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.
42 typedef struct ALverbState
{
43 // Must be first in all effects!
46 // All delay lines are allocated as a single buffer to reduce memory
47 // fragmentation and management code.
48 ALfloat
*SampleBuffer
;
49 // Master effect low-pass filter (2 chained 1-pole filters).
52 // Initial effect delay and decorrelation.
54 // The tap points for the initial delay. First tap goes to early
55 // reflections, the last four decorrelate to late reverb.
58 // Total gain for early reflections.
60 // Early reflections are done with 4 delay lines.
64 // The gain for each output channel based on 3D panning.
65 ALfloat PanGain
[OUTPUTCHANNELS
];
68 // Total gain for late reverb.
70 // Attenuation to compensate for modal density and decay rate.
72 // The feed-back and feed-forward all-pass coefficient.
74 // Mixing matrix coefficient.
76 // Late reverb has 4 parallel all-pass filters.
80 // In addition to 4 cyclical delay lines.
84 // The cyclical delay lines are 1-pole low-pass filtered.
87 // The gain for each output channel based on 3D panning.
88 ALfloat PanGain
[OUTPUTCHANNELS
];
90 // The current read offset for all delay lines.
94 // All delay line lengths are specified in seconds.
96 // The lengths of the early delay lines.
97 static const ALfloat EARLY_LINE_LENGTH
[4] =
99 0.0015f
, 0.0045f
, 0.0135f
, 0.0405f
102 // The lengths of the late all-pass delay lines.
103 static const ALfloat ALLPASS_LINE_LENGTH
[4] =
105 0.0151f
, 0.0167f
, 0.0183f
, 0.0200f
,
108 // The lengths of the late cyclical delay lines.
109 static const ALfloat LATE_LINE_LENGTH
[4] =
111 0.0211f
, 0.0311f
, 0.0461f
, 0.0680f
114 // The late cyclical delay lines have a variable length dependent on the
115 // effect's density parameter (inverted for some reason) and this multiplier.
116 static const ALfloat LATE_LINE_MULTIPLIER
= 4.0f
;
118 // Input into the late reverb is decorrelated between four channels. Their
119 // timings are dependent on a fraction and multiplier. See VerbUpdate() for
120 // the calculations involved.
121 static const ALfloat DECO_FRACTION
= 1.0f
/ 32.0f
;
122 static const ALfloat DECO_MULTIPLIER
= 2.0f
;
124 // The maximum length of initial delay for the master delay line (a sum of
125 // the maximum early reflection and late reverb delays).
126 static const ALfloat MASTER_LINE_LENGTH
= 0.3f
+ 0.1f
;
128 // Find the next power of 2. Actually, this will return the input value if
129 // it is already a power of 2.
130 static ALuint
NextPowerOf2(ALuint value
)
146 // Basic delay line input/output routines.
147 static __inline ALfloat
DelayLineOut(DelayLine
*Delay
, ALuint offset
)
149 return Delay
->Line
[offset
&Delay
->Mask
];
152 static __inline ALvoid
DelayLineIn(DelayLine
*Delay
, ALuint offset
, ALfloat in
)
154 Delay
->Line
[offset
&Delay
->Mask
] = in
;
157 // Delay line output routine for early reflections.
158 static __inline ALfloat
EarlyDelayLineOut(ALverbState
*State
, ALuint index
)
160 return State
->Early
.Coeff
[index
] *
161 DelayLineOut(&State
->Early
.Delay
[index
],
162 State
->Offset
- State
->Early
.Offset
[index
]);
165 // Given an input sample, this function produces stereo output for early
167 static __inline ALvoid
EarlyReflection(ALverbState
*State
, ALfloat in
, ALfloat
*out
)
169 ALfloat d
[4], v
, f
[4];
171 // Obtain the decayed results of each early delay line.
172 d
[0] = EarlyDelayLineOut(State
, 0);
173 d
[1] = EarlyDelayLineOut(State
, 1);
174 d
[2] = EarlyDelayLineOut(State
, 2);
175 d
[3] = EarlyDelayLineOut(State
, 3);
177 /* The following uses a lossless scattering junction from waveguide
178 * theory. It actually amounts to a householder mixing matrix, which
179 * will produce a maximally diffuse response, and means this can probably
180 * be considered a simple feedback delay network (FDN).
