2 * OpenAL cross platform audio library
3 * Copyright (C) 1999-2007 by authors.
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.,
17 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
32 #include "alListener.h"
33 #include "alAuxEffectSlot.h"
37 #include "mastering.h"
38 #include "uhjfilter.h"
39 #include "bformatdec.h"
40 #include "static_assert.h"
41 #include "ringbuffer.h"
43 #include "fpu_modes.h"
45 #include "mixer_defs.h"
46 #include "bsinc_inc.h"
48 #include "backends/base.h"
51 extern inline ALfloat
minf(ALfloat a
, ALfloat b
);
52 extern inline ALfloat
maxf(ALfloat a
, ALfloat b
);
53 extern inline ALfloat
clampf(ALfloat val
, ALfloat min
, ALfloat max
);
55 extern inline ALdouble
mind(ALdouble a
, ALdouble b
);
56 extern inline ALdouble
maxd(ALdouble a
, ALdouble b
);
57 extern inline ALdouble
clampd(ALdouble val
, ALdouble min
, ALdouble max
);
59 extern inline ALuint
minu(ALuint a
, ALuint b
);
60 extern inline ALuint
maxu(ALuint a
, ALuint b
);
61 extern inline ALuint
clampu(ALuint val
, ALuint min
, ALuint max
);
63 extern inline ALint
mini(ALint a
, ALint b
);
64 extern inline ALint
maxi(ALint a
, ALint b
);
65 extern inline ALint
clampi(ALint val
, ALint min
, ALint max
);
67 extern inline ALint64
mini64(ALint64 a
, ALint64 b
);
68 extern inline ALint64
maxi64(ALint64 a
, ALint64 b
);
69 extern inline ALint64
clampi64(ALint64 val
, ALint64 min
, ALint64 max
);
71 extern inline ALuint64
minu64(ALuint64 a
, ALuint64 b
);
72 extern inline ALuint64
maxu64(ALuint64 a
, ALuint64 b
);
73 extern inline ALuint64
clampu64(ALuint64 val
, ALuint64 min
, ALuint64 max
);
75 extern inline size_t minz(size_t a
, size_t b
);
76 extern inline size_t maxz(size_t a
, size_t b
);
77 extern inline size_t clampz(size_t val
, size_t min
, size_t max
);
79 extern inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALfloat mu
);
80 extern inline ALfloat
cubic(ALfloat val1
, ALfloat val2
, ALfloat val3
, ALfloat val4
, ALfloat mu
);
82 extern inline void aluVectorSet(aluVector
*restrict vector
, ALfloat x
, ALfloat y
, ALfloat z
, ALfloat w
);
84 extern inline void aluMatrixfSetRow(aluMatrixf
*matrix
, ALuint row
,
85 ALfloat m0
, ALfloat m1
, ALfloat m2
, ALfloat m3
);
86 extern inline void aluMatrixfSet(aluMatrixf
*matrix
,
87 ALfloat m00
, ALfloat m01
, ALfloat m02
, ALfloat m03
,
88 ALfloat m10
, ALfloat m11
, ALfloat m12
, ALfloat m13
,
89 ALfloat m20
, ALfloat m21
, ALfloat m22
, ALfloat m23
,
90 ALfloat m30
, ALfloat m31
, ALfloat m32
, ALfloat m33
);
94 ALfloat ConeScale
= 1.0f
;
96 /* Localized Z scalar for mono sources */
97 ALfloat ZScale
= 1.0f
;
99 /* Force default speed of sound for distance-related reverb decay. */
100 ALboolean OverrideReverbSpeedOfSound
= AL_FALSE
;
102 const aluMatrixf IdentityMatrixf
= {{
103 { 1.0f
, 0.0f
, 0.0f
, 0.0f
},
104 { 0.0f
, 1.0f
, 0.0f
, 0.0f
},
105 { 0.0f
, 0.0f
, 1.0f
, 0.0f
},
106 { 0.0f
, 0.0f
, 0.0f
, 1.0f
},
111 enum Channel channel
;
116 static HrtfDirectMixerFunc MixDirectHrtf
= MixDirectHrtf_C
;
119 void DeinitVoice(ALvoice
*voice
)
121 al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice
->Update
, NULL
));
125 static inline HrtfDirectMixerFunc
SelectHrtfMixer(void)
128 if((CPUCapFlags
&CPU_CAP_NEON
))
129 return MixDirectHrtf_Neon
;
132 if((CPUCapFlags
&CPU_CAP_SSE
))
133 return MixDirectHrtf_SSE
;
136 return MixDirectHrtf_C
;
140 /* Prior to VS2013, MSVC lacks the round() family of functions. */
141 #if defined(_MSC_VER) && _MSC_VER < 1800
142 static float roundf(float val
)
145 return ceilf(val
-0.5f
);
146 return floorf(val
+0.5f
);
150 /* This RNG method was created based on the math found in opusdec. It's quick,
151 * and starting with a seed value of 22222, is suitable for generating
154 static inline ALuint
dither_rng(ALuint
*seed
)
156 *seed
= (*seed
* 96314165) + 907633515;
161 static inline void aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
163 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
164 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
165 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
168 static inline ALfloat
aluDotproduct(const aluVector
*vec1
, const aluVector
*vec2
)
170 return vec1
->v
[0]*vec2
->v
[0] + vec1
->v
[1]*vec2
->v
[1] + vec1
->v
[2]*vec2
->v
[2];
173 static ALfloat
aluNormalize(ALfloat
*vec
)
175 ALfloat length
= sqrtf(vec
[0]*vec
[0] + vec
[1]*vec
[1] + vec
[2]*vec
[2]);
178 ALfloat inv_length
= 1.0f
/length
;
179 vec
[0] *= inv_length
;
180 vec
[1] *= inv_length
;
181 vec
[2] *= inv_length
;
186 static void aluMatrixfFloat3(ALfloat
*vec
, ALfloat w
, const aluMatrixf
*mtx
)
188 ALfloat v
[4] = { vec
[0], vec
[1], vec
[2], w
};
190 vec
[0] = v
[0]*mtx
->m
[0][0] + v
[1]*mtx
->m
[1][0] + v
[2]*mtx
->m
[2][0] + v
[3]*mtx
->m
[3][0];
191 vec
[1] = v
[0]*mtx
->m
[0][1] + v
[1]*mtx
->m
[1][1] + v
[2]*mtx
->m
[2][1] + v
[3]*mtx
->m
[3][1];
192 vec
[2] = v
[0]*mtx
->m
[0][2] + v
[1]*mtx
->m
[1][2] + v
[2]*mtx
->m
[2][2] + v
[3]*mtx
->m
[3][2];
195 static aluVector
aluMatrixfVector(const aluMatrixf
*mtx
, const aluVector
*vec
)
198 v
.v
[0] = vec
->v
[0]*mtx
->m
[0][0] + vec
->v
[1]*mtx
->m
[1][0] + vec
->v
[2]*mtx
->m
[2][0] + vec
->v
[3]*mtx
->m
[3][0];
199 v
.v
[1] = vec
->v
[0]*mtx
->m
[0][1] + vec
->v
[1]*mtx
->m
[1][1] + vec
->v
[2]*mtx
->m
[2][1] + vec
->v
[3]*mtx
->m
[3][1];
200 v
.v
[2] = vec
->v
[0]*mtx
->m
[0][2] + vec
->v
[1]*mtx
->m
[1][2] + vec
->v
[2]*mtx
->m
[2][2] + vec
->v
[3]*mtx
->m
[3][2];
201 v
.v
[3] = vec
->v
[0]*mtx
->m
[0][3] + vec
->v
[1]*mtx
->m
[1][3] + vec
->v
[2]*mtx
->m
[2][3] + vec
->v
[3]*mtx
->m
[3][3];
208 MixDirectHrtf
= SelectHrtfMixer();
212 static void ProcessHrtf(ALCdevice
*device
, ALsizei SamplesToDo
)
214 DirectHrtfState
*state
;
219 ambiup_process(device
->AmbiUp
,
220 device
->Dry
.Buffer
, device
->Dry
.NumChannels
, device
->FOAOut
.Buffer
,
224 lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
225 ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
226 assert(lidx
!= -1 && ridx
!= -1);
228 state
= device
->Hrtf
;
229 for(c
= 0;c
< device
->Dry
.NumChannels
;c
++)
231 MixDirectHrtf(device
->RealOut
.Buffer
[lidx
], device
->RealOut
.Buffer
[ridx
],
232 device
->Dry
.Buffer
[c
], state
->Offset
, state
->IrSize
