2 * Reverb for the OpenAL cross platform audio library
3 * Copyright (C) 2008-2009 by Christopher Fitzgerald.
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
29 #include "alAuxEffectSlot.h"
34 typedef struct DelayLine
36 // The delay lines use sample lengths that are powers of 2 to allow
37 // bitmasking instead of modulus wrapping.
44 // All delay lines are allocated as a single buffer to reduce memory
45 // fragmentation and management code.
46 ALfloat
*SampleBuffer
;
47 // Master effect gain.
49 // Initial effect delay and decorrelation.
51 // The tap points for the initial delay. First tap goes to early
52 // reflections, the last four decorrelate to late reverb.
55 // Gain for early reflections.
57 // Early reflections are done with 4 delay lines.
63 // Gain for late reverb.
65 // Attenuation to compensate for modal density and decay rate.
67 // The feed-back and feed-forward all-pass coefficient.
69 // Mixing matrix coefficient.
71 // Late reverb has 4 parallel all-pass filters.
75 // In addition to 4 cyclical delay lines.
79 // The cyclical delay lines are low-pass filtered.
80 ALfloat LpCoeff
[4][2];
83 // The current read offset for all delay lines.
87 // All delay line lengths are specified in seconds.
89 // The lengths of the early delay lines.
90 static const ALfloat EARLY_LINE_LENGTH
[4] =
92 0.0015f
, 0.0045f
, 0.0135f
, 0.0405f
95 // The lengths of the late all-pass delay lines.
96 static const ALfloat ALLPASS_LINE_LENGTH
[4] =
98 0.0151f
, 0.0167f
, 0.0183f
, 0.0200f
,
101 // The lengths of the late cyclical delay lines.
102 static const ALfloat LATE_LINE_LENGTH
[4] =
104 0.0211f
, 0.0311f
, 0.0461f
, 0.0680f
107 // The late cyclical delay lines have a variable length dependent on the
108 // effect's density parameter (inverted for some reason) and this multiplier.
109 static const ALfloat LATE_LINE_MULTIPLIER
= 4.0f
;
111 // Input into the late reverb is decorrelated between four channels. Their
112 // timings are dependent on a fraction and multiplier. See VerbUpdate() for
113 // the calculations involved.
114 static const ALfloat DECO_FRACTION
= 1.0f
/ 32.0f
;
115 static const ALfloat DECO_MULTIPLIER
= 2.0f
;
117 // The maximum length of initial delay for the master delay line (a sum of
118 // the maximum early reflection and late reverb delays).
119 static const ALfloat MASTER_LINE_LENGTH
= 0.3f
+ 0.1f
;
121 // Find the next power of 2. Actually, this will return the input value if
122 // it is already a power of 2.
123 static ALuint
NextPowerOf2(ALuint value
)
139 // Basic delay line input/output routines.
140 static __inline ALfloat
DelayLineOut(DelayLine
*Delay
, ALuint offset
)
142 return Delay
->Line
[offset
&Delay
->Mask
];
145 static __inline ALvoid
DelayLineIn(DelayLine
*Delay
, ALuint offset
, ALfloat in
)
147 Delay
->Line
[offset
&Delay
->Mask
] = in
;
150 // Delay line output routine for early reflections.
151 static __inline ALfloat
EarlyDelayLineOut(ALverbState
*State
, ALuint index
)
153 return State
->Early
.Coeff
[index
] *
154 DelayLineOut(&State
->Early
.Delay
[index
],
155 State
->Offset
- State
->Early
.Offset
[index
]);
158 // Given an input sample, this function produces stereo output for early
160 static __inline ALvoid
EarlyReflection(ALverbState
*State
, ALfloat in
, ALfloat
*out
)
162 ALfloat d
[4], v
, f
[4];
164 // Obtain the decayed results of each early delay line.
165 d
[0] = EarlyDelayLineOut(State
, 0);
166 d
[1] = EarlyDelayLineOut(State
, 1);
167 d
[2] = EarlyDelayLineOut(State
, 2);
168 d
[3] = EarlyDelayLineOut(State
, 3);
170 /* The following uses a lossless scattering junction from waveguide
171 * theory. It actually amounts to a householder mixing matrix, which
172 * will produce a maximally diffuse response, and means this can probably
173 * be considered a simple feedback delay network (FDN).
