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 "mixer/defs.h"
44 #include "fpu_modes.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
},
110 static void ClearArray(ALfloat f
[MAX_OUTPUT_CHANNELS
])
113 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
118 enum Channel channel
;
123 static HrtfDirectMixerFunc MixDirectHrtf
= MixDirectHrtf_C
;
126 void DeinitVoice(ALvoice
*voice
)
128 al_free(ATOMIC_EXCHANGE_PTR_SEQ(&voice
->Update
, NULL
));
132 static inline HrtfDirectMixerFunc
SelectHrtfMixer(void)
135 if((CPUCapFlags
&CPU_CAP_NEON
))
136 return MixDirectHrtf_Neon
;
139 if((CPUCapFlags
&CPU_CAP_SSE
))
140 return MixDirectHrtf_SSE
;
143 return MixDirectHrtf_C
;
147 /* Prior to VS2013, MSVC lacks the round() family of functions. */
148 #if defined(_MSC_VER) && _MSC_VER < 1800
149 static float roundf(float val
)
152 return ceilf(val
-0.5f
);
153 return floorf(val
+0.5f
);
157 /* This RNG method was created based on the math found in opusdec. It's quick,
158 * and starting with a seed value of 22222, is suitable for generating
161 static inline ALuint
dither_rng(ALuint
*seed
)
163 *seed
= (*seed
* 96314165) + 907633515;
168 static inline void aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
170 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
171 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
172 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
175 static inline ALfloat
aluDotproduct(const aluVector
*vec1
, const aluVector
*vec2
)
177 return vec1
->v
[0]*vec2
->v
[0] + vec1
->v
[1]*vec2
->v
[1] + vec1
->v
[2]*vec2
->v
[2];
180 static ALfloat
aluNormalize(ALfloat
*vec
)
182 ALfloat length
= sqrtf(vec
[0]*vec
[0] + vec
[1]*vec
[1] + vec
[2]*vec
[2]);
183 if(length
> FLT_EPSILON
)
185 ALfloat inv_length
= 1.0f
/length
;
186 vec
[0] *= inv_length
;
187 vec
[1] *= inv_length
;
188 vec
[2] *= inv_length
;
191 vec
[0] = vec
[1] = vec
[2] = 0.0f
;
195 static void aluMatrixfFloat3(ALfloat
*vec
, ALfloat w
, const aluMatrixf
*mtx
)
197 ALfloat v
[4] = { vec
[0], vec
[1], vec
[2], w
};
199 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];
200 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];
201 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];
204 static aluVector
aluMatrixfVector(const aluMatrixf
*mtx
, const aluVector
*vec
)
207 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];
208 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];
209 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];
210 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];
217 MixDirectHrtf
= SelectHrtfMixer();
221 static void SendSourceStoppedEvent(ALCcontext
*context
, ALuint id
)
223 ALbitfieldSOFT enabledevt
;
228 enabledevt
= ATOMIC_LOAD(&context
->EnabledEvts
, almemory_order_acquire
);
229 if(!(enabledevt
&EventType_SourceStateChange
)) return;
231 evt
.EnumType
= EventType_SourceStateChange
;
232 evt
.Type
= AL_EVENT_TYPE_SOURCE_STATE_CHANGED_SOFT
;
234 evt
.Param
= AL_STOPPED
;
236 /* Normally snprintf would be used, but this is called from the mixer and
237 * that function's not real-time safe, so we have to construct it manually.
239 strcpy(evt
.Message
, "Source ID "); strpos
= 10;
241 while(scale
> 0 && scale
> id
)
245 evt
.Message
[strpos
++] = '0' + ((id
/scale
)%10);
248 strcpy(evt
.Message
+strpos
, " state changed to AL_STOPPED");
250 if(ll_ringbuffer_write(context
->AsyncEvents
, (const char*)&evt
, 1) == 1)
251 alsem_post(&context
->EventSem
);
255 static void ProcessHrtf(ALCdevice
*device
, ALsizei SamplesToDo
)
257 DirectHrtfState
*state
;
262 ambiup_process(device
->AmbiUp
,
263 device
->Dry
.Buffer
, device
->Dry
.NumChannels
, device
->FOAOut
.Buffer
,
267 lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
268 ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
269 assert(lidx
!= -1 && ridx
!= -1);
271 state
= device
->Hrtf
;
272 for(c
= 0;c
< device
->Dry
.NumChannels
;c
++)
274 MixDirectHrtf(device
->RealOut
.Buffer
[lidx
], device
->RealOut
.Buffer
[ridx
],
275 device
->Dry
.Buffer
[c
], state
->Offset
, state
->IrSize
,
276 state
->Chan
[c
].Coeffs
, state
->Chan
[c
].Values
, SamplesToDo
279 state
->Offset
+= SamplesToDo
;
282 static void ProcessAmbiDec(ALCdevice
*device
, ALsizei SamplesToDo
)
284 if(device
->Dry
.Buffer
!= device
->FOAOut
.Buffer
)
285 bformatdec_upSample(device
->AmbiDecoder
,
286 device
->Dry
.Buffer
, device
->FOAOut
.Buffer
, device
->FOAOut
.NumChannels
,
289 bformatdec_process(device
->AmbiDecoder
,
290 device
->RealOut
.Buffer
, device
->RealOut
.NumChannels
, device
->Dry
.Buffer
,
295 static void ProcessAmbiUp(ALCdevice
*device
, ALsizei SamplesToDo
)
297 ambiup_process(device
->AmbiUp
,
298 device
->RealOut
.Buffer
, device
->RealOut
.NumChannels
, device
->FOAOut
.Buffer
,
303 static void ProcessUhj(ALCdevice
*device
, ALsizei SamplesToDo
)
305 int lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
306 int ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
307 if(LIKELY(lidx
!= -1 && ridx
!= -1))
309 /* Encode to stereo-compatible 2-channel UHJ output. */
310 EncodeUhj2(device
->Uhj_Encoder
,
311 device
->RealOut
.Buffer
[lidx
], device
->RealOut
.Buffer
[ridx
],
312 device
->Dry
.Buffer
, SamplesToDo
317 static void ProcessBs2b(ALCdevice
*device
, ALsizei SamplesToDo
)
319 int lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
320 int ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
321 if(LIKELY(lidx
!= -1 && ridx
!= -1))
323 /* Apply binaural/crossfeed filter */
324 bs2b_cross_feed(device
->Bs2b
, device
->RealOut
.Buffer
[lidx
],
325 device
->RealOut
.Buffer
[ridx
], SamplesToDo
);
329 void aluSelectPostProcess(ALCdevice
*device
)
331 if(device
->HrtfHandle
)
332 device
->PostProcess
= ProcessHrtf
;
333 else if(device
->AmbiDecoder
)
334 device
->PostProcess
= ProcessAmbiDec
;
335 else if(device
->AmbiUp
)
336 device
->PostProcess
= ProcessAmbiUp
;
337 else if(device
->Uhj_Encoder
)
338 device
->PostProcess
= ProcessUhj
;
339 else if(device
->Bs2b
)
340 device
->PostProcess
= ProcessBs2b
;
342 device
->PostProcess
= NULL
;
346 /* Prepares the interpolator for a given rate (determined by increment). A
347 * result of AL_FALSE indicates that the filter output will completely cut
350 * With a bit of work, and a trade of memory for CPU cost, this could be
351 * modified for use with an interpolated increment for buttery-smooth pitch
354 void BsincPrepare(const ALuint increment
, BsincState
*state
, const BSincTable
*table
)
359 if(increment
> FRACTIONONE
)
361 sf
= (ALfloat
)FRACTIONONE
/ increment
;
362 sf
= maxf(0.0f
, (BSINC_SCALE_COUNT
-1) * (sf
-table
->scaleBase
) * table
->scaleRange
);
364 /* The interpolation factor is fit to this diagonally-symmetric curve
365 * to reduce the transition ripple caused by interpolating different
366 * scales of the sinc function.