,
233 state
->Chan
[c
].Coeffs
, state
->Chan
[c
].Values
, SamplesToDo
236 state
->Offset
+= SamplesToDo
;
239 static void ProcessAmbiDec(ALCdevice
*device
, ALsizei SamplesToDo
)
241 if(device
->Dry
.Buffer
!= device
->FOAOut
.Buffer
)
242 bformatdec_upSample(device
->AmbiDecoder
,
243 device
->Dry
.Buffer
, device
->FOAOut
.Buffer
, device
->FOAOut
.NumChannels
,
246 bformatdec_process(device
->AmbiDecoder
,
247 device
->RealOut
.Buffer
, device
->RealOut
.NumChannels
, device
->Dry
.Buffer
,
252 static void ProcessAmbiUp(ALCdevice
*device
, ALsizei SamplesToDo
)
254 ambiup_process(device
->AmbiUp
,
255 device
->RealOut
.Buffer
, device
->RealOut
.NumChannels
, device
->FOAOut
.Buffer
,
260 static void ProcessUhj(ALCdevice
*device
, ALsizei SamplesToDo
)
262 int lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
263 int ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
264 if(LIKELY(lidx
!= -1 && ridx
!= -1))
266 /* Encode to stereo-compatible 2-channel UHJ output. */
267 EncodeUhj2(device
->Uhj_Encoder
,
268 device
->RealOut
.Buffer
[lidx
], device
->RealOut
.Buffer
[ridx
],
269 device
->Dry
.Buffer
, SamplesToDo
274 static void ProcessBs2b(ALCdevice
*device
, ALsizei SamplesToDo
)
276 int lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
277 int ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
278 if(LIKELY(lidx
!= -1 && ridx
!= -1))
280 /* Apply binaural/crossfeed filter */
281 bs2b_cross_feed(device
->Bs2b
, device
->RealOut
.Buffer
[lidx
],
282 device
->RealOut
.Buffer
[ridx
], SamplesToDo
);
286 void aluSelectPostProcess(ALCdevice
*device
)
288 if(device
->HrtfHandle
)
289 device
->PostProcess
= ProcessHrtf
;
290 else if(device
->AmbiDecoder
)
291 device
->PostProcess
= ProcessAmbiDec
;
292 else if(device
->AmbiUp
)
293 device
->PostProcess
= ProcessAmbiUp
;
294 else if(device
->Uhj_Encoder
)
295 device
->PostProcess
= ProcessUhj
;
296 else if(device
->Bs2b
)
297 device
->PostProcess
= ProcessBs2b
;
299 device
->PostProcess
= NULL
;
303 /* Prepares the interpolator for a given rate (determined by increment). A
304 * result of AL_FALSE indicates that the filter output will completely cut
307 * With a bit of work, and a trade of memory for CPU cost, this could be
308 * modified for use with an interpolated increment for buttery-smooth pitch
311 void BsincPrepare(const ALuint increment
, BsincState
*state
, const BSincTable
*table
)
316 if(increment
> FRACTIONONE
)
318 sf
= (ALfloat
)FRACTIONONE
/ increment
;
319 sf
= maxf(0.0f
, (BSINC_SCALE_COUNT
-1) * (sf
-table
->scaleBase
) * table
->scaleRange
);
321 /* The interpolation factor is fit to this diagonally-symmetric curve
322 * to reduce the transition ripple caused by interpolating different
323 * scales of the sinc function.
325 sf
= 1.0f
- cosf(asinf(sf
- si
));
330 si
= BSINC_SCALE_COUNT
- 1;
334 state
->m
= table
->m
[si
];
335 state
->l
= -((state
->m
/2) - 1);
336 state
->filter
= table
->Tab
+ table
->filterOffset
[si
];
340 static bool CalcContextParams(ALCcontext
*Context
)
342 ALlistener
*Listener
= Context
->Listener
;
343 struct ALcontextProps
*props
;
345 props
= ATOMIC_EXCHANGE_PTR(&Context
->Update
, NULL
, almemory_order_acq_rel
);
346 if(!props
) return false;
348 Listener
->Params
.MetersPerUnit
= props
->MetersPerUnit
;
350 Listener
->Params
.DopplerFactor
= props
->DopplerFactor
;
351 Listener
->Params
.SpeedOfSound
= props
->SpeedOfSound
* props
->DopplerVelocity
;
352 if(!OverrideReverbSpeedOfSound
)
353 Listener
->Params
.ReverbSpeedOfSound
= Listener
->Params
.SpeedOfSound
*
354 Listener
->Params
.MetersPerUnit
;
356 Listener
->Params
.SourceDistanceModel
= props
->SourceDistanceModel
;
357 Listener
->Params
.DistanceModel
= props
->DistanceModel
;
359 ATOMIC_REPLACE_HEAD(struct ALcontextProps
*, &Context
->FreeContextProps
, props
);
363 static bool CalcListenerParams(ALCcontext
*Context
)
365 ALlistener
*Listener
= Context
->Listener
;
366 ALfloat N
[3], V
[3], U
[3], P
[3];
367 struct ALlistenerProps
*props
;
370 props
= ATOMIC_EXCHANGE_PTR(&Listener
->Update
, NULL
, almemory_order_acq_rel
);
371 if(!props
) return false;
374 N
[0] = props
->Forward
[0];
375 N
[1] = props
->Forward
[1];
376 N
[2] = props
->Forward
[2];
382 /* Build and normalize right-vector */
383 aluCrossproduct(N
, V
, U
);
386 aluMatrixfSet(&Listener
->Params
.Matrix
,
387 U
[0], V
[0], -N
[0], 0.0,
388 U
[1], V
[1], -N
[1], 0.0,
389 U
[2], V
[2], -N
[2], 0.0,
393 P
[0] = props
->Position
[0];
394 P
[1] = props
->Position
[1];
395 P
[2] = props
->Position
[2];
396 aluMatrixfFloat3(P
, 1.0, &Listener
->Params
.Matrix
);
397 aluMatrixfSetRow(&Listener
->Params
.Matrix
, 3, -P
[0], -P
[1], -P
[2], 1.0f
);
399 aluVectorSet(&vel
, props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
);
400 Listener
->Params
.Velocity
= aluMatrixfVector(&Listener
->Params
.Matrix
, &vel
);
402 Listener
->Params
.Gain
= props
->Gain
* Context
->GainBoost
;
404 ATOMIC_REPLACE_HEAD(struct ALlistenerProps
*, &Context
->FreeListenerProps
, props
);
408 static bool CalcEffectSlotParams(ALeffectslot
*slot
, ALCcontext
*context
, bool force
)
410 struct ALeffectslotProps
*props
;
411 ALeffectState
*state
;
413 props
= ATOMIC_EXCHANGE_PTR(&slot
->Update
, NULL
, almemory_order_acq_rel
);
414 if(!props
&& !force
) return false;
418 slot
->Params
.Gain
= props
->Gain
;
419 slot
->Params
.AuxSendAuto
= props
->AuxSendAuto
;
420 slot
->Params
.EffectType
= props
->Type
;
421 slot
->Params
.EffectProps
= props
->Props
;
422 if(IsReverbEffect(props
->Type
))
424 slot
->Params
.RoomRolloff
= props
->Props
.Reverb
.RoomRolloffFactor
;
425 slot
->Params
.DecayTime
= props
->Props
.Reverb
.DecayTime
;
426 slot
->Params
.DecayHFRatio
= props
->Props
.Reverb
.DecayHFRatio
;
427 slot
->Params
.DecayHFLimit
= props
->Props
.Reverb
.DecayHFLimit
;
428 slot
->Params
.AirAbsorptionGainHF
= props
->Props
.Reverb
.AirAbsorptionGainHF
;
432 slot
->Params
.RoomRolloff
= 0.0f
;
433 slot
->Params
.DecayTime
= 0.0f
;
434 slot
->Params
.DecayHFRatio
= 0.0f
;
435 slot
->Params
.DecayHFLimit
= AL_FALSE
;
436 slot
->Params
.AirAbsorptionGainHF
= 1.0f
;
439 /* Swap effect states. No need to play with the ref counts since they
440 * keep the same number of refs.
