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"
33 typedef struct DelayLine
35 // The delay lines use sample lengths that are powers of 2 to allow
36 // bitmasking instead of modulus wrapping.
41 typedef struct ALverbState
{
42 // Must be first in all effects!
45 // All delay lines are allocated as a single buffer to reduce memory
46 // fragmentation and management code.
47 ALfloat
*SampleBuffer
;
48 // Master effect low-pass filter (2 chained 1-pole filters).
51 // Initial effect delay and decorrelation.
53 // The tap points for the initial delay. First tap goes to early
54 // reflections, the last four decorrelate to late reverb.
57 // Total gain for early reflections.
59 // Early reflections are done with 4 delay lines.
63 // The gain for each output channel based on 3D panning.
64 ALfloat PanGain
[OUTPUTCHANNELS
];
67 // Total gain for late reverb.
69 // Attenuation to compensate for modal density and decay rate.
71 // The feed-back and feed-forward all-pass coefficient.
73 // Mixing matrix coefficient.
75 // Late reverb has 4 parallel all-pass filters.
79 // In addition to 4 cyclical delay lines.
83 // The cyclical delay lines are 1-pole low-pass filtered.
86 // The gain for each output channel based on 3D panning.
87 ALfloat PanGain
[OUTPUTCHANNELS
];
89 // The current read offset for all delay lines.
93 // All delay line lengths are specified in seconds.
95 // The lengths of the early delay lines.
96 static const ALfloat EARLY_LINE_LENGTH
[4] =
98 0.0015f
, 0.0045f
, 0.0135f
, 0.0405f
101 // The lengths of the late all-pass delay lines.
102 static const ALfloat ALLPASS_LINE_LENGTH
[4] =
104 0.0151f
, 0.0167f
, 0.0183f
, 0.0200f
,
107 // The lengths of the late cyclical delay lines.
108 static const ALfloat LATE_LINE_LENGTH
[4] =
110 0.0211f
, 0.0311f
, 0.0461f
, 0.0680f
113 // The late cyclical delay lines have a variable length dependent on the
114 // effect's density parameter (inverted for some reason) and this multiplier.
115 static const ALfloat LATE_LINE_MULTIPLIER
= 4.0f
;
117 // Input into the late reverb is decorrelated between four channels. Their
118 // timings are dependent on a fraction and multiplier. See VerbUpdate() for
119 // the calculations involved.
120 static const ALfloat DECO_FRACTION
= 1.0f
/ 32.0f
;
121 static const ALfloat DECO_MULTIPLIER
= 2.0f
;
123 // The maximum length of initial delay for the master delay line (a sum of
124 // the maximum early reflection and late reverb delays).
125 static const ALfloat MASTER_LINE_LENGTH
= 0.3f
+ 0.1f
;
127 // Find the next power of 2. Actually, this will return the input value if
128 // it is already a power of 2.
129 static ALuint
NextPowerOf2(ALuint value
)
145 // Basic delay line input/output routines.
146 static __inline ALfloat
DelayLineOut(DelayLine
*Delay
, ALuint offset
)
148 return Delay
->Line
[offset
&Delay
->Mask
];
151 static __inline ALvoid
DelayLineIn(DelayLine
*Delay
, ALuint offset
, ALfloat in
)
153 Delay
->Line
[offset
&Delay
->Mask
] = in
;
156 // Delay line output routine for early reflections.
157 static __inline ALfloat
EarlyDelayLineOut(ALverbState
*State
, ALuint index
)
159 return State
->Early
.Coeff
[index
] *
160 DelayLineOut(&State
->Early
.Delay
[index
],
161 State
->Offset
- State
->Early
.Offset
[index
]);
164 // Given an input sample, this function produces stereo output for early
166 static __inline ALvoid
EarlyReflection(ALverbState
*State
, ALfloat in
, ALfloat
*out
)
168 ALfloat d
[4], v
, f
[4];
170 // Obtain the decayed results of each early delay line.