188 v
= (d
[0] + d
[1] + d
[2] + d
[3]) * 0.5f
;
189 // The junction is loaded with the input here.
192 // Calculate the feed values for the delay lines.
198 // Refeed the delay lines.
199 DelayLineIn(&State
->Early
.Delay
[0], State
->Offset
, f
[0]);
200 DelayLineIn(&State
->Early
.Delay
[1], State
->Offset
, f
[1]);
201 DelayLineIn(&State
->Early
.Delay
[2], State
->Offset
, f
[2]);
202 DelayLineIn(&State
->Early
.Delay
[3], State
->Offset
, f
[3]);
204 // Output the results of the junction for all four lines.
205 out
[0] = State
->Early
.Gain
* f
[0];
206 out
[1] = State
->Early
.Gain
* f
[1];
207 out
[2] = State
->Early
.Gain
* f
[2];
208 out
[3] = State
->Early
.Gain
* f
[3];
211 // All-pass input/output routine for late reverb.
212 static __inline ALfloat
LateAllPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
216 out
= State
->Late
.ApCoeff
[index
] *
217 DelayLineOut(&State
->Late
.ApDelay
[index
],
218 State
->Offset
- State
->Late
.ApOffset
[index
]);
219 out
-= (State
->Late
.ApFeedCoeff
* in
);
220 DelayLineIn(&State
->Late
.ApDelay
[index
], State
->Offset
,
221 (State
->Late
.ApFeedCoeff
* out
) + in
);
225 // Delay line output routine for late reverb.
226 static __inline ALfloat
LateDelayLineOut(ALverbState
*State
, ALuint index
)
228 return State
->Late
.Coeff
[index
] *
229 DelayLineOut(&State
->Late
.Delay
[index
],
230 State
->Offset
- State
->Late
.Offset
[index
]);
233 // Low-pass filter input/output routine for late reverb.
234 static __inline ALfloat
LateLowPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
236 State
->Late
.LpSample
[index
] = in
+
237 ((State
->Late
.LpSample
[index
] - in
) * State
->Late
.LpCoeff
[index
]);
238 return State
->Late
.LpSample
[index
];
241 // Given four decorrelated input samples, this function produces stereo
242 // output for late reverb.
243 static __inline ALvoid
LateReverb(ALverbState
*State
, ALfloat
*in
, ALfloat
*out
)
247 // Obtain the decayed results of the cyclical delay lines, and add the
248 // corresponding input channels attenuated by density. Then pass the
249 // results through the low-pass filters.
250 d
[0] = LateLowPassInOut(State
, 0, (State
->Late
.DensityGain
* in
[0]) +
251 LateDelayLineOut(State
, 0));
252 d
[1] = LateLowPassInOut(State
, 1, (State
->Late
.DensityGain
* in
[1]) +
253 LateDelayLineOut(State
, 1));
254 d
[2] = LateLowPassInOut(State
, 2, (State
->Late
.DensityGain
* in
[2]) +
255 LateDelayLineOut(State
, 2));
256 d
[3] = LateLowPassInOut(State
, 3, (State
->Late
.DensityGain
* in
[3]) +
257 LateDelayLineOut(State
, 3));
259 // To help increase diffusion, run each line through an all-pass filter.
260 // The order of the all-pass filters is selected so that the shortest
261 // all-pass filter will feed the shortest delay line.
262 d
[0] = LateAllPassInOut(State
, 1, d
[0]);
263 d
[1] = LateAllPassInOut(State
, 3, d
[1]);
264 d
[2] = LateAllPassInOut(State
, 0, d
[2]);
265 d
[3] = LateAllPassInOut(State
, 2, d
[3]);
267 /* Late reverb is done with a modified feedback delay network (FDN)
268 * topology. Four input lines are each fed through their own all-pass
269 * filter and then into the mixing matrix. The four outputs of the
270 * mixing matrix are then cycled back to the inputs. Each output feeds
271 * a different input to form a circlular feed cycle.