181 v
= (d
[0] + d
[1] + d
[2] + d
[3]) * 0.5f
;
182 // The junction is loaded with the input here.
185 // Calculate the feed values for the delay lines.
191 // Refeed the delay lines.
192 DelayLineIn(&State
->Early
.Delay
[0], State
->Offset
, f
[0]);
193 DelayLineIn(&State
->Early
.Delay
[1], State
->Offset
, f
[1]);
194 DelayLineIn(&State
->Early
.Delay
[2], State
->Offset
, f
[2]);
195 DelayLineIn(&State
->Early
.Delay
[3], State
->Offset
, f
[3]);
197 // To decorrelate the output for stereo separation, the two outputs are
198 // obtained from the inner delay lines.
199 // Output is instant by using the inputs to them instead of taking the
200 // result of the two delay lines directly (f[0] and f[3] instead of d[1]
202 out
[0] = State
->Early
.Gain
* f
[0];
203 out
[1] = State
->Early
.Gain
* f
[3];
206 // All-pass input/output routine for late reverb.
207 static __inline ALfloat
LateAllPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
211 out
= State
->Late
.ApCoeff
[index
] *
212 DelayLineOut(&State
->Late
.ApDelay
[index
],
213 State
->Offset
- State
->Late
.ApOffset
[index
]);
214 out
-= (State
->Late
.ApFeedCoeff
* in
);
215 DelayLineIn(&State
->Late
.ApDelay
[index
], State
->Offset
,
216 (State
->Late
.ApFeedCoeff
* out
) + in
);
220 // Delay line output routine for late reverb.
221 static __inline ALfloat
LateDelayLineOut(ALverbState
*State
, ALuint index
)
223 return State
->Late
.Coeff
[index
] *
224 DelayLineOut(&State
->Late
.Delay
[index
],
225 State
->Offset
- State
->Late
.Offset
[index
]);
228 // Low-pass filter input/output routine for late reverb.
229 static __inline ALfloat
LateLowPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
231 State
->Late
.LpSample
[index
] = (State
->Late
.LpCoeff
[index
][0] * in
) +
232 (State
->Late
.LpCoeff
[index
][1] * State
->Late
.LpSample
[index
]);
233 return State
->Late
.LpSample
[index
];
236 // Given four decorrelated input samples, this function produces stereo
237 // output for late reverb.
238 static __inline ALvoid
LateReverb(ALverbState
*State
, ALfloat
*in
, ALfloat
*out
)
242 // Obtain the decayed results of the cyclical delay lines, and add the
243 // corresponding input channels attenuated by density. Then pass the
244 // results through the low-pass filters.
245 d
[0] = LateLowPassInOut(State
, 0, (State
->Late
.DensityGain
* in
[0]) +
246 LateDelayLineOut(State
, 0));
247 d
[1] = LateLowPassInOut(State
, 1, (State
->Late
.DensityGain
* in
[1]) +
248 LateDelayLineOut(State
, 1));
249 d
[2] = LateLowPassInOut(State
, 2, (State
->Late
.DensityGain
* in
[2]) +
250 LateDelayLineOut(State
, 2));
251 d
[3] = LateLowPassInOut(State
, 3, (State
->Late
.DensityGain
* in
[3]) +
252 LateDelayLineOut(State
, 3));
254 // To help increase diffusion, run each line through an all-pass filter.
255 // The order of the all-pass filters is selected so that the shortest
256 // all-pass filter will feed the shortest delay line.
257 d
[0] = LateAllPassInOut(State
, 1, d
[0]);
258 d
[1] = LateAllPassInOut(State
, 3, d
[1]);
259 d
[2] = LateAllPassInOut(State
, 0, d
[2]);
260 d
[3] = LateAllPassInOut(State
, 2, d
[3]);
262 /* Late reverb is done with a modified feedback delay network (FDN)
263 * topology. Four input lines are each fed through their own all-pass
264 * filter and then into the mixing matrix. The four outputs of the
265 * mixing matrix are then cycled back to the inputs. Each output feeds
266 * a different input to form a circlular feed cycle.