368 sf
= 1.0f
- cosf(asinf(sf
- si
));
373 si
= BSINC_SCALE_COUNT
- 1;
377 state
->m
= table
->m
[si
];
378 state
->l
= -((state
->m
/2) - 1);
379 state
->filter
= table
->Tab
+ table
->filterOffset
[si
];
383 static bool CalcContextParams(ALCcontext
*Context
)
385 ALlistener
*Listener
= Context
->Listener
;
386 struct ALcontextProps
*props
;
388 props
= ATOMIC_EXCHANGE_PTR(&Context
->Update
, NULL
, almemory_order_acq_rel
);
389 if(!props
) return false;
391 Listener
->Params
.MetersPerUnit
= props
->MetersPerUnit
;
393 Listener
->Params
.DopplerFactor
= props
->DopplerFactor
;
394 Listener
->Params
.SpeedOfSound
= props
->SpeedOfSound
* props
->DopplerVelocity
;
395 if(!OverrideReverbSpeedOfSound
)
396 Listener
->Params
.ReverbSpeedOfSound
= Listener
->Params
.SpeedOfSound
*
397 Listener
->Params
.MetersPerUnit
;
399 Listener
->Params
.SourceDistanceModel
= props
->SourceDistanceModel
;
400 Listener
->Params
.DistanceModel
= props
->DistanceModel
;
402 ATOMIC_REPLACE_HEAD(struct ALcontextProps
*, &Context
->FreeContextProps
, props
);
406 static bool CalcListenerParams(ALCcontext
*Context
)
408 ALlistener
*Listener
= Context
->Listener
;
409 ALfloat N
[3], V
[3], U
[3], P
[3];
410 struct ALlistenerProps
*props
;
413 props
= ATOMIC_EXCHANGE_PTR(&Listener
->Update
, NULL
, almemory_order_acq_rel
);
414 if(!props
) return false;
417 N
[0] = props
->Forward
[0];
418 N
[1] = props
->Forward
[1];
419 N
[2] = props
->Forward
[2];
425 /* Build and normalize right-vector */
426 aluCrossproduct(N
, V
, U
);
429 aluMatrixfSet(&Listener
->Params
.Matrix
,
430 U
[0], V
[0], -N
[0], 0.0,
431 U
[1], V
[1], -N
[1], 0.0,
432 U
[2], V
[2], -N
[2], 0.0,
436 P
[0] = props
->Position
[0];
437 P
[1] = props
->Position
[1];
438 P
[2] = props
->Position
[2];
439 aluMatrixfFloat3(P
, 1.0, &Listener
->Params
.Matrix
);
440 aluMatrixfSetRow(&Listener
->Params
.Matrix
, 3, -P
[0], -P
[1], -P
[2], 1.0f
);
442 aluVectorSet(&vel
, props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
);
443 Listener
->Params
.Velocity
= aluMatrixfVector(&Listener
->Params
.Matrix
, &vel
);
445 Listener
->Params
.Gain
= props
->Gain
* Context
->GainBoost
;
447 ATOMIC_REPLACE_HEAD(struct ALlistenerProps
*, &Context
->FreeListenerProps
, props
);
451 static bool CalcEffectSlotParams(ALeffectslot
*slot
, ALCcontext
*context
, bool force
)
453 struct ALeffectslotProps
*props
;
454 ALeffectState
*state
;
456 props
= ATOMIC_EXCHANGE_PTR(&slot
->Update
, NULL
, almemory_order_acq_rel
);
457 if(!props
&& !force
) return false;
461 slot
->Params
.Gain
= props
->Gain
;
462 slot
->Params
.AuxSendAuto
= props
->AuxSendAuto
;
463 slot
->Params
.EffectType
= props
->Type
;
464 slot
->Params
.EffectProps
= props
->Props
;
465 if(IsReverbEffect(props
->Type
))
467 slot
->Params
.RoomRolloff
= props
->Props
.Reverb
.RoomRolloffFactor
;
468 slot
->Params
.DecayTime
= props
->Props
.Reverb
.DecayTime
;
469 slot
->Params
.DecayLFRatio
= props
->Props
.Reverb
.DecayLFRatio
;
470 slot
->Params
.DecayHFRatio
= props
->Props
.Reverb
.DecayHFRatio
;
471 slot
->Params
.DecayHFLimit
= props
->Props
.Reverb
.DecayHFLimit
;
472 slot
->Params
.AirAbsorptionGainHF
= props
->Props
.Reverb
.AirAbsorptionGainHF
;
476 slot
->Params
.RoomRolloff
= 0.0f
;
477 slot
->Params
.DecayTime
= 0.0f
;
478 slot
->Params
.DecayLFRatio
= 0.0f
;
479 slot
->Params
.DecayHFRatio
= 0.0f
;
480 slot
->Params
.DecayHFLimit
= AL_FALSE
;
481 slot
->Params
.AirAbsorptionGainHF
= 1.0f
;
484 /* Swap effect states. No need to play with the ref counts since they
485 * keep the same number of refs.
487 state
= props
->State
;
488 props
->State
= slot
->Params
.EffectState
;
489 slot
->Params
.EffectState
= state
;
491 ATOMIC_REPLACE_HEAD(struct ALeffectslotProps
*, &context
->FreeEffectslotProps
, props
);
494 state
= slot
->Params
.EffectState
;
496 V(state
,update
)(context
, slot
, &slot
->Params
.EffectProps
);
501 static const struct ChanMap MonoMap
[1] = {
502 { FrontCenter
, 0.0f
, 0.0f
}
504 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
505 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) }
507 { FrontLeft
, DEG2RAD( -45.0f
), DEG2RAD(0.0f
) },
508 { FrontRight
, DEG2RAD( 45.0f
), DEG2RAD(0.0f
) },
509 { BackLeft
, DEG2RAD(-135.0f
), DEG2RAD(0.0f
) },
510 { BackRight
, DEG2RAD( 135.0f
), DEG2RAD(0.0f
) }
512 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
513 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
514 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
516 { SideLeft
, DEG2RAD(-110.0f
), DEG2RAD(0.0f
) },
517 { SideRight
, DEG2RAD( 110.0f
), DEG2RAD(0.0f
) }
519 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
520 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
521 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
523 { BackCenter
, DEG2RAD(180.0f
), DEG2RAD(0.0f
) },
524 { SideLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
525 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
527 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
528 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
529 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
531 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
532 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) },
533 { SideLeft
, DEG2RAD( -90.0f
), DEG2RAD(0.0f
) },
534 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
537 static void CalcPanningAndFilters(ALvoice
*voice
, const ALfloat Azi
, const ALfloat Elev
,
538 const ALfloat Distance
, const ALfloat Spread
,
539 const ALfloat DryGain
, const ALfloat DryGainHF
,
540 const ALfloat DryGainLF
, const ALfloat
*WetGain
,
541 const ALfloat
*WetGainLF
, const ALfloat
*WetGainHF
,
542 ALeffectslot
**SendSlots
, const ALbuffer
*Buffer
,
543 const struct ALvoiceProps
*props
, const ALlistener
*Listener
,
544 const ALCdevice
*Device
)
546 struct ChanMap StereoMap
[2] = {
547 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
548 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) }
550 bool DirectChannels
= props
->DirectChannels
;
551 const ALsizei NumSends
= Device
->NumAuxSends
;
552 const ALuint Frequency
= Device
->Frequency
;
553 const struct ChanMap
*chans
= NULL
;
554 ALsizei num_channels
= 0;
555 bool isbformat
= false;
556 ALfloat downmix_gain
= 1.0f
;
559 switch(Buffer
->FmtChannels
)
564 /* Mono buffers are never played direct. */
565 DirectChannels
= false;
569 /* Convert counter-clockwise to clockwise. */
570 StereoMap
[0].angle
= -props
->StereoPan
[0];
571 StereoMap
[1].angle
= -props
->StereoPan
[1];
575 downmix_gain
= 1.0f
/ 2.0f
;
581 downmix_gain
= 1.0f
/ 2.0f
;
587 downmix_gain
= 1.0f
/ 4.0f
;
593 /* NOTE: Excludes LFE. */
594 downmix_gain
= 1.0f
/ 5.0f
;
600 /* NOTE: Excludes LFE. */
601 downmix_gain
= 1.0f
/ 6.0f
;
607 /* NOTE: Excludes LFE. */
608 downmix_gain
= 1.0f
/ 7.0f
;
614 DirectChannels
= false;
620 DirectChannels
= false;
624 for(c
= 0;c
< num_channels
;c
++)
626 memset(&voice
->Direct
.Params
[c
].Hrtf
.Target
, 0,
627 sizeof(voice
->Direct
.Params
[c
].Hrtf
.Target
));
628 ClearArray(voice
->Direct
.Params
[c
].Gains
.Target
);
630 for(i
= 0;i
< NumSends
;i
++)
632 for(c
= 0;c
< num_channels
;c
++)
633 ClearArray(voice
->Send
[i
].Params
[c
].Gains
.Target
);
636 voice
->Flags
&= ~(VOICE_HAS_HRTF
| VOICE_HAS_NFC
);
639 /* Special handling for B-Format sources. */
641 if(Distance
> FLT_EPSILON
)
643 /* Panning a B-Format sound toward some direction is easy. Just pan
644 * the first (W) channel as a normal mono sound and silence the
647 ALfloat coeffs
[MAX_AMBI_COEFFS
];
649 if(Device
->AvgSpeakerDist
> 0.0f
)
651 ALfloat mdist
= Distance
* Listener
->Params
.MetersPerUnit
;
652 ALfloat w0
= SPEEDOFSOUNDMETRESPERSEC
/
653 (mdist
* (ALfloat
)Device
->Frequency
);
654 ALfloat w1
= SPEEDOFSOUNDMETRESPERSEC
/
655 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
656 /* Clamp w0 for really close distances, to prevent excessive
659 w0
= minf(w0
, w1
*4.0f
);
661 /* Only need to adjust the first channel of a B-Format source. */
662 NfcFilterAdjust(&voice
->Direct
.Params
[0].NFCtrlFilter
, w0
);
664 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
665 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
666 voice
->Flags
|= VOICE_HAS_NFC
;
669 if(Device
->Render_Mode
== StereoPair
)
670 CalcAnglePairwiseCoeffs(Azi
, Elev
, Spread
, coeffs
);
672 CalcAngleCoeffs(Azi
, Elev
, Spread
, coeffs
);
674 /* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */
675 ComputeDryPanGains(&Device
->Dry
, coeffs
, DryGain
*1.414213562f
,
676 voice
->Direct
.Params
[0].Gains
.Target
);
677 for(i
= 0;i
< NumSends
;i
++)
679 const ALeffectslot
*Slot
= SendSlots
[i
];
681 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
682 coeffs
, WetGain
[i
]*1.414213562f
, voice
->Send
[i
].Params
[0].Gains
.Target
688 /* Local B-Format sources have their XYZ channels rotated according
689 * to the orientation.
