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 "uhjfilter.h"
38 #include "bformatdec.h"
39 #include "static_assert.h"
41 #include "mixer_defs.h"
42 #include "bsinc_inc.c"
44 #include "backends/base.h"
47 extern inline ALfloat
minf(ALfloat a
, ALfloat b
);
48 extern inline ALfloat
maxf(ALfloat a
, ALfloat b
);
49 extern inline ALfloat
clampf(ALfloat val
, ALfloat min
, ALfloat max
);
51 extern inline ALdouble
mind(ALdouble a
, ALdouble b
);
52 extern inline ALdouble
maxd(ALdouble a
, ALdouble b
);
53 extern inline ALdouble
clampd(ALdouble val
, ALdouble min
, ALdouble max
);
55 extern inline ALuint
minu(ALuint a
, ALuint b
);
56 extern inline ALuint
maxu(ALuint a
, ALuint b
);
57 extern inline ALuint
clampu(ALuint val
, ALuint min
, ALuint max
);
59 extern inline ALint
mini(ALint a
, ALint b
);
60 extern inline ALint
maxi(ALint a
, ALint b
);
61 extern inline ALint
clampi(ALint val
, ALint min
, ALint max
);
63 extern inline ALint64
mini64(ALint64 a
, ALint64 b
);
64 extern inline ALint64
maxi64(ALint64 a
, ALint64 b
);
65 extern inline ALint64
clampi64(ALint64 val
, ALint64 min
, ALint64 max
);
67 extern inline ALuint64
minu64(ALuint64 a
, ALuint64 b
);
68 extern inline ALuint64
maxu64(ALuint64 a
, ALuint64 b
);
69 extern inline ALuint64
clampu64(ALuint64 val
, ALuint64 min
, ALuint64 max
);
71 extern inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALfloat mu
);
72 extern inline ALfloat
resample_fir4(ALfloat val0
, ALfloat val1
, ALfloat val2
, ALfloat val3
, ALsizei frac
);
74 extern inline void aluVectorSet(aluVector
*restrict vector
, ALfloat x
, ALfloat y
, ALfloat z
, ALfloat w
);
76 extern inline void aluMatrixfSetRow(aluMatrixf
*matrix
, ALuint row
,
77 ALfloat m0
, ALfloat m1
, ALfloat m2
, ALfloat m3
);
78 extern inline void aluMatrixfSet(aluMatrixf
*matrix
,
79 ALfloat m00
, ALfloat m01
, ALfloat m02
, ALfloat m03
,
80 ALfloat m10
, ALfloat m11
, ALfloat m12
, ALfloat m13
,
81 ALfloat m20
, ALfloat m21
, ALfloat m22
, ALfloat m23
,
82 ALfloat m30
, ALfloat m31
, ALfloat m32
, ALfloat m33
);
86 ALfloat ConeScale
= 1.0f
;
88 /* Localized Z scalar for mono sources */
89 ALfloat ZScale
= 1.0f
;
91 const aluMatrixf IdentityMatrixf
= {{
92 { 1.0f
, 0.0f
, 0.0f
, 0.0f
},
93 { 0.0f
, 1.0f
, 0.0f
, 0.0f
},
94 { 0.0f
, 0.0f
, 1.0f
, 0.0f
},
95 { 0.0f
, 0.0f
, 0.0f
, 1.0f
},
100 enum Channel channel
;
105 static HrtfDirectMixerFunc MixDirectHrtf
= MixDirectHrtf_C
;
108 void DeinitVoice(ALvoice
*voice
)
110 struct ALvoiceProps
*props
;
113 props
= ATOMIC_EXCHANGE_PTR_SEQ(&voice
->Update
, NULL
);
114 if(props
) al_free(props
);
116 props
= ATOMIC_EXCHANGE_PTR(&voice
->FreeList
, NULL
, almemory_order_relaxed
);
119 struct ALvoiceProps
*next
;
120 next
= ATOMIC_LOAD(&props
->next
, almemory_order_relaxed
);
125 /* This is excessively spammy if it traces every voice destruction, so just
126 * warn if it was unexpectedly large.
129 WARN("Freed "SZFMT
" voice property objects\n", count
);
133 static inline HrtfDirectMixerFunc
SelectHrtfMixer(void)
136 if((CPUCapFlags
&CPU_CAP_NEON
))
137 return MixDirectHrtf_Neon
;
140 if((CPUCapFlags
&CPU_CAP_SSE
))
141 return MixDirectHrtf_SSE
;
144 return MixDirectHrtf_C
;
148 /* Prior to VS2013, MSVC lacks the round() family of functions. */
149 #if defined(_MSC_VER) && _MSC_VER < 1800
150 static float roundf(float val
)
153 return ceilf(val
-0.5f
);
154 return floorf(val
+0.5f
);
158 /* This RNG method was created based on the math found in opusdec. It's quick,
159 * and starting with a seed value of 22222, is suitable for generating
162 static inline ALuint
dither_rng(ALuint
*seed
)
164 *seed
= (*seed
* 96314165) + 907633515;
169 static inline void aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
171 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
172 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
173 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
176 static inline ALfloat
aluDotproduct(const aluVector
*vec1
, const aluVector
*vec2
)
178 return vec1
->v
[0]*vec2
->v
[0] + vec1
->v
[1]*vec2
->v
[1] + vec1
->v
[2]*vec2
->v
[2];
181 static ALfloat
aluNormalize(ALfloat
*vec
)
183 ALfloat length
= sqrtf(vec
[0]*vec
[0] + vec
[1]*vec
[1] + vec
[2]*vec
[2]);
186 ALfloat inv_length
= 1.0f
/length
;
187 vec
[0] *= inv_length
;
188 vec
[1] *= inv_length
;
189 vec
[2] *= inv_length
;
194 static void aluMatrixfFloat3(ALfloat
*vec
, ALfloat w
, const aluMatrixf
*mtx
)
196 ALfloat v
[4] = { vec
[0], vec
[1], vec
[2], w
};
198 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];
199 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];
200 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];
203 static aluVector
aluMatrixfVector(const aluMatrixf
*mtx
, const aluVector
*vec
)
206 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];
207 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];
208 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];
209 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];
216 MixDirectHrtf
= SelectHrtfMixer();
219 /* Prepares the interpolator for a given rate (determined by increment). A
220 * result of AL_FALSE indicates that the filter output will completely cut
223 * With a bit of work, and a trade of memory for CPU cost, this could be
224 * modified for use with an interpolated increment for buttery-smooth pitch
227 ALboolean
BsincPrepare(const ALuint increment
, BsincState
*state
)
229 ALboolean uncut
= AL_TRUE
;
233 if(increment
> FRACTIONONE
)
235 sf
= (ALfloat
)FRACTIONONE
/ increment
;
236 if(sf
< bsinc
.scaleBase
)
238 /* Signal has been completely cut. The return result can be used
239 * to skip the filter (and output zeros) as an optimization.
247 sf
= (BSINC_SCALE_COUNT
- 1) * (sf
- bsinc
.scaleBase
) * bsinc
.scaleRange
;
249 /* The interpolation factor is fit to this diagonally-symmetric
250 * curve to reduce the transition ripple caused by interpolating
251 * different scales of the sinc function.