268 * The mixing matrix used is a 4D skew-symmetric rotation matrix derived
269 * using a single unitary rotational parameter:
271 * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2
276 * The rotation is constructed from the effect's diffusion parameter,
277 * yielding: 1 = x^2 + 3 y^2; where a, b, and c are the coefficient y
278 * with differing signs, and d is the coefficient x. The matrix is thus:
280 * [ x, y, -y, y ] x = 1 - (0.5 diffusion^3)
281 * [ -y, x, y, y ] y = sqrt((1 - x^2) / 3)
285 * To reduce the number of multiplies, the x coefficient is applied with
286 * the cyclical delay line coefficients. Thus only the y coefficient is
287 * applied when mixing, and is modified to be: y / x.
289 f
[0] = d
[0] + (State
->Late
.MixCoeff
* ( d
[1] - d
[2] + d
[3]));
290 f
[1] = d
[1] + (State
->Late
.MixCoeff
* (-d
[0] + d
[2] + d
[3]));
291 f
[2] = d
[2] + (State
->Late
.MixCoeff
* ( d
[0] - d
[1] + d
[3]));
292 f
[3] = d
[3] + (State
->Late
.MixCoeff
* (-d
[0] - d
[1] - d
[2]));
294 // Output is tapped at the input to the shortest two cyclical delay
295 // lines, attenuated by the late reverb gain (which is attenuated by the
296 // mixing coefficient x).
297 out
[0] = State
->Late
.Gain
* f
[0];
298 out
[1] = State
->Late
.Gain
* f
[1];
300 // The delay lines are fed circularly in the order:
301 // 0 -> 1 -> 3 -> 2 -> 0 ...
302 DelayLineIn(&State
->Late
.Delay
[0], State
->Offset
, f
[2]);
303 DelayLineIn(&State
->Late
.Delay
[1], State
->Offset
, f
[0]);
304 DelayLineIn(&State
->Late
.Delay
[2], State
->Offset
, f
[3]);
305 DelayLineIn(&State
->Late
.Delay
[3], State
->Offset
, f
[1]);
308 // This creates the reverb state. It should be called only when the reverb
309 // effect is loaded into a slot that doesn't already have a reverb effect.
310 ALverbState
*VerbCreate(ALCcontext
*Context
)
312 ALverbState
*State
= NULL
;
313 ALuint samples
, length
[13], totalLength
, index
;
315 State
= malloc(sizeof(ALverbState
));
319 // All line lengths are powers of 2, calculated from their lengths, with
320 // an additional sample in case of rounding errors.
322 // See VerbUpdate() for an explanation of the additional calculation
323 // added to the master line length.
325 ((MASTER_LINE_LENGTH
+
326 (LATE_LINE_LENGTH
[0] * (1.0f
+ LATE_LINE_MULTIPLIER
) *
327 (DECO_FRACTION
* ((DECO_MULTIPLIER
* DECO_MULTIPLIER
*
328 DECO_MULTIPLIER
) - 1.0f
)))) *
329 Context
->Frequency
) + 1;
330 length
[0] = NextPowerOf2(samples
);
331 totalLength
= length
[0];
332 for(index
= 0;index
< 4;index
++)
334 samples
= (ALuint
)(EARLY_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
335 length
[1 + index
] = NextPowerOf2(samples
);
336 totalLength
+= length
[1 + index
];
338 for(index
= 0;index
< 4;index
++)
340 samples
= (ALuint
)(ALLPASS_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
341 length
[5 + index
] = NextPowerOf2(samples
);
342 totalLength
+= length
[5 + index
];
344 for(index
= 0;index
< 4;index
++)
346 samples
= (ALuint
)(LATE_LINE_LENGTH
[index
] *
347 (1.0f
+ LATE_LINE_MULTIPLIER
) * Context
->Frequency
) + 1;
348 length
[9 + index
] = NextPowerOf2(samples
);
349 totalLength
+= length
[9 + index
];
352 // All lines share a single sample buffer.