691 const ALfloat sqrt_2
= sqrtf(2.0f
);
692 const ALfloat sqrt_3
= sqrtf(3.0f
);
693 ALfloat N
[3], V
[3], U
[3];
696 if(Device
->AvgSpeakerDist
> 0.0f
)
698 /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
699 * is what we want for FOA input. The first channel may have
700 * been previously re-adjusted if panned, so reset it.
702 NfcFilterAdjust(&voice
->Direct
.Params
[0].NFCtrlFilter
, 0.0f
);
704 voice
->Direct
.ChannelsPerOrder
[0] = 1;
705 voice
->Direct
.ChannelsPerOrder
[1] = mini(voice
->Direct
.Channels
-1, 3);
706 for(i
= 2;i
< MAX_AMBI_ORDER
+1;i
++)
707 voice
->Direct
.ChannelsPerOrder
[i
] = 0;
708 voice
->Flags
|= VOICE_HAS_NFC
;
712 N
[0] = props
->Orientation
[0][0];
713 N
[1] = props
->Orientation
[0][1];
714 N
[2] = props
->Orientation
[0][2];
716 V
[0] = props
->Orientation
[1][0];
717 V
[1] = props
->Orientation
[1][1];
718 V
[2] = props
->Orientation
[1][2];
720 if(!props
->HeadRelative
)
722 const aluMatrixf
*lmatrix
= &Listener
->Params
.Matrix
;
723 aluMatrixfFloat3(N
, 0.0f
, lmatrix
);
724 aluMatrixfFloat3(V
, 0.0f
, lmatrix
);
726 /* Build and normalize right-vector */
727 aluCrossproduct(N
, V
, U
);
730 /* Build a rotate + conversion matrix (FuMa -> ACN+N3D). NOTE: This
731 * matrix is transposed, for the inputs to align on the rows and
732 * outputs on the columns.
734 aluMatrixfSet(&matrix
,
735 // ACN0 ACN1 ACN2 ACN3
736 sqrt_2
, 0.0f
, 0.0f
, 0.0f
, // Ambi W
737 0.0f
, -N
[0]*sqrt_3
, N
[1]*sqrt_3
, -N
[2]*sqrt_3
, // Ambi X
738 0.0f
, U
[0]*sqrt_3
, -U
[1]*sqrt_3
, U
[2]*sqrt_3
, // Ambi Y
739 0.0f
, -V
[0]*sqrt_3
, V
[1]*sqrt_3
, -V
[2]*sqrt_3
// Ambi Z
742 voice
->Direct
.Buffer
= Device
->FOAOut
.Buffer
;
743 voice
->Direct
.Channels
= Device
->FOAOut
.NumChannels
;
744 for(c
= 0;c
< num_channels
;c
++)
745 ComputeFirstOrderGains(&Device
->FOAOut
, matrix
.m
[c
], DryGain
,
746 voice
->Direct
.Params
[c
].Gains
.Target
);
747 for(i
= 0;i
< NumSends
;i
++)
749 const ALeffectslot
*Slot
= SendSlots
[i
];
752 for(c
= 0;c
< num_channels
;c
++)
753 ComputeFirstOrderGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
754 matrix
.m
[c
], WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
760 else if(DirectChannels
)
762 /* Direct source channels always play local. Skip the virtual channels
763 * and write inputs to the matching real outputs.
765 voice
->Direct
.Buffer
= Device
->RealOut
.Buffer
;
766 voice
->Direct
.Channels
= Device
->RealOut
.NumChannels
;
768 for(c
= 0;c
< num_channels
;c
++)
770 int idx
= GetChannelIdxByName(&Device
->RealOut
, chans
[c
].channel
);
771 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
774 /* Auxiliary sends still use normal channel panning since they mix to
775 * B-Format, which can't channel-match.
777 for(c
= 0;c
< num_channels
;c
++)
779 ALfloat coeffs
[MAX_AMBI_COEFFS
];
780 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, 0.0f
, coeffs
);
782 for(i
= 0;i
< NumSends
;i
++)
784 const ALeffectslot
*Slot
= SendSlots
[i
];
786 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
787 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
792 else if(Device
->Render_Mode
== HrtfRender
)
794 /* Full HRTF rendering. Skip the virtual channels and render to the
797 voice
->Direct
.Buffer
= Device
->RealOut
.Buffer
;
798 voice
->Direct
.Channels
= Device
->RealOut
.NumChannels
;
800 if(Distance
> FLT_EPSILON
)
802 ALfloat coeffs
[MAX_AMBI_COEFFS
];
804 /* Get the HRIR coefficients and delays just once, for the given
807 GetHrtfCoeffs(Device
->HrtfHandle
, Elev
, Azi
, Spread
,
808 voice
->Direct
.Params
[0].Hrtf
.Target
.Coeffs
,
809 voice
->Direct
.Params
[0].Hrtf
.Target
.Delay
);
810 voice
->Direct
.Params
[0].Hrtf
.Target
.Gain
= DryGain
* downmix_gain
;
812 /* Remaining channels use the same results as the first. */
813 for(c
= 1;c
< num_channels
;c
++)
816 if(chans
[c
].channel
!= LFE
)
817 voice
->Direct
.Params
[c
].Hrtf
.Target
= voice
->Direct
.Params
[0].Hrtf
.Target
;
820 /* Calculate the directional coefficients once, which apply to all
821 * input channels of the source sends.
823 CalcAngleCoeffs(Azi
, Elev
, Spread
, coeffs
);
825 for(i
= 0;i
< NumSends
;i
++)
827 const ALeffectslot
*Slot
= SendSlots
[i
];
829 for(c
= 0;c
< num_channels
;c
++)
832 if(chans
[c
].channel
!= LFE
)
833 ComputePanningGainsBF(Slot
->ChanMap
,
834 Slot
->NumChannels
, coeffs
, WetGain
[i
] * downmix_gain
,
835 voice
->Send
[i
].Params
[c
].Gains
.Target
842 /* Local sources on HRTF play with each channel panned to its
843 * relative location around the listener, providing "virtual
844 * speaker" responses.