253 sf
= 1.0f
- cosf(asinf(sf
- si
));
259 si
= BSINC_SCALE_COUNT
- 1;
263 state
->m
= bsinc
.m
[si
];
264 state
->l
= -((state
->m
/2) - 1);
265 state
->filter
= bsinc
.Tab
+ bsinc
.filterOffset
[si
];
270 static ALboolean
CalcListenerParams(ALCcontext
*Context
)
272 ALlistener
*Listener
= Context
->Listener
;
273 ALfloat N
[3], V
[3], U
[3], P
[3];
274 struct ALlistenerProps
*props
;
277 props
= ATOMIC_EXCHANGE_PTR(&Listener
->Update
, NULL
, almemory_order_acq_rel
);
278 if(!props
) return AL_FALSE
;
281 N
[0] = props
->Forward
[0];
282 N
[1] = props
->Forward
[1];
283 N
[2] = props
->Forward
[2];
289 /* Build and normalize right-vector */
290 aluCrossproduct(N
, V
, U
);
293 aluMatrixfSet(&Listener
->Params
.Matrix
,
294 U
[0], V
[0], -N
[0], 0.0,
295 U
[1], V
[1], -N
[1], 0.0,
296 U
[2], V
[2], -N
[2], 0.0,
300 P
[0] = props
->Position
[0];
301 P
[1] = props
->Position
[1];
302 P
[2] = props
->Position
[2];
303 aluMatrixfFloat3(P
, 1.0, &Listener
->Params
.Matrix
);
304 aluMatrixfSetRow(&Listener
->Params
.Matrix
, 3, -P
[0], -P
[1], -P
[2], 1.0f
);
306 aluVectorSet(&vel
, props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
);
307 Listener
->Params
.Velocity
= aluMatrixfVector(&Listener
->Params
.Matrix
, &vel
);
309 Listener
->Params
.Gain
= props
->Gain
* Context
->GainBoost
;
310 Listener
->Params
.MetersPerUnit
= props
->MetersPerUnit
;
312 Listener
->Params
.DopplerFactor
= props
->DopplerFactor
;
313 Listener
->Params
.SpeedOfSound
= props
->SpeedOfSound
* props
->DopplerVelocity
;
315 Listener
->Params
.SourceDistanceModel
= props
->SourceDistanceModel
;
316 Listener
->Params
.DistanceModel
= props
->DistanceModel
;
318 ATOMIC_REPLACE_HEAD(struct ALlistenerProps
*, &Listener
->FreeList
, props
);
322 static ALboolean
CalcEffectSlotParams(ALeffectslot
*slot
, ALCdevice
*device
)
324 struct ALeffectslotProps
*props
;
325 ALeffectState
*state
;
327 props
= ATOMIC_EXCHANGE_PTR(&slot
->Update
, NULL
, almemory_order_acq_rel
);
328 if(!props
) return AL_FALSE
;
330 slot
->Params
.Gain
= props
->Gain
;
331 slot
->Params
.AuxSendAuto
= props
->AuxSendAuto
;
332 slot
->Params
.EffectType
= props
->Type
;
333 if(IsReverbEffect(slot
->Params
.EffectType
))
335 slot
->Params
.RoomRolloff
= props
->Props
.Reverb
.RoomRolloffFactor
;
336 slot
->Params
.DecayTime
= props
->Props
.Reverb
.DecayTime
;
337 slot
->Params
.DecayHFRatio
= props
->Props
.Reverb
.DecayHFRatio
;
338 slot
->Params
.DecayHFLimit
= props
->Props
.Reverb
.DecayHFLimit
;
339 slot
->Params
.AirAbsorptionGainHF
= props
->Props
.Reverb
.AirAbsorptionGainHF
;
343 slot
->Params
.RoomRolloff
= 0.0f
;
344 slot
->Params
.DecayTime
= 0.0f
;
345 slot
->Params
.DecayHFRatio
= 0.0f
;
346 slot
->Params
.DecayHFLimit
= AL_FALSE
;
347 slot
->Params
.AirAbsorptionGainHF
= 1.0f
;
350 /* Swap effect states. No need to play with the ref counts since they keep
351 * the same number of refs.
353 state
= props
->State
;
354 props
->State
= slot
->Params
.EffectState
;
355 slot
->Params
.EffectState
= state
;
357 V(state
,update
)(device
, slot
, &props
->Props
);
359 ATOMIC_REPLACE_HEAD(struct ALeffectslotProps
*, &slot
->FreeList
, props
);
364 static const struct ChanMap MonoMap
[1] = {
365 { FrontCenter
, 0.0f
, 0.0f
}
367 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
368 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) }
370 { FrontLeft
, DEG2RAD( -45.0f
), DEG2RAD(0.0f
) },
371 { FrontRight
, DEG2RAD( 45.0f
), DEG2RAD(0.0f
) },
372 { BackLeft
, DEG2RAD(-135.0f
), DEG2RAD(0.0f
) },
373 { BackRight
, DEG2RAD( 135.0f
), DEG2RAD(0.0f
) }
375 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
376 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
377 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
379 { SideLeft
, DEG2RAD(-110.0f
), DEG2RAD(0.0f
) },
380 { SideRight
, DEG2RAD( 110.0f
), DEG2RAD(0.0f
) }
382 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
383 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
384 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
386 { BackCenter
, DEG2RAD(180.0f
), DEG2RAD(0.0f
) },
387 { SideLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
388 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
390 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
391 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
392 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
394 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
395 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) },
396 { SideLeft
, DEG2RAD( -90.0f
), DEG2RAD(0.0f
) },
397 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
400 static void CalcPanningAndFilters(ALvoice
*voice
, const ALfloat Distance
, const ALfloat
*Dir
,
401 const ALfloat Spread
, const ALfloat DryGain
,
402 const ALfloat DryGainHF
, const ALfloat DryGainLF
,
403 const ALfloat
*WetGain
, const ALfloat
*WetGainLF
,
404 const ALfloat
*WetGainHF
, ALeffectslot
**SendSlots
,
405 const ALbuffer
*Buffer
, const struct ALvoiceProps
*props
,
406 const ALlistener
*Listener
, const ALCdevice
*Device
)
408 struct ChanMap StereoMap
[2] = {
409 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
410 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) }
412 bool DirectChannels
= props
->DirectChannels
;
413 const ALsizei NumSends
= Device
->NumAuxSends
;
414 const ALuint Frequency
= Device
->Frequency
;
415 const struct ChanMap
*chans
= NULL
;
416 ALsizei num_channels
= 0;
417 bool isbformat
= false;
418 ALfloat downmix_gain
= 1.0f
;
421 switch(Buffer
->FmtChannels
)
426 /* Mono buffers are never played direct. */
427 DirectChannels
= false;
431 /* Convert counter-clockwise to clockwise. */
432 StereoMap
[0].angle
= -props
->StereoPan
[0];
433 StereoMap
[1].angle
= -props
->StereoPan
[1];
437 downmix_gain
= 1.0f
/ 2.0f
;
443 downmix_gain
= 1.0f
/ 2.0f
;
449 downmix_gain
= 1.0f
/ 4.0f
;
455 /* NOTE: Excludes LFE. */
456 downmix_gain
= 1.0f
/ 5.0f
;
462 /* NOTE: Excludes LFE. */
463 downmix_gain
= 1.0f
/ 6.0f
;
469 /* NOTE: Excludes LFE. */
470 downmix_gain
= 1.0f
/ 7.0f
;
476 DirectChannels
= false;
482 DirectChannels
= false;
486 voice
->Flags
&= ~(VOICE_HAS_HRTF
| VOICE_HAS_NFC
);
489 /* Special handling for B-Format sources. */
491 if(Distance
> FLT_EPSILON
)
493 /* Panning a B-Format sound toward some direction is easy. Just pan
494 * the first (W) channel as a normal mono sound and silence the
497 ALfloat coeffs
[MAX_AMBI_COEFFS
];
499 if(Device
->AvgSpeakerDist
> 0.0f
&& Listener
->Params
.MetersPerUnit
> 0.0f
)
501 ALfloat mdist
= Distance
* Listener
->Params
.MetersPerUnit
;
502 ALfloat w0
= SPEEDOFSOUNDMETRESPERSEC
/
503 (mdist
* (ALfloat
)Device
->Frequency
);
504 ALfloat w1
= SPEEDOFSOUNDMETRESPERSEC
/
505 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
506 /* Clamp w0 for really close distances, to prevent excessive
509 w0
= minf(w0
, w1
*4.0f
);
511 /* Only need to adjust the first channel of a B-Format source. */
512 NfcFilterAdjust1(&voice
->Direct
.Params
[0].NFCtrlFilter
[0], w0
);
513 NfcFilterAdjust2(&voice
->Direct
.Params
[0].NFCtrlFilter
[1], w0
);
514 NfcFilterAdjust3(&voice
->Direct
.Params
[0].NFCtrlFilter
[2], w0
);
516 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
517 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
518 voice
->Flags
|= VOICE_HAS_NFC
;
521 if(Device
->Render_Mode
== StereoPair
)
523 ALfloat ev
= asinf(Dir
[1]);
524 ALfloat az
= atan2f(Dir
[0], -Dir
[2]);
525 CalcAnglePairwiseCoeffs(az
, ev
, Spread
, coeffs
);
528 CalcDirectionCoeffs(Dir
, Spread
, coeffs
);
530 /* NOTE: W needs to be scaled by sqrt(2) due to FuMa normalization. */
531 ComputePanningGains(Device
->Dry
, coeffs
, DryGain
*1.414213562f
,
532 voice
->Direct
.Params
[0].Gains
.Target
);
533 for(c
= 1;c
< num_channels
;c
++)
535 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
536 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
539 for(i
= 0;i
< NumSends
;i
++)
541 const ALeffectslot
*Slot
= SendSlots
[i
];
543 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
544 coeffs
, WetGain
[i
]*1.414213562f
, voice
->Send
[i
].Params
[0].Gains
.Target
547 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
548 voice
->Send
[i
].Params
[0].Gains
.Target
[j
] = 0.0f
;
549 for(c
= 1;c
< num_channels
;c
++)
551 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
552 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
558 /* Local B-Format sources have their XYZ channels rotated according
559 * to the orientation.