273 * The mixing matrix used is a 4D skew-symmetric rotation matrix derived
274 * using a single unitary rotational parameter:
276 * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2
281 * The rotation is constructed from the effect's diffusion parameter,
282 * yielding: 1 = x^2 + 3 y^2; where a, b, and c are the coefficient y
283 * with differing signs, and d is the coefficient x. The matrix is thus:
285 * [ x, y, -y, y ] x = 1 - (0.5 diffusion^3)
286 * [ -y, x, y, y ] y = sqrt((1 - x^2) / 3)
290 * To reduce the number of multiplies, the x coefficient is applied with
291 * the cyclical delay line coefficients. Thus only the y coefficient is
292 * applied when mixing, and is modified to be: y / x.
294 f
[0] = d
[0] + (State
->Late
.MixCoeff
* ( d
[1] - d
[2] + d
[3]));
295 f
[1] = d
[1] + (State
->Late
.MixCoeff
* (-d
[0] + d
[2] + d
[3]));
296 f
[2] = d
[2] + (State
->Late
.MixCoeff
* ( d
[0] - d
[1] + d
[3]));
297 f
[3] = d
[3] + (State
->Late
.MixCoeff
* (-d
[0] - d
[1] - d
[2]));
299 // Output the results of the matrix for all four cyclical delay lines,
300 // attenuated by the late reverb gain (which is attenuated by the 'x'
302 out
[0] = State
->Late
.Gain
* f
[0];
303 out
[1] = State
->Late
.Gain
* f
[1];
304 out
[2] = State
->Late
.Gain
* f
[2];
305 out
[3] = State
->Late
.Gain
* f
[3];
307 // The delay lines are fed circularly in the order:
308 // 0 -> 1 -> 3 -> 2 -> 0 ...
309 DelayLineIn(&State
->Late
.Delay
[0], State
->Offset
, f
[2]);
310 DelayLineIn(&State
->Late
.Delay
[1], State
->Offset
, f
[0]);
311 DelayLineIn(&State
->Late
.Delay
[2], State
->Offset
, f
[3]);
312 DelayLineIn(&State
->Late
.Delay
[3], State
->Offset
, f
[1]);
315 // Process the reverb for a given input sample, resulting in separate four-
316 // channel output for both early reflections and late reverb.
317 static __inline ALvoid
ReverbInOut(ALverbState
*State
, ALfloat in
, ALfloat
*early
, ALfloat
*late
)
321 // Low-pass filter the incoming sample.
322 in
= lpFilter2P(&State
->LpFilter
, 0, in
);
324 // Feed the initial delay line.
325 DelayLineIn(&State
->Delay
, State
->Offset
, in
);
327 // Calculate the early reflection from the first delay tap.
328 in
= DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[0]);
329 EarlyReflection(State
, in
, early
);
331 // Calculate the late reverb from the last four delay taps.
332 taps
[0] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[1]);
333 taps
[1] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[2]);
334 taps
[2] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[3]);
335 taps
[3] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[4]);
336 LateReverb(State
, taps
, late
);
338 // Step all delays forward one sample.
342 // This destroys the reverb state. It should be called only when the effect
343 // slot has a different (or no) effect loaded over the reverb effect.
344 ALvoid
VerbDestroy(ALeffectState
*effect
)
346 ALverbState
*State
= (ALverbState
*)effect
;
349 free(State
->SampleBuffer
);
350 State
->SampleBuffer
= NULL
;
355 // NOTE: Temp, remove later.
356 static __inline ALint
aluCart2LUTpos(ALfloat re
, ALfloat im
)
359 ALfloat denom
= aluFabs(re
) + aluFabs(im
);
361 pos
= (ALint
)(QUADRANT_NUM
*aluFabs(im
) / denom
+ 0.5);
364 pos
= 2 * QUADRANT_NUM
- pos
;
370 // This updates the reverb state. This is called any time the reverb effect
371 // is loaded into a slot.