442 state
= props
->State
;
443 props
->State
= slot
->Params
.EffectState
;
444 slot
->Params
.EffectState
= state
;
446 ATOMIC_REPLACE_HEAD(struct ALeffectslotProps
*, &context
->FreeEffectslotProps
, props
);
449 state
= slot
->Params
.EffectState
;
451 V(state
,update
)(context
, slot
, &slot
->Params
.EffectProps
);
456 static const struct ChanMap MonoMap
[1] = {
457 { FrontCenter
, 0.0f
, 0.0f
}
459 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
460 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) }
462 { FrontLeft
, DEG2RAD( -45.0f
), DEG2RAD(0.0f
) },
463 { FrontRight
, DEG2RAD( 45.0f
), DEG2RAD(0.0f
) },
464 { BackLeft
, DEG2RAD(-135.0f
), DEG2RAD(0.0f
) },
465 { BackRight
, DEG2RAD( 135.0f
), DEG2RAD(0.0f
) }
467 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
468 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
469 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
471 { SideLeft
, DEG2RAD(-110.0f
), DEG2RAD(0.0f
) },
472 { SideRight
, DEG2RAD( 110.0f
), DEG2RAD(0.0f
) }
474 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
475 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
476 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
478 { BackCenter
, DEG2RAD(180.0f
), DEG2RAD(0.0f
) },
479 { SideLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
480 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
482 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
483 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
484 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
486 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
487 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) },
488 { SideLeft
, DEG2RAD( -90.0f
), DEG2RAD(0.0f
) },
489 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
492 static void CalcPanningAndFilters(ALvoice
*voice
, const ALfloat Distance
, const ALfloat
*Dir
,
493 const ALfloat Spread
, const ALfloat DryGain
,
494 const ALfloat DryGainHF
, const ALfloat DryGainLF
,
495 const ALfloat
*WetGain
, const ALfloat
*WetGainLF
,
496 const ALfloat
*WetGainHF
, ALeffectslot
**SendSlots
,
497 const ALbuffer
*Buffer
, const struct ALvoiceProps
*props
,
498 const ALlistener
*Listener
, const ALCdevice
*Device
)
500 struct ChanMap StereoMap
[2] = {
501 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
502 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) }
504 bool DirectChannels
= props
->DirectChannels
;
505 const ALsizei NumSends
= Device
->NumAuxSends
;
506 const ALuint Frequency
= Device
->Frequency
;
507 const struct ChanMap
*chans
= NULL
;
508 ALsizei num_channels
= 0;
509 bool isbformat
= false;
510 ALfloat downmix_gain
= 1.0f
;
513 switch(Buffer
->FmtChannels
)
518 /* Mono buffers are never played direct. */
519 DirectChannels
= false;
523 /* Convert counter-clockwise to clockwise. */
524 StereoMap
[0].angle
= -props
->StereoPan
[0];
525 StereoMap
[1].angle
= -props
->StereoPan
[1];
529 downmix_gain
= 1.0f
/ 2.0f
;
535 downmix_gain
= 1.0f
/ 2.0f
;
541 downmix_gain
= 1.0f
/ 4.0f
;
547 /* NOTE: Excludes LFE. */
548 downmix_gain
= 1.0f
/ 5.0f
;
554 /* NOTE: Excludes LFE. */
555 downmix_gain
= 1.0f
/ 6.0f
;
561 /* NOTE: Excludes LFE. */
562 downmix_gain
= 1.0f
/ 7.0f
;
568 DirectChannels
= false;
574 DirectChannels
= false;
578 voice
->Flags
&= ~(VOICE_HAS_HRTF
| VOICE_HAS_NFC
);
581 /* Special handling for B-Format sources. */
583 if(Distance
> FLT_EPSILON
)
585 /* Panning a B-Format sound toward some direction is easy. Just pan
586 * the first (W) channel as a normal mono sound and silence the
589 ALfloat coeffs
[MAX_AMBI_COEFFS
];
591 if(Device
->AvgSpeakerDist
> 0.0f
)
593 ALfloat mdist
= Distance
* Listener
->Params
.MetersPerUnit
;
594 ALfloat w0
= SPEEDOFSOUNDMETRESPERSEC
/
595 (mdist
* (ALfloat
)Device
->Frequency
);
596 ALfloat w1
= SPEEDOFSOUNDMETRESPERSEC
/
597 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
598 /* Clamp w0 for really close distances, to prevent excessive
601 w0
= minf(w0
, w1
*4.0f
);
603 /* Only need to adjust the first channel of a B-Format source. */
604 NfcFilterAdjust(&voice
->Direct
.Params
[0].NFCtrlFilter
, w0
);
606 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
607 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
608 voice
->Flags
|= VOICE_HAS_NFC
;
611 if(Device
->Render_Mode
== StereoPair
)
613 ALfloat ev
= asinf(Dir
[1]);
614 ALfloat az
= atan2f(Dir
[0], -Dir
[2]);
615 CalcAnglePairwiseCoeffs(az
, ev
, Spread
, coeffs
);
618 CalcDirectionCoeffs(Dir
, Spread
, coeffs
);
620 /* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */
621 ComputeDryPanGains(&Device
->Dry
, coeffs
, DryGain
*1.414213562f
,
622 voice
->Direct
.Params
[0].Gains
.Target
);
623 for(c
= 1;c
< num_channels
;c
++)
625 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
626 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
629 for(i
= 0;i
< NumSends
;i
++)
631 const ALeffectslot
*Slot
= SendSlots
[i
];
633 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
634 coeffs
, WetGain
[i
]*1.414213562f
, voice
->Send
[i
].Params
[0].Gains
.Target
637 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
638 voice
->Send
[i
].Params
[0].Gains
.Target
[j
] = 0.0f
;
639 for(c
= 1;c
< num_channels
;c
++)
641 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
642 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
648 /* Local B-Format sources have their XYZ channels rotated according
649 * to the orientation.
651 ALfloat N
[3], V
[3], U
[3];
655 if(Device
->AvgSpeakerDist
> 0.0f
)
657 /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
658 * is what we want for FOA input. The first channel may have
659 * been previously re-adjusted if panned, so reset it.
661 NfcFilterAdjust(&voice
->Direct
.Params
[0].NFCtrlFilter
, 0.0f
);
663 voice
->Direct
.ChannelsPerOrder
[0] = 1;
664 voice
->Direct
.ChannelsPerOrder
[1] = mini(voice
->Direct
.Channels
-1, 3);
665 for(i
= 2;i
< MAX_AMBI_ORDER
+1;i
++)
666 voice
->Direct
.ChannelsPerOrder
[i
] = 0;
667 voice
->Flags
|= VOICE_HAS_NFC
;
671 N
[0] = props
->Orientation
[0][0];
672 N
[1] = props
->Orientation
[0][1];
673 N
[2] = props
->Orientation
[0][2];
675 V
[0] = props
->Orientation
[1][0];
676 V
[1] = props
->Orientation
[1][1];
677 V
[2] = props
->Orientation
[1][2];
679 if(!props
->HeadRelative
)
681 const aluMatrixf
*lmatrix
= &Listener
->Params
.Matrix
;
682 aluMatrixfFloat3(N
, 0.0f
, lmatrix
);
683 aluMatrixfFloat3(V
, 0.0f
, lmatrix
);
685 /* Build and normalize right-vector */
686 aluCrossproduct(N
, V
, U
);
689 /* Build a rotate + conversion matrix (FuMa -> ACN+N3D). */
690 scale
= 1.732050808f
;
691 aluMatrixfSet(&matrix
,
692 1.414213562f
, 0.0f
, 0.0f
, 0.0f
,
693 0.0f
, -N
[0]*scale
, N
[1]*scale
, -N
[2]*scale
,
694 0.0f
, U
[0]*scale
, -U
[1]*scale
, U
[2]*scale
,
695 0.0f
, -V
[0]*scale
, V
[1]*scale
, -V
[2]*scale
698 voice
->Direct
.Buffer
= Device
->FOAOut
.Buffer
;
699 voice
->Direct
.Channels
= Device
->FOAOut
.NumChannels
;
700 for(c
= 0;c
< num_channels
;c
++)
701 ComputeFirstOrderGains(&Device
->FOAOut
, matrix
.m
[c
], DryGain
,
702 voice
->Direct
.Params
[c
].Gains
.Target
);
703 for(i
= 0;i
< NumSends
;i
++)
705 const ALeffectslot
*Slot
= SendSlots
[i
];
708 for(c
= 0;c
< num_channels
;c
++)
709 ComputeFirstOrderGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
710 matrix
.m
[c
], WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
715 for(c
= 0;c
< num_channels
;c
++)
716 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
717 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
722 else if(DirectChannels
)
724 /* Direct source channels always play local. Skip the virtual channels
725 * and write inputs to the matching real outputs.
727 voice
->Direct
.Buffer
= Device
->RealOut
.Buffer
;
728 voice
->Direct
.Channels
= Device
->RealOut
.NumChannels
;
730 for(c
= 0;c
< num_channels
;c
++)
733 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
734 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
735 if((idx
=GetChannelIdxByName(&Device
->RealOut
, chans
[c
].channel
)) != -1)
736 voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
739 /* Auxiliary sends still use normal channel panning since they mix to
740 * B-Format, which can't channel-match.
742 for(c
= 0;c
< num_channels
;c
++)
744 ALfloat coeffs
[MAX_AMBI_COEFFS
];
745 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, 0.0f
, coeffs
);
747 for(i
= 0;i
< NumSends
;i
++)
749 const ALeffectslot
*Slot
= SendSlots
[i
];
751 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
752 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
755 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
756 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
760 else if(Device
->Render_Mode
== HrtfRender
)
762 /* Full HRTF rendering. Skip the virtual channels and render to the
765 voice
->Direct
.Buffer
= Device
->RealOut
.Buffer
;
766 voice
->Direct
.Channels
= Device
->RealOut
.NumChannels
;
768 if(Distance
> FLT_EPSILON
)
770 ALfloat coeffs
[MAX_AMBI_COEFFS
];
774 az
= atan2f(Dir
[0], -Dir
[2]);
776 /* Get the HRIR coefficients and delays just once, for the given
779 GetHrtfCoeffs(Device
->HrtfHandle
, ev
, az
, Spread
,
780 voice
->Direct
.Params
[0].Hrtf
.Target
.Coeffs
,
781 voice
->Direct
.Params
[0].Hrtf
.Target
.Delay
);
782 voice
->Direct
.Params
[0].Hrtf
.Target
.Gain
= DryGain
* downmix_gain
;
784 /* Remaining channels use the same results as the first. */
785 for(c
= 1;c
< num_channels
;c
++)
788 if(chans
[c
].channel
== LFE
)
789 memset(&voice
->Direct
.Params
[c
].Hrtf
.Target
, 0,
790 sizeof(voice
->Direct
.Params
[c
].Hrtf
.Target
));
792 voice
->Direct
.Params
[c
].Hrtf
.Target
= voice
->Direct
.Params
[0].Hrtf
.Target
;
795 /* Calculate the directional coefficients once, which apply to all
796 * input channels of the source sends.