171 d
[0] = EarlyDelayLineOut(State
, 0);
172 d
[1] = EarlyDelayLineOut(State
, 1);
173 d
[2] = EarlyDelayLineOut(State
, 2);
174 d
[3] = EarlyDelayLineOut(State
, 3);
176 /* The following uses a lossless scattering junction from waveguide
177 * theory. It actually amounts to a householder mixing matrix, which
178 * will produce a maximally diffuse response, and means this can probably
179 * be considered a simple feedback delay network (FDN).
187 v
= (d
[0] + d
[1] + d
[2] + d
[3]) * 0.5f
;
188 // The junction is loaded with the input here.
191 // Calculate the feed values for the delay lines.
197 // Refeed the delay lines.
198 DelayLineIn(&State
->Early
.Delay
[0], State
->Offset
, f
[0]);
199 DelayLineIn(&State
->Early
.Delay
[1], State
->Offset
, f
[1]);
200 DelayLineIn(&State
->Early
.Delay
[2], State
->Offset
, f
[2]);
201 DelayLineIn(&State
->Early
.Delay
[3], State
->Offset
, f
[3]);
203 // Output the results of the junction for all four lines.
204 out
[0] = State
->Early
.Gain
* f
[0];
205 out
[1] = State
->Early
.Gain
* f
[1];
206 out
[2] = State
->Early
.Gain
* f
[2];
207 out
[3] = State
->Early
.Gain
* f
[3];
210 // All-pass input/output routine for late reverb.
211 static __inline ALfloat
LateAllPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
215 out
= State
->Late
.ApCoeff
[index
] *
216 DelayLineOut(&State
->Late
.ApDelay
[index
],
217 State
->Offset
- State
->Late
.ApOffset
[index
]);
218 out
-= (State
->Late
.ApFeedCoeff
* in
);
219 DelayLineIn(&State
->Late
.ApDelay
[index
], State
->Offset
,
220 (State
->Late
.ApFeedCoeff
* out
) + in
);
224 // Delay line output routine for late reverb.
225 static __inline ALfloat
LateDelayLineOut(ALverbState
*State
, ALuint index
)
227 return State
->Late
.Coeff
[index
] *
228 DelayLineOut(&State
->Late
.Delay
[index
],
229 State
->Offset
- State
->Late
.Offset
[index
]);
232 // Low-pass filter input/output routine for late reverb.
233 static __inline ALfloat
LateLowPassInOut(ALverbState
*State
, ALuint index
, ALfloat in
)
235 State
->Late
.LpSample
[index
] = in
+
236 ((State
->Late
.LpSample
[index
] - in
) * State
->Late
.LpCoeff
[index
]);
237 return State
->Late
.LpSample
[index
];
240 // Given four decorrelated input samples, this function produces stereo
241 // output for late reverb.
242 static __inline ALvoid
LateReverb(ALverbState
*State
, ALfloat
*in
, ALfloat
*out
)
246 // Obtain the decayed results of the cyclical delay lines, and add the
247 // corresponding input channels attenuated by density. Then pass the
248 // results through the low-pass filters.
249 d
[0] = LateLowPassInOut(State
, 0, (State
->Late
.DensityGain
* in
[0]) +
250 LateDelayLineOut(State
, 0));
251 d
[1] = LateLowPassInOut(State
, 1, (State
->Late
.DensityGain
* in
[1]) +
252 LateDelayLineOut(State
, 1));
253 d
[2] = LateLowPassInOut(State
, 2, (State
->Late
.DensityGain
* in
[2]) +
254 LateDelayLineOut(State
, 2));
255 d
[3] = LateLowPassInOut(State
, 3, (State
->Late
.DensityGain
* in
[3]) +
256 LateDelayLineOut(State
, 3));
258 // To help increase diffusion, run each line through an all-pass filter.
259 // The order of the all-pass filters is selected so that the shortest
260 // all-pass filter will feed the shortest delay line.
261 d
[0] = LateAllPassInOut(State
, 1, d
[0]);
262 d
[1] = LateAllPassInOut(State
, 3, d
[1]);
263 d
[2] = LateAllPassInOut(State
, 0, d
[2]);
264 d
[3] = LateAllPassInOut(State
, 2, d
[3]);
266 /* Late reverb is done with a modified feedback delay network (FDN)
267 * topology. Four input lines are each fed through their own all-pass
268 * filter and then into the mixing matrix. The four outputs of the
269 * mixing matrix are then cycled back to the inputs. Each output feeds
270 * a different input to form a circlular feed cycle.