372 ALvoid
VerbUpdate(ALeffectState
*effect
, ALCcontext
*Context
, ALeffect
*Effect
)
374 ALverbState
*State
= (ALverbState
*)effect
;
376 ALfloat length
, mixCoeff
, cw
, g
, coeff
;
377 ALfloat hfRatio
= Effect
->Reverb
.DecayHFRatio
;
379 // Calculate the master low-pass filter (from the master effect HF gain).
380 cw
= cos(2.0 * M_PI
* Effect
->Reverb
.HFReference
/ Context
->Frequency
);
381 g
= __max(Effect
->Reverb
.GainHF
, 0.0001f
);
382 State
->LpFilter
.coeff
= 0.0f
;
383 if(g
< 0.9999f
) // 1-epsilon
384 State
->LpFilter
.coeff
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) / (1 - g
);
386 // Calculate the initial delay taps.
387 length
= Effect
->Reverb
.ReflectionsDelay
;
388 State
->Tap
[0] = (ALuint
)(length
* Context
->Frequency
);
390 length
+= Effect
->Reverb
.LateReverbDelay
;
392 /* The four inputs to the late reverb are decorrelated to smooth the
393 * initial reverb and reduce harsh echos. The timings are calculated as
394 * multiples of a fraction of the smallest cyclical delay time. This
395 * result is then adjusted so that the first tap occurs immediately (all
396 * taps are reduced by the shortest fraction).
398 * offset[index] = ((FRACTION MULTIPLIER^index) - 1) delay
400 for(index
= 0;index
< 4;index
++)
402 length
+= LATE_LINE_LENGTH
[0] *
403 (1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
)) *
404 (DECO_FRACTION
* (pow(DECO_MULTIPLIER
, (ALfloat
)index
) - 1.0f
));
405 State
->Tap
[1 + index
] = (ALuint
)(length
* Context
->Frequency
);
408 // Calculate the early reflections gain (from the master effect gain, and
409 // reflections gain parameters).
410 State
->Early
.Gain
= Effect
->Reverb
.Gain
* Effect
->Reverb
.ReflectionsGain
;
412 // Calculate the gain (coefficient) for each early delay line.
413 for(index
= 0;index
< 4;index
++)
414 State
->Early
.Coeff
[index
] = pow(10.0f
, EARLY_LINE_LENGTH
[index
] /
415 Effect
->Reverb
.LateReverbDelay
*
418 // Calculate the first mixing matrix coefficient (x).
419 mixCoeff
= 1.0f
- (0.5f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
));
421 // Calculate the late reverb gain (from the master effect gain, and late
422 // reverb gain parameters). Since the output is tapped prior to the
423 // application of the delay line coefficients, this gain needs to be
424 // attenuated by the 'x' mix coefficient from above.
425 State
->Late
.Gain
= Effect
->Reverb
.Gain
* Effect
->Reverb
.LateReverbGain
* mixCoeff
;
427 /* To compensate for changes in modal density and decay time of the late
428 * reverb signal, the input is attenuated based on the maximal energy of
429 * the outgoing signal. This is calculated as the ratio between a
430 * reference value and the current approximation of energy for the output
433 * Reverb output matches exponential decay of the form Sum(a^n), where a
434 * is the attenuation coefficient, and n is the sample ranging from 0 to
435 * infinity. The signal energy can thus be approximated using the area
436 * under this curve, calculated as: 1 / (1 - a).
438 * The reference energy is calculated from a signal at the lowest (effect
439 * at 1.0) density with a decay time of one second.
441 * The coefficient is calculated as the average length of the cyclical
442 * delay lines. This produces a better result than calculating the gain
443 * for each line individually (most likely a side effect of diffusion).
445 * The final result is the square root of the ratio bound to a maximum
446 * value of 1 (no amplification).