798 CalcDirectionCoeffs(Dir
, Spread
, coeffs
);
800 for(i
= 0;i
< NumSends
;i
++)
802 const ALeffectslot
*Slot
= SendSlots
[i
];
804 for(c
= 0;c
< num_channels
;c
++)
807 if(chans
[c
].channel
== LFE
)
808 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
809 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
811 ComputePanningGainsBF(Slot
->ChanMap
,
812 Slot
->NumChannels
, coeffs
, WetGain
[i
] * downmix_gain
,
813 voice
->Send
[i
].Params
[c
].Gains
.Target
817 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
818 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
823 /* Local sources on HRTF play with each channel panned to its
824 * relative location around the listener, providing "virtual
825 * speaker" responses.
827 for(c
= 0;c
< num_channels
;c
++)
829 ALfloat coeffs
[MAX_AMBI_COEFFS
];
831 if(chans
[c
].channel
== LFE
)
834 memset(&voice
->Direct
.Params
[c
].Hrtf
.Target
, 0,
835 sizeof(voice
->Direct
.Params
[c
].Hrtf
.Target
));
836 for(i
= 0;i
< NumSends
;i
++)
838 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
839 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
844 /* Get the HRIR coefficients and delays for this channel
847 GetHrtfCoeffs(Device
->HrtfHandle
,
848 chans
[c
].elevation
, chans
[c
].angle
, Spread
,
849 voice
->Direct
.Params
[c
].Hrtf
.Target
.Coeffs
,
850 voice
->Direct
.Params
[c
].Hrtf
.Target
.Delay
852 voice
->Direct
.Params
[c
].Hrtf
.Target
.Gain
= DryGain
;
854 /* Normal panning for auxiliary sends. */
855 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
857 for(i
= 0;i
< NumSends
;i
++)
859 const ALeffectslot
*Slot
= SendSlots
[i
];
861 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
862 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
865 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
866 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
871 voice
->Flags
|= VOICE_HAS_HRTF
;
875 /* Non-HRTF rendering. Use normal panning to the output. */
877 if(Distance
> FLT_EPSILON
)
879 ALfloat coeffs
[MAX_AMBI_COEFFS
];
882 /* Calculate NFC filter coefficient if needed. */
883 if(Device
->AvgSpeakerDist
> 0.0f
)
885 ALfloat mdist
= Distance
* Listener
->Params
.MetersPerUnit
;
886 ALfloat w1
= SPEEDOFSOUNDMETRESPERSEC
/
887 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
888 w0
= SPEEDOFSOUNDMETRESPERSEC
/
889 (mdist
* (ALfloat
)Device
->Frequency
);
890 /* Clamp w0 for really close distances, to prevent excessive
893 w0
= minf(w0
, w1
*4.0f
);
895 /* Adjust NFC filters. */
896 for(c
= 0;c
< num_channels
;c
++)
897 NfcFilterAdjust(&voice
->Direct
.Params
[c
].NFCtrlFilter
, w0
);
899 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
900 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
901 voice
->Flags
|= VOICE_HAS_NFC
;
904 /* Calculate the directional coefficients once, which apply to all
907 if(Device
->Render_Mode
== StereoPair
)
909 ALfloat ev
= asinf(Dir
[1]);
910 ALfloat az
= atan2f(Dir
[0], -Dir
[2]);
911 CalcAnglePairwiseCoeffs(az
, ev
, Spread
, coeffs
);
914 CalcDirectionCoeffs(Dir
, Spread
, coeffs
);
916 for(c
= 0;c
< num_channels
;c
++)
918 /* Special-case LFE */
919 if(chans
[c
].channel
== LFE
)
921 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
922 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
923 if(Device
->Dry
.Buffer
== Device
->RealOut
.Buffer
)
925 int idx
= GetChannelIdxByName(&Device
->RealOut
, chans
[c
].channel
);
926 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
931 ComputeDryPanGains(&Device
->Dry
,
932 coeffs
, DryGain
* downmix_gain
, voice
->Direct
.Params
[c
].Gains
.Target
936 for(i
= 0;i
< NumSends
;i
++)
938 const ALeffectslot
*Slot
= SendSlots
[i
];
940 for(c
= 0;c
< num_channels
;c
++)
943 if(chans
[c
].channel
== LFE
)
944 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
945 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
947 ComputePanningGainsBF(Slot
->ChanMap
,
948 Slot
->NumChannels
, coeffs
, WetGain
[i
] * downmix_gain
,
949 voice
->Send
[i
].Params
[c
].Gains
.Target
953 for(c
= 0;c
< num_channels
;c
++)
955 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
956 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
964 if(Device
->AvgSpeakerDist
> 0.0f
)
966 /* If the source distance is 0, set w0 to w1 to act as a pass-
967 * through. We still want to pass the signal through the
968 * filters so they keep an appropriate history, in case the
969 * source moves away from the listener.
971 w0
= SPEEDOFSOUNDMETRESPERSEC
/
972 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
974 for(c
= 0;c
< num_channels
;c
++)
975 NfcFilterAdjust(&voice
->Direct
.Params
[c
].NFCtrlFilter
, w0
);
977 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
978 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
979 voice
->Flags
|= VOICE_HAS_NFC
;
982 for(c
= 0;c
< num_channels
;c
++)
984 ALfloat coeffs
[MAX_AMBI_COEFFS
];
986 /* Special-case LFE */
987 if(chans
[c
].channel
== LFE
)
989 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
990 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
991 if(Device
->Dry
.Buffer
== Device
->RealOut
.Buffer
)
993 int idx
= GetChannelIdxByName(&Device
->RealOut
, chans
[c
].channel
);
994 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
997 for(i
= 0;i
< NumSends
;i
++)
999 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
1000 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
1005 if(Device
->Render_Mode
== StereoPair
)
1006 CalcAnglePairwiseCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
1008 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
1009 ComputeDryPanGains(&Device
->Dry
,
1010 coeffs
, DryGain
, voice
->Direct
.Params
[c
].Gains
.Target
1013 for(i
= 0;i
< NumSends
;i
++)
1015 const ALeffectslot
*Slot
= SendSlots
[i
];
1017 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
1018 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
1021 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
1022 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
1029 ALfloat hfScale
= props
->Direct
.HFReference
/ Frequency
;
1030 ALfloat lfScale
= props
->Direct
.LFReference
/ Frequency
;
1031 ALfloat gainHF
= maxf(DryGainHF
, 0.001f
); /* Limit -60dB */
1032 ALfloat gainLF
= maxf(DryGainLF
, 0.001f
);
1034 voice
->Direct
.FilterType
= AF_None
;
1035 if(gainHF
!= 1.0f
) voice
->Direct
.FilterType
|= AF_LowPass
;
1036 if(gainLF
!= 1.0f
) voice
->Direct
.FilterType
|= AF_HighPass
;
1037 ALfilterState_setParams(
1038 &voice
->Direct
.Params
[0].LowPass
, ALfilterType_HighShelf
,
1039 gainHF
, hfScale
, calc_rcpQ_from_slope(gainHF
, 1.0f
)
1041 ALfilterState_setParams(
1042 &voice
->Direct
.Params
[0].HighPass
, ALfilterType_LowShelf
,
1043 gainLF
, lfScale
, calc_rcpQ_from_slope(gainLF
, 1.0f
)
1045 for(c
= 1;c
< num_channels
;c
++)
1047 ALfilterState_copyParams(&voice
->Direct
.Params
[c
].LowPass
,
1048 &voice
->Direct
.Params
[0].LowPass
);
1049 ALfilterState_copyParams(&voice
->Direct
.Params
[c
].HighPass
,
1050 &voice
->Direct
.Params
[0].HighPass
);
1053 for(i
= 0;i
< NumSends
;i
++)
1055 ALfloat hfScale
= props
->Send
[i
].HFReference
/ Frequency
;
1056 ALfloat lfScale
= props
->Send
[i
].LFReference
/ Frequency
;
1057 ALfloat gainHF
= maxf(WetGainHF
[i
], 0.001f
);
1058 ALfloat gainLF
= maxf(WetGainLF
