Reverb.Diffusion;
+
+    // The EFX specification does not make it clear whether the air
+    // absorption parameter should always take effect.  Both Generic Software
+    // and Generic Hardware only apply it when HF limit is flagged, so that's
+    // what is done here.
+    // If the HF limit parameter is flagged, calculate an appropriate limit
+    // based on the air absorption parameter.
+    if(Effect->Reverb.DecayHFLimit)
+    {
+        ALfloat limitRatio;
+
+        // The following is my best guess at how to limit the HF ratio by the
+        // air absorption parameter.
+        // For each of the last 4 delays, 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 / (log10(Effect->Reverb.AirAbsorptionGainHF) *
+                             SPEEDOFSOUNDMETRESPERSEC*Effect->Reverb.DecayTime/
+                             -60.0f * 20.0f);
+        // 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.
+        hfRatio = __min(hfRatio, limitRatio);
+    }
+
+    cw = cos(2.0f*3.141592654f * LOWPASSFREQCUTOFF / Context->Frequency);
+
+    for(index = 0;index < 8;index++)
+    {
+        // Calculate the length (in seconds) of each delay line.
+        length = LATE_LINE_LENGTH[index];
+        if(index >= 4)
+        {
+            index2 = index - 3;
+
+            length *= 1.0f + (Effect->Reverb.Density * LATE_LINE_MULTIPLIER);
+
+            // Calculate the delay offset for the variable-length delay
+            // lines.
+            State->Late.Offset[index] = (ALuint)(length * Context->Frequency);
+
+            // Calculate the decay equation for each low-pass filter.
+            g = pow(10.0f, length / (Effect->Reverb.DecayTime * hfRatio) *
+                           -60.0f / 20.0f);
+            g  = __max(g, 0.1f);
+            g *= g;
+            // Calculate the gain (coefficient) for each low-pass filter.
+            lpcoeff = 0.0f;
+            if(g < 0.9999f) // 1-epsilon
+                lpcoeff = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
+
+            // Very low decay times will produce minimal output, so apply an
+            // upper bound to the coefficient.
+            State->Late.LpCoeff[index2] = __min(lpcoeff, 0.98f);
+        }
+
+        // Calculate the gain (coefficient) for each line.
+        State->Late.Coeff[index] = pow(10.0f, length / Effect->Reverb.DecayTime *
+                                              -60.0f / 20.0f);
+    }
+
+    // This just calculates the coefficient for the late reverb input low-
+    // pass filter.  It is calculated based the average (hence -30 instead
+    // of -60) length of the inner two variable-length delay lines.
+    length = LATE_LINE_LENGTH[5] * (1.0f + Effect->Reverb.Density * LATE_LINE_MULTIPLIER) +
+             LATE_LINE_LENGTH[6] * (1.0f + Effect->Reverb.Density * LATE_LINE_MULTIPLIER);
+
+    g = pow(10.0f, length / (Effect->Reverb.DecayTime * hfRatio) * -30.0f / 20.0f);
+    g  = __max(g, 0.1f);
+    g *= g;
+
+    lpcoeff = 0.0f;
+    if(g < 0.9999f) // 1-epsilon
+        lpcoeff = (1 - g*cw - aluSqrt(2*g*(1-cw) - g*g*(1 - cw*cw))) / (1 - g);
+
+    State->Late.LpCoeff[0] = __min(lpcoeff, 0.98f);
+}
+
+// This processes the reverb state, given the input samples and an output
+// buffer.
+ALvoid VerbProcess(ALverbState *State, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[OUTPUTCHANNELS])
+{
+    ALuint index;
+    ALfloat in, early[2], late[2], out[2];
+
+    for(index = 0;index < SamplesToDo;index++)
+    {
+        // Feed the initial delay line.
+        DelayLineIn(&State->Delay, State->Offset, SamplesIn[index]);
+
+        // Calculate the early reflection from the first delay tap.
+        in = DelayLineOut(&State->Delay, State->Offset - State->Tap[0]);
+        EarlyReflection(State, in, early);
+
+        // Calculate the late reverb from the second delay tap.
+        in = DelayLineOut(&State->Delay, State->Offset - State->Tap[1]);
+        LateReverb(State, in, late);
+
+        // Mix early reflections and late reverb.
+        out[0] = State->Gain * (early[0] + late[0]);
+        out[1] = State->Gain * (early[1] + late[1]);
+
+        // Step all delays forward one sample.
+        State->Offset++;
+
+        // Output the results.
+        SamplesOut[index][FRONT_LEFT]  += out[0];
+        SamplesOut[index][FRONT_RIGHT] += out[1];
+        SamplesOut[index][SIDE_LEFT]   += out[0];
+        SamplesOut[index][SIDE_RIGHT]  += out[1];
+        SamplesOut[index][BACK_LEFT]   += out[0];
+        SamplesOut[index][BACK_RIGHT]  += out[1];
+    }
+}