448 length
= (LATE_LINE_LENGTH
[0] + LATE_LINE_LENGTH
[1] +
449 LATE_LINE_LENGTH
[2] + LATE_LINE_LENGTH
[3]);
450 g
= length
* (1.0f
+ LATE_LINE_MULTIPLIER
) * 0.25f
;
451 g
= pow(10.0f
, g
* -60.0f
/ 20.0f
);
452 g
= 1.0f
/ (1.0f
- (g
* g
));
453 length
*= 1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
) * 0.25f
;
454 length
= pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
* -60.0f
/ 20.0f
);
455 length
= 1.0f
/ (1.0f
- (length
* length
));
456 State
->Late
.DensityGain
= __min(aluSqrt(g
/ length
), 1.0f
);
458 // Calculate the all-pass feed-back and feed-forward coefficient.
459 State
->Late
.ApFeedCoeff
= 0.6f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
);
461 // Calculate the mixing matrix coefficient (y / x).
462 g
= aluSqrt((1.0f
- (mixCoeff
* mixCoeff
)) / 3.0f
);
463 State
->Late
.MixCoeff
= g
/ mixCoeff
;
465 for(index
= 0;index
< 4;index
++)
467 // Calculate the gain (coefficient) for each all-pass line.
468 State
->Late
.ApCoeff
[index
] = pow(10.0f
, ALLPASS_LINE_LENGTH
[index
] /
469 Effect
->Reverb
.DecayTime
*
473 // If the HF limit parameter is flagged, calculate an appropriate limit
474 // based on the air absorption parameter.
475 if(Effect
->Reverb
.DecayHFLimit
&& Effect
->Reverb
.AirAbsorptionGainHF
< 1.0f
)
479 // For each of the cyclical delays, find the attenuation due to air
480 // absorption in dB (converting delay time to meters using the speed
481 // of sound). Then reversing the decay equation, solve for HF ratio.
482 // The delay length is cancelled out of the equation, so it can be
483 // calculated once for all lines.
484 limitRatio
= 1.0f
/ (log10(Effect
->Reverb
.AirAbsorptionGainHF
) *
485 SPEEDOFSOUNDMETRESPERSEC
*
486 Effect
->Reverb
.DecayTime
/ -60.0f
* 20.0f
);
487 // Need to limit the result to a minimum of 0.1, just like the HF
489 limitRatio
= __max(limitRatio
, 0.1f
);
491 // Using the limit calculated above, apply the upper bound to the
493 hfRatio
= __min(hfRatio
, limitRatio
);
496 // Calculate the low-pass filter frequency.
497 cw
= cos(2.0f
* M_PI
* Effect
->Reverb
.HFReference
/ Context
->Frequency
);
499 for(index
= 0;index
< 4;index
++)
501 // Calculate the length (in seconds) of each cyclical delay line.
502 length
= LATE_LINE_LENGTH
[index
] * (1.0f
+ (Effect
->Reverb
.Density
*
503 LATE_LINE_MULTIPLIER
));
504 // Calculate the delay offset for the cyclical delay lines.
505 State
->Late
.Offset
[index
] = (ALuint
)(length
* Context
->Frequency
);
507 // Calculate the gain (coefficient) for each cyclical line.
508 State
->Late
.Coeff
[index
] = pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
*
511 // Eventually this should boost the high frequencies when the ratio
516 // Calculate the decay equation for each low-pass filter.
517 g
= pow(10.0f
, length
/ (Effect
->Reverb
.DecayTime
* hfRatio
) *
518 -60.0f
/ 20.0f
) / State
->Late
.Coeff
[index
];
522 // Calculate the gain (coefficient) for each low-pass filter.
523 if(g
< 0.9999f
) // 1-epsilon
524 coeff
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) / (1 - g
);
526 // Very low decay times will produce minimal output, so apply an
527 // upper bound to the coefficient.
528 coeff
= __min(coeff
, 0.98f
);
530 State
->Late
.LpCoeff
[index
] = coeff
;
532 // Attenuate the cyclical line coefficients by the mixing coefficient
534 State
->Late
.Coeff
[index
] *= mixCoeff
;
537 // Calculate the 3D-panning gains for the early reflections and late
538 // reverb (for EAX mode).