[i
], 0.001f
);
1060 voice
->Send
[i
].FilterType
= AF_None
;
1061 if(gainHF
!= 1.0f
) voice
->Send
[i
].FilterType
|= AF_LowPass
;
1062 if(gainLF
!= 1.0f
) voice
->Send
[i
].FilterType
|= AF_HighPass
;
1063 ALfilterState_setParams(
1064 &voice
->Send
[i
].Params
[0].LowPass
, ALfilterType_HighShelf
,
1065 gainHF
, hfScale
, calc_rcpQ_from_slope(gainHF
, 1.0f
)
1067 ALfilterState_setParams(
1068 &voice
->Send
[i
].Params
[0].HighPass
, ALfilterType_LowShelf
,
1069 gainLF
, lfScale
, calc_rcpQ_from_slope(gainLF
, 1.0f
)
1071 for(c
= 1;c
< num_channels
;c
++)
1073 ALfilterState_copyParams(&voice
->Send
[i
].Params
[c
].LowPass
,
1074 &voice
->Send
[i
].Params
[0].LowPass
);
1075 ALfilterState_copyParams(&voice
->Send
[i
].Params
[c
].HighPass
,
1076 &voice
->Send
[i
].Params
[0].HighPass
);
1081 static void CalcNonAttnSourceParams(ALvoice
*voice
, const struct ALvoiceProps
*props
, const ALbuffer
*ALBuffer
, const ALCcontext
*ALContext
)
1083 static const ALfloat dir
[3] = { 0.0f
, 0.0f
, -1.0f
};
1084 const ALCdevice
*Device
= ALContext
->Device
;
1085 const ALlistener
*Listener
= ALContext
->Listener
;
1086 ALfloat DryGain
, DryGainHF
, DryGainLF
;
1087 ALfloat WetGain
[MAX_SENDS
];
1088 ALfloat WetGainHF
[MAX_SENDS
];
1089 ALfloat WetGainLF
[MAX_SENDS
];
1090 ALeffectslot
*SendSlots
[MAX_SENDS
];
1094 voice
->Direct
.Buffer
= Device
->Dry
.Buffer
;
1095 voice
->Direct
.Channels
= Device
->Dry
.NumChannels
;
1096 for(i
= 0;i
< Device
->NumAuxSends
;i
++)
1098 SendSlots
[i
] = props
->Send
[i
].Slot
;
1099 if(!SendSlots
[i
] && i
== 0)
1100 SendSlots
[i
] = ALContext
->DefaultSlot
;
1101 if(!SendSlots
[i
] || SendSlots
[i
]->Params
.EffectType
== AL_EFFECT_NULL
)
1103 SendSlots
[i
] = NULL
;
1104 voice
->Send
[i
].Buffer
= NULL
;
1105 voice
->Send
[i
].Channels
= 0;
1109 voice
->Send
[i
].Buffer
= SendSlots
[i
]->WetBuffer
;
1110 voice
->Send
[i
].Channels
= SendSlots
[i
]->NumChannels
;
1114 /* Calculate the stepping value */
1115 Pitch
= (ALfloat
)ALBuffer
->Frequency
/(ALfloat
)Device
->Frequency
* props
->Pitch
;
1116 if(Pitch
> (ALfloat
)MAX_PITCH
)
1117 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1119 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1120 if(props
->Resampler
== BSinc24Resampler
)
1121 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc24
);
1122 else if(props
->Resampler
== BSinc12Resampler
)
1123 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc12
);
1124 voice
->Resampler
= SelectResampler(props
->Resampler
);
1126 /* Calculate gains */
1127 DryGain
= clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
);
1128 DryGain
*= props
->Direct
.Gain
* Listener
->Params
.Gain
;
1129 DryGain
= minf(DryGain
, GAIN_MIX_MAX
);
1130 DryGainHF
= props
->Direct
.GainHF
;
1131 DryGainLF
= props
->Direct
.GainLF
;
1132 for(i
= 0;i
< Device
->NumAuxSends
;i
++)
1134 WetGain
[i
] = clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
);
1135 WetGain
[i
] *= props
->Send
[i
].Gain
* Listener
->Params
.Gain
;
1136 WetGain
[i
] = minf(WetGain
[i
], GAIN_MIX_MAX
);
1137 WetGainHF
[i
] = props
->Send
[i
].GainHF
;
1138 WetGainLF
[i
] = props
->Send
[i
].GainLF
;
1141 CalcPanningAndFilters(voice
, 0.0f
, dir
, 0.0f
, DryGain
, DryGainHF
, DryGainLF
, WetGain
,
1142 WetGainLF
, WetGainHF
, SendSlots
, ALBuffer
, props
, Listener
, Device
);
1145 static void CalcAttnSourceParams(ALvoice
*voice
, const struct ALvoiceProps
*props
, const ALbuffer
*ALBuffer
, const ALCcontext
*ALContext
)
1147 const ALCdevice
*Device
= ALContext
->Device
;
1148 const ALlistener
*Listener
= ALContext
->Listener
;
1149 const ALsizei NumSends
= Device
->NumAuxSends
;
1150 aluVector Position
, Velocity
, Direction
, SourceToListener
;
1151 ALfloat Distance
, ClampedDist
, DopplerFactor
;
1152 ALeffectslot
*SendSlots
[MAX_SENDS
];
1153 ALfloat RoomRolloff
[MAX_SENDS
];
1154 ALfloat DecayDistance
[MAX_SENDS
];
1155 ALfloat DecayHFDistance
[MAX_SENDS
];
1156 ALfloat DryGain
, DryGainHF
, DryGainLF
;
1157 ALfloat WetGain
[MAX_SENDS
];
1158 ALfloat WetGainHF
[MAX_SENDS
];
1159 ALfloat WetGainLF
[MAX_SENDS
];
1166 /* Set mixing buffers and get send parameters. */
1167 voice
->Direct
.Buffer
= Device
->Dry
.Buffer
;
1168 voice
->Direct
.Channels
= Device
->Dry
.NumChannels
;
1169 for(i
= 0;i
< NumSends
;i
++)
1171 SendSlots
[i
] = props
->Send
[i
].Slot
;
1172 if(!SendSlots
[i
] && i
== 0)
1173 SendSlots
[i
] = ALContext
->DefaultSlot
;
1174 if(!SendSlots
[i
] || SendSlots
[i
]->Params
.EffectType
== AL_EFFECT_NULL
)
1176 SendSlots
[i
] = NULL
;
1177 RoomRolloff
[i
] = 0.0f
;
1178 DecayDistance
[i
] = 0.0f
;
1179 DecayHFDistance
[i
] = 0.0f
;
1181 else if(SendSlots
[i
]->Params
.AuxSendAuto
)
1183 RoomRolloff
[i
] = SendSlots
[i
]->Params
.RoomRolloff
+ props
->RoomRolloffFactor
;
1184 DecayDistance
[i
] = SendSlots
[i
]->Params
.DecayTime
*
1185 Listener
->Params
.ReverbSpeedOfSound
;
1186 DecayHFDistance
[i
] = DecayDistance
[i
] * SendSlots
[i
]->Params
.DecayHFRatio
;
1187 if(SendSlots
[i
]->Params
.DecayHFLimit
)
1189 ALfloat airAbsorption
= SendSlots
[i
]->Params
.AirAbsorptionGainHF
;
1190 if(airAbsorption
< 1.0f
)
1192 ALfloat limitRatio
= log10f(REVERB_DECAY_GAIN
) / log10f(airAbsorption
);
1193 DecayHFDistance
[i
] = minf(limitRatio
, DecayHFDistance
[i
]);
1199 /* If the slot's auxiliary send auto is off, the data sent to the
1200 * effect slot is the same as the dry path, sans filter effects */
1201 RoomRolloff
[i
] = props
->RolloffFactor
;
1202 DecayDistance
[i
] = 0.0f
;
1203 DecayHFDistance
[i
] = 0.0f
;
1208 voice
->Send
[i
].Buffer
= NULL
;
1209 voice
->Send
[i
].Channels
= 0;
1213 voice
->Send
[i
].Buffer
= SendSlots
[i
]->WetBuffer
;
1214 voice
->Send
[i
].Channels
= SendSlots
[i
]->NumChannels
;
1218 /* Transform source to listener space (convert to head relative) */
1219 aluVectorSet(&Position
, props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
);
1220 aluVectorSet(&Direction
, props
->Direction
[0], props
->Direction
[1], props
->Direction
[2], 0.0f
);
1221 aluVectorSet(&Velocity
, props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
);
1222 if(props
->HeadRelative
== AL_FALSE
)
1224 const aluMatrixf
*Matrix
= &Listener
->Params
.Matrix
;
1225 /* Transform source vectors */
1226 Position
= aluMatrixfVector(Matrix
, &Position
);
1227 Velocity
= aluMatrixfVector(Matrix
, &Velocity
);
1228 Direction
= aluMatrixfVector(Matrix
, &Direction
);
1232 const aluVector
*lvelocity
= &Listener
->Params
.Velocity
;
1233 /* Offset the source velocity to be relative of the listener velocity */
1234 Velocity
.v
[0] += lvelocity
->v
[0];
1235 Velocity
.v
[1] += lvelocity
->v
[1];
1236 Velocity
.v
[2] += lvelocity
->v
[2];
1239 directional
= aluNormalize(Direction
.v
) > FLT_EPSILON
;
1240 SourceToListener
.v
[0] = -Position
.v
[0];
1241 SourceToListener
.v
[1] = -Position
.v
[1];
1242 SourceToListener
.v
[2] = -Position
.v
[2];
1243 SourceToListener
.v
[3] = 0.0f
;
1244 Distance
= aluNormalize(SourceToListener
.v
);
1246 /* Initial source gain */
1247 DryGain
= props
->Gain
;
1250 for(i
= 0;i
< NumSends
;i
++)
1252 WetGain
[i
] = props
->Gain
;
1253 WetGainHF
[i
] = 1.0f
;
1254 WetGainLF
[i
] = 1.0f
;
1257 /* Calculate distance attenuation */
1258 ClampedDist
= Distance
;
1260 switch(Listener
->Params
.SourceDistanceModel
?