272 * The mixing matrix used is a 4D skew-symmetric rotation matrix derived
273 * using a single unitary rotational parameter:
275 * [ d, a, b, c ] 1 = a^2 + b^2 + c^2 + d^2
280 * The rotation is constructed from the effect's diffusion parameter,
281 * yielding: 1 = x^2 + 3 y^2; where a, b, and c are the coefficient y
282 * with differing signs, and d is the coefficient x. The matrix is thus:
284 * [ x, y, -y, y ] x = 1 - (0.5 diffusion^3)
285 * [ -y, x, y, y ] y = sqrt((1 - x^2) / 3)
289 * To reduce the number of multiplies, the x coefficient is applied with
290 * the cyclical delay line coefficients. Thus only the y coefficient is
291 * applied when mixing, and is modified to be: y / x.
293 f
[0] = d
[0] + (State
->Late
.MixCoeff
* ( d
[1] - d
[2] + d
[3]));
294 f
[1] = d
[1] + (State
->Late
.MixCoeff
* (-d
[0] + d
[2] + d
[3]));
295 f
[2] = d
[2] + (State
->Late
.MixCoeff
* ( d
[0] - d
[1] + d
[3]));
296 f
[3] = d
[3] + (State
->Late
.MixCoeff
* (-d
[0] - d
[1] - d
[2]));
298 // Output the results of the matrix for all four cyclical delay lines,
299 // attenuated by the late reverb gain (which is attenuated by the 'x'
301 out
[0] = State
->Late
.Gain
* f
[0];
302 out
[1] = State
->Late
.Gain
* f
[1];
303 out
[2] = State
->Late
.Gain
* f
[2];
304 out
[3] = State
->Late
.Gain
* f
[3];
306 // The delay lines are fed circularly in the order:
307 // 0 -> 1 -> 3 -> 2 -> 0 ...
308 DelayLineIn(&State
->Late
.Delay
[0], State
->Offset
, f
[2]);
309 DelayLineIn(&State
->Late
.Delay
[1], State
->Offset
, f
[0]);
310 DelayLineIn(&State
->Late
.Delay
[2], State
->Offset
, f
[3]);
311 DelayLineIn(&State
->Late
.Delay
[3], State
->Offset
, f
[1]);
314 // Process the reverb for a given input sample, resulting in separate four-
315 // channel output for both early reflections and late reverb.
316 static __inline ALvoid
ReverbInOut(ALverbState
*State
, ALfloat in
, ALfloat
*early
, ALfloat
*late
)
320 // Low-pass filter the incoming sample.
321 in
= in
+ ((State
->LpSamples
[0] - in
) * State
->LpCoeff
);
322 State
->LpSamples
[0] = in
;
323 in
= in
+ ((State
->LpSamples
[1] - in
) * State
->LpCoeff
);
324 State
->LpSamples
[1] = in
;
326 // Feed the initial delay line.
327 DelayLineIn(&State
->Delay
, State
->Offset
, in
);
329 // Calculate the early reflection from the first delay tap.
330 in
= DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[0]);
331 EarlyReflection(State
, in
, early
);
333 // Calculate the late reverb from the last four delay taps.
334 taps
[0] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[1]);
335 taps
[1] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[2]);
336 taps
[2] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[3]);
337 taps
[3] = DelayLineOut(&State
->Delay
, State
->Offset
- State
->Tap
[4]);
338 LateReverb(State
, taps
, late
);
340 // Step all delays forward one sample.
344 // This destroys the reverb state. It should be called only when the effect
345 // slot has a different (or no) effect loaded over the reverb effect.
346 ALvoid
VerbDestroy(ALeffectState
*effect
)
348 ALverbState
*State
= (ALverbState
*)effect
;
351 free(State
->SampleBuffer
);
352 State
->SampleBuffer
= NULL
;
357 // NOTE: Temp, remove later.
