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 #define aluSqrt(x) ((ALfloat)sqrtf((float)(x)))
36 #define aluSqrt(x) ((ALfloat)sqrt((double)(x)))
40 #if defined(max) && !defined(__max)
43 #if defined(min) && !defined(__min)
47 typedef struct DelayLine
49 // The delay lines use sample lengths that are powers of 2 to allow
50 // bitmasking instead of modulus wrapping.
57 // All delay lines are allocated as a single buffer to reduce memory
58 // fragmentation and management code.
59 ALfloat
*SampleBuffer
;
60 // Master effect gain.
62 // Initial effect delay and decorrelation.
64 // The tap points for the initial delay. First tap goes to early
65 // reflections, the last four decorrelate to late reverb.
68 // Gain for early reflections.
70 // Early reflections are done with 4 delay lines.
76 // Gain for late reverb.
78 // Attenuation to compensate for modal density and decay rate.
80 // The feed-back and feed-forward all-pass coefficient.
82 // Mixing matrix coefficient.
84 // Late reverb has 4 parallel all-pass filters.
88 // In addition to 4 cyclical delay lines.
92 // The cyclical delay lines are low-pass filtered.
93 ALfloat LpCoeff
[4][2];
96 // The current read offset for all delay lines.
100 // All delay line lengths are specified in seconds.
102 // The lengths of the early delay lines.
103 static const ALfloat EARLY_LINE_LENGTH
[4] =
105 0.0015f
, 0.0045f
, 0.0135f
, 0.0405f
108 // The lengths of the late all-pass delay lines.
109 static const ALfloat ALLPASS_LINE_LENGTH
[4] =
111 0.0151f
, 0.0167f
, 0.0183f
, 0.0200f
,
114 // The lengths of the late cyclical delay lines.
115 static const ALfloat LATE_LINE_LENGTH
[4] =
117 0.0211f
, 0.0311f
, 0.0461f
, 0.0680f
120 // The late cyclical delay lines have a variable length dependent on the
121 // effect's density parameter (inverted for some reason) and this multiplier.
122 static const ALfloat LATE_LINE_MULTIPLIER
= 4.0f
;
124 // Input into the late reverb is decorrelated between four channels. Their
125 // timings are dependent on a fraction and multiplier. See VerbUpdate() for
126 // the calculations involved.
127 static const ALfloat DECO_FRACTION
= 1.0f
/ 32.0f
;
128 static const ALfloat DECO_MULTIPLIER
= 2.0f
;
130 // The maximum length of initial delay for the master delay line (a sum of
131 // the maximum early reflection and late reverb delays).
132 static const ALfloat MASTER_LINE_LENGTH
= 0.3f
+ 0.1f
;
134 // Find the next power of 2. Actually, this will return the input value if
135 // it is already a power of 2.
136 static ALuint
NextPowerOf2(ALuint value
)
152 // Basic delay line input/output routines.
153 static __inline ALfloat
DelayLineOut(DelayLine
*Delay
, ALuint offset
)
155 return Delay
->Line
[offset
&Delay
->Mask
];
158 static __inline ALvoid
DelayLineIn(DelayLine
*Delay
, ALuint offset
, ALfloat in
)
160 Delay
->Line
[offset
&Delay
->Mask
] = in
;
163 // Delay line output routine for early reflections.
164 static __inline ALfloat
EarlyDelayLineOut(ALverbState
*State
, ALuint index
)
166 return State
->Early
.Coeff
[index
] *
167 DelayLineOut(&State
->Early
.Delay
[index
],
168 State
->Offset
- State
->Early
.Offset
[index
]);
171 // Given an input sample, this function produces stereo output for early
173 static __inline ALvoid
EarlyReflection(ALverbState
*State
, ALfloat in
, ALfloat
*out
)
175 ALfloat d
[4], v
, f
[4];
177 // Obtain the decayed results of each early delay line.