1261 props
->DistanceModel
: Listener
->Params
.DistanceModel
)
1263 case InverseDistanceClamped
:
1264 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1265 if(props
->MaxDistance
< props
->RefDistance
)
1268 case InverseDistance
:
1269 if(!(props
->RefDistance
> 0.0f
))
1270 ClampedDist
= props
->RefDistance
;
1273 ALfloat dist
= lerp(props
->RefDistance
, ClampedDist
, props
->RolloffFactor
);
1274 if(dist
> 0.0f
) DryGain
*= props
->RefDistance
/ dist
;
1275 for(i
= 0;i
< NumSends
;i
++)
1277 dist
= lerp(props
->RefDistance
, ClampedDist
, RoomRolloff
[i
]);
1278 if(dist
> 0.0f
) WetGain
[i
] *= props
->RefDistance
/ dist
;
1283 case LinearDistanceClamped
:
1284 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1285 if(props
->MaxDistance
< props
->RefDistance
)
1288 case LinearDistance
:
1289 if(!(props
->MaxDistance
!= props
->RefDistance
))
1290 ClampedDist
= props
->RefDistance
;
1293 ALfloat attn
= props
->RolloffFactor
* (ClampedDist
-props
->RefDistance
) /
1294 (props
->MaxDistance
-props
->RefDistance
);
1295 DryGain
*= maxf(1.0f
- attn
, 0.0f
);
1296 for(i
= 0;i
< NumSends
;i
++)
1298 attn
= RoomRolloff
[i
] * (ClampedDist
-props
->RefDistance
) /
1299 (props
->MaxDistance
-props
->RefDistance
);
1300 WetGain
[i
] *= maxf(1.0f
- attn
, 0.0f
);
1305 case ExponentDistanceClamped
:
1306 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1307 if(props
->MaxDistance
< props
->RefDistance
)
1310 case ExponentDistance
:
1311 if(!(ClampedDist
> 0.0f
&& props
->RefDistance
> 0.0f
))
1312 ClampedDist
= props
->RefDistance
;
1315 DryGain
*= powf(ClampedDist
/props
->RefDistance
, -props
->RolloffFactor
);
1316 for(i
= 0;i
< NumSends
;i
++)
1317 WetGain
[i
] *= powf(ClampedDist
/props
->RefDistance
, -RoomRolloff
[i
]);
1321 case DisableDistance
:
1322 ClampedDist
= props
->RefDistance
;
1326 /* Distance-based air absorption */
1327 if(ClampedDist
> props
->RefDistance
&& props
->RolloffFactor
> 0.0f
)
1329 ALfloat meters_base
= (ClampedDist
-props
->RefDistance
) * props
->RolloffFactor
*
1330 Listener
->Params
.MetersPerUnit
;
1331 if(props
->AirAbsorptionFactor
> 0.0f
)
1333 ALfloat hfattn
= powf(AIRABSORBGAINHF
, meters_base
* props
->AirAbsorptionFactor
);
1334 DryGainHF
*= hfattn
;
1335 for(i
= 0;i
< NumSends
;i
++)
1336 WetGainHF
[i
] *= hfattn
;
1339 if(props
->WetGainAuto
)
1341 /* Apply a decay-time transformation to the wet path, based on the
1342 * source distance in meters. The initial decay of the reverb
1343 * effect is calculated and applied to the wet path.
1345 for(i
= 0;i
< NumSends
;i
++)
1349 if(!(DecayDistance
[i
] > 0.0f
))
1352 gain
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayDistance
[i
]);
1354 /* Yes, the wet path's air absorption is applied with
1355 * WetGainAuto on, rather than WetGainHFAuto.
1359 ALfloat gainhf
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayHFDistance
[i
]);
1360 WetGainHF
[i
] *= minf(gainhf
/ gain
, 1.0f
);
1366 /* Calculate directional soundcones */
1367 if(directional
&& props
->InnerAngle
< 360.0f
)
1373 Angle
= acosf(aluDotproduct(&Direction
, &SourceToListener
));
1374 Angle
= RAD2DEG(Angle
* ConeScale
* 2.0f
);
1375 if(!(Angle
> props
->InnerAngle
))
1380 else if(Angle
< props
->OuterAngle
)
1382 ALfloat scale
= ( Angle
-props
->InnerAngle
) /
1383 (props
->OuterAngle
-props
->InnerAngle
);
1384 ConeVolume
= lerp(1.0f
, props
->OuterGain
, scale
);
1385 ConeHF
= lerp(1.0f
, props
->OuterGainHF
, scale
);
1389 ConeVolume
= props
->OuterGain
;
1390 ConeHF
= props
->OuterGainHF
;
1393 DryGain
*= ConeVolume
;
1394 if(props
->DryGainHFAuto
)
1395 DryGainHF
*= ConeHF
;
1396 if(props
->WetGainAuto
)
1398 for(i
= 0;i
< NumSends
;i
++)
1399 WetGain
[i
] *= ConeVolume
;
1401 if(props
->WetGainHFAuto
)
1403 for(i
= 0;i
< NumSends
;i
++)
1404 WetGainHF
[i
] *= ConeHF
;
1408 /* Apply gain and frequency filters */
1409 DryGain
= clampf(DryGain
, props
->MinGain
, props
->MaxGain
);
1410 DryGain
= minf(DryGain
*props
->Direct
.Gain
*Listener
->Params
.Gain
, GAIN_MIX_MAX
);
1411 DryGainHF
*= props
->Direct
.GainHF
;
1412 DryGainLF
*= props
->Direct
.GainLF
;
1413 for(i
= 0;i
< NumSends
;i
++)
1415 WetGain
[i
] = clampf(WetGain
[i
], props
->MinGain
, props
->MaxGain
);
1416 WetGain
[i
] = minf(WetGain
[i
]*props
->Send
[i
].Gain
*Listener
->Params
.Gain
, GAIN_MIX_MAX
);
1417 WetGainHF
[i
] *= props
->Send
[i
].GainHF
;
1418 WetGainLF
[i
] *= props
->Send
[i
].GainLF
;
1422 /* Initial source pitch */
1423 Pitch
= props
->Pitch
;
1425 /* Calculate velocity-based doppler effect */
1426 DopplerFactor
= props
->DopplerFactor
* Listener
->Params
.DopplerFactor
;
1427 if(DopplerFactor
> 0.0f
)
1429 const aluVector
*lvelocity
= &Listener
->Params
.Velocity
;
1430 const ALfloat SpeedOfSound
= Listener
->Params
.SpeedOfSound
;
1433 vss
= aluDotproduct(&Velocity
, &SourceToListener
) * DopplerFactor
;
1434 vls
= aluDotproduct(lvelocity
, &SourceToListener
) * DopplerFactor
;
1436 if(!(vls
< SpeedOfSound
))
1438 /* Listener moving away from the source at the speed of sound.
1439 * Sound waves can't catch it.
1443 else if(!(vss
< SpeedOfSound
))
1445 /* Source moving toward the listener at the speed of sound. Sound
1446 * waves bunch up to extreme frequencies.
1452 /* Source and listener movement is nominal. Calculate the proper
1455 Pitch
*= (SpeedOfSound
-vls
) / (SpeedOfSound
-vss
);
1459 /* Adjust pitch based on the buffer and output frequencies, and calculate
1460 * fixed-point stepping value.