358 static __inline ALint
aluCart2LUTpos(ALfloat re
, ALfloat im
)
361 ALfloat denom
= aluFabs(re
) + aluFabs(im
);
363 pos
= (ALint
)(QUADRANT_NUM
*aluFabs(im
) / denom
+ 0.5);
366 pos
= 2 * QUADRANT_NUM
- pos
;
372 // This updates the reverb state. This is called any time the reverb effect
373 // is loaded into a slot.
374 ALvoid
VerbUpdate(ALeffectState
*effect
, ALCcontext
*Context
, ALeffect
*Effect
)
376 ALverbState
*State
= (ALverbState
*)effect
;
378 ALfloat length
, mixCoeff
, cw
, g
, coeff
;
379 ALfloat hfRatio
= Effect
->Reverb
.DecayHFRatio
;
381 // Calculate the master low-pass filter (from the master effect HF gain).
382 cw
= cos(2.0 * M_PI
* Effect
->Reverb
.HFReference
/ Context
->Frequency
);
383 g
= __max(Effect
->Reverb
.GainHF
, 0.0001f
);
384 State
->LpCoeff
= 0.0f
;
385 if(g
< 0.9999f
) // 1-epsilon
386 State
->LpCoeff
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) / (1 - g
);
388 // Calculate the initial delay taps.
389 length
= Effect
->Reverb
.ReflectionsDelay
;
390 State
->Tap
[0] = (ALuint
)(length
* Context
->Frequency
);
392 length
+= Effect
->Reverb
.LateReverbDelay
;
394 /* The four inputs to the late reverb are decorrelated to smooth the
395 * initial reverb and reduce harsh echos. The timings are calculated as
396 * multiples of a fraction of the smallest cyclical delay time. This
397 * result is then adjusted so that the first tap occurs immediately (all
398 * taps are reduced by the shortest fraction).
400 * offset[index] = ((FRACTION MULTIPLIER^index) - 1) delay
402 for(index
= 0;index
< 4;index
++)
404 length
+= LATE_LINE_LENGTH
[0] *
405 (1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
)) *
406 (DECO_FRACTION
* (pow(DECO_MULTIPLIER
, (ALfloat
)index
) - 1.0f
));
407 State
->Tap
[1 + index
] = (ALuint
)(length
* Context
->Frequency
);
410 // Calculate the early reflections gain (from the slot gain, master
411 // effect gain, and reflections gain parameters).
412 State
->Early
.Gain
= Effect
->Reverb
.Gain
* Effect
->Reverb
.ReflectionsGain
;
414 // Calculate the gain (coefficient) for each early delay line.
415 for(index
= 0;index
< 4;index
++)
416 State
->Early
.Coeff
[index
] = pow(10.0f
, EARLY_LINE_LENGTH
[index
] /
417 Effect
->Reverb
.LateReverbDelay
*
420 // Calculate the first mixing matrix coefficient (x).
421 mixCoeff
= 1.0f
- (0.5f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
));
423 // Calculate the late reverb gain (from the slot gain, master effect
424 // gain, and late reverb gain parameters). Since the output is tapped
425 // prior to the application of the delay line coefficients, this gain
426 // needs to be attenuated by the 'x' mix coefficient from above.
427 State
->Late
.Gain
= Effect
->Reverb
.Gain
* Effect
->Reverb
.LateReverbGain
* mixCoeff
;
429 /* To compensate for changes in modal density and decay time of the late
430 * reverb signal, the input is attenuated based on the maximal energy of
431 * the outgoing signal. This is calculated as the ratio between a
432 * reference value and the current approximation of energy for the output
435 * Reverb output matches exponential decay of the form Sum(a^n), where a
436 * is the attenuation coefficient, and n is the sample ranging from 0 to
437 * infinity. The signal energy can thus be approximated using the area
438 * under this curve, calculated as: 1 / (1 - a).
440 * The reference energy is calculated from a signal at the lowest (effect
441 * at 1.0) density with a decay time of one second.
443 * The coefficient is calculated as the average length of the cyclical
444 * delay lines. This produces a better result than calculating the gain
445 * for each line individually (most likely a side effect of diffusion).
447 * The final result is the square root of the ratio bound to a maximum
448 * value of 1 (no amplification).