561 ALfloat N
[3], V
[3], U
[3];
565 if(Device
->AvgSpeakerDist
> 0.0f
)
567 /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
568 * is what we want for FOA input. The first channel may have
569 * been previously re-adjusted if panned, so reset it.
571 NfcFilterAdjust1(&voice
->Direct
.Params
[0].NFCtrlFilter
[0], 0.0f
);
572 NfcFilterAdjust2(&voice
->Direct
.Params
[0].NFCtrlFilter
[1], 0.0f
);
573 NfcFilterAdjust3(&voice
->Direct
.Params
[0].NFCtrlFilter
[2], 0.0f
);
575 voice
->Direct
.ChannelsPerOrder
[0] = 1;
576 voice
->Direct
.ChannelsPerOrder
[1] = mini(voice
->Direct
.Channels
-1, 3);
577 for(i
= 2;i
< MAX_AMBI_ORDER
+1;i
++)
578 voice
->Direct
.ChannelsPerOrder
[2] = 0;
579 voice
->Flags
|= VOICE_HAS_NFC
;
583 N
[0] = props
->Orientation
[0][0];
584 N
[1] = props
->Orientation
[0][1];
585 N
[2] = props
->Orientation
[0][2];
587 V
[0] = props
->Orientation
[1][0];
588 V
[1] = props
->Orientation
[1][1];
589 V
[2] = props
->Orientation
[1][2];
591 if(!props
->HeadRelative
)
593 const aluMatrixf
*lmatrix
= &Listener
->Params
.Matrix
;
594 aluMatrixfFloat3(N
, 0.0f
, lmatrix
);
595 aluMatrixfFloat3(V
, 0.0f
, lmatrix
);
597 /* Build and normalize right-vector */
598 aluCrossproduct(N
, V
, U
);
601 /* Build a rotate + conversion matrix (FuMa -> ACN+N3D). */
602 scale
= 1.732050808f
;
603 aluMatrixfSet(&matrix
,
604 1.414213562f
, 0.0f
, 0.0f
, 0.0f
,
605 0.0f
, -N
[0]*scale
, N
[1]*scale
, -N
[2]*scale
,
606 0.0f
, U
[0]*scale
, -U
[1]*scale
, U
[2]*scale
,
607 0.0f
, -V
[0]*scale
, V
[1]*scale
, -V
[2]*scale
610 voice
->Direct
.Buffer
= Device
->FOAOut
.Buffer
;
611 voice
->Direct
.Channels
= Device
->FOAOut
.NumChannels
;
612 for(c
= 0;c
< num_channels
;c
++)
613 ComputeFirstOrderGains(Device
->FOAOut
, matrix
.m
[c
], DryGain
,
614 voice
->Direct
.Params
[c
].Gains
.Target
);
615 for(i
= 0;i
< NumSends
;i
++)
617 const ALeffectslot
*Slot
= SendSlots
[i
];
620 for(c
= 0;c
< num_channels
;c
++)
621 ComputeFirstOrderGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
622 matrix
.m
[c
], WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
627 for(c
= 0;c
< num_channels
;c
++)
628 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
629 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
634 else if(DirectChannels
)
636 /* Direct source channels always play local. Skip the virtual channels
637 * and write inputs to the matching real outputs.
639 voice
->Direct
.Buffer
= Device
->RealOut
.Buffer
;
640 voice
->Direct
.Channels
= Device
->RealOut
.NumChannels
;
642 for(c
= 0;c
< num_channels
;c
++)
645 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
646 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
647 if((idx
=GetChannelIdxByName(Device
->RealOut
, chans
[c
].channel
)) != -1)
648 voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
651 /* Auxiliary sends still use normal channel panning since they mix to
652 * B-Format, which can't channel-match.
654 for(c
= 0;c
< num_channels
;c
++)
656 ALfloat coeffs
[MAX_AMBI_COEFFS
];
657 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, 0.0f
, coeffs
);
659 for(i
= 0;i
< NumSends
;i
++)
661 const ALeffectslot
*Slot
= SendSlots
[i
];
663 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
664 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
667 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
668 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
672 else if(Device
->Render_Mode
== HrtfRender
)
674 /* Full HRTF rendering. Skip the virtual channels and render to the
677 voice
->Direct
.Buffer
= Device
->RealOut
.Buffer
;
678 voice
->Direct
.Channels
= Device
->RealOut
.NumChannels
;
680 if(Distance
> FLT_EPSILON
)
682 ALfloat coeffs
[MAX_AMBI_COEFFS
];
686 az
= atan2f(Dir
[0], -Dir
[2]);
688 /* Get the HRIR coefficients and delays just once, for the given
691 GetHrtfCoeffs(Device
->HrtfHandle
, ev
, az
, Spread
,
692 voice
->Direct
.Params
[0].Hrtf
.Target
.Coeffs
,
693 voice
->Direct
.Params
[0].Hrtf
.Target
.Delay
);
694 voice
->Direct
.Params
[0].Hrtf
.Target
.Gain
= DryGain
* downmix_gain
;
696 /* Remaining channels use the same results as the first. */
697 for(c
= 1;c
< num_channels
;c
++)
700 if(chans
[c
].channel
== LFE
)
701 memset(&voice
->Direct
.Params
[c
].Hrtf
.Target
, 0,
702 sizeof(voice
->Direct
.Params
[c
].Hrtf
.Target
));
704 voice
->Direct
.Params
[c
].Hrtf
.Target
= voice
->Direct
.Params
[0].Hrtf
.Target
;
707 /* Calculate the directional coefficients once, which apply to all
708 * input channels of the source sends.
710 CalcDirectionCoeffs(Dir
, Spread
, coeffs
);
712 for(i
= 0;i
< NumSends
;i
++)
714 const ALeffectslot
*Slot
= SendSlots
[i
];
716 for(c
= 0;c
< num_channels
;c
++)
719 if(chans
[c
].channel
== LFE
)
720 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
721 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
723 ComputePanningGainsBF(Slot
->ChanMap
,
724 Slot
->NumChannels
, coeffs
, WetGain
[i
] * downmix_gain
,
725 voice
->Send
[i
].Params
[c
].Gains
.Target
729 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
730 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
735 /* Local sources on HRTF play with each channel panned to its
736 * relative location around the listener, providing "virtual
737 * speaker" responses.