540 ALfloat earlyPan
[3] = { Effect
->Reverb
.ReflectionsPan
[0], Effect
->Reverb
.ReflectionsPan
[1], Effect
->Reverb
.ReflectionsPan
[2] };
541 ALfloat latePan
[3] = { Effect
->Reverb
.LateReverbPan
[0], Effect
->Reverb
.LateReverbPan
[1], Effect
->Reverb
.LateReverbPan
[2] };
542 ALfloat
*speakerGain
, dirGain
, ambientGain
;
546 length
= earlyPan
[0]*earlyPan
[0] + earlyPan
[1]*earlyPan
[1] + earlyPan
[2]*earlyPan
[2];
549 length
= 1.0f
/ aluSqrt(length
);
550 earlyPan
[0] *= length
;
551 earlyPan
[1] *= length
;
552 earlyPan
[2] *= length
;
554 length
= latePan
[0]*latePan
[0] + latePan
[1]*latePan
[1] + latePan
[2]*latePan
[2];
557 length
= 1.0f
/ aluSqrt(length
);
558 latePan
[0] *= length
;
559 latePan
[1] *= length
;
560 latePan
[2] *= length
;
563 // This code applies directional reverb just like the mixer applies
564 // directional sources. It diffuses the sound toward all speakers
565 // as the magnitude of the panning vector drops, which is only an
566 // approximation of the expansion of sound across the speakers from
567 // the panning direction.
568 pos
= aluCart2LUTpos(earlyPan
[2], earlyPan
[0]);
569 speakerGain
= &Context
->PanningLUT
[OUTPUTCHANNELS
* pos
];
570 dirGain
= aluSqrt((earlyPan
[0] * earlyPan
[0]) + (earlyPan
[2] * earlyPan
[2]));
571 ambientGain
= (1.0 - dirGain
);
572 for(index
= 0;index
< OUTPUTCHANNELS
;index
++)
573 State
->Early
.PanGain
[index
] = dirGain
* speakerGain
[index
] + ambientGain
;
575 pos
= aluCart2LUTpos(latePan
[2], latePan
[0]);
576 speakerGain
= &Context
->PanningLUT
[OUTPUTCHANNELS
* pos
];
577 dirGain
= aluSqrt((latePan
[0] * latePan
[0]) + (latePan
[2] * latePan
[2]));
578 ambientGain
= (1.0 - dirGain
);
579 for(index
= 0;index
< OUTPUTCHANNELS
;index
++)
580 State
->Late
.PanGain
[index
] = dirGain
* speakerGain
[index
] + ambientGain
;
584 // This processes the reverb state, given the input samples and an output
586 ALvoid
VerbProcess(ALeffectState
*effect
, const ALeffectslot
*Slot
, ALuint SamplesToDo
, const ALfloat
*SamplesIn
, ALfloat (*SamplesOut
)[OUTPUTCHANNELS
])
588 ALverbState
*State
= (ALverbState
*)effect
;
590 ALfloat early
[4], late
[4], out
[4];
591 ALfloat gain
= Slot
->Gain
;
593 for(index
= 0;index
< SamplesToDo
;index
++)
595 // Process reverb for this sample.
596 ReverbInOut(State
, SamplesIn
[index
], early
, late
);
598 // Mix early reflections and late reverb.
599 out
[0] = (early
[0] + late
[0]) * gain
;
600 out
[1] = (early
[1] + late
[1]) * gain
;
601 out
[2] = (early
[2] + late
[2]) * gain
;
602 out
[3] = (early
[3] + late
[3]) * gain
;
604 // Output the results.