846 for(c
= 0;c
< num_channels
;c
++)
848 ALfloat coeffs
[MAX_AMBI_COEFFS
];
850 if(chans
[c
].channel
== LFE
)
856 /* Get the HRIR coefficients and delays for this channel
859 GetHrtfCoeffs(Device
->HrtfHandle
,
860 chans
[c
].elevation
, chans
[c
].angle
, Spread
,
861 voice
->Direct
.Params
[c
].Hrtf
.Target
.Coeffs
,
862 voice
->Direct
.Params
[c
].Hrtf
.Target
.Delay
864 voice
->Direct
.Params
[c
].Hrtf
.Target
.Gain
= DryGain
;
866 /* Normal panning for auxiliary sends. */
867 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
869 for(i
= 0;i
< NumSends
;i
++)
871 const ALeffectslot
*Slot
= SendSlots
[i
];
873 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
874 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
880 voice
->Flags
|= VOICE_HAS_HRTF
;
884 /* Non-HRTF rendering. Use normal panning to the output. */
886 if(Distance
> FLT_EPSILON
)
888 ALfloat coeffs
[MAX_AMBI_COEFFS
];
891 /* Calculate NFC filter coefficient if needed. */
892 if(Device
->AvgSpeakerDist
> 0.0f
)
894 ALfloat mdist
= Distance
* Listener
->Params
.MetersPerUnit
;
895 ALfloat w1
= SPEEDOFSOUNDMETRESPERSEC
/
896 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
897 w0
= SPEEDOFSOUNDMETRESPERSEC
/
898 (mdist
* (ALfloat
)Device
->Frequency
);
899 /* Clamp w0 for really close distances, to prevent excessive
902 w0
= minf(w0
, w1
*4.0f
);
904 /* Adjust NFC filters. */
905 for(c
= 0;c
< num_channels
;c
++)
906 NfcFilterAdjust(&voice
->Direct
.Params
[c
].NFCtrlFilter
, w0
);
908 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
909 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
910 voice
->Flags
|= VOICE_HAS_NFC
;
913 /* Calculate the directional coefficients once, which apply to all
916 if(Device
->Render_Mode
== StereoPair
)
917 CalcAnglePairwiseCoeffs(Azi
, Elev
, Spread
, coeffs
);
919 CalcAngleCoeffs(Azi
, Elev
, Spread
, coeffs
);
921 for(c
= 0;c
< num_channels
;c
++)
923 /* Special-case LFE */
924 if(chans
[c
].channel
== LFE
)
926 if(Device
->Dry
.Buffer
== Device
->RealOut
.Buffer
)
928 int idx
= GetChannelIdxByName(&Device
->RealOut
, chans
[c
].channel
);
929 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
934 ComputeDryPanGains(&Device
->Dry
,
935 coeffs
, DryGain
* downmix_gain
, voice
->Direct
.Params
[c
].Gains
.Target
939 for(i
= 0;i
< NumSends
;i
++)
941 const ALeffectslot
*Slot
= SendSlots
[i
];
943 for(c
= 0;c
< num_channels
;c
++)
946 if(chans
[c
].channel
!= LFE
)
947 ComputePanningGainsBF(Slot
->ChanMap
,
948 Slot
->NumChannels
, coeffs
, WetGain
[i
] * downmix_gain
,
949 voice
->Send
[i
].Params
[c
].Gains
.Target
958 if(Device
->AvgSpeakerDist
> 0.0f
)
960 /* If the source distance is 0, set w0 to w1 to act as a pass-
961 * through. We still want to pass the signal through the
962 * filters so they keep an appropriate history, in case the
963 * source moves away from the listener.
965 w0
= SPEEDOFSOUNDMETRESPERSEC
/
966 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
968 for(c
= 0;c
< num_channels
;c
++)
969 NfcFilterAdjust(&voice
->Direct
.Params
[c
].NFCtrlFilter
, w0
);
971 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
972 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
973 voice
->Flags
|= VOICE_HAS_NFC
;
976 for(c
= 0;c
< num_channels
;c
++)
978 ALfloat coeffs
[MAX_AMBI_COEFFS
];
980 /* Special-case LFE */
981 if(chans
[c
].channel
== LFE
)
983 if(Device
->Dry
.Buffer
== Device
->RealOut
.Buffer
)
985 int idx
= GetChannelIdxByName(&Device
->RealOut
, chans
[c
].channel
);
986 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
991 if(Device
->Render_Mode
== StereoPair
)
992 CalcAnglePairwiseCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
994 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
995 ComputeDryPanGains(&Device
->Dry
,
996 coeffs
, DryGain
, voice
->Direct
.Params
[c
].Gains
.Target
999 for(i
= 0;i
< NumSends
;i
++)
1001 const ALeffectslot
*Slot
= SendSlots
[i
];
1003 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
1004 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
1012 ALfloat hfScale
= props
->Direct
.HFReference
/ Frequency
;
1013 ALfloat lfScale
= props
->Direct
.LFReference
/ Frequency
;
1014 ALfloat gainHF
= maxf(DryGainHF
, 0.001f
); /* Limit -60dB */
1015 ALfloat gainLF
= maxf(DryGainLF
, 0.001f
);
1017 voice
->Direct
.FilterType
= AF_None
;
1018 if(gainHF
!= 1.0f
) voice
->Direct
.FilterType
|= AF_LowPass
;
1019 if(gainLF
!= 1.0f
) voice
->Direct
.FilterType
|= AF_HighPass
;
1020 BiquadState_setParams(
1021 &voice
->Direct
.Params
[0].LowPass
, BiquadType_HighShelf
,
1022 gainHF
, hfScale
, calc_rcpQ_from_slope(gainHF
, 1.0f
)
1024 BiquadState_setParams(
1025 &voice
->Direct
.Params
[0].HighPass
, BiquadType_LowShelf
,
1026 gainLF
, lfScale
, calc_rcpQ_from_slope(gainLF
, 1.0f
)
1028 for(c
= 1;c
< num_channels
;c
++)
1030 BiquadState_copyParams(&voice
->Direct
.Params
[c
].LowPass
,
1031 &voice
->Direct
.Params
[0].LowPass
);
1032 BiquadState_copyParams(&voice
->Direct
.Params
[c
].HighPass
,
1033 &voice
->Direct
.Params
[0].HighPass
);
1036 for(i
= 0;i
< NumSends
;i
++)
1038 ALfloat hfScale
= props
->Send
[i
].HFReference
/ Frequency
;
1039 ALfloat lfScale
= props
->Send
[i
].LFReference
/ Frequency
;
1040 ALfloat gainHF
= maxf(WetGainHF
[i
], 0.001f
);
1041 ALfloat gainLF
= maxf(WetGainLF
[i
], 0.001f
);
1043 voice
->Send
[i
].FilterType
= AF_None
;
1044 if(gainHF
!= 1.0f
) voice
->Send
[i
].FilterType
|= AF_LowPass
;
1045 if(gainLF
!= 1.0f
) voice
->Send
[i
].FilterType
|= AF_HighPass
;
1046 BiquadState_setParams(
1047 &voice
->Send
[i
].Params
[0].LowPass
, BiquadType_HighShelf
,
1048 gainHF
, hfScale
, calc_rcpQ_from_slope(gainHF
, 1.0f
)
1050 BiquadState_setParams(
1051 &voice
->Send
[i
].Params
[0].HighPass
, BiquadType_LowShelf
,
1052 gainLF
, lfScale
, calc_rcpQ_from_slope(gainLF
, 1.0f
)
1054 for(c
= 1;c
< num_channels
;c
++)
1056 BiquadState_copyParams(&voice
->Send
[i
].Params
[c
].LowPass
,
1057 &voice
->Send
[i
].Params
[0].LowPass
);
1058 BiquadState_copyParams(&voice
->Send
[i
].Params
[c
].HighPass
,
1059 &voice
->Send
[i
].Params
[0].HighPass
);
1064 static void CalcNonAttnSourceParams(ALvoice
*voice
, const struct ALvoiceProps
*props
, const ALbuffer
*ALBuffer
, const ALCcontext
*ALContext
)
1066 const ALCdevice
*Device
= ALContext
->Device
;
1067 const ALlistener
*Listener
= ALContext
->Listener
;
1068 ALfloat DryGain
, DryGainHF
, DryGainLF
;
1069 ALfloat WetGain
[MAX_SENDS
];
1070 ALfloat WetGainHF
[MAX_SENDS
];
1071 ALfloat WetGainLF
[MAX_SENDS