353 State
->SampleBuffer
= malloc(totalLength
* sizeof(ALfloat
));
354 if(!State
->SampleBuffer
)
359 for(index
= 0; index
< totalLength
;index
++)
360 State
->SampleBuffer
[index
] = 0.0f
;
362 // Each one has its mask and start address calculated one time.
364 State
->Delay
.Mask
= length
[0] - 1;
365 State
->Delay
.Line
= &State
->SampleBuffer
[0];
366 totalLength
= length
[0];
374 State
->Early
.Gain
= 0.0f
;
375 for(index
= 0;index
< 4;index
++)
377 State
->Early
.Coeff
[index
] = 0.0f
;
378 State
->Early
.Delay
[index
].Mask
= length
[1 + index
] - 1;
379 State
->Early
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
380 totalLength
+= length
[1 + index
];
382 // The early delay lines have their read offsets calculated once.
383 State
->Early
.Offset
[index
] = (ALuint
)(EARLY_LINE_LENGTH
[index
] *
387 State
->Late
.Gain
= 0.0f
;
388 State
->Late
.DensityGain
= 0.0f
;
389 State
->Late
.ApFeedCoeff
= 0.0f
;
390 State
->Late
.MixCoeff
= 0.0f
;
392 for(index
= 0;index
< 4;index
++)
394 State
->Late
.ApCoeff
[index
] = 0.0f
;
395 State
->Late
.ApDelay
[index
].Mask
= length
[5 + index
] - 1;
396 State
->Late
.ApDelay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
397 totalLength
+= length
[5 + index
];
399 // The late all-pass lines have their read offsets calculated once.
400 State
->Late
.ApOffset
[index
] = (ALuint
)(ALLPASS_LINE_LENGTH
[index
] *
404 for(index
= 0;index
< 4;index
++)
406 State
->Late
.Coeff
[index
] = 0.0f
;
407 State
->Late
.Delay
[index
].Mask
= length
[9 + index
] - 1;
408 State
->Late
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
409 totalLength
+= length
[9 + index
];
411 State
->Late
.Offset
[index
] = 0;
413 State
->Late
.LpCoeff
[index
][0] = 0.0f
;
414 State
->Late
.LpCoeff
[index
][1] = 0.0f
;
415 State
->Late
.LpSample
[index
] = 0.0f
;
422 // This destroys the reverb state. It should be called only when the effect
423 // slot has a different (or no) effect loaded over the reverb effect.
424 ALvoid
VerbDestroy(ALverbState
*State
)
428 free(State
->SampleBuffer
);
429 State
->SampleBuffer
= NULL
;
434 // This updates the reverb state. This is called any time the reverb effect
435 // is loaded into a slot.
436 ALvoid
VerbUpdate(ALCcontext
*Context
, ALeffectslot
*Slot
, ALeffect
*Effect
)
438 ALverbState
*State
= Slot
->ReverbState
;
440 ALfloat length
, mixCoeff
, cw
, g
, lpCoeff
;
441 ALfloat hfRatio
= Effect
->Reverb
.DecayHFRatio
;
443 // Calculate the master gain (from the slot and master effect gain).
444 State
->Gain
= Slot
->Gain
* Effect
->Reverb
.Gain
;
446 // Calculate the initial delay taps.
447 length
= Effect
->Reverb
.ReflectionsDelay
;
448 State
->Tap
[0] = (ALuint
)(length
* Context
->Frequency
);
450 length
+= Effect
->Reverb
.LateReverbDelay
;
452 /* The four inputs to the late reverb are decorrelated to smooth the
453 * initial reverb and reduce harsh echos. The timings are calculated as
454 * multiples of a fraction of the smallest cyclical delay time. This
455 * result is then adjusted so that the first tap occurs immediately (all
456 * taps are reduced by the shortest fraction).
458 * offset[index] = ((FRACTION MULTIPLIER^index) - 1) delay
460 for(index
= 0;index
< 4;index
++)
462 length
+= LATE_LINE_LENGTH
[0] *
463 (1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
)) *
464 (DECO_FRACTION
* (pow(DECO_MULTIPLIER
, (ALfloat
)index
) - 1.0f
));
465 State
->Tap
[1 + index
] = (ALuint
)(length
* Context
->Frequency
);
468 // Set the early reflections gain.