178 d
[0] = EarlyDelayLineOut(State
, 0);
179 d
[1] = EarlyDelayLineOut(State
, 1);
180 d
[2] = EarlyDelayLineOut(State
, 2);
181 d
[3] = EarlyDelayLineOut(State
, 3);
183 /* The following uses a lossless scattering junction from waveguide
184 * theory. It actually amounts to a householder mixing matrix, which
185 * will produce a maximally diffuse response, and means this can probably
186 * be considered a simple feedback delay network (FDN).
194 v
= (d
[0] + d
[1] + d
[2] + d
[3]) * 0.5f
;
195 // The junction is loaded with the input here.
198 // Calculate the feed values for the delay lines.
204 // Refeed the delay lines.
205 DelayLineIn(&State
->Early
.Delay
[0], State
->Offset
, f
[0]);
206 DelayLineIn(&State
->Early
.Delay
[1], State
->Offset
, f
[1]);
207 DelayLineIn(&State
->Early
.Delay
[2], State
->Offset
, f
[2]);
208 DelayLineIn(&State
->Early
.Delay
[3], State
->Offset
, f
[3]);
210 // To decorrelate the output for stereo separation, the two outputs are
211 // obtained from the inner delay lines.
212 // Output is instant by using the inputs to them instead of taking the
213 // result of the two delay lines directly (f[0] and f[3] instead of d[1]
215 out
[0] = State
->Early
.Gain
* f
[0];
216 out
[1] = State
->Early
.Gain
* f
[3];
219 // All-pass input/output routine for late reverb.
220 static __inline ALfloat
LateAllPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
224 out
= State
->Late
.ApCoeff
[index
] *
225 DelayLineOut(&State
->Late
.ApDelay
[index
],
226 State
->Offset
- State
->Late
.ApOffset
[index
]);
227 out
-= (State
->Late
.ApFeedCoeff
* in
);
228 DelayLineIn(&State
->Late
.ApDelay
[index
], State
->Offset
,
229 (State
->Late
.ApFeedCoeff
* out
) + in
);
233 // Delay line output routine for late reverb.
234 static __inline ALfloat
LateDelayLineOut(ALverbState
*State
, ALuint index
)
236 return State
->Late
.Coeff
[index
] *
237 DelayLineOut(&State
->Late
.Delay
[index
],
238 State
->Offset
- State
->Late
.Offset
[index
]);
241 // Low-pass filter input/output routine for late reverb.
242 static __inline ALfloat
LateLowPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
244 State
->Late
.LpSample
[index
] = (State
->Late
.LpCoeff
[index
][0] * in
) +
245 (State
->Late
.LpCoeff
[index
][1] * State
->Late
.LpSample
[index
]);
246 return State
->Late
.LpSample
[index
];
249 // Given four decorrelated input samples, this function produces stereo
250 // output for late reverb.
251 static __inline ALvoid
LateReverb(ALverbState
*State
, ALfloat
*in
, ALfloat
*out
)
255 // Obtain the decayed results of the cyclical delay lines, and add the
256 // corresponding input channels attenuated by density. Then pass the
257 // results through the low-pass filters.
258 d
[0] = LateLowPassInOut(State
, 0, (State
->Late
.DensityGain
* in
[0]) +
259 LateDelayLineOut(State
, 0));
260 d
[1] = LateLowPassInOut(State
, 1, (State
->Late
.DensityGain
* in
[1]) +
261 LateDelayLineOut(State
, 1));
262 d
[2] = LateLowPassInOut(State
, 2, (State
->Late
.DensityGain
* in
[2]) +
263 LateDelayLineOut(State
, 2));
264 d
[3] = LateLowPassInOut(State
, 3, (State
->Late
.DensityGain
* in
[3]) +
265 LateDelayLineOut(State
, 3));
267 // To help increase diffusion, run each line through an all-pass filter.
268 // The order of the all-pass filters is selected so that the shortest
269 // all-pass filter will feed the shortest delay line.
270 d
[0] = LateAllPassInOut(State
, 1, d
[0]);
271 d
[1] = LateAllPassInOut(State
, 3, d
[1]);
272 d
[2] = LateAllPassInOut(State
, 0, d
[2]);
273 d
[3] = LateAllPassInOut(State
, 2, d
[3]);
275 /* Late reverb is done with a modified feedback delay network (FDN)
276 * topology. Four input lines are each fed through their own all-pass
277 * filter and then into the mixing matrix. The four outputs of the
278 * mixing matrix are then cycled back to the inputs. Each output feeds
279 * a different input to form a circlular feed cycle.