1462 Pitch
*= (ALfloat
)ALBuffer
->Frequency
/(ALfloat
)Device
->Frequency
;
1463 if(Pitch
> (ALfloat
)MAX_PITCH
)
1464 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1466 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1467 if(props
->Resampler
== BSinc24Resampler
)
1468 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc24
);
1469 else if(props
->Resampler
== BSinc12Resampler
)
1470 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc12
);
1471 voice
->Resampler
= SelectResampler(props
->Resampler
);
1473 if(Distance
> FLT_EPSILON
)
1475 dir
[0] = -SourceToListener
.v
[0];
1476 /* Clamp Y, in case rounding errors caused it to end up outside of
1479 dir
[1] = clampf(-SourceToListener
.v
[1], -1.0f
, 1.0f
);
1480 dir
[2] = -SourceToListener
.v
[2] * ZScale
;
1488 if(props
->Radius
> Distance
)
1489 spread
= F_TAU
- Distance
/props
->Radius
*F_PI
;
1490 else if(Distance
> FLT_EPSILON
)
1491 spread
= asinf(props
->Radius
/ Distance
) * 2.0f
;
1495 CalcPanningAndFilters(voice
, Distance
, dir
, spread
, DryGain
, DryGainHF
, DryGainLF
, WetGain
,
1496 WetGainLF
, WetGainHF
, SendSlots
, ALBuffer
, props
, Listener
, Device
);
1499 static void CalcSourceParams(ALvoice
*voice
, ALCcontext
*context
, bool force
)
1501 ALbufferlistitem
*BufferListItem
;
1502 struct ALvoiceProps
*props
;
1504 props
= ATOMIC_EXCHANGE_PTR(&voice
->Update
, NULL
, almemory_order_acq_rel
);
1505 if(!props
&& !force
) return;
1509 memcpy(voice
->Props
, props
,
1510 FAM_SIZE(struct ALvoiceProps
, Send
, context
->Device
->NumAuxSends
)
1513 ATOMIC_REPLACE_HEAD(struct ALvoiceProps
*, &context
->FreeVoiceProps
, props
);
1515 props
= voice
->Props
;
1517 BufferListItem
= ATOMIC_LOAD(&voice
->current_buffer
, almemory_order_relaxed
);
1518 while(BufferListItem
!= NULL
)
1520 const ALbuffer
*buffer
;
1521 if(BufferListItem
->num_buffers
>= 1 && (buffer
=BufferListItem
->buffers
[0]) != NULL
)
1523 if(props
->SpatializeMode
== SpatializeOn
||
1524 (props
->SpatializeMode
== SpatializeAuto
&& buffer
->FmtChannels
== FmtMono
))
1525 CalcAttnSourceParams(voice
, props
, buffer
, context
);
1527 CalcNonAttnSourceParams(voice
, props
, buffer
, context
);
1530 BufferListItem
= ATOMIC_LOAD(&BufferListItem
->next
, almemory_order_acquire
);
1535 static void UpdateContextSources(ALCcontext
*ctx
, const struct ALeffectslotArray
*slots
)
1537 ALvoice
**voice
, **voice_end
;
1541 IncrementRef(&ctx
->UpdateCount
);
1542 if(!ATOMIC_LOAD(&ctx
->HoldUpdates
, almemory_order_acquire
))
1544 bool cforce
= CalcContextParams(ctx
);
1545 bool force
= CalcListenerParams(ctx
) | cforce
;
1546 for(i
= 0;i
< slots
->count
;i
++)
1547 force
|= CalcEffectSlotParams(slots
->slot
[i
], ctx
, cforce
);
1549 voice
= ctx
->Voices
;
1550 voice_end
= voice
+ ctx
->VoiceCount
;
1551 for(;voice
!= voice_end
;++voice
)
1553 source
= ATOMIC_LOAD(&(*voice
)->Source
, almemory_order_acquire
);
1554 if(source
) CalcSourceParams(*voice
, ctx
, force
);
1557 IncrementRef(&ctx
->UpdateCount
);
1561 static void ApplyStablizer(FrontStablizer
*Stablizer
, ALfloat (*restrict Buffer
)[BUFFERSIZE
],
1562 int lidx
, int ridx
, int cidx
, ALsizei SamplesToDo
,
1563 ALsizei NumChannels
)
1565 ALfloat (*restrict lsplit
)[BUFFERSIZE
] = ASSUME_ALIGNED(Stablizer
->LSplit
, 16);
1566 ALfloat (*restrict rsplit
)[BUFFERSIZE
] = ASSUME_ALIGNED(Stablizer
->RSplit
, 16);
1569 /* Apply an all-pass to all channels, except the front-left and front-
1570 * right, so they maintain the same relative phase.
1572 for(i
= 0;i
< NumChannels
;i
++)
1574 if(i
== lidx
|| i
== ridx
)
1576 splitterap_process(&Stablizer
->APFilter
[i
], Buffer
[i
], SamplesToDo
);
1579 bandsplit_process(&Stablizer
->LFilter
, lsplit
[1], lsplit
[0], Buffer
[lidx
], SamplesToDo
);
1580 bandsplit_process(&Stablizer
->RFilter
, rsplit
[1], rsplit
[0], Buffer
[ridx
], SamplesToDo
);
1582 for(i
= 0;i
< SamplesToDo
;i
++)
1584 ALfloat lfsum
, hfsum
;
1587 lfsum
= lsplit
[0][i
] + rsplit
[0][i
];
1588 hfsum
= lsplit
[1][i
] + rsplit
[1][i
];
1589 s
= lsplit
[0][i
] + lsplit
[1][i
] - rsplit
[0][i
] - rsplit
[1][i
];
1591 /* This pans the separate low- and high-frequency sums between being on
1592 * the center channel and the left/right channels. The low-frequency
1593 * sum is 1/3rd toward center (2/3rds on left/right) and the high-
1594 * frequency sum is 1/4th toward center (3/4ths on left/right). These
1595 * values can be tweaked.
1597 m
= lfsum
*cosf(1.0f
/3.0f
* F_PI_2
) + hfsum
*cosf(1.0f
/4.0f
* F_PI_2
);
1598 c
= lfsum
*sinf(1.0f
/3.0f
* F_PI_2
) + hfsum
*sinf(1.0f
/4.0f
* F_PI_2
);
1600 /* The generated center channel signal adds to the existing signal,
1601 * while the modified left and right channels replace.
1603 Buffer
[lidx
][i
] = (m
+ s
) * 0.5f
;
1604 Buffer
[ridx
][i
] = (m
- s
) * 0.5f
;
1605 Buffer
[cidx
][i
] += c
* 0.5f
;
1609 static void ApplyDistanceComp(ALfloat (*restrict Samples
)[BUFFERSIZE
], DistanceComp
*distcomp
,
1610 ALfloat
*restrict Values
, ALsizei SamplesToDo
, ALsizei numchans
)
1614 Values
= ASSUME_ALIGNED(Values
, 16);
1615 for(c
= 0;c
< numchans
;c
++)
1617 ALfloat
*restrict inout
= ASSUME_ALIGNED(Samples
[c
], 16);
1618 const ALfloat gain
= distcomp
[c
].Gain
;
1619 const ALsizei base
= distcomp
[c
].Length
;
1620 ALfloat
*restrict distbuf
= ASSUME_ALIGNED(distcomp
[c
].Buffer
, 16);
1626 for(i
= 0;i
< SamplesToDo
;i
++)
1632 if(SamplesToDo
>= base
)
1634 for(i
= 0;i
< base
;i
++)
1635 Values
[i
] = distbuf
[i
];
1636 for(;i
< SamplesToDo
;i
++)
1637 Values
[i
] = inout
[i
-base
];
1638 memcpy(distbuf
, &inout
[SamplesToDo
-base
], base
*sizeof(ALfloat
));
1642 for(i
= 0;i
< SamplesToDo
;i
++)
1643 Values
[i
] = distbuf
[i
];
1644 memmove(distbuf
, distbuf
+SamplesToDo
, (base
-SamplesToDo
)*sizeof(ALfloat
));
1645 memcpy(distbuf
+base
-SamplesToDo
, inout
, SamplesToDo
*sizeof(ALfloat
));
1647 for(i
= 0;i
< SamplesToDo
;i
++)
1648 inout
[i
] = Values
[i
]*gain
;
1652 static void ApplyDither(ALfloat (*restrict Samples
)[BUFFERSIZE
], ALuint
*dither_seed
,
1653 const ALfloat quant_scale
, const ALsizei SamplesToDo
,
1654 const ALsizei numchans
)
1656 const ALfloat invscale
= 1.0f
/ quant_scale
;
1657 ALuint seed
= *dither_seed
;
1660 /* Dithering. Step 1, generate whitenoise (uniform distribution of random
1661 * values between -1 and +1). Step 2 is to add the noise to the samples,
1662 * before rounding and after scaling up to the desired quantization depth.
1664 for(c
= 0;c
< numchans
;c
++)
1666 ALfloat
*restrict samples
= Samples
[c
];
1667 for(i
= 0;i
< SamplesToDo
;i
++)
1669 ALfloat val
= samples
[i
] * quant_scale
;
1670 ALuint rng0
= dither_rng(&seed
);
1671 ALuint rng1
= dither_rng(&seed
);
1672 val
+= (ALfloat
)(rng0
*(1.0/UINT_MAX
) - rng1
*(1.0/UINT_MAX
));
1673 samples
[i
] = roundf(val
) * invscale
;
1676 *dither_seed
= seed
;
1680 static inline ALfloat
Conv_ALfloat(ALfloat val
)
1682 static inline ALint
Conv_ALint(ALfloat val
)
1684 /* Floats only have a 24-bit mantissa, so [-16777216, +16777216] is the max
1685 * integer range normalized floats can be safely converted to (a bit of the
1686 * exponent helps out, effectively giving 25 bits).