450 length
= (LATE_LINE_LENGTH
[0] + LATE_LINE_LENGTH
[1] +
451 LATE_LINE_LENGTH
[2] + LATE_LINE_LENGTH
[3]);
452 g
= length
* (1.0f
+ LATE_LINE_MULTIPLIER
) * 0.25f
;
453 g
= pow(10.0f
, g
* -60.0f
/ 20.0f
);
454 g
= 1.0f
/ (1.0f
- (g
* g
));
455 length
*= 1.0f
+ (Effect
->Reverb
.Density
* LATE_LINE_MULTIPLIER
) * 0.25f
;
456 length
= pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
* -60.0f
/ 20.0f
);
457 length
= 1.0f
/ (1.0f
- (length
* length
));
458 State
->Late
.DensityGain
= __min(aluSqrt(g
/ length
), 1.0f
);
460 // Calculate the all-pass feed-back and feed-forward coefficient.
461 State
->Late
.ApFeedCoeff
= 0.6f
* pow(Effect
->Reverb
.Diffusion
, 3.0f
);
463 // Calculate the mixing matrix coefficient (y / x).
464 g
= aluSqrt((1.0f
- (mixCoeff
* mixCoeff
)) / 3.0f
);
465 State
->Late
.MixCoeff
= g
/ mixCoeff
;
467 for(index
= 0;index
< 4;index
++)
469 // Calculate the gain (coefficient) for each all-pass line.
470 State
->Late
.ApCoeff
[index
] = pow(10.0f
, ALLPASS_LINE_LENGTH
[index
] /
471 Effect
->Reverb
.DecayTime
*
475 // If the HF limit parameter is flagged, calculate an appropriate limit
476 // based on the air absorption parameter.
477 if(Effect
->Reverb
.DecayHFLimit
&& Effect
->Reverb
.AirAbsorptionGainHF
< 1.0f
)
481 // For each of the cyclical delays, find the attenuation due to air
482 // absorption in dB (converting delay time to meters using the speed
483 // of sound). Then reversing the decay equation, solve for HF ratio.
484 // The delay length is cancelled out of the equation, so it can be
485 // calculated once for all lines.
486 limitRatio
= 1.0f
/ (log10(Effect
->Reverb
.AirAbsorptionGainHF
) *
487 SPEEDOFSOUNDMETRESPERSEC
*
488 Effect
->Reverb
.DecayTime
/ -60.0f
* 20.0f
);
489 // Need to limit the result to a minimum of 0.1, just like the HF
491 limitRatio
= __max(limitRatio
, 0.1f
);
493 // Using the limit calculated above, apply the upper bound to the
495 hfRatio
= __min(hfRatio
, limitRatio
);
498 // Calculate the low-pass filter frequency.
499 cw
= cos(2.0f
* M_PI
* Effect
->Reverb
.HFReference
/ Context
->Frequency
);
501 for(index
= 0;index
< 4;index
++)
503 // Calculate the length (in seconds) of each cyclical delay line.
504 length
= LATE_LINE_LENGTH
[index
] * (1.0f
+ (Effect
->Reverb
.Density
*
505 LATE_LINE_MULTIPLIER
));
506 // Calculate the delay offset for the cyclical delay lines.
507 State
->Late
.Offset
[index
] = (ALuint
)(length
* Context
->Frequency
);
509 // Calculate the gain (coefficient) for each cyclical line.
510 State
->Late
.Coeff
[index
] = pow(10.0f
, length
/ Effect
->Reverb
.DecayTime
*
513 // Eventually this should boost the high frequencies when the ratio
518 // Calculate the decay equation for each low-pass filter.
519 g
= pow(10.0f
, length
/ (Effect
->Reverb
.DecayTime
* hfRatio
) *
520 -60.0f
/ 20.0f
) / State
->Late
.Coeff
[index
];
524 // Calculate the gain (coefficient) for each low-pass filter.
525 if(g
< 0.9999f
) // 1-epsilon
526 coeff
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) / (1 - g
);
528 // Very low decay times will produce minimal output, so apply an
529 // upper bound to the coefficient.