605 SamplesOut
[index
][FRONT_LEFT
] += out
[0];
606 SamplesOut
[index
][FRONT_RIGHT
] += out
[1];
607 SamplesOut
[index
][FRONT_CENTER
] += out
[3];
608 SamplesOut
[index
][SIDE_LEFT
] += out
[0];
609 SamplesOut
[index
][SIDE_RIGHT
] += out
[1];
610 SamplesOut
[index
][BACK_LEFT
] += out
[0];
611 SamplesOut
[index
][BACK_RIGHT
] += out
[1];
612 SamplesOut
[index
][BACK_CENTER
] += out
[2];
616 // This processes the EAX reverb state, given the input samples and an output
618 ALvoid
EAXVerbProcess(ALeffectState
*effect
, const ALeffectslot
*Slot
, ALuint SamplesToDo
, const ALfloat
*SamplesIn
, ALfloat (*SamplesOut
)[OUTPUTCHANNELS
])
620 ALverbState
*State
= (ALverbState
*)effect
;
622 ALfloat early
[4], late
[4];
623 ALfloat gain
= Slot
->Gain
;
625 for(index
= 0;index
< SamplesToDo
;index
++)
627 // Process reverb for this sample.
628 ReverbInOut(State
, SamplesIn
[index
], early
, late
);
630 // Unfortunately, while the number and configuration of gains for
631 // panning adjust according to OUTPUTCHANNELS, the output from the
632 // reverb engine is not so scalable.
633 SamplesOut
[index
][FRONT_LEFT
] +=
634 (State
->Early
.PanGain
[FRONT_LEFT
]*early
[0] +
635 State
->Late
.PanGain
[FRONT_LEFT
]*late
[0]) * gain
;
636 SamplesOut
[index
][FRONT_RIGHT
] +=
637 (State
->Early
.PanGain
[FRONT_RIGHT
]*early
[1] +
638 State
->Late
.PanGain
[FRONT_RIGHT
]*late
[1]) * gain
;
639 SamplesOut
[index
][FRONT_CENTER
] +=
640 (State
->Early
.PanGain
[FRONT_CENTER
]*early
[3] +
641 State
->Late
.PanGain
[FRONT_CENTER
]*late
[3]) * gain
;
642 SamplesOut
[index
][SIDE_LEFT
] +=
643 (State
->Early
.PanGain
[SIDE_LEFT
]*early
[0] +
644 State
->Late
.PanGain
[SIDE_LEFT
]*late
[0]) * gain
;
645 SamplesOut
[index
][SIDE_RIGHT
] +=
646 (State
->Early
.PanGain
[SIDE_RIGHT
]*early
[1] +
647 State
->Late
.PanGain
[SIDE_RIGHT
]*late
[1]) * gain
;
648 SamplesOut
[index
][BACK_LEFT
] +=
649 (State
->Early
.PanGain
[BACK_LEFT
]*early
[0] +
650 State
->Late
.PanGain
[BACK_LEFT
]*late
[0]) * gain
;
651 SamplesOut
[index
][BACK_RIGHT
] +=
652 (State
->Early
.PanGain
[BACK_RIGHT
]*early
[1] +
653 State
->Late
.PanGain
[BACK_RIGHT
]*late
[1]) * gain
;
654 SamplesOut
[index
][BACK_CENTER
] +=
655 (State
->Early
.PanGain
[BACK_CENTER
]*early
[2] +
656 State
->Late
.PanGain
[BACK_CENTER
]*late
[2]) * gain
;
660 // This creates the reverb state. It should be called only when the reverb
661 // effect is loaded into a slot that doesn't already have a reverb effect.
662 ALeffectState
*VerbCreate(ALCcontext
*Context
)
664 ALverbState
*State
= NULL
;
665 ALuint samples
, length
[13], totalLength
, index
;
667 State
= malloc(sizeof(ALverbState
));
670 alSetError(AL_OUT_OF_MEMORY
);
674 State
->state
.Destroy
= VerbDestroy
;
675 State
->state
.Update
= VerbUpdate
;
676 State
->state
.Process
= VerbProcess
;
678 // All line lengths are powers of 2, calculated from their lengths, with
679 // an additional sample in case of rounding errors.
681 // See VerbUpdate() for an explanation of the additional calculation
682 // added to the master line length.