281 * The mixing matrix used is a 4D skew-symmetric rotation matrix derived
282 * using a single unitary rotational parameter:
284 * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2
289 * The rotation is constructed from the effect's diffusion parameter,
290 * yielding: 1 = x^2 + 3 y^2; where a, b, and c are the coefficient y
291 * with differing signs, and d is the coefficient x. The matrix is thus:
293 * [ x, y, -y, y ] x = 1 - (0.5 diffusion^3)
294 * [ -y, x, y, y ] y = sqrt((1 - x^2) / 3)
298 * To reduce the number of multiplies, the x coefficient is applied with
299 * the cyclical delay line coefficients. Thus only the y coefficient is
300 * applied when mixing, and is modified to be: y / x.
302 f
[0] = d
[0] + (State
->Late
.MixCoeff
* ( d
[1] - d
[2] + d
[3]));
303 f
[1] = d
[1] + (State
->Late
.MixCoeff
* (-d
[0] + d
[2] + d
[3]));
304 f
[2] = d
[2] + (State
->Late
.MixCoeff
* ( d
[0] - d
[1] + d
[3]));
305 f
[3] = d
[3] + (State
->Late
.MixCoeff
* (-d
[0] - d
[1] - d
[2]));
307 // Output is tapped at the input to the shortest two cyclical delay
308 // lines, attenuated by the late reverb gain (which is attenuated by the
309 // mixing coefficient x).
310 out
[0] = State
->Late
.Gain
* f
[0];
311 out
[1] = State
->Late
.Gain
* f
[1];
313 // The delay lines are fed circularly in the order:
314 // 0 -> 1 -> 3 -> 2 -> 0 ...
315 DelayLineIn(&State
->Late
.Delay
[0], State
->Offset
, f
[2]);
316 DelayLineIn(&State
->Late
.Delay
[1], State
->Offset
, f
[0]);
317 DelayLineIn(&State
->Late
.Delay
[2], State
->Offset
, f
[3]);
318 DelayLineIn(&State
->Late
.Delay
[3], State
->Offset
, f
[1]);
321 // This creates the reverb state. It should be called only when the reverb
322 // effect is loaded into a slot that doesn't already have a reverb effect.
323 ALverbState
*VerbCreate(ALCcontext
*Context
)
325 ALverbState
*State
= NULL
;
326 ALuint samples
, length
[13], totalLength
, index
;
328 State
= malloc(sizeof(ALverbState
));
332 // All line lengths are powers of 2, calculated from their lengths, with
333 // an additional sample in case of rounding errors.
335 // See VerbUpdate() for an explanation of the additional calculation
336 // added to the master line length.
338 ((MASTER_LINE_LENGTH
+
339 (LATE_LINE_LENGTH
[0] * (1.0f
+ LATE_LINE_MULTIPLIER
) *
340 (DECO_FRACTION
* ((DECO_MULTIPLIER
* DECO_MULTIPLIER
*
341 DECO_MULTIPLIER
) - 1.0f
)))) *
342 Context
->Frequency
) + 1;
343 length
[0] = NextPowerOf2(samples
);
344 totalLength
= length
[0];
345 for(index
= 0;index
< 4;index
++)
347 samples
= (ALuint
)(EARLY_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
348 length
[1 + index
] = NextPowerOf2(samples
);
349 totalLength
+= length
[1 + index
];
351 for(index
= 0;index
< 4;index
++)
353 samples
= (ALuint
)(ALLPASS_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
354 length
[5 + index
] = NextPowerOf2(samples
);
355 totalLength
+= length
[5 + index
];
357 for(index
= 0;index
< 4;index
++)
359 samples
= (ALuint
)(LATE_LINE_LENGTH
[index
] *
360 (1.0f
+ LATE_LINE_MULTIPLIER
) * Context
->Frequency
) + 1;
361 length
[9 + index
] = NextPowerOf2(samples
);
362 totalLength
+= length
[9 + index
];
365 // All lines share a single sample buffer.