530 coeff
= __min(coeff
, 0.98f
);
532 State
->Late
.LpCoeff
[index
] = coeff
;
534 // Attenuate the cyclical line coefficients by the mixing coefficient
536 State
->Late
.Coeff
[index
] *= mixCoeff
;
539 // Calculate the 3D-panning gains for the early reflections and late
540 // reverb (for EAX mode).
542 ALfloat earlyPan
[3] = { Effect
->Reverb
.ReflectionsPan
[0], Effect
->Reverb
.ReflectionsPan
[1], Effect
->Reverb
.ReflectionsPan
[2] };
543 ALfloat latePan
[3] = { Effect
->Reverb
.LateReverbPan
[0], Effect
->Reverb
.LateReverbPan
[1], Effect
->Reverb
.LateReverbPan
[2] };
544 ALfloat
*speakerGain
, dirGain
, ambientGain
;
548 length
= earlyPan
[0]*earlyPan
[0] + earlyPan
[1]*earlyPan
[1] + earlyPan
[2]*earlyPan
[2];
551 length
= 1.0f
/ aluSqrt(length
);
552 earlyPan
[0] *= length
;
553 earlyPan
[1] *= length
;
554 earlyPan
[2] *= length
;
556 length
= latePan
[0]*latePan
[0] + latePan
[1]*latePan
[1] + latePan
[2]*latePan
[2];
559 length
= 1.0f
/ aluSqrt(length
);
560 latePan
[0] *= length
;
561 latePan
[1] *= length
;
562 latePan
[2] *= length
;
565 // This code applies directional reverb just like the mixer applies
566 // directional sources. It diffuses the sound toward all speakers
567 // as the magnitude of the panning vector drops, which is only an
568 // approximation of the expansion of sound across the speakers from
569 // the panning direction.
570 pos
= aluCart2LUTpos(earlyPan
[2], earlyPan
[0]);
571 speakerGain
= &Context
->PanningLUT
[OUTPUTCHANNELS
* pos
];
572 dirGain
= aluSqrt((earlyPan
[0] * earlyPan
[0]) + (earlyPan
[2] * earlyPan
[2]));
573 ambientGain
= (1.0 - dirGain
);
574 for(index
= 0;index
< OUTPUTCHANNELS
;index
++)
575 State
->Early
.PanGain
[index
] = dirGain
* speakerGain
[index
] + ambientGain
;
577 pos
= aluCart2LUTpos(latePan
[2], latePan
[0]);
578 speakerGain
= &Context
->PanningLUT
[OUTPUTCHANNELS
* pos
];
579 dirGain
= aluSqrt((latePan
[0] * latePan
[0]) + (latePan
[2] * latePan
[2]));
580 ambientGain
= (1.0 - dirGain
);
581 for(index
= 0;index
< OUTPUTCHANNELS
;index
++)
582 State
->Late
.PanGain
[index
] = dirGain
* speakerGain
[index
] + ambientGain
;
586 // This processes the reverb state, given the input samples and an output
588 ALvoid
VerbProcess(ALeffectState
*effect
, const ALeffectslot
*Slot
, ALuint SamplesToDo
, const ALfloat
*SamplesIn
, ALfloat (*SamplesOut
)[OUTPUTCHANNELS
])
590 ALverbState
*State
= (ALverbState
*)effect
;
592 ALfloat early
[4], late
[4], out
[4];
593 ALfloat gain
= Slot
->Gain
;
595 for(index
= 0;index
< SamplesToDo
;index
++)
597 // Process reverb for this sample.
598 ReverbInOut(State
, SamplesIn
[index
], early
, late
);
600 // Mix early reflections and late reverb.
601 out
[0] = (early
[0] + late
[0]) * gain
;
602 out
[1] = (early
[1] + late
[1]) * gain
;
603 out
[2] = (early
[2] + late
[2]) * gain
;
604 out
[3] = (early
[3] + late
[3]) * gain
;
606 // Output the results.