739 for(c
= 0;c
< num_channels
;c
++)
741 ALfloat coeffs
[MAX_AMBI_COEFFS
];
743 if(chans
[c
].channel
== LFE
)
746 memset(&voice
->Direct
.Params
[c
].Hrtf
.Target
, 0,
747 sizeof(voice
->Direct
.Params
[c
].Hrtf
.Target
));
748 for(i
= 0;i
< NumSends
;i
++)
750 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
751 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
756 /* Get the HRIR coefficients and delays for this channel
759 GetHrtfCoeffs(Device
->HrtfHandle
,
760 chans
[c
].elevation
, chans
[c
].angle
, Spread
,
761 voice
->Direct
.Params
[c
].Hrtf
.Target
.Coeffs
,
762 voice
->Direct
.Params
[c
].Hrtf
.Target
.Delay
764 voice
->Direct
.Params
[c
].Hrtf
.Target
.Gain
= DryGain
;
766 /* Normal panning for auxiliary sends. */
767 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
769 for(i
= 0;i
< NumSends
;i
++)
771 const ALeffectslot
*Slot
= SendSlots
[i
];
773 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
774 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
777 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
778 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
783 voice
->Flags
|= VOICE_HAS_HRTF
;
787 /* Non-HRTF rendering. Use normal panning to the output. */
789 if(Distance
> FLT_EPSILON
)
791 ALfloat coeffs
[MAX_AMBI_COEFFS
];
794 /* Calculate NFC filter coefficient if needed. */
795 if(Device
->AvgSpeakerDist
> 0.0f
&& Listener
->Params
.MetersPerUnit
> 0.0f
)
797 ALfloat mdist
= Distance
* Listener
->Params
.MetersPerUnit
;
798 ALfloat w1
= SPEEDOFSOUNDMETRESPERSEC
/
799 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
800 w0
= SPEEDOFSOUNDMETRESPERSEC
/
801 (mdist
* (ALfloat
)Device
->Frequency
);
802 /* Clamp w0 for really close distances, to prevent excessive
805 w0
= minf(w0
, w1
*4.0f
);
807 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
808 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
809 voice
->Flags
|= VOICE_HAS_NFC
;
812 /* Calculate the directional coefficients once, which apply to all
815 if(Device
->Render_Mode
== StereoPair
)
817 ALfloat ev
= asinf(Dir
[1]);
818 ALfloat az
= atan2f(Dir
[0], -Dir
[2]);
819 CalcAnglePairwiseCoeffs(az
, ev
, Spread
, coeffs
);
822 CalcDirectionCoeffs(Dir
, Spread
, coeffs
);
824 for(c
= 0;c
< num_channels
;c
++)
826 /* Adjust NFC filters if needed. */
827 if((voice
->Flags
&VOICE_HAS_NFC
))
829 NfcFilterAdjust1(&voice
->Direct
.Params
[c
].NFCtrlFilter
[0], w0
);
830 NfcFilterAdjust2(&voice
->Direct
.Params
[c
].NFCtrlFilter
[1], w0
);
831 NfcFilterAdjust3(&voice
->Direct
.Params
[c
].NFCtrlFilter
[2], w0
);
834 /* Special-case LFE */
835 if(chans
[c
].channel
== LFE
)
837 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
838 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
839 if(Device
->Dry
.Buffer
== Device
->RealOut
.Buffer
)
841 int idx
= GetChannelIdxByName(Device
->RealOut
, chans
[c
].channel
);
842 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
847 ComputePanningGains(Device
->Dry
,
848 coeffs
, DryGain
* downmix_gain
, voice
->Direct
.Params
[c
].Gains
.Target
852 for(i
= 0;i
< NumSends
;i
++)
854 const ALeffectslot
*Slot
= SendSlots
[i
];
856 for(c
= 0;c
< num_channels
;c
++)
859 if(chans
[c
].channel
== LFE
)
860 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
861 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
863 ComputePanningGainsBF(Slot
->ChanMap
,
864 Slot
->NumChannels
, coeffs
, WetGain
[i
] * downmix_gain
,
865 voice
->Send
[i
].Params
[c
].Gains
.Target
869 for(c
= 0;c
< num_channels
;c
++)
871 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
872 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
880 if(Device
->AvgSpeakerDist
> 0.0f
)
882 /* If the source distance is 0, set w0 to w1 to act as a pass-
883 * through. We still want to pass the signal through the
884 * filters so they keep an appropriate history, in case the
885 * source moves away from the listener.
887 w0
= SPEEDOFSOUNDMETRESPERSEC
/
888 (Device
->AvgSpeakerDist
* (ALfloat
)Device
->Frequency
);
890 for(i
= 0;i
< MAX_AMBI_ORDER
+1;i
++)
891 voice
->Direct
.ChannelsPerOrder
[i
] = Device
->Dry
.NumChannelsPerOrder
[i
];
892 voice
->Flags
|= VOICE_HAS_NFC
;
895 for(c
= 0;c
< num_channels
;c
++)
897 ALfloat coeffs
[MAX_AMBI_COEFFS
];
899 if((voice
->Flags
&VOICE_HAS_NFC
))
901 NfcFilterAdjust1(&voice
->Direct
.Params
[c
].NFCtrlFilter
[0], w0
);
902 NfcFilterAdjust2(&voice
->Direct
.Params
[c
].NFCtrlFilter
[1], w0
);
903 NfcFilterAdjust3(&voice
->Direct
.Params
[c
].NFCtrlFilter
[2], w0
);
906 /* Special-case LFE */
907 if(chans
[c
].channel
== LFE
)
909 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
910 voice
->Direct
.Params
[c
].Gains
.Target
[j
] = 0.0f
;
911 if(Device
->Dry
.Buffer
== Device
->RealOut
.Buffer
)
913 int idx
= GetChannelIdxByName(Device
->RealOut
, chans
[c
].channel
);
914 if(idx
!= -1) voice
->Direct
.Params
[c
].Gains
.Target
[idx
] = DryGain
;
917 for(i
= 0;i
< NumSends
;i
++)
919 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
920 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
925 if(Device
->Render_Mode
== StereoPair
)
926 CalcAnglePairwiseCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
928 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
, coeffs
);
929 ComputePanningGains(Device
->Dry
,
930 coeffs
, DryGain
, voice
->Direct
.Params
[c
].Gains
.Target
933 for(i
= 0;i
< NumSends
;i
++)
935 const ALeffectslot
*Slot
= SendSlots
[i
];
937 ComputePanningGainsBF(Slot
->ChanMap
, Slot
->NumChannels
,
938 coeffs
, WetGain
[i
], voice
->Send
[i
].Params
[c
].Gains
.Target
941 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
942 voice
->Send
[i
].Params
[c
].Gains
.Target
[j
] = 0.0f
;
949 ALfloat hfScale
= props
->Direct
.HFReference
/ Frequency
;
950 ALfloat lfScale
= props
->Direct
.LFReference
/ Frequency
;
951 ALfloat gainHF
= maxf(DryGainHF
, 0.001f
); /* Limit -60dB */
952 ALfloat gainLF
= maxf(DryGainLF
, 0.001f
);
954 voice
->Direct
.FilterType
= AF_None
;
955 if(gainHF
!= 1.0f
) voice
->Direct
.FilterType
|= AF_LowPass
;
956 if(gainLF
!= 1.0f
) voice
->Direct
.FilterType
|= AF_HighPass
;
957 ALfilterState_setParams(
958 &voice
->Direct
.Params
[0].LowPass
, ALfilterType_HighShelf
,
959 gainHF
, hfScale
, calc_rcpQ_from_slope(gainHF
, 1.0f
)
961 ALfilterState_setParams(
962 &voice
->Direct
.Params
[0].HighPass
, ALfilterType_LowShelf
,
963 gainLF
, lfScale
, calc_rcpQ_from_slope(gainLF
, 1.0f
)
965 for(c
= 1;c
< num_channels
;c
++)
967 ALfilterState_copyParams(&voice
->Direct
.Params
[c
].LowPass
,
968 &voice
->Direct
.Params
[0].LowPass
);
969 ALfilterState_copyParams(&voice
->Direct
.Params
[c
].HighPass
,
970 &voice
->Direct
.Params
[0].HighPass
);
973 for(i
= 0;i
< NumSends
;i
++)
975 ALfloat hfScale
= props
->Send
[i
].HFReference
/ Frequency
;
976 ALfloat lfScale
= props
->Send
[i
].LFReference
/ Frequency
;
977 ALfloat gainHF
= maxf(WetGainHF
[i
], 0.001f
);
978 ALfloat gainLF
= maxf(WetGainLF
[i
], 0.001f
);
980 voice
->Send
[i
].FilterType
= AF_None
;
981 if(gainHF