366 State
->SampleBuffer
= malloc(totalLength
* sizeof(ALfloat
));
367 if(!State
->SampleBuffer
)
372 for(index
= 0; index
< totalLength
;index
++)
373 State
->SampleBuffer
[index
] = 0.0f
;
375 // Each one has its mask and start address calculated one time.
377 State
->Delay
.Mask
= length
[0] - 1;
378 State
->Delay
.Line
= &State
->SampleBuffer
[0];
379 totalLength
= length
[0];
387 State
->Early
.Gain
= 0.0f
;
388 for(index
= 0;index
< 4;index
++)
390 State
->Early
.Coeff
[index
] = 0.0f
;
391 State
->Early
.Delay
[index
].Mask
= length
[1 + index
] - 1;
392 State
->Early
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
393 totalLength
+= length
[1 + index
];
395 // The early delay lines have their read offsets calculated once.
396 State
->Early
.Offset
[index
] = (ALuint
)(EARLY_LINE_LENGTH
[index
] *
400 State
->Late
.Gain
= 0.0f
;
401 State
->Late
.DensityGain
= 0.0f
;
402 State
->Late
.ApFeedCoeff
= 0.0f
;
403 State
->Late
.MixCoeff
= 0.0f
;
405 for(index
= 0;index
< 4;index
++)
407 State
->Late
.ApCoeff
[index
] = 0.0f
;
408 State
->Late
.ApDelay
[index
].Mask
= length
[5 + index
] - 1;
409 State
->Late
.ApDelay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
410 totalLength
+= length
[5 + index
];
412 // The late all-pass lines have their read offsets calculated once.
413 State
->Late
.ApOffset
[index
] = (ALuint
)(ALLPASS_LINE_LENGTH
[index
] *
417 for(index
= 0;index
< 4;index
++)
419 State
->Late
.Coeff
[index
] = 0.0f
;
420 State
->Late
.Delay
[index
].Mask
= length
[9 + index
] - 1;
421 State
->Late
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
422 totalLength
+= length
[9 + index
];
424 State
->Late
.Offset
[index
] = 0;
426 State
->Late
.LpCoeff
[index
][0] = 0.0f
;
427 State
->Late
.LpCoeff
[index
][1] = 0.0f
;
428 State
->Late
.LpSample
[index
] = 0.0f
;
435 // This destroys the reverb state. It should be called only when the effect
436 // slot has a different (or no) effect loaded over the reverb effect.
437 ALvoid
VerbDestroy(ALverbState
*State
)
441 free(State
->SampleBuffer
);
442 State
->SampleBuffer
= NULL
;
447 // This updates the reverb state. This is called any time the reverb effect
448 // is loaded into a slot.
449 ALvoid
VerbUpdate(ALCcontext
*Context
, ALeffectslot
*Slot
, ALeffect
*Effect
)
451 ALverbState
*State
= Slot
->ReverbState
;
453 ALfloat length
, mixCoeff
, cw
, g
, lpCoeff
;
454 ALfloat hfRatio
= Effect
->Reverb
.DecayHFRatio
;
456 // Calculate the master gain (from the slot and master effect gain).
457 State
->Gain
= Slot
->Gain
* Effect
->Reverb
.Gain
;
459 // Calculate the initial delay taps.
460 length
= Effect
->Reverb
.ReflectionsDelay
;
461 State
->Tap
[0] = (ALuint
)(length
* Context
->Frequency
);
463 length
+= Effect
->Reverb
.LateReverbDelay
;
465 /* The four inputs to the late reverb are decorrelated to smooth the
466 * initial reverb and reduce harsh echos. The timings are calculated as
467 * multiples of a fraction of the smallest cyclical delay time. This
468 * result is then adjusted so that the first tap occurs immediately (all
469 * taps are reduced by the shortest fraction).
471 * offset[index] = ((FRACTION MULTIPLIER^index) - 1) delay
473 for(index
= 0;index
< 4;index
++)
475 length
+= LATE_LINE_LENGTH
[0] *
476 (1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
)) *
477 (DECO_FRACTION
* (pow(DECO_MULTIPLIER
, (ALfloat
)index
) - 1.0f
));
478 State
->Tap
[1 + index
] = (ALuint
)(length
* Context
->Frequency
);
481 // Set the early reflections gain.