];
1072 ALeffectslot
*SendSlots
[MAX_SENDS
];
1076 voice
->Direct
.Buffer
= Device
->Dry
.Buffer
;
1077 voice
->Direct
.Channels
= Device
->Dry
.NumChannels
;
1078 for(i
= 0;i
< Device
->NumAuxSends
;i
++)
1080 SendSlots
[i
] = props
->Send
[i
].Slot
;
1081 if(!SendSlots
[i
] && i
== 0)
1082 SendSlots
[i
] = ALContext
->DefaultSlot
;
1083 if(!SendSlots
[i
] || SendSlots
[i
]->Params
.EffectType
== AL_EFFECT_NULL
)
1085 SendSlots
[i
] = NULL
;
1086 voice
->Send
[i
].Buffer
= NULL
;
1087 voice
->Send
[i
].Channels
= 0;
1091 voice
->Send
[i
].Buffer
= SendSlots
[i
]->WetBuffer
;
1092 voice
->Send
[i
].Channels
= SendSlots
[i
]->NumChannels
;
1096 /* Calculate the stepping value */
1097 Pitch
= (ALfloat
)ALBuffer
->Frequency
/(ALfloat
)Device
->Frequency
* props
->Pitch
;
1098 if(Pitch
> (ALfloat
)MAX_PITCH
)
1099 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1101 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1102 if(props
->Resampler
== BSinc24Resampler
)
1103 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc24
);
1104 else if(props
->Resampler
== BSinc12Resampler
)
1105 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc12
);
1106 voice
->Resampler
= SelectResampler(props
->Resampler
);
1108 /* Calculate gains */
1109 DryGain
= clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
);
1110 DryGain
*= props
->Direct
.Gain
* Listener
->Params
.Gain
;
1111 DryGain
= minf(DryGain
, GAIN_MIX_MAX
);
1112 DryGainHF
= props
->Direct
.GainHF
;
1113 DryGainLF
= props
->Direct
.GainLF
;
1114 for(i
= 0;i
< Device
->NumAuxSends
;i
++)
1116 WetGain
[i
] = clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
);
1117 WetGain
[i
] *= props
->Send
[i
].Gain
* Listener
->Params
.Gain
;
1118 WetGain
[i
] = minf(WetGain
[i
], GAIN_MIX_MAX
);
1119 WetGainHF
[i
] = props
->Send
[i
].GainHF
;
1120 WetGainLF
[i
] = props
->Send
[i
].GainLF
;
1123 CalcPanningAndFilters(voice
, 0.0f
, 0.0f
, 0.0f
, 0.0f
, DryGain
, DryGainHF
, DryGainLF
, WetGain
,
1124 WetGainLF
, WetGainHF
, SendSlots
, ALBuffer
, props
, Listener
, Device
);
1127 static void CalcAttnSourceParams(ALvoice
*voice
, const struct ALvoiceProps
*props
, const ALbuffer
*ALBuffer
, const ALCcontext
*ALContext
)
1129 const ALCdevice
*Device
= ALContext
->Device
;
1130 const ALlistener
*Listener
= ALContext
->Listener
;
1131 const ALsizei NumSends
= Device
->NumAuxSends
;
1132 aluVector Position
, Velocity
, Direction
, SourceToListener
;
1133 ALfloat Distance
, ClampedDist
, DopplerFactor
;
1134 ALeffectslot
*SendSlots
[MAX_SENDS
];
1135 ALfloat RoomRolloff
[MAX_SENDS
];
1136 ALfloat DecayDistance
[MAX_SENDS
];
1137 ALfloat DecayLFDistance
[MAX_SENDS
];
1138 ALfloat DecayHFDistance
[MAX_SENDS
];
1139 ALfloat DryGain
, DryGainHF
, DryGainLF
;
1140 ALfloat WetGain
[MAX_SENDS
];
1141 ALfloat WetGainHF
[MAX_SENDS
];
1142 ALfloat WetGainLF
[MAX_SENDS
];
1149 /* Set mixing buffers and get send parameters. */
1150 voice
->Direct
.Buffer
= Device
->Dry
.Buffer
;
1151 voice
->Direct
.Channels
= Device
->Dry
.NumChannels
;
1152 for(i
= 0;i
< NumSends
;i
++)
1154 SendSlots
[i
] = props
->Send
[i
].Slot
;
1155 if(!SendSlots
[i
] && i
== 0)
1156 SendSlots
[i
] = ALContext
->DefaultSlot
;
1157 if(!SendSlots
[i
] || SendSlots
[i
]->Params
.EffectType
== AL_EFFECT_NULL
)
1159 SendSlots
[i
] = NULL
;
1160 RoomRolloff
[i
] = 0.0f
;
1161 DecayDistance
[i
] = 0.0f
;
1162 DecayLFDistance
[i
] = 0.0f
;
1163 DecayHFDistance
[i
] = 0.0f
;
1165 else if(SendSlots
[i
]->Params
.AuxSendAuto
)
1167 RoomRolloff
[i
] = SendSlots
[i
]->Params
.RoomRolloff
+ props
->RoomRolloffFactor
;
1168 /* Calculate the distances to where this effect's decay reaches
1171 DecayDistance
[i
] = SendSlots
[i
]->Params
.DecayTime
*
1172 Listener
->Params
.ReverbSpeedOfSound
;
1173 DecayLFDistance
[i
] = DecayDistance
[i
] * SendSlots
[i
]->Params
.DecayLFRatio
;
1174 DecayHFDistance
[i
] = DecayDistance
[i
] * SendSlots
[i
]->Params
.DecayHFRatio
;
1175 if(SendSlots
[i
]->Params
.DecayHFLimit
)
1177 ALfloat airAbsorption
= SendSlots
[i
]->Params
.AirAbsorptionGainHF
;
1178 if(airAbsorption
< 1.0f
)
1180 /* Calculate the distance to where this effect's air
1181 * absorption reaches -60dB, and limit the effect's HF
1182 * decay distance (so it doesn't take any longer to decay
1183 * than the air would allow).
1185 ALfloat absorb_dist
= log10f(REVERB_DECAY_GAIN
) / log10f(airAbsorption
);
1186 DecayHFDistance
[i
] = minf(absorb_dist
, DecayHFDistance
[i
]);
1192 /* If the slot's auxiliary send auto is off, the data sent to the
1193 * effect slot is the same as the dry path, sans filter effects */
1194 RoomRolloff
[i
] = props
->RolloffFactor
;
1195 DecayDistance
[i
] = 0.0f
;
1196 DecayLFDistance
[i
] = 0.0f
;
1197 DecayHFDistance
[i
] = 0.0f
;
1202 voice
->Send
[i
].Buffer
= NULL
;
1203 voice
->Send
[i
].Channels
= 0;
1207 voice
->Send
[i
].Buffer
= SendSlots
[i
]->WetBuffer
;
1208 voice
->Send
[i
].Channels
= SendSlots
[i
]->NumChannels
;
1212 /* Transform source to listener space (convert to head relative) */
1213 aluVectorSet(&Position
, props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
);
1214 aluVectorSet(&Direction
, props
->Direction
[0], props
->Direction
[1], props
->Direction
[2], 0.0f
);
1215 aluVectorSet(&Velocity
, props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
);
1216 if(props
->HeadRelative
== AL_FALSE
)
1218 const aluMatrixf
*Matrix
= &Listener
->Params
.Matrix
;
1219 /* Transform source vectors */
1220 Position
= aluMatrixfVector(Matrix
, &Position
);
1221 Velocity
= aluMatrixfVector(Matrix
, &Velocity
);
1222 Direction
= aluMatrixfVector(Matrix
, &Direction
);
1226 const aluVector
*lvelocity
= &Listener
->Params
.Velocity
;
1227 /* Offset the source velocity to be relative of the listener velocity */
1228 Velocity
.v
[0] += lvelocity
->v
[0];
1229 Velocity
.v
[1] += lvelocity
->v
[1];
1230 Velocity
.v
[2] += lvelocity
->v
[2];
1233 directional
= aluNormalize(Direction
.v
) > 0.0f
;
1234 SourceToListener
.v
[0] = -Position
.v
[0];
1235 SourceToListener
.v
[1] = -Position
.v
[1];
1236 SourceToListener
.v
[2] = -Position
.v
[2];
1237 SourceToListener
.v
[3] = 0.0f
;
1238 Distance
= aluNormalize(SourceToListener
.v
);
1240 /* Initial source gain */
1241 DryGain
= props
->Gain
;
1244 for(i
= 0;i
< NumSends
;i
++)
1246 WetGain
[i
] = props
->Gain
;
1247 WetGainHF
[i
] = 1.0f
;
1248 WetGainLF
[i
] = 1.0f
;
1251 /* Calculate distance attenuation */
1252 ClampedDist
= Distance
;
1254 switch(Listener
->Params
.SourceDistanceModel
?