607 SamplesOut
[index
][FRONT_LEFT
] += out
[0];
608 SamplesOut
[index
][FRONT_RIGHT
] += out
[1];
609 SamplesOut
[index
][FRONT_CENTER
] += out
[3];
610 SamplesOut
[index
][SIDE_LEFT
] += out
[0];
611 SamplesOut
[index
][SIDE_RIGHT
] += out
[1];
612 SamplesOut
[index
][BACK_LEFT
] += out
[0];
613 SamplesOut
[index
][BACK_RIGHT
] += out
[1];
614 SamplesOut
[index
][BACK_CENTER
] += out
[2];
618 // This processes the EAX reverb state, given the input samples and an output
620 ALvoid
EAXVerbProcess(ALeffectState
*effect
, const ALeffectslot
*Slot
, ALuint SamplesToDo
, const ALfloat
*SamplesIn
, ALfloat (*SamplesOut
)[OUTPUTCHANNELS
])
622 ALverbState
*State
= (ALverbState
*)effect
;
624 ALfloat early
[4], late
[4];
625 ALfloat gain
= Slot
->Gain
;
627 for(index
= 0;index
< SamplesToDo
;index
++)
629 // Process reverb for this sample.
630 ReverbInOut(State
, SamplesIn
[index
], early
, late
);
632 // Unfortunately, while the number and configuration of gains for
633 // panning adjust according to OUTPUTCHANNELS, the output from the
634 // reverb engine is not so scalable.
635 SamplesOut
[index
][FRONT_LEFT
] +=
636 (State
->Early
.PanGain
[FRONT_LEFT
]*early
[0] +
637 State
->Late
.PanGain
[FRONT_LEFT
]*late
[0]) * gain
;
638 SamplesOut
[index
][FRONT_RIGHT
] +=
639 (State
->Early
.PanGain
[FRONT_RIGHT
]*early
[1] +
640 State
->Late
.PanGain
[FRONT_RIGHT
]*late
[1]) * gain
;
641 SamplesOut
[index
][FRONT_CENTER
] +=
642 (State
->Early
.PanGain
[FRONT_CENTER
]*early
[3] +
643 State
->Late
.PanGain
[FRONT_CENTER
]*late
[3]) * gain
;
644 SamplesOut
[index
][SIDE_LEFT
] +=
645 (State
->Early
.PanGain
[SIDE_LEFT
]*early
[0] +
646 State
->Late
.PanGain
[SIDE_LEFT
]*late
[0]) * gain
;
647 SamplesOut
[index
][SIDE_RIGHT
] +=
648 (State
->Early
.PanGain
[SIDE_RIGHT
]*early
[1] +
649 State
->Late
.PanGain
[SIDE_RIGHT
]*late
[1]) * gain
;
650 SamplesOut
[index
][BACK_LEFT
] +=
651 (State
->Early
.PanGain
[BACK_LEFT
]*early
[0] +
652 State
->Late
.PanGain
[BACK_LEFT
]*late
[0]) * gain
;
653 SamplesOut
[index
][BACK_RIGHT
] +=
654 (State
->Early
.PanGain
[BACK_RIGHT
]*early
[1] +
655 State
->Late
.PanGain
[BACK_RIGHT
]*late
[1]) * gain
;
656 SamplesOut
[index
][BACK_CENTER
] +=
657 (State
->Early
.PanGain
[BACK_CENTER
]*early
[2] +
658 State
->Late
.PanGain
[BACK_CENTER
]*late
[2]) * gain
;
662 // This creates the reverb state. It should be called only when the reverb
663 // effect is loaded into a slot that doesn't already have a reverb effect.
664 ALeffectState
*VerbCreate(ALCcontext
*Context
)
666 ALverbState
*State
= NULL
;
667 ALuint samples
, length
[13], totalLength
, index
;
669 State
= malloc(sizeof(ALverbState
));
673 State
->state
.Destroy
= VerbDestroy
;
674 State
->state
.Update
= VerbUpdate
;
675 State
->state
.Process
= VerbProcess
;
677 // All line lengths are powers of 2, calculated from their lengths, with
678 // an additional sample in case of rounding errors.
680 // See VerbUpdate() for an explanation of the additional calculation
681 // added to the master line length.