469 State
->Early
.Gain
= Effect
->Reverb
.ReflectionsGain
;
471 // Calculate the gain (coefficient) for each early delay line.
472 for(index
= 0;index
< 4;index
++)
473 State
->Early
.Coeff
[index
] = pow(10.0f
, EARLY_LINE_LENGTH
[index
] /
474 Effect
->Reverb
.LateReverbDelay
*
477 // Calculate the first mixing matrix coefficient (x).
478 mixCoeff
= 1.0f
- (0.5f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
));
480 // Set the late reverb gain. Since the output is tapped prior to the
481 // application of the delay line coefficients, this gain needs to be
482 // attenuated by the mix coefficient from above.
483 State
->Late
.Gain
= Effect
->Reverb
.LateReverbGain
* mixCoeff
;
485 /* To compensate for changes in modal density and decay time of the late
486 * reverb signal, the input is attenuated based on the maximal energy of
487 * the outgoing signal. This is calculated as the ratio between a
488 * reference value and the current approximation of energy for the output
491 * Reverb output matches exponential decay of the form Sum(a^n), where a
492 * is the attenuation coefficient, and n is the sample ranging from 0 to
493 * infinity. The signal energy can thus be approximated using the area
494 * under this curve, calculated as: 1 / (1 - a).
496 * The reference energy is calculated from a signal at the lowest (effect
497 * at 1.0) density with a decay time of one second.
499 * The coefficient is calculated as the average length of the cyclical
500 * delay lines. This produces a better result than calculating the gain
501 * for each line individually (most likely a side effect of diffusion).
503 * The final result is the square root of the ratio bound to a maximum
504 * value of 1 (no amplification) and attenuated by 1 / sqrt(2) to
505 * compensate for the four decorrelated inputs.
507 length
= (LATE_LINE_LENGTH
[0] + LATE_LINE_LENGTH
[1] +
508 LATE_LINE_LENGTH
[2] + LATE_LINE_LENGTH
[3]);
509 g
= length
* (1.0f
+ LATE_LINE_MULTIPLIER
) * 0.25f
;
510 g
= pow(10.0f
, g
* -60.0f
/ 20.0f
);
511 g
= 1.0f
/ (1.0f
- (g
*g
));
512 length
*= 1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
) * 0.25f
;
513 length
= pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
* -60.0f
/ 20.0f
);
514 length
= 1.0f
/ (1.0f
- (length
*length
));
515 State
->Late
.DensityGain
= 0.707106f
* __min(aluSqrt(g
/ length
), 1.0f
);
517 // Calculate the all-pass feed-back and feed-forward coefficient.
518 State
->Late
.ApFeedCoeff
= 0.6f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
);
520 // Calculate the mixing matrix coefficient (y / x).
521 g
= aluSqrt((1.0f
- (mixCoeff
* mixCoeff
)) / 3.0f
);
522 State
->Late
.MixCoeff
= g
/ mixCoeff
;
524 for(index
= 0;index
< 4;index
++)
526 // Calculate the gain (coefficient) for each all-pass line.
527 State
->Late
.ApCoeff
[index
] = pow(10.0f
, ALLPASS_LINE_LENGTH
[index
] /
528 Effect
->Reverb
.DecayTime
*
532 // If the HF limit parameter is flagged, calculate an appropriate limit
533 // based on the air absorption parameter.
534 if(Effect
->Reverb
.DecayHFLimit
&& Effect
->Reverb
.AirAbsorptionGainHF
< 1.0f
)
538 // For each of the cyclical delays, find the attenuation due to air
539 // absorption in dB (converting delay time to meters using the speed
540 // of sound). Then reversing the decay equation, solve for HF ratio.
541 // The delay length is cancelled out of the equation, so it can be
542 // calculated once for all lines.