1255 props
->DistanceModel
: Listener
->Params
.DistanceModel
)
1257 case InverseDistanceClamped
:
1258 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1259 if(props
->MaxDistance
< props
->RefDistance
)
1262 case InverseDistance
:
1263 if(!(props
->RefDistance
> 0.0f
))
1264 ClampedDist
= props
->RefDistance
;
1267 ALfloat dist
= lerp(props
->RefDistance
, ClampedDist
, props
->RolloffFactor
);
1268 if(dist
> 0.0f
) DryGain
*= props
->RefDistance
/ dist
;
1269 for(i
= 0;i
< NumSends
;i
++)
1271 dist
= lerp(props
->RefDistance
, ClampedDist
, RoomRolloff
[i
]);
1272 if(dist
> 0.0f
) WetGain
[i
] *= props
->RefDistance
/ dist
;
1277 case LinearDistanceClamped
:
1278 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1279 if(props
->MaxDistance
< props
->RefDistance
)
1282 case LinearDistance
:
1283 if(!(props
->MaxDistance
!= props
->RefDistance
))
1284 ClampedDist
= props
->RefDistance
;
1287 ALfloat attn
= props
->RolloffFactor
* (ClampedDist
-props
->RefDistance
) /
1288 (props
->MaxDistance
-props
->RefDistance
);
1289 DryGain
*= maxf(1.0f
- attn
, 0.0f
);
1290 for(i
= 0;i
< NumSends
;i
++)
1292 attn
= RoomRolloff
[i
] * (ClampedDist
-props
->RefDistance
) /
1293 (props
->MaxDistance
-props
->RefDistance
);
1294 WetGain
[i
] *= maxf(1.0f
- attn
, 0.0f
);
1299 case ExponentDistanceClamped
:
1300 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1301 if(props
->MaxDistance
< props
->RefDistance
)
1304 case ExponentDistance
:
1305 if(!(ClampedDist
> 0.0f
&& props
->RefDistance
> 0.0f
))
1306 ClampedDist
= props
->RefDistance
;
1309 DryGain
*= powf(ClampedDist
/props
->RefDistance
, -props
->RolloffFactor
);
1310 for(i
= 0;i
< NumSends
;i
++)
1311 WetGain
[i
] *= powf(ClampedDist
/props
->RefDistance
, -RoomRolloff
[i
]);
1315 case DisableDistance
:
1316 ClampedDist
= props
->RefDistance
;
1320 /* Calculate directional soundcones */
1321 if(directional
&& props
->InnerAngle
< 360.0f
)
1327 Angle
= acosf(aluDotproduct(&Direction
, &SourceToListener
));
1328 Angle
= RAD2DEG(Angle
* ConeScale
* 2.0f
);
1329 if(!(Angle
> props
->InnerAngle
))
1334 else if(Angle
< props
->OuterAngle
)
1336 ALfloat scale
= ( Angle
-props
->InnerAngle
) /
1337 (props
->OuterAngle
-props
->InnerAngle
);
1338 ConeVolume
= lerp(1.0f
, props
->OuterGain
, scale
);
1339 ConeHF
= lerp(1.0f
, props
->OuterGainHF
, scale
);
1343 ConeVolume
= props
->OuterGain
;
1344 ConeHF
= props
->OuterGainHF
;
1347 DryGain
*= ConeVolume
;
1348 if(props
->DryGainHFAuto
)
1349 DryGainHF
*= ConeHF
;
1350 if(props
->WetGainAuto
)
1352 for(i
= 0;i
< NumSends
;i
++)
1353 WetGain
[i
] *= ConeVolume
;
1355 if(props
->WetGainHFAuto
)
1357 for(i
= 0;i
< NumSends
;i
++)
1358 WetGainHF
[i
] *= ConeHF
;
1362 /* Apply gain and frequency filters */
1363 DryGain
= clampf(DryGain
, props
->MinGain
, props
->MaxGain
);
1364 DryGain
= minf(DryGain
*props
->Direct
.Gain
*Listener
->Params
.Gain
, GAIN_MIX_MAX
);
1365 DryGainHF
*= props
->Direct
.GainHF
;
1366 DryGainLF
*= props
->Direct
.GainLF
;
1367 for(i
= 0;i
< NumSends
;i
++)
1369 WetGain
[i
] = clampf(WetGain
[i
], props
->MinGain
, props
->MaxGain
);
1370 WetGain
[i
] = minf(WetGain
[i
]*props
->Send
[i
].Gain
*Listener
->Params
.Gain
, GAIN_MIX_MAX
);
1371 WetGainHF
[i
] *= props
->Send
[i
].GainHF
;
1372 WetGainLF
[i
] *= props
->Send
[i
].GainLF
;
1375 /* Distance-based air absorption and initial send decay. */
1376 if(ClampedDist
> props
->RefDistance
&& props
->RolloffFactor
> 0.0f
)
1378 ALfloat meters_base
= (ClampedDist
-props
->RefDistance
) * props
->RolloffFactor
*
1379 Listener
->Params
.MetersPerUnit
;
1380 if(props
->AirAbsorptionFactor
> 0.0f
)
1382 ALfloat hfattn
= powf(AIRABSORBGAINHF
, meters_base
* props
->AirAbsorptionFactor
);
1383 DryGainHF
*= hfattn
;
1384 for(i
= 0;i
< NumSends
;i
++)
1385 WetGainHF
[i
] *= hfattn
;
1388 if(props
->WetGainAuto
)
1390 /* Apply a decay-time transformation to the wet path, based on the
1391 * source distance in meters. The initial decay of the reverb
1392 * effect is calculated and applied to the wet path.
1394 for(i
= 0;i
< NumSends
;i
++)
1396 ALfloat gain
, gainhf
, gainlf
;
1398 if(!(DecayDistance
[i
] > 0.0f
))
1401 gain
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayDistance
[i
]);
1403 /* Yes, the wet path's air absorption is applied with
1404 * WetGainAuto on, rather than WetGainHFAuto.
1408 gainhf
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayHFDistance
[i
]);
1409 WetGainHF
[i
] *= minf(gainhf
/ gain
, 1.0f
);
1410 gainlf
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayLFDistance
[i
]);
1411 WetGainLF
[i
] *= minf(gainlf
/ gain
, 1.0f
);
1418 /* Initial source pitch */
1419 Pitch
= props
->Pitch
;
1421 /* Calculate velocity-based doppler effect */
1422 DopplerFactor
= props
->DopplerFactor
* Listener
->Params
.DopplerFactor
;
1423 if(DopplerFactor
> 0.0f
)
1425 const aluVector
*lvelocity
= &Listener
->Params
.Velocity
;
1426 const ALfloat SpeedOfSound
= Listener
->Params
.SpeedOfSound
;
1429 vss
= aluDotproduct(&Velocity
, &SourceToListener
) * DopplerFactor
;
1430 vls
= aluDotproduct(lvelocity
, &SourceToListener
) * DopplerFactor
;
1432 if(!(vls
< SpeedOfSound
))
1434 /* Listener moving away from the source at the speed of sound.
1435 * Sound waves can't catch it.
1439 else if(!(vss
< SpeedOfSound
))
1441 /* Source moving toward the listener at the speed of sound. Sound
1442 * waves bunch up to extreme frequencies.
1448 /* Source and listener movement is nominal. Calculate the proper
1451 Pitch
*= (SpeedOfSound
-vls
) / (SpeedOfSound
-vss
);
1455 /* Adjust pitch based on the buffer and output frequencies, and calculate
1456 * fixed-point stepping value.