684 ((MASTER_LINE_LENGTH
+
685 (LATE_LINE_LENGTH
[0] * (1.0f
+ LATE_LINE_MULTIPLIER
) *
686 (DECO_FRACTION
* ((DECO_MULTIPLIER
* DECO_MULTIPLIER
*
687 DECO_MULTIPLIER
) - 1.0f
)))) *
688 Context
->Frequency
) + 1;
689 length
[0] = NextPowerOf2(samples
);
690 totalLength
= length
[0];
691 for(index
= 0;index
< 4;index
++)
693 samples
= (ALuint
)(EARLY_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
694 length
[1 + index
] = NextPowerOf2(samples
);
695 totalLength
+= length
[1 + index
];
697 for(index
= 0;index
< 4;index
++)
699 samples
= (ALuint
)(ALLPASS_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
700 length
[5 + index
] = NextPowerOf2(samples
);
701 totalLength
+= length
[5 + index
];
703 for(index
= 0;index
< 4;index
++)
705 samples
= (ALuint
)(LATE_LINE_LENGTH
[index
] *
706 (1.0f
+ LATE_LINE_MULTIPLIER
) * Context
->Frequency
) + 1;
707 length
[9 + index
] = NextPowerOf2(samples
);
708 totalLength
+= length
[9 + index
];
711 // All lines share a single sample buffer and have their masks and start
712 // addresses calculated once.
713 State
->SampleBuffer
= malloc(totalLength
* sizeof(ALfloat
));
714 if(!State
->SampleBuffer
)
717 alSetError(AL_OUT_OF_MEMORY
);
720 for(index
= 0; index
< totalLength
;index
++)
721 State
->SampleBuffer
[index
] = 0.0f
;
723 State
->LpFilter
.coeff
= 0.0f
;
724 State
->LpFilter
.history
[0] = 0.0f
;
725 State
->LpFilter
.history
[1] = 0.0f
;
726 State
->Delay
.Mask
= length
[0] - 1;
727 State
->Delay
.Line
= &State
->SampleBuffer
[0];
728 totalLength
= length
[0];
736 State
->Early
.Gain
= 0.0f
;
737 for(index
= 0;index
< 4;index
++)
739 State
->Early
.Coeff
[index
] = 0.0f
;
740 State
->Early
.Delay
[index
].Mask
= length
[1 + index
] - 1;
741 State
->Early
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
742 totalLength
+= length
[1 + index
];
744 // The early delay lines have their read offsets calculated once.
745 State
->Early
.Offset
[index
] = (ALuint
)(EARLY_LINE_LENGTH
[index
] *
749 State
->Late
.Gain
= 0.0f
;
750 State
->Late
.DensityGain
= 0.0f
;
751 State
->Late
.ApFeedCoeff
= 0.0f
;
752 State
->Late
.MixCoeff
= 0.0f
;
754 for(index
= 0;index
< 4;index
++)
756 State
->Late
.ApCoeff
[index
] = 0.0f
;
757 State
->Late
.ApDelay
[index
].Mask
= length
[5 + index
] - 1;
758 State
->Late
.ApDelay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
759 totalLength
+= length
[5 + index
];
761 // The late all-pass lines have their read offsets calculated once.
762 State
->Late
.ApOffset
[index
] = (ALuint
)(ALLPASS_LINE_LENGTH
[index
] *
766 for(index
= 0;index
< 4;index
++)
768 State
->Late
.Coeff
[index
] = 0.0f
;
769 State
->Late
.Delay
[index
].Mask
= length
[9 + index
] - 1;
770 State
->Late
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
771 totalLength
+= length
[9 + index
];
773 State
->Late
.Offset
[index
] = 0;
775 State
->Late
.LpCoeff
[index
] = 0.0f
;
776 State
->Late
.LpSample
[index
] = 0.0f
;
779 // Panning is applied as an independent gain for each output channel.
780 for(index
= 0;index
< OUTPUTCHANNELS
;index
++)
782 State
->Early
.PanGain
[index
] = 0.0f
;
783 State
->Late
.PanGain
[index
] = 0.0f
;
787 return &State
->state
;
790 ALeffectState
*EAXVerbCreate(ALCcontext
*Context
)
792 ALeffectState
*State
= VerbCreate(Context
);
793 if(State
) State
->Process
= EAXVerbProcess
;