543 limitRatio
= 1.0f
/ (log10(Effect
->Reverb
.AirAbsorptionGainHF
) *
544 SPEEDOFSOUNDMETRESPERSEC
*
545 Effect
->Reverb
.DecayTime
/ -60.0f
* 20.0f
);
546 // Need to limit the result to a minimum of 0.1, just like the HF
548 limitRatio
= __max(limitRatio
, 0.1f
);
550 // Using the limit calculated above, apply the upper bound to the
552 hfRatio
= __min(hfRatio
, limitRatio
);
555 // Calculate the filter frequency for low-pass or high-pass depending on
556 // whether the HF ratio is above 1.
557 cw
= 2.0f
* M_PI
* LOWPASSFREQCUTOFF
/ Context
->Frequency
;
562 for(index
= 0;index
< 4;index
++)
564 // Calculate the length (in seconds) of each cyclical delay line.
565 length
= LATE_LINE_LENGTH
[index
] * (1.0f
+ (Effect
->Reverb
.Density
*
566 LATE_LINE_MULTIPLIER
));
567 // Calculate the delay offset for the cyclical delay lines.
568 State
->Late
.Offset
[index
] = (ALuint
)(length
* Context
->Frequency
);
570 // Calculate the gain (coefficient) for each cyclical line.
571 State
->Late
.Coeff
[index
] = pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
*
574 // Calculate the decay equation for each low-pass filter.
575 g
= pow(10.0f
, length
/ (Effect
->Reverb
.DecayTime
* hfRatio
) *
578 g
= State
->Late
.Coeff
[index
] / g
;
580 g
= g
/ State
->Late
.Coeff
[index
];
584 // Calculate the gain (coefficient) for each low-pass filter.
586 if(g
< 0.9999f
) // 1-epsilon
587 lpCoeff
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) / (1 - g
);
589 // Very low decay times will produce minimal output, so apply an
590 // upper bound to the coefficient.
591 lpCoeff
= __min(lpCoeff
, 0.98f
);
593 // Calculate the filter coefficients for high-pass or low-pass
594 // dependent on HF ratio being above 1.
596 State
->Late
.LpCoeff
[index
][0] = 1.0f
+ lpCoeff
;
597 State
->Late
.LpCoeff
[index
][1] = -lpCoeff
;
599 State
->Late
.LpCoeff
[index
][0] = 1.0f
- lpCoeff
;
600 State
->Late
.LpCoeff
[index
][1] = lpCoeff
;
603 // Attenuate the cyclical line coefficients by the mixing coefficient
605 State
->Late
.Coeff
[index
] *= mixCoeff
;
609 // This processes the reverb state, given the input samples and an output
611 ALvoid
VerbProcess(ALverbState
*State
, ALuint SamplesToDo
, const ALfloat
*SamplesIn
, ALfloat (*SamplesOut
)[OUTPUTCHANNELS
])
614 ALfloat in
[4], early
[2], late
[2], out
[2];
616 for(index
= 0;index
< SamplesToDo
;index
++)
618 // Feed the initial delay line.
619 DelayLineIn(&State
->Delay
, State
->Offset
, SamplesIn
[index
]);
621 // Calculate the early reflection from the first delay tap.
622 in
[0] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[0]);
623 EarlyReflection(State
, in
[0], early
);
625 // Calculate the late reverb from the last four delay taps.
626 in
[0] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[1]);
627 in
[1] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[2]);
628 in
[2] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[3]);
629 in
[3] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[4]);
630 LateReverb(State
, in
, late
);
632 // Mix early reflections and late reverb.
633 out
[0] = State
->Gain
* (early
[0] + late
[0]);
634 out
[1] = State
->Gain
* (early
[1] + late
[1]);
636 // Step all delays forward one sample.
639 // Output the results.
640 SamplesOut
[index
][FRONT_LEFT
] += out
[0];
641 SamplesOut
[index
][FRONT_RIGHT
] += out
[1];
642 SamplesOut
[index
][SIDE_LEFT
] += out
[0];
643 SamplesOut
[index
][SIDE_RIGHT
] += out
[1];
644 SamplesOut
[index
][BACK_LEFT
] += out
[0];
645 SamplesOut
[index
][BACK_RIGHT
] += out
[1];