!= 1.0f
) voice
->Send
[i
].FilterType
|= AF_LowPass
;
982 if(gainLF
!= 1.0f
) voice
->Send
[i
].FilterType
|= AF_HighPass
;
983 ALfilterState_setParams(
984 &voice
->Send
[i
].Params
[0].LowPass
, ALfilterType_HighShelf
,
985 gainHF
, hfScale
, calc_rcpQ_from_slope(gainHF
, 1.0f
)
987 ALfilterState_setParams(
988 &voice
->Send
[i
].Params
[0].HighPass
, ALfilterType_LowShelf
,
989 gainLF
, lfScale
, calc_rcpQ_from_slope(gainLF
, 1.0f
)
991 for(c
= 1;c
< num_channels
;c
++)
993 ALfilterState_copyParams(&voice
->Send
[i
].Params
[c
].LowPass
,
994 &voice
->Send
[i
].Params
[0].LowPass
);
995 ALfilterState_copyParams(&voice
->Send
[i
].Params
[c
].HighPass
,
996 &voice
->Send
[i
].Params
[0].HighPass
);
1001 static void CalcNonAttnSourceParams(ALvoice
*voice
, const struct ALvoiceProps
*props
, const ALbuffer
*ALBuffer
, const ALCcontext
*ALContext
)
1003 static const ALfloat dir
[3] = { 0.0f
, 0.0f
, -1.0f
};
1004 const ALCdevice
*Device
= ALContext
->Device
;
1005 const ALlistener
*Listener
= ALContext
->Listener
;
1006 ALfloat DryGain
, DryGainHF
, DryGainLF
;
1007 ALfloat WetGain
[MAX_SENDS
];
1008 ALfloat WetGainHF
[MAX_SENDS
];
1009 ALfloat WetGainLF
[MAX_SENDS
];
1010 ALeffectslot
*SendSlots
[MAX_SENDS
];
1014 voice
->Direct
.Buffer
= Device
->Dry
.Buffer
;
1015 voice
->Direct
.Channels
= Device
->Dry
.NumChannels
;
1016 for(i
= 0;i
< Device
->NumAuxSends
;i
++)
1018 SendSlots
[i
] = props
->Send
[i
].Slot
;
1019 if(!SendSlots
[i
] && i
== 0)
1020 SendSlots
[i
] = ALContext
->DefaultSlot
;
1021 if(!SendSlots
[i
] || SendSlots
[i
]->Params
.EffectType
== AL_EFFECT_NULL
)
1023 SendSlots
[i
] = NULL
;
1024 voice
->Send
[i
].Buffer
= NULL
;
1025 voice
->Send
[i
].Channels
= 0;
1029 voice
->Send
[i
].Buffer
= SendSlots
[i
]->WetBuffer
;
1030 voice
->Send
[i
].Channels
= SendSlots
[i
]->NumChannels
;
1034 /* Calculate the stepping value */
1035 Pitch
= (ALfloat
)ALBuffer
->Frequency
/(ALfloat
)Device
->Frequency
* props
->Pitch
;
1036 if(Pitch
> (ALfloat
)MAX_PITCH
)
1037 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1039 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1040 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
);
1041 voice
->Resampler
= SelectResampler(props
->Resampler
);
1043 /* Calculate gains */
1044 DryGain
= clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
);
1045 DryGain
*= props
->Direct
.Gain
* Listener
->Params
.Gain
;
1046 DryGain
= minf(DryGain
, GAIN_MIX_MAX
);
1047 DryGainHF
= props
->Direct
.GainHF
;
1048 DryGainLF
= props
->Direct
.GainLF
;
1049 for(i
= 0;i
< Device
->NumAuxSends
;i
++)
1051 WetGain
[i
] = clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
);
1052 WetGain
[i
] *= props
->Send
[i
].Gain
* Listener
->Params
.Gain
;
1053 WetGain
[i
] = minf(WetGain
[i
], GAIN_MIX_MAX
);
1054 WetGainHF
[i
] = props
->Send
[i
].GainHF
;
1055 WetGainLF
[i
] = props
->Send
[i
].GainLF
;
1058 CalcPanningAndFilters(voice
, 0.0f
, dir
, 0.0f
, DryGain
, DryGainHF
, DryGainLF
, WetGain
,
1059 WetGainLF
, WetGainHF
, SendSlots
, ALBuffer
, props
, Listener
, Device
);
1062 static void CalcAttnSourceParams(ALvoice
*voice
, const struct ALvoiceProps
*props
, const ALbuffer
*ALBuffer
, const ALCcontext
*ALContext
)
1064 const ALCdevice
*Device
= ALContext
->Device
;
1065 const ALlistener
*Listener
= ALContext
->Listener
;
1066 const ALsizei NumSends
= Device
->NumAuxSends
;
1067 aluVector Position
, Velocity
, Direction
, SourceToListener
;
1068 ALfloat Distance
, ClampedDist
, DopplerFactor
;
1069 ALeffectslot
*SendSlots
[MAX_SENDS
];
1070 ALfloat RoomRolloff
[MAX_SENDS
];
1071 ALfloat DecayDistance
[MAX_SENDS
];
1072 ALfloat DecayHFDistance
[MAX_SENDS
];
1073 ALfloat DryGain
, DryGainHF
, DryGainLF
;
1074 ALfloat WetGain
[MAX_SENDS
];
1075 ALfloat WetGainHF
[MAX_SENDS
];
1076 ALfloat WetGainLF
[MAX_SENDS
];
1083 /* Set mixing buffers and get send parameters. */
1084 voice
->Direct
.Buffer
= Device
->Dry
.Buffer
;
1085 voice
->Direct
.Channels
= Device
->Dry
.NumChannels
;
1086 for(i
= 0;i
< NumSends
;i
++)
1088 SendSlots
[i
] = props
->Send
[i
].Slot
;
1089 if(!SendSlots
[i
] && i
== 0)
1090 SendSlots
[i
] = ALContext
->DefaultSlot
;
1091 if(!SendSlots
[i
] || SendSlots
[i
]->Params
.EffectType
== AL_EFFECT_NULL
)
1093 SendSlots
[i
] = NULL
;
1094 RoomRolloff
[i
] = 0.0f
;
1095 DecayDistance
[i
] = 0.0f
;
1096 DecayHFDistance
[i
] = 0.0f
;
1098 else if(SendSlots
[i
]->Params
.AuxSendAuto
)
1100 RoomRolloff
[i
] = SendSlots
[i
]->Params
.RoomRolloff
+ props
->RoomRolloffFactor
;
1101 DecayDistance
[i
] = SendSlots
[i
]->Params
.DecayTime
* SPEEDOFSOUNDMETRESPERSEC
;
1102 DecayHFDistance
[i
] = DecayDistance
[i
] * SendSlots
[i
]->Params
.DecayHFRatio
;
1103 if(SendSlots
[i
]->Params
.DecayHFLimit
)
1105 ALfloat airAbsorption
= SendSlots
[i
]->Params
.AirAbsorptionGainHF
;
1106 if(airAbsorption
< 1.0f
)
1108 ALfloat limitRatio
= log10f(REVERB_DECAY_GAIN
) / log10f(airAbsorption
);
1109 DecayHFDistance
[i
] = minf(limitRatio
, DecayHFDistance
[i
]);
1115 /* If the slot's auxiliary send auto is off, the data sent to the
1116 * effect slot is the same as the dry path, sans filter effects */
1117 RoomRolloff
[i
] = props
->RolloffFactor
;
1118 DecayDistance
[i
] = 0.0f
;
1119 DecayHFDistance
[i
] = 0.0f
;
1124 voice
->Send
[i
].Buffer
= NULL
;
1125 voice
->Send
[i
].Channels
= 0;
1129 voice
->Send
[i
].Buffer
= SendSlots
[i
]->WetBuffer
;
1130 voice
->Send
[i
].Channels
= SendSlots
[i
]->NumChannels
;
1134 /* Transform source to listener space (convert to head relative) */
1135 aluVectorSet(&Position
, props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
);
1136 aluVectorSet(&Direction
, props
->Direction
[0], props
->Direction
[1], props
->Direction
[2], 0.0f
);
1137 aluVectorSet(&Velocity
, props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
);
1138 if(props
->HeadRelative
== AL_FALSE
)
1140 const aluMatrixf
*Matrix
= &Listener
->Params
.Matrix
;
1141 /* Transform source vectors */
1142 Position
= aluMatrixfVector(Matrix
, &Position
);
1143 Velocity
= aluMatrixfVector(Matrix
, &Velocity
);
1144 Direction
= aluMatrixfVector(Matrix
, &Direction
);
1148 const aluVector
*lvelocity
= &Listener
->Params
.Velocity
;
1149 /* Offset the source velocity to be relative of the listener velocity */
1150 Velocity
.v
[0] += lvelocity
->v
[0];
1151 Velocity
.v
[1] += lvelocity
->v
[1];
1152 Velocity
.v
[2] += lvelocity
->v
[2];
1155 directional
= aluNormalize(Direction
.v
) > FLT_EPSILON
;
1156 SourceToListener
.v
[0] = -Position
.v
[0];
1157 SourceToListener
.v
[1] = -Position
.v
[1];
1158 SourceToListener
.v
[2] = -Position
.v
[2];
1159 SourceToListener
.v
[3] = 0.0f
;
1160 Distance
= aluNormalize(SourceToListener
.v
);
1162 /* Initial source gain */
1163 DryGain
= props
->Gain
;
1166 for(i
= 0;i
< NumSends
;i
++)
1168 WetGain
[i
] = props
->Gain
;
1169 WetGainHF
[i
] = 1.0f
;
1170 WetGainLF
[i
] = 1.0f
;
1173 /* Calculate distance attenuation */
1174 ClampedDist
= Distance
;
1176 switch(Listener
->Params
.SourceDistanceModel
?