683 ((MASTER_LINE_LENGTH
+
684 (LATE_LINE_LENGTH
[0] * (1.0f
+ LATE_LINE_MULTIPLIER
) *
685 (DECO_FRACTION
* ((DECO_MULTIPLIER
* DECO_MULTIPLIER
*
686 DECO_MULTIPLIER
) - 1.0f
)))) *
687 Context
->Frequency
) + 1;
688 length
[0] = NextPowerOf2(samples
);
689 totalLength
= length
[0];
690 for(index
= 0;index
< 4;index
++)
692 samples
= (ALuint
)(EARLY_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
693 length
[1 + index
] = NextPowerOf2(samples
);
694 totalLength
+= length
[1 + index
];
696 for(index
= 0;index
< 4;index
++)
698 samples
= (ALuint
)(ALLPASS_LINE_LENGTH
[index
] * Context
->Frequency
) + 1;
699 length
[5 + index
] = NextPowerOf2(samples
);
700 totalLength
+= length
[5 + index
];
702 for(index
= 0;index
< 4;index
++)
704 samples
= (ALuint
)(LATE_LINE_LENGTH
[index
] *
705 (1.0f
+ LATE_LINE_MULTIPLIER
) * Context
->Frequency
) + 1;
706 length
[9 + index
] = NextPowerOf2(samples
);
707 totalLength
+= length
[9 + index
];
710 // All lines share a single sample buffer and have their masks and start
711 // addresses calculated once.
712 State
->SampleBuffer
= malloc(totalLength
* sizeof(ALfloat
));
713 if(!State
->SampleBuffer
)
718 for(index
= 0; index
< totalLength
;index
++)
719 State
->SampleBuffer
[index
] = 0.0f
;
721 State
->LpCoeff
= 0.0f
;
722 State
->LpSamples
[0] = 0.0f
;
723 State
->LpSamples
[1] = 0.0f
;
724 State
->Delay
.Mask
= length
[0] - 1;
725 State
->Delay
.Line
= &State
->SampleBuffer
[0];
726 totalLength
= length
[0];
734 State
->Early
.Gain
= 0.0f
;
735 for(index
= 0;index
< 4;index
++)
737 State
->Early
.Coeff
[index
] = 0.0f
;
738 State
->Early
.Delay
[index
].Mask
= length
[1 + index
] - 1;
739 State
->Early
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
740 totalLength
+= length
[1 + index
];
742 // The early delay lines have their read offsets calculated once.
743 State
->Early
.Offset
[index
] = (ALuint
)(EARLY_LINE_LENGTH
[index
] *
747 State
->Late
.Gain
= 0.0f
;
748 State
->Late
.DensityGain
= 0.0f
;
749 State
->Late
.ApFeedCoeff
= 0.0f
;
750 State
->Late
.MixCoeff
= 0.0f
;
752 for(index
= 0;index
< 4;index
++)
754 State
->Late
.ApCoeff
[index
] = 0.0f
;
755 State
->Late
.ApDelay
[index
].Mask
= length
[5 + index
] - 1;
756 State
->Late
.ApDelay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
757 totalLength
+= length
[5 + index
];
759 // The late all-pass lines have their read offsets calculated once.
760 State
->Late
.ApOffset
[index
] = (ALuint
)(ALLPASS_LINE_LENGTH
[index
] *
764 for(index
= 0;index
< 4;index
++)
766 State
->Late
.Coeff
[index
] = 0.0f
;
767 State
->Late
.Delay
[index
].Mask
= length
[9 + index
] - 1;
768 State
->Late
.Delay
[index
].Line
= &State
->SampleBuffer
[totalLength
];
769 totalLength
+= length
[9 + index
];
771 State
->Late
.Offset
[index
] = 0;
773 State
->Late
.LpCoeff
[index
] = 0.0f
;
774 State
->Late
.LpSample
[index
] = 0.0f
;
777 // Panning is applied as an independent gain for each output channel.
778 for(index
= 0;index
< OUTPUTCHANNELS
;index
++)
780 State
->Early
.PanGain
[index
] = 0.0f
;
781 State
->Late
.PanGain
[index
] = 0.0f
;
785 return &State
->state
;
788 ALeffectState
*EAXVerbCreate(ALCcontext
*Context
)
790 ALeffectState
*State
= VerbCreate(Context
);
791 if(State
) State
->Process
= EAXVerbProcess
;