1458 Pitch
*= (ALfloat
)ALBuffer
->Frequency
/(ALfloat
)Device
->Frequency
;
1459 if(Pitch
> (ALfloat
)MAX_PITCH
)
1460 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1462 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1463 if(props
->Resampler
== BSinc24Resampler
)
1464 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc24
);
1465 else if(props
->Resampler
== BSinc12Resampler
)
1466 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
, &bsinc12
);
1467 voice
->Resampler
= SelectResampler(props
->Resampler
);
1471 /* Clamp Y, in case rounding errors caused it to end up outside of
1474 ev
= asinf(clampf(-SourceToListener
.v
[1], -1.0f
, 1.0f
));
1475 /* Double negation on Z cancels out; negate once for changing source-
1476 * to-listener to listener-to-source, and again for right-handed coords
1479 az
= atan2f(-SourceToListener
.v
[0], SourceToListener
.v
[2]*ZScale
);
1484 if(props
->Radius
> Distance
)
1485 spread
= F_TAU
- Distance
/props
->Radius
*F_PI
;
1486 else if(Distance
> 0.0f
)
1487 spread
= asinf(props
->Radius
/ Distance
) * 2.0f
;
1491 CalcPanningAndFilters(voice
, az
, ev
, Distance
, spread
, DryGain
, DryGainHF
, DryGainLF
, WetGain
,
1492 WetGainLF
, WetGainHF
, SendSlots
, ALBuffer
, props
, Listener
, Device
);
1495 static void CalcSourceParams(ALvoice
*voice
, ALCcontext
*context
, bool force
)
1497 ALbufferlistitem
*BufferListItem
;
1498 struct ALvoiceProps
*props
;
1500 props
= ATOMIC_EXCHANGE_PTR(&voice
->Update
, NULL
, almemory_order_acq_rel
);
1501 if(!props
&& !force
) return;
1505 memcpy(voice
->Props
, props
,
1506 FAM_SIZE(struct ALvoiceProps
, Send
, context
->Device
->NumAuxSends
)
1509 ATOMIC_REPLACE_HEAD(struct ALvoiceProps
*, &context
->FreeVoiceProps
, props
);
1511 props
= voice
->Props
;
1513 BufferListItem
= ATOMIC_LOAD(&voice
->current_buffer
, almemory_order_relaxed
);
1514 while(BufferListItem
!= NULL
)
1516 const ALbuffer
*buffer
= NULL
;
1518 while(!buffer
&& i
< BufferListItem
->num_buffers
)
1519 buffer
= BufferListItem
->buffers
[i
];
1522 if(props
->SpatializeMode
== SpatializeOn
||
1523 (props
->SpatializeMode
== SpatializeAuto
&& buffer
->FmtChannels
== FmtMono
))
1524 CalcAttnSourceParams(voice
, props
, buffer
, context
);
1526 CalcNonAttnSourceParams(voice
, props
, buffer
, context
);
1529 BufferListItem
= ATOMIC_LOAD(&BufferListItem
->next
, almemory_order_acquire
);
1534 static void ProcessParamUpdates(ALCcontext
*ctx
, const struct ALeffectslotArray
*slots
)
1536 ALvoice
**voice
, **voice_end
;
1540 IncrementRef(&ctx
->UpdateCount
);
1541 if(!ATOMIC_LOAD(&ctx
->HoldUpdates
, almemory_order_acquire
))
1543 bool cforce
= CalcContextParams(ctx
);
1544 bool force
= CalcListenerParams(ctx
) | cforce
;
1545 for(i
= 0;i
< slots
->count
;i
++)
1546 force
|= CalcEffectSlotParams(slots
->slot
[i
], ctx
, cforce
);
1548 voice
= ctx
->Voices
;
1549 voice_end
= voice
+ ctx
->VoiceCount
;
1550 for(;voice
!= voice_end
;++voice
)
1552 source
= ATOMIC_LOAD(&(*voice
)->Source
, almemory_order_acquire
);
1553 if(source
) CalcSourceParams(*voice
, ctx
, force
);
1556 IncrementRef(&ctx
->UpdateCount
);
1560 static void ApplyStablizer(FrontStablizer
*Stablizer
, ALfloat (*restrict Buffer
)[BUFFERSIZE
],
1561 int lidx
, int ridx
, int cidx
, ALsizei SamplesToDo
,
1562 ALsizei NumChannels
)
1564 ALfloat (*restrict lsplit
)[BUFFERSIZE
] = ASSUME_ALIGNED(Stablizer
->LSplit
, 16);
1565 ALfloat (*restrict rsplit
)[BUFFERSIZE
] = ASSUME_ALIGNED(Stablizer
->RSplit
, 16);
1568 /* Apply an all-pass to all channels, except the front-left and front-
1569 * right, so they maintain the same relative phase.
1571 for(i
= 0;i
< NumChannels
;i
++)
1573 if(i
== lidx
|| i
== ridx
)
1575 splitterap_process(&Stablizer
->APFilter
[i
], Buffer
[i
], SamplesToDo
);
1578 bandsplit_process(&Stablizer
->LFilter
, lsplit
[1], lsplit
[0], Buffer
[lidx
], SamplesToDo
);
1579 bandsplit_process(&Stablizer
->RFilter
, rsplit
[1], rsplit
[0], Buffer
[ridx
], SamplesToDo
);
1581 for(i
= 0;i
< SamplesToDo
;i
++)
1583 ALfloat lfsum
, hfsum
;
1586 lfsum
= lsplit
[0][i
] + rsplit
[0][i
];
1587 hfsum
= lsplit
[1][i
] + rsplit
[1][i
];
1588 s
= lsplit
[0][i
] + lsplit
[1][i
] - rsplit
[0][i
] - rsplit
[1][i
];
1590 /* This pans the separate low- and high-frequency sums between being on
1591 * the center channel and the left/right channels. The low-frequency
1592 * sum is 1/3rd toward center (2/3rds on left/right) and the high-
1593 * frequency sum is 1/4th toward center (3/4ths on left/right). These
1594 * values can be tweaked.
1596 m
= lfsum
*cosf(1.0f
/3.0f
* F_PI_2
) + hfsum
*cosf(1.0f
/4.0f
* F_PI_2
);
1597 c
= lfsum
*sinf(1.0f
/3.0f
* F_PI_2
) + hfsum
*sinf(1.0f
/4.0f
* F_PI_2
);
1599 /* The generated center channel signal adds to the existing signal,
1600 * while the modified left and right channels replace.
1602 Buffer
[lidx
][i
] = (m
+ s
) * 0.5f
;
1603 Buffer
[ridx
][i
] = (m
- s
) * 0.5f
;
1604 Buffer
[cidx
][i
] += c
* 0.5f
;
1608 static void ApplyDistanceComp(ALfloat (*restrict Samples
)[BUFFERSIZE
], DistanceComp
*distcomp
,
1609 ALfloat
*restrict Values
, ALsizei SamplesToDo
, ALsizei numchans
)
1613 Values
= ASSUME_ALIGNED(Values
, 16);
1614 for(c
= 0;c
< numchans
;c
++)
1616 ALfloat
*restrict inout
= ASSUME_ALIGNED(Samples
[c
], 16);
1617 const ALfloat gain
= distcomp
[c
].Gain
;
1618 const ALsizei base
= distcomp
[c
].Length
;
1619 ALfloat
*restrict distbuf
= ASSUME_ALIGNED(distcomp
[c
].Buffer
, 16);
1625 for(i
= 0;i
< SamplesToDo
;i
++)
1631 if(SamplesToDo
>= base
)
1633 for(i
= 0;i
< base
;i
++)
1634 Values
[i
] = distbuf
[i
];
1635 for(;i
< SamplesToDo
;i
++)
1636 Values
[i
] = inout
[i
-base
];
1637 memcpy(distbuf
, &inout
[SamplesToDo
-base
], base
*sizeof(ALfloat
));
1641 for(i
= 0;i
< SamplesToDo
;i
++)
1642 Values
[i
] = distbuf
[i
];
1643 memmove(distbuf
, distbuf
+SamplesToDo
, (base
-SamplesToDo
)*sizeof(ALfloat
));
1644 memcpy(distbuf
+base
-SamplesToDo
, inout
, SamplesToDo
*sizeof(ALfloat
));
1646 for(i
= 0;i
< SamplesToDo
;i
++)
1647 inout
[i
] = Values
[i
]*gain
;
1651 static void ApplyDither(ALfloat (*restrict Samples
)[BUFFERSIZE
], ALuint
*dither_seed
,
1652 const ALfloat quant_scale
, const ALsizei SamplesToDo
,
1653 const ALsizei numchans
)
1655 const ALfloat invscale
= 1.0f
/ quant_scale
;
1656 ALuint seed
= *dither_seed
;
1659 /* Dithering. Step 1, generate whitenoise (uniform distribution of random
1660 * values between -1 and +1). Step 2 is to add the noise to the samples,
1661 * before rounding and after scaling up to the desired quantization depth.
1663 for(c
= 0;c
< numchans
;c
++)
1665 ALfloat
*restrict samples
= Samples
[c
];
1666 for(i
= 0;i
< SamplesToDo
;i
++)
1668 ALfloat val
= samples
[i
] * quant_scale
;
1669 ALuint rng0
= dither_rng(&seed
);
1670 ALuint rng1
= dither_rng(&seed
);
1671 val
+= (ALfloat
)(rng0
*(1.0/UINT_MAX
) - rng1
*(1.0/UINT_MAX
));
1672 samples
[i
] = roundf(val
) * invscale
;
1675 *dither_seed
= seed
;
1679 static inline ALfloat
Conv_ALfloat(ALfloat val
)
1681 static inline ALint
Conv_ALint(ALfloat val
)
1683 /* Floats only have a 24-bit mantissa, so [-16777216, +16777216] is the max
1684 * integer range normalized floats can be safely converted to (a bit of the
1685 * exponent helps out, effectively giving 25 bits).