1177 props
->DistanceModel
: Listener
->Params
.DistanceModel
)
1179 case InverseDistanceClamped
:
1180 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1181 if(props
->MaxDistance
< props
->RefDistance
)
1184 case InverseDistance
:
1185 if(!(props
->RefDistance
> 0.0f
))
1186 ClampedDist
= props
->RefDistance
;
1189 ALfloat dist
= lerp(props
->RefDistance
, ClampedDist
, props
->RolloffFactor
);
1190 if(dist
> 0.0f
) DryGain
*= props
->RefDistance
/ dist
;
1191 for(i
= 0;i
< NumSends
;i
++)
1193 dist
= lerp(props
->RefDistance
, ClampedDist
, RoomRolloff
[i
]);
1194 if(dist
> 0.0f
) WetGain
[i
] *= props
->RefDistance
/ dist
;
1199 case LinearDistanceClamped
:
1200 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1201 if(props
->MaxDistance
< props
->RefDistance
)
1204 case LinearDistance
:
1205 if(!(props
->MaxDistance
!= props
->RefDistance
))
1206 ClampedDist
= props
->RefDistance
;
1209 ALfloat attn
= props
->RolloffFactor
* (ClampedDist
-props
->RefDistance
) /
1210 (props
->MaxDistance
-props
->RefDistance
);
1211 DryGain
*= maxf(1.0f
- attn
, 0.0f
);
1212 for(i
= 0;i
< NumSends
;i
++)
1214 attn
= RoomRolloff
[i
] * (ClampedDist
-props
->RefDistance
) /
1215 (props
->MaxDistance
-props
->RefDistance
);
1216 WetGain
[i
] *= maxf(1.0f
- attn
, 0.0f
);
1221 case ExponentDistanceClamped
:
1222 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1223 if(props
->MaxDistance
< props
->RefDistance
)
1226 case ExponentDistance
:
1227 if(!(ClampedDist
> 0.0f
&& props
->RefDistance
> 0.0f
))
1228 ClampedDist
= props
->RefDistance
;
1231 DryGain
*= powf(ClampedDist
/props
->RefDistance
, -props
->RolloffFactor
);
1232 for(i
= 0;i
< NumSends
;i
++)
1233 WetGain
[i
] *= powf(ClampedDist
/props
->RefDistance
, -RoomRolloff
[i
]);
1237 case DisableDistance
:
1238 ClampedDist
= props
->RefDistance
;
1242 /* Distance-based air absorption */
1243 if(ClampedDist
> props
->RefDistance
&& props
->RolloffFactor
> 0.0f
)
1245 ALfloat meters_base
= (ClampedDist
-props
->RefDistance
) * props
->RolloffFactor
*
1246 Listener
->Params
.MetersPerUnit
;
1247 if(props
->AirAbsorptionFactor
> 0.0f
)
1249 ALfloat hfattn
= powf(AIRABSORBGAINHF
, meters_base
* props
->AirAbsorptionFactor
);
1250 DryGainHF
*= hfattn
;
1251 for(i
= 0;i
< NumSends
;i
++)
1252 WetGainHF
[i
] *= hfattn
;
1255 if(props
->WetGainAuto
)
1257 /* Apply a decay-time transformation to the wet path, based on the
1258 * source distance in meters. The initial decay of the reverb
1259 * effect is calculated and applied to the wet path.
1261 for(i
= 0;i
< NumSends
;i
++)
1265 if(!(DecayDistance
[i
] > 0.0f
))
1268 gain
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayDistance
[i
]);
1270 /* Yes, the wet path's air absorption is applied with
1271 * WetGainAuto on, rather than WetGainHFAuto.
1275 ALfloat gainhf
= powf(REVERB_DECAY_GAIN
, meters_base
/DecayHFDistance
[i
]);
1276 WetGainHF
[i
] *= minf(gainhf
/ gain
, 1.0f
);
1282 /* Calculate directional soundcones */
1283 if(directional
&& props
->InnerAngle
< 360.0f
)
1289 Angle
= acosf(aluDotproduct(&Direction
, &SourceToListener
));
1290 Angle
= RAD2DEG(Angle
* ConeScale
* 2.0f
);
1291 if(!(Angle
> props
->InnerAngle
))
1296 else if(Angle
< props
->OuterAngle
)
1298 ALfloat scale
= ( Angle
-props
->InnerAngle
) /
1299 (props
->OuterAngle
-props
->InnerAngle
);
1300 ConeVolume
= lerp(1.0f
, props
->OuterGain
, scale
);
1301 ConeHF
= lerp(1.0f
, props
->OuterGainHF
, scale
);
1305 ConeVolume
= props
->OuterGain
;
1306 ConeHF
= props
->OuterGainHF
;
1309 DryGain
*= ConeVolume
;
1310 if(props
->DryGainHFAuto
)
1311 DryGainHF
*= ConeHF
;
1312 if(props
->WetGainAuto
)
1314 for(i
= 0;i
< NumSends
;i
++)
1315 WetGain
[i
] *= ConeVolume
;
1317 if(props
->WetGainHFAuto
)
1319 for(i
= 0;i
< NumSends
;i
++)
1320 WetGainHF
[i
] *= ConeHF
;
1324 /* Apply gain and frequency filters */
1325 DryGain
= clampf(DryGain
, props
->MinGain
, props
->MaxGain
);
1326 DryGain
= minf(DryGain
*props
->Direct
.Gain
*Listener
->Params
.Gain
, GAIN_MIX_MAX
);
1327 DryGainHF
*= props
->Direct
.GainHF
;
1328 DryGainLF
*= props
->Direct
.GainLF
;
1329 for(i
= 0;i
< NumSends
;i
++)
1331 WetGain
[i
] = clampf(WetGain
[i
], props
->MinGain
, props
->MaxGain
);
1332 WetGain
[i
] = minf(WetGain
[i
]*props
->Send
[i
].Gain
*Listener
->Params
.Gain
, GAIN_MIX_MAX
);
1333 WetGainHF
[i
] *= props
->Send
[i
].GainHF
;
1334 WetGainLF
[i
] *= props
->Send
[i
].GainLF
;
1338 /* Initial source pitch */
1339 Pitch
= props
->Pitch
;
1341 /* Calculate velocity-based doppler effect */
1342 DopplerFactor
= props
->DopplerFactor
* Listener
->Params
.DopplerFactor
;
1343 if(DopplerFactor
> 0.0f
)
1345 const aluVector
*lvelocity
= &Listener
->Params
.Velocity
;
1346 const ALfloat SpeedOfSound
= Listener
->Params
.SpeedOfSound
;
1349 vss
= aluDotproduct(&Velocity
, &SourceToListener
) * DopplerFactor
;
1350 vls
= aluDotproduct(lvelocity
, &SourceToListener
) * DopplerFactor
;
1352 if(!(vls
< SpeedOfSound
))
1354 /* Listener moving away from the source at the speed of sound.
1355 * Sound waves can't catch it.
1359 else if(!(vss
< SpeedOfSound
))
1361 /* Source moving toward the listener at the speed of sound. Sound
1362 * waves bunch up to extreme frequencies.
1368 /* Source and listener movement is nominal. Calculate the proper
1371 Pitch
*= (SpeedOfSound
-vls
) / (SpeedOfSound
-vss
);
1375 /* Adjust pitch based on the buffer and output frequencies, and calculate
1376 * fixed-point stepping value.