482 State
->Early
.Gain
= Effect
->Reverb
.ReflectionsGain
;
484 // Calculate the gain (coefficient) for each early delay line.
485 for(index
= 0;index
< 4;index
++)
486 State
->Early
.Coeff
[index
] = pow(10.0f
, EARLY_LINE_LENGTH
[index
] /
487 Effect
->Reverb
.LateReverbDelay
*
490 // Calculate the first mixing matrix coefficient (x).
491 mixCoeff
= 1.0f
- (0.5f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
));
493 // Set the late reverb gain. Since the output is tapped prior to the
494 // application of the delay line coefficients, this gain needs to be
495 // attenuated by the mix coefficient from above.
496 State
->Late
.Gain
= Effect
->Reverb
.LateReverbGain
* mixCoeff
;
498 /* To compensate for changes in modal density and decay time of the late
499 * reverb signal, the input is attenuated based on the maximal energy of
500 * the outgoing signal. This is calculated as the ratio between a
501 * reference value and the current approximation of energy for the output
504 * Reverb output matches exponential decay of the form Sum(a^n), where a
505 * is the attenuation coefficient, and n is the sample ranging from 0 to
506 * infinity. The signal energy can thus be approximated using the area
507 * under this curve, calculated as: 1 / (1 - a).
509 * The reference energy is calculated from a signal at the lowest (effect
510 * at 1.0) density with a decay time of one second.
512 * The coefficient is calculated as the average length of the cyclical
513 * delay lines. This produces a better result than calculating the gain
514 * for each line individually (most likely a side effect of diffusion).
516 * The final result is the square root of the ratio bound to a maximum
517 * value of 1 (no amplification) and attenuated by 1 / sqrt(2) to
518 * compensate for the four decorrelated inputs.
520 length
= (LATE_LINE_LENGTH
[0] + LATE_LINE_LENGTH
[1] +
521 LATE_LINE_LENGTH
[2] + LATE_LINE_LENGTH
[3]);
522 g
= length
* (1.0f
+ LATE_LINE_MULTIPLIER
) * 0.25f
;
523 g
= pow(10.0f
, g
* -60.0f
/ 20.0f
);
524 g
= 1.0f
/ (1.0f
- g
);
525 length
*= 1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
) * 0.25f
;
526 length
= pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
* -60.0f
/ 20.0f
);
527 length
= 1.0f
/ (1.0f
- length
);
528 State
->Late
.DensityGain
= 0.707106f
* __min(aluSqrt(g
/ length
), 1.0f
);
530 // Calculate the all-pass feed-back and feed-forward coefficient.
531 State
->Late
.ApFeedCoeff
= 0.6f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
);
533 // Calculate the mixing matrix coefficient (y / x).
534 g
= aluSqrt((1.0f
- (mixCoeff
* mixCoeff
)) / 3.0f
);
535 State
->Late
.MixCoeff
= g
/ mixCoeff
;
537 for(index
= 0;index
< 4;index
++)
539 // Calculate the gain (coefficient) for each all-pass line.
540 State
->Late
.ApCoeff
[index
] = pow(10.0f
, ALLPASS_LINE_LENGTH
[index
] /
541 Effect
->Reverb
.DecayTime
*
545 // If the HF limit parameter is flagged, calculate an appropriate limit
546 // based on the air absorption parameter.
547 if(Effect
->Reverb
.DecayHFLimit
&& Effect
->Reverb
.AirAbsorptionGainHF
< 1.0f
)
551 // For each of the cyclical delays, find the attenuation due to air
552 // absorption in dB (converting delay time to meters using the speed
553 // of sound). Then reversing the decay equation, solve for HF ratio.
554 // The delay length is cancelled out of the equation, so it can be
555 // calculated once for all lines.