1688 return fastf2i(clampf(val
*16777216.0f
, -16777216.0f
, 16777215.0f
))<<7;
1690 static inline ALshort
Conv_ALshort(ALfloat val
)
1691 { return fastf2i(clampf(val
*32768.0f
, -32768.0f
, 32767.0f
)); }
1692 static inline ALbyte
Conv_ALbyte(ALfloat val
)
1693 { return fastf2i(clampf(val
*128.0f
, -128.0f
, 127.0f
)); }
1695 /* Define unsigned output variations. */
1696 #define DECL_TEMPLATE(T, func, O) \
1697 static inline T Conv_##T(ALfloat val) { return func(val)+O; }
1699 DECL_TEMPLATE(ALubyte
, Conv_ALbyte
, 128)
1700 DECL_TEMPLATE(ALushort
, Conv_ALshort
, 32768)
1701 DECL_TEMPLATE(ALuint
, Conv_ALint
, 2147483648u)
1703 #undef DECL_TEMPLATE
1705 #define DECL_TEMPLATE(T, A) \
1706 static void Write##A(const ALfloat (*restrict InBuffer)[BUFFERSIZE], \
1707 ALvoid *OutBuffer, ALsizei Offset, ALsizei SamplesToDo, \
1711 for(j = 0;j < numchans;j++) \
1713 const ALfloat *restrict in = ASSUME_ALIGNED(InBuffer[j], 16); \
1714 T *restrict out = (T*)OutBuffer + Offset*numchans + j; \
1716 for(i = 0;i < SamplesToDo;i++) \
1717 out[i*numchans] = Conv_##T(in[i]); \
1721 DECL_TEMPLATE(ALfloat
, F32
)
1722 DECL_TEMPLATE(ALuint
, UI32
)
1723 DECL_TEMPLATE(ALint
, I32
)
1724 DECL_TEMPLATE(ALushort
, UI16
)
1725 DECL_TEMPLATE(ALshort
, I16
)
1726 DECL_TEMPLATE(ALubyte
, UI8
)
1727 DECL_TEMPLATE(ALbyte
, I8
)
1729 #undef DECL_TEMPLATE
1732 void aluMixData(ALCdevice
*device
, ALvoid
*OutBuffer
, ALsizei NumSamples
)
1734 ALsizei SamplesToDo
;
1735 ALsizei SamplesDone
;
1740 for(SamplesDone
= 0;SamplesDone
< NumSamples
;)
1742 SamplesToDo
= mini(NumSamples
-SamplesDone
, BUFFERSIZE
);
1743 for(c
= 0;c
< device
->Dry
.NumChannels
;c
++)
1744 memset(device
->Dry
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1745 if(device
->Dry
.Buffer
!= device
->FOAOut
.Buffer
)
1746 for(c
= 0;c
< device
->FOAOut
.NumChannels
;c
++)
1747 memset(device
->FOAOut
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1748 if(device
->Dry
.Buffer
!= device
->RealOut
.Buffer
)
1749 for(c
= 0;c
< device
->RealOut
.NumChannels
;c
++)
1750 memset(device
->RealOut
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1752 IncrementRef(&device
->MixCount
);
1754 ctx
= ATOMIC_LOAD(&device
->ContextList
, almemory_order_acquire
);
1757 const struct ALeffectslotArray
*auxslots
;
1759 auxslots
= ATOMIC_LOAD(&ctx
->ActiveAuxSlots
, almemory_order_acquire
);
1760 UpdateContextSources(ctx
, auxslots
);
1762 for(i
= 0;i
< auxslots
->count
;i
++)
1764 ALeffectslot
*slot
= auxslots
->slot
[i
];
1765 for(c
= 0;c
< slot
->NumChannels
;c
++)
1766 memset(slot
->WetBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1769 /* source processing */
1770 for(i
= 0;i
< ctx
->VoiceCount
;i
++)
1772 ALvoice
*voice
= ctx
->Voices
[i
];
1773 ALsource
*source
= ATOMIC_LOAD(&voice
->Source
, almemory_order_acquire
);
1774 if(source
&& ATOMIC_LOAD(&voice
->Playing
, almemory_order_relaxed
) &&
1777 if(!MixSource(voice
, source
->id
, ctx
, SamplesToDo
))
1779 ATOMIC_STORE(&voice
->Source
, NULL
, almemory_order_relaxed
);
1780 ATOMIC_STORE(&voice
->Playing
, false, almemory_order_release
);
1785 /* effect slot processing */
1786 for(i
= 0;i
< auxslots
->count
;i
++)
1788 const ALeffectslot
*slot
= auxslots
->slot
[i
];
1789 ALeffectState
*state
= slot
->Params
.EffectState
;
1790 V(state
,process
)(SamplesToDo
, slot
->WetBuffer
, state
->OutBuffer
,
1791 state
->OutChannels
);
1797 /* Increment the clock time. Every second's worth of samples is
1798 * converted and added to clock base so that large sample counts don't
1799 * overflow during conversion. This also guarantees an exact, stable
1801 device
->SamplesDone
+= SamplesToDo
;
1802 device
->ClockBase
+= (device
->SamplesDone
/device
->Frequency
) * DEVICE_CLOCK_RES
;
1803 device
->SamplesDone
%= device
->Frequency
;
1804 IncrementRef(&device
->MixCount
);
1806 /* Apply post-process for finalizing the Dry mix to the RealOut
1807 * (Ambisonic decode, UHJ encode, etc).
1809 if(LIKELY(device
->PostProcess
))
1810 device
->PostProcess(device
, SamplesToDo
);
1814 ALfloat (*Buffer
)[BUFFERSIZE
] = device
->RealOut
.Buffer
;
1815 ALsizei Channels
= device
->RealOut
.NumChannels
;
1817 if(device
->Stablizer
)
1819 int lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
1820 int ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
1821 int cidx
= GetChannelIdxByName(&device
->RealOut
, FrontCenter
);
1822 assert(lidx
>= 0 && ridx
>= 0 && cidx
>= 0);
1824 ApplyStablizer(device
->Stablizer
, Buffer
, lidx
, ridx
, cidx
,
1825 SamplesToDo
, Channels
);
1828 ApplyDistanceComp(Buffer
, device
->ChannelDelay
, device
->TempBuffer
[0],
1829 SamplesToDo
, Channels
);
1832 ApplyCompression(device
->Limiter
, Channels
, SamplesToDo
, Buffer
);
1834 if(device
->DitherDepth
> 0.0f
)
1835 ApplyDither(Buffer
, &device
->DitherSeed
, device
->DitherDepth
, SamplesToDo
,
1838 switch(device
->FmtType
)
1841 WriteI8(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1844 WriteUI8(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1847 WriteI16(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1850 WriteUI16(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1853 WriteI32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1856 WriteUI32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1859 WriteF32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1864 SamplesDone
+= SamplesToDo
;
1870 void aluHandleDisconnect(ALCdevice
*device
, const char *msg
, ...)
1877 if(!ATOMIC_EXCHANGE(&device
->Connected
, AL_FALSE
, almemory_order_acq_rel
))
1880 evt
.EnumType
= EventType_Disconnected
;
1881 evt
.Type
= AL_EVENT_TYPE_DISCONNECTED_SOFT
;
1885 va_start(args
, msg
);
1886 msglen
= vsnprintf(evt
.Message
, sizeof(evt
.Message
), msg
, args
);
1889 if(msglen
< 0 || (size_t)msglen
>= sizeof(evt
.Message
))
1891 evt
.Message
[sizeof(evt
.Message
)-1] = 0;
1892 msglen
= (int)strlen(evt
.Message
);
1898 msg
= "<internal error constructing message>";
1899 msglen
= (int)strlen(msg
);
1902 ctx
= ATOMIC_LOAD_SEQ(&device
->ContextList
);
1907 if((ATOMIC_LOAD(&ctx
->EnabledEvts
, almemory_order_acquire
)&EventType_Disconnected
) &&
1908 ll_ringbuffer_write_space(ctx
->AsyncEvents
) > 0)
1910 ll_ringbuffer_write(ctx
->AsyncEvents
, (const char*)&evt
, 1);
1911 alsem_post(&ctx
->EventSem
);
1914 for(i
= 0;i
< ctx
->VoiceCount
;i
++)
1916 ALvoice
*voice
= ctx
->Voices
[i
];
1918 ATOMIC_STORE(&voice
->Source
, NULL
, almemory_order_relaxed
);
1919 ATOMIC_STORE(&voice
->Playing
, false, almemory_order_release
);