1378 Pitch
*= (ALfloat
)ALBuffer
->Frequency
/(ALfloat
)Device
->Frequency
;
1379 if(Pitch
> (ALfloat
)MAX_PITCH
)
1380 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1382 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1383 BsincPrepare(voice
->Step
, &voice
->ResampleState
.bsinc
);
1384 voice
->Resampler
= SelectResampler(props
->Resampler
);
1386 if(Distance
> FLT_EPSILON
)
1388 dir
[0] = -SourceToListener
.v
[0];
1389 /* Clamp Y, in case rounding errors caused it to end up outside of
1392 dir
[1] = clampf(-SourceToListener
.v
[1], -1.0f
, 1.0f
);
1393 dir
[2] = -SourceToListener
.v
[2] * ZScale
;
1401 if(props
->Radius
> Distance
)
1402 spread
= F_TAU
- Distance
/props
->Radius
*F_PI
;
1403 else if(Distance
> FLT_EPSILON
)
1404 spread
= asinf(props
->Radius
/ Distance
) * 2.0f
;
1408 CalcPanningAndFilters(voice
, Distance
, dir
, spread
, DryGain
, DryGainHF
, DryGainLF
, WetGain
,
1409 WetGainLF
, WetGainHF
, SendSlots
, ALBuffer
, props
, Listener
, Device
);
1412 static void CalcSourceParams(ALvoice
*voice
, ALCcontext
*context
, ALboolean force
)
1414 ALbufferlistitem
*BufferListItem
;
1415 struct ALvoiceProps
*props
;
1417 props
= ATOMIC_EXCHANGE_PTR(&voice
->Update
, NULL
, almemory_order_acq_rel
);
1418 if(!props
&& !force
) return;
1422 memcpy(voice
->Props
, props
,
1423 FAM_SIZE(struct ALvoiceProps
, Send
, context
->Device
->NumAuxSends
)
1426 ATOMIC_REPLACE_HEAD(struct ALvoiceProps
*, &voice
->FreeList
, props
);
1428 props
= voice
->Props
;
1430 BufferListItem
= ATOMIC_LOAD(&voice
->current_buffer
, almemory_order_relaxed
);
1431 while(BufferListItem
!= NULL
)
1433 const ALbuffer
*buffer
;
1434 if((buffer
=BufferListItem
->buffer
) != NULL
)
1436 if(props
->SpatializeMode
== SpatializeOn
||
1437 (props
->SpatializeMode
== SpatializeAuto
&& buffer
->FmtChannels
== FmtMono
))
1438 CalcAttnSourceParams(voice
, props
, buffer
, context
);
1440 CalcNonAttnSourceParams(voice
, props
, buffer
, context
);
1443 BufferListItem
= ATOMIC_LOAD(&BufferListItem
->next
, almemory_order_acquire
);
1448 static void UpdateContextSources(ALCcontext
*ctx
, const struct ALeffectslotArray
*slots
)
1450 ALvoice
**voice
, **voice_end
;
1454 IncrementRef(&ctx
->UpdateCount
);
1455 if(!ATOMIC_LOAD(&ctx
->HoldUpdates
, almemory_order_acquire
))
1457 ALboolean force
= CalcListenerParams(ctx
);
1458 for(i
= 0;i
< slots
->count
;i
++)
1459 force
|= CalcEffectSlotParams(slots
->slot
[i
], ctx
->Device
);
1461 voice
= ctx
->Voices
;
1462 voice_end
= voice
+ ctx
->VoiceCount
;
1463 for(;voice
!= voice_end
;++voice
)
1465 source
= ATOMIC_LOAD(&(*voice
)->Source
, almemory_order_acquire
);
1466 if(source
) CalcSourceParams(*voice
, ctx
, force
);
1469 IncrementRef(&ctx
->UpdateCount
);
1473 static void ApplyStablizer(FrontStablizer
*Stablizer
, ALfloat (*restrict Buffer
)[BUFFERSIZE
],
1474 int lidx
, int ridx
, int cidx
, ALsizei SamplesToDo
,
1475 ALsizei NumChannels
)
1477 ALfloat (*restrict lsplit
)[BUFFERSIZE
] = ASSUME_ALIGNED(Stablizer
->LSplit
, 16);
1478 ALfloat (*restrict rsplit
)[BUFFERSIZE
] = ASSUME_ALIGNED(Stablizer
->RSplit
, 16);
1481 /* Apply an all-pass to all channels, except the front-left and front-
1482 * right, so they maintain the same relative phase.
1484 for(i
= 0;i
< NumChannels
;i
++)
1486 if(i
== lidx
|| i
== ridx
)
1488 splitterap_process(&Stablizer
->APFilter
[i
], Buffer
[i
], SamplesToDo
);
1491 bandsplit_process(&Stablizer
->LFilter
, lsplit
[1], lsplit
[0], Buffer
[lidx
], SamplesToDo
);
1492 bandsplit_process(&Stablizer
->RFilter
, rsplit
[1], rsplit
[0], Buffer
[ridx
], SamplesToDo
);
1494 for(i
= 0;i
< SamplesToDo
;i
++)
1496 ALfloat lfsum
, hfsum
;
1499 lfsum
= lsplit
[0][i
] + rsplit
[0][i
];
1500 hfsum
= lsplit
[1][i
] + rsplit
[1][i
];
1501 s
= lsplit
[0][i
] + lsplit
[1][i
] - rsplit
[0][i
] - rsplit
[1][i
];
1503 /* This pans the separate low- and high-frequency sums between being on
1504 * the center channel and the left/right channels. The low-frequency
1505 * sum is 1/3rd toward center (2/3rds on left/right) and the high-
1506 * frequency sum is 1/4th toward center (3/4ths on left/right). These
1507 * values can be tweaked.
1509 m
= lfsum
*cosf(1.0f
/3.0f
* F_PI_2
) + hfsum
*cosf(1.0f
/4.0f
* F_PI_2
);
1510 c
= lfsum
*sinf(1.0f
/3.0f
* F_PI_2
) + hfsum
*sinf(1.0f
/4.0f
* F_PI_2
);
1512 /* The generated center channel signal adds to the existing signal,
1513 * while the modified left and right channels replace.
1515 Buffer
[lidx
][i
] = (m
+ s
) * 0.5f
;
1516 Buffer
[ridx
][i
] = (m
- s
) * 0.5f
;
1517 Buffer
[cidx
][i
] += c
* 0.5f
;
1521 static void ApplyDistanceComp(ALfloatBUFFERSIZE
*restrict Samples
, DistanceComp
*distcomp
,
1522 ALfloat
*restrict Values
, ALsizei SamplesToDo
, ALsizei numchans
)
1526 Values
= ASSUME_ALIGNED(Values
, 16);
1527 for(c
= 0;c
< numchans
;c
++)
1529 ALfloat
*restrict inout
= ASSUME_ALIGNED(Samples
[c
], 16);
1530 const ALfloat gain
= distcomp
[c
].Gain
;
1531 const ALsizei base
= distcomp
[c
].Length
;
1532 ALfloat
*restrict distbuf
= ASSUME_ALIGNED(distcomp
[c
].Buffer
, 16);
1538 for(i
= 0;i
< SamplesToDo
;i
++)
1544 if(SamplesToDo
>= base
)
1546 for(i
= 0;i
< base
;i
++)
1547 Values
[i
] = distbuf
[i
];
1548 for(;i
< SamplesToDo
;i
++)
1549 Values
[i
] = inout
[i
-base
];
1550 memcpy(distbuf
, &inout
[SamplesToDo
-base
], base
*sizeof(ALfloat
));
1554 for(i
= 0;i
< SamplesToDo
;i
++)
1555 Values
[i
] = distbuf
[i
];
1556 memmove(distbuf
, distbuf
+SamplesToDo
, (base
-SamplesToDo
)*sizeof(ALfloat
));
1557 memcpy(distbuf
+base
-SamplesToDo
, inout
, SamplesToDo
*sizeof(ALfloat
));
1559 for(i
= 0;i
< SamplesToDo
;i
++)
1560 inout
[i
] = Values
[i
]*gain
;
1564 static void ApplyDither(ALfloatBUFFERSIZE
*restrict Samples
, ALuint
*dither_seed
,
1565 const ALfloat quant_scale
, const ALsizei SamplesToDo
,
1566 const ALsizei numchans
)
1568 const ALfloat invscale
= 1.0f
/ quant_scale
;
1569 ALuint seed
= *dither_seed
;
1572 /* Dithering. Step 1, generate whitenoise (uniform distribution of random
1573 * values between -1 and +1). Step 2 is to add the noise to the samples,
1574 * before rounding and after scaling up to the desired quantization depth.