556 limitRatio
= 1.0f
/ (log10(Effect
->Reverb
.AirAbsorptionGainHF
) *
557 SPEEDOFSOUNDMETRESPERSEC
*
558 Effect
->Reverb
.DecayTime
/ -60.0f
* 20.0f
);
559 // Need to limit the result to a minimum of 0.1, just like the HF
561 limitRatio
= __max(limitRatio
, 0.1f
);
563 // Using the limit calculated above, apply the upper bound to the
565 hfRatio
= __min(hfRatio
, limitRatio
);
568 // Calculate the filter frequency for low-pass or high-pass depending on
569 // whether the HF ratio is above 1.
570 cw
= 2.0f
* M_PI
* LOWPASSFREQCUTOFF
/ Context
->Frequency
;
575 for(index
= 0;index
< 4;index
++)
577 // Calculate the length (in seconds) of each cyclical delay line.
578 length
= LATE_LINE_LENGTH
[index
] * (1.0f
+ (Effect
->Reverb
.Density
*
579 LATE_LINE_MULTIPLIER
));
580 // Calculate the delay offset for the cyclical delay lines.
581 State
->Late
.Offset
[index
] = (ALuint
)(length
* Context
->Frequency
);
583 // Calculate the gain (coefficient) for each cyclical line.
584 State
->Late
.Coeff
[index
] = pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
*
587 // Calculate the decay equation for each low-pass filter.
588 g
= pow(10.0f
, length
/ (Effect
->Reverb
.DecayTime
* hfRatio
) *
591 g
= State
->Late
.Coeff
[index
] / g
;
593 g
= g
/ State
->Late
.Coeff
[index
];
597 // Calculate the gain (coefficient) for each low-pass filter.
599 if(g
< 0.9999f
) // 1-epsilon
600 lpCoeff
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) / (1 - g
);
602 // Very low decay times will produce minimal output, so apply an
603 // upper bound to the coefficient.
604 lpCoeff
= __min(lpCoeff
, 0.98f
);
606 // Calculate the filter coefficients for high-pass or low-pass
607 // dependent on HF ratio being above 1.
609 State
->Late
.LpCoeff
[index
][0] = 1.0f
+ lpCoeff
;
610 State
->Late
.LpCoeff
[index
][1] = -lpCoeff
;
612 State
->Late
.LpCoeff
[index
][0] = 1.0f
- lpCoeff
;
613 State
->Late
.LpCoeff
[index
][1] = lpCoeff
;
616 // Attenuate the cyclical line coefficients by the mixing coefficient
618 State
->Late
.Coeff
[index
] *= mixCoeff
;
622 // This processes the reverb state, given the input samples and an output
624 ALvoid
VerbProcess(ALverbState
*State
, ALuint SamplesToDo
, const ALfloat
*SamplesIn
, ALfloat (*SamplesOut
)[OUTPUTCHANNELS
])
627 ALfloat in
[4], early
[2], late
[2], out
[2];
629 for(index
= 0;index
< SamplesToDo
;index
++)
631 // Feed the initial delay line.
632 DelayLineIn(&State
->Delay
, State
->Offset
, SamplesIn
[index
]);
634 // Calculate the early reflection from the first delay tap.
635 in
[0] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[0]);
636 EarlyReflection(State
, in
[0], early
);
638 // Calculate the late reverb from the last four delay taps.
639 in
[0] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[1]);
640 in
[1] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[2]);
641 in
[2] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[3]);
642 in
[3] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[4]);
643 LateReverb(State
, in
, late
);
645 // Mix early reflections and late reverb.
646 out
[0] = State
->Gain
* (early
[0] + late
[0]);
647 out
[1] = State
->Gain
* (early
[1] + late
[1]);
649 // Step all delays forward one sample.
652 // Output the results.
653 SamplesOut
[index
][FRONT_LEFT
] += out
[0];
654 SamplesOut
[index
][FRONT_RIGHT
] += out
[1];
655 SamplesOut
[index
][SIDE_LEFT
] += out
[0];
656 SamplesOut
[index
][SIDE_RIGHT
] += out
[1];
657 SamplesOut
[index
][BACK_LEFT
] += out
[0];
658 SamplesOut
[index
][BACK_RIGHT
] += out
[1];