1687 return fastf2i(clampf(val
*16777216.0f
, -16777216.0f
, 16777215.0f
))<<7;
1689 static inline ALshort
Conv_ALshort(ALfloat val
)
1690 { return fastf2i(clampf(val
*32768.0f
, -32768.0f
, 32767.0f
)); }
1691 static inline ALbyte
Conv_ALbyte(ALfloat val
)
1692 { return fastf2i(clampf(val
*128.0f
, -128.0f
, 127.0f
)); }
1694 /* Define unsigned output variations. */
1695 #define DECL_TEMPLATE(T, func, O) \
1696 static inline T Conv_##T(ALfloat val) { return func(val)+O; }
1698 DECL_TEMPLATE(ALubyte
, Conv_ALbyte
, 128)
1699 DECL_TEMPLATE(ALushort
, Conv_ALshort
, 32768)
1700 DECL_TEMPLATE(ALuint
, Conv_ALint
, 2147483648u)
1702 #undef DECL_TEMPLATE
1704 #define DECL_TEMPLATE(T, A) \
1705 static void Write##A(const ALfloat (*restrict InBuffer)[BUFFERSIZE], \
1706 ALvoid *OutBuffer, ALsizei Offset, ALsizei SamplesToDo, \
1710 for(j = 0;j < numchans;j++) \
1712 const ALfloat *restrict in = ASSUME_ALIGNED(InBuffer[j], 16); \
1713 T *restrict out = (T*)OutBuffer + Offset*numchans + j; \
1715 for(i = 0;i < SamplesToDo;i++) \
1716 out[i*numchans] = Conv_##T(in[i]); \
1720 DECL_TEMPLATE(ALfloat
, F32
)
1721 DECL_TEMPLATE(ALuint
, UI32
)
1722 DECL_TEMPLATE(ALint
, I32
)
1723 DECL_TEMPLATE(ALushort
, UI16
)
1724 DECL_TEMPLATE(ALshort
, I16
)
1725 DECL_TEMPLATE(ALubyte
, UI8
)
1726 DECL_TEMPLATE(ALbyte
, I8
)
1728 #undef DECL_TEMPLATE
1731 void aluMixData(ALCdevice
*device
, ALvoid
*OutBuffer
, ALsizei NumSamples
)
1733 ALsizei SamplesToDo
;
1734 ALsizei SamplesDone
;
1739 for(SamplesDone
= 0;SamplesDone
< NumSamples
;)
1741 SamplesToDo
= mini(NumSamples
-SamplesDone
, BUFFERSIZE
);
1742 for(c
= 0;c
< device
->Dry
.NumChannels
;c
++)
1743 memset(device
->Dry
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1744 if(device
->Dry
.Buffer
!= device
->FOAOut
.Buffer
)
1745 for(c
= 0;c
< device
->FOAOut
.NumChannels
;c
++)
1746 memset(device
->FOAOut
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1747 if(device
->Dry
.Buffer
!= device
->RealOut
.Buffer
)
1748 for(c
= 0;c
< device
->RealOut
.NumChannels
;c
++)
1749 memset(device
->RealOut
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1751 IncrementRef(&device
->MixCount
);
1753 ctx
= ATOMIC_LOAD(&device
->ContextList
, almemory_order_acquire
);
1756 const struct ALeffectslotArray
*auxslots
;
1758 auxslots
= ATOMIC_LOAD(&ctx
->ActiveAuxSlots
, almemory_order_acquire
);
1759 ProcessParamUpdates(ctx
, auxslots
);
1761 for(i
= 0;i
< auxslots
->count
;i
++)
1763 ALeffectslot
*slot
= auxslots
->slot
[i
];
1764 for(c
= 0;c
< slot
->NumChannels
;c
++)
1765 memset(slot
->WetBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1768 /* source processing */
1769 for(i
= 0;i
< ctx
->VoiceCount
;i
++)
1771 ALvoice
*voice
= ctx
->Voices
[i
];
1772 ALsource
*source
= ATOMIC_LOAD(&voice
->Source
, almemory_order_acquire
);
1773 if(source
&& ATOMIC_LOAD(&voice
->Playing
, almemory_order_relaxed
) &&
1776 if(!MixSource(voice
, source
->id
, ctx
, SamplesToDo
))
1778 ATOMIC_STORE(&voice
->Source
, NULL
, almemory_order_relaxed
);
1779 ATOMIC_STORE(&voice
->Playing
, false, almemory_order_release
);
1780 SendSourceStoppedEvent(ctx
, source
->id
);
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
);
1794 ctx
= ATOMIC_LOAD(&ctx
->next
, almemory_order_relaxed
);
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
);
1812 if(device
->Stablizer
)
1814 int lidx
= GetChannelIdxByName(&device
->RealOut
, FrontLeft
);
1815 int ridx
= GetChannelIdxByName(&device
->RealOut
, FrontRight
);
1816 int cidx
= GetChannelIdxByName(&device
->RealOut
, FrontCenter
);
1817 assert(lidx
>= 0 && ridx
>= 0 && cidx
>= 0);
1819 ApplyStablizer(device
->Stablizer
, device
->RealOut
.Buffer
, lidx
, ridx
, cidx
,
1820 SamplesToDo
, device
->RealOut
.NumChannels
);
1823 ApplyDistanceComp(device
->RealOut
.Buffer
, device
->ChannelDelay
, device
->TempBuffer
[0],
1824 SamplesToDo
, device
->RealOut
.NumChannels
);
1827 ApplyCompression(device
->Limiter
, device
->RealOut
.NumChannels
, SamplesToDo
,
1828 device
->RealOut
.Buffer
);
1830 if(device
->DitherDepth
> 0.0f
)
1831 ApplyDither(device
->RealOut
.Buffer
, &device
->DitherSeed
, device
->DitherDepth
,
1832 SamplesToDo
, device
->RealOut
.NumChannels
);
1836 ALfloat (*Buffer
)[BUFFERSIZE
] = device
->RealOut
.Buffer
;
1837 ALsizei Channels
= device
->RealOut
.NumChannels
;
1839 switch(device
->FmtType
)
1842 WriteI8(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1845 WriteUI8(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1848 WriteI16(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1851 WriteUI16(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1854 WriteI32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1857 WriteUI32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1860 WriteF32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1865 SamplesDone
+= SamplesToDo
;
1871 void aluHandleDisconnect(ALCdevice
*device
, const char *msg
, ...)
1878 if(!ATOMIC_EXCHANGE(&device
->Connected
, AL_FALSE
, almemory_order_acq_rel
))
1881 evt
.EnumType
= EventType_Disconnected
;
1882 evt
.Type
= AL_EVENT_TYPE_DISCONNECTED_SOFT
;
1886 va_start(args
, msg
);
1887 msglen
= vsnprintf(evt
.Message
, sizeof(evt
.Message
), msg
, args
);
1890 if(msglen
< 0 || (size_t)msglen
>= sizeof(evt
.Message
))
1892 evt
.Message
[sizeof(evt
.Message
)-1] = 0;
1893 msglen
= (int)strlen(evt
.Message
);
1899 msg
= "<internal error constructing message>";
1900 msglen
= (int)strlen(msg
);
1903 ctx
= ATOMIC_LOAD_SEQ(&device
->ContextList
);
1906 ALbitfieldSOFT enabledevt
= ATOMIC_LOAD(&ctx
->EnabledEvts
, almemory_order_acquire
);
1909 if((enabledevt
&EventType_Disconnected
) &&
1910 ll_ringbuffer_write(ctx
->AsyncEvents
, (const char*)&evt
, 1) == 1)
1911 alsem_post(&ctx
->EventSem
);
1913 for(i
= 0;i
< ctx
->VoiceCount
;i
++)
1915 ALvoice
*voice
= ctx
->Voices
[i
];
1918 source
= ATOMIC_EXCHANGE_PTR(&voice
->Source
, NULL
, almemory_order_relaxed
);
1919 if(source
&& ATOMIC_LOAD(&voice
->Playing
, almemory_order_relaxed
))
1921 /* If the source's voice was playing, it's now effectively
1922 * stopped (the source state will be updated the next time it's
1925 SendSourceStoppedEvent(ctx
, source
->id
);
1927 ATOMIC_STORE(&voice
->Playing
, false, almemory_order_release
);
1930 ctx
= ATOMIC_LOAD(&ctx
->next
, almemory_order_relaxed
);