1576 for(c
= 0;c
< numchans
;c
++)
1578 ALfloat
*restrict samples
= Samples
[c
];
1579 for(i
= 0;i
< SamplesToDo
;i
++)
1581 ALfloat val
= samples
[i
] * quant_scale
;
1582 ALuint rng0
= dither_rng(&seed
);
1583 ALuint rng1
= dither_rng(&seed
);
1584 val
+= (ALfloat
)(rng0
*(1.0/UINT_MAX
) - rng1
*(1.0/UINT_MAX
));
1585 samples
[i
] = roundf(val
) * invscale
;
1588 *dither_seed
= seed
;
1592 static inline ALfloat
Conv_ALfloat(ALfloat val
)
1594 static inline ALint
Conv_ALint(ALfloat val
)
1596 /* Floats only have a 24-bit mantissa, so [-16777216, +16777216] is the max
1597 * integer range normalized floats can be safely converted to (a bit of the
1598 * exponent helps out, effectively giving 25 bits).
1600 return fastf2i(clampf(val
*16777216.0f
, -16777216.0f
, 16777215.0f
))<<7;
1602 static inline ALshort
Conv_ALshort(ALfloat val
)
1603 { return fastf2i(clampf(val
*32768.0f
, -32768.0f
, 32767.0f
)); }
1604 static inline ALbyte
Conv_ALbyte(ALfloat val
)
1605 { return fastf2i(clampf(val
*128.0f
, -128.0f
, 127.0f
)); }
1607 /* Define unsigned output variations. */
1608 #define DECL_TEMPLATE(T, func, O) \
1609 static inline T Conv_##T(ALfloat val) { return func(val)+O; }
1611 DECL_TEMPLATE(ALubyte
, Conv_ALbyte
, 128)
1612 DECL_TEMPLATE(ALushort
, Conv_ALshort
, 32768)
1613 DECL_TEMPLATE(ALuint
, Conv_ALint
, 2147483648u)
1615 #undef DECL_TEMPLATE
1617 #define DECL_TEMPLATE(T, A) \
1618 static void Write##A(const ALfloatBUFFERSIZE *InBuffer, ALvoid *OutBuffer, \
1619 ALsizei Offset, ALsizei SamplesToDo, ALsizei numchans) \
1622 for(j = 0;j < numchans;j++) \
1624 const ALfloat *restrict in = ASSUME_ALIGNED(InBuffer[j], 16); \
1625 T *restrict out = (T*)OutBuffer + Offset*numchans + j; \
1627 for(i = 0;i < SamplesToDo;i++) \
1628 out[i*numchans] = Conv_##T(in[i]); \
1632 DECL_TEMPLATE(ALfloat
, F32
)
1633 DECL_TEMPLATE(ALuint
, UI32
)
1634 DECL_TEMPLATE(ALint
, I32
)
1635 DECL_TEMPLATE(ALushort
, UI16
)
1636 DECL_TEMPLATE(ALshort
, I16
)
1637 DECL_TEMPLATE(ALubyte
, UI8
)
1638 DECL_TEMPLATE(ALbyte
, I8
)
1640 #undef DECL_TEMPLATE
1643 void aluMixData(ALCdevice
*device
, ALvoid
*OutBuffer
, ALsizei NumSamples
)
1645 ALsizei SamplesToDo
;
1646 ALsizei SamplesDone
;
1651 for(SamplesDone
= 0;SamplesDone
< NumSamples
;)
1653 SamplesToDo
= mini(NumSamples
-SamplesDone
, BUFFERSIZE
);
1654 for(c
= 0;c
< device
->Dry
.NumChannels
;c
++)
1655 memset(device
->Dry
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1656 if(device
->Dry
.Buffer
!= device
->FOAOut
.Buffer
)
1657 for(c
= 0;c
< device
->FOAOut
.NumChannels
;c
++)
1658 memset(device
->FOAOut
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1659 if(device
->Dry
.Buffer
!= device
->RealOut
.Buffer
)
1660 for(c
= 0;c
< device
->RealOut
.NumChannels
;c
++)
1661 memset(device
->RealOut
.Buffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1663 IncrementRef(&device
->MixCount
);
1665 ctx
= ATOMIC_LOAD(&device
->ContextList
, almemory_order_acquire
);
1668 const struct ALeffectslotArray
*auxslots
;
1670 auxslots
= ATOMIC_LOAD(&ctx
->ActiveAuxSlots
, almemory_order_acquire
);
1671 UpdateContextSources(ctx
, auxslots
);
1673 for(i
= 0;i
< auxslots
->count
;i
++)
1675 ALeffectslot
*slot
= auxslots
->slot
[i
];
1676 for(c
= 0;c
< slot
->NumChannels
;c
++)
1677 memset(slot
->WetBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1680 /* source processing */
1681 for(i
= 0;i
< ctx
->VoiceCount
;i
++)
1683 ALvoice
*voice
= ctx
->Voices
[i
];
1684 ALsource
*source
= ATOMIC_LOAD(&voice
->Source
, almemory_order_acquire
);
1685 if(source
&& ATOMIC_LOAD(&voice
->Playing
, almemory_order_relaxed
) &&
1688 if(!MixSource(voice
, source
, device
, SamplesToDo
))
1690 ATOMIC_STORE(&voice
->Source
, NULL
, almemory_order_relaxed
);
1691 ATOMIC_STORE(&voice
->Playing
, false, almemory_order_release
);
1696 /* effect slot processing */
1697 for(i
= 0;i
< auxslots
->count
;i
++)
1699 const ALeffectslot
*slot
= auxslots
->slot
[i
];
1700 ALeffectState
*state
= slot
->Params
.EffectState
;
1701 V(state
,process
)(SamplesToDo
, slot
->WetBuffer
, state
->OutBuffer
,
1702 state
->OutChannels
);
1708 /* Increment the clock time. Every second's worth of samples is
1709 * converted and added to clock base so that large sample counts don't
1710 * overflow during conversion. This also guarantees an exact, stable
1712 device
->SamplesDone
+= SamplesToDo
;
1713 device
->ClockBase
+= (device
->SamplesDone
/device
->Frequency
) * DEVICE_CLOCK_RES
;
1714 device
->SamplesDone
%= device
->Frequency
;
1715 IncrementRef(&device
->MixCount
);
1717 if(device
->HrtfHandle
)
1719 DirectHrtfState
*state
;
1723 ambiup_process(device
->AmbiUp
,
1724 device
->Dry
.Buffer
, device
->Dry
.NumChannels
,
1725 SAFE_CONST(ALfloatBUFFERSIZE
*,device
->FOAOut
.Buffer
), SamplesToDo
1728 lidx
= GetChannelIdxByName(device
->RealOut
, FrontLeft
);
1729 ridx
= GetChannelIdxByName(device
->RealOut
, FrontRight
);
1730 assert(lidx
!= -1 && ridx
!= -1);
1732 state
= device
->Hrtf
;
1733 for(c
= 0;c
< device
->Dry
.NumChannels
;c
++)
1735 MixDirectHrtf(device
->RealOut
.Buffer
[lidx
], device
->RealOut
.Buffer
[ridx
],
1736 device
->Dry
.Buffer
[c
], state
->Offset
, state
->IrSize
,
1737 SAFE_CONST(ALfloat2
*,state
->Chan
[c
].Coeffs
),
1738 state
->Chan
[c
].Values
, SamplesToDo
1741 state
->Offset
+= SamplesToDo
;
1743 else if(device
->AmbiDecoder
)
1745 if(device
->Dry
.Buffer
!= device
->FOAOut
.Buffer
)
1746 bformatdec_upSample(device
->AmbiDecoder
,
1747 device
->Dry
.Buffer
, SAFE_CONST(ALfloatBUFFERSIZE
*,device
->FOAOut
.Buffer
),
1748 device
->FOAOut
.NumChannels
, SamplesToDo
1750 bformatdec_process(device
->AmbiDecoder
,
1751 device
->RealOut
.Buffer
, device
->RealOut
.NumChannels
,
1752 SAFE_CONST(ALfloatBUFFERSIZE
*,device
->Dry
.Buffer
), SamplesToDo
1755 else if(device
->AmbiUp
)
1757 ambiup_process(device
->AmbiUp
,
1758 device
->RealOut
.Buffer
, device
->RealOut
.NumChannels
,
1759 SAFE_CONST(ALfloatBUFFERSIZE
*,device
->FOAOut
.Buffer
), SamplesToDo
1762 else if(device
->Uhj_Encoder
)
1764 int lidx
= GetChannelIdxByName(device
->RealOut
, FrontLeft
);
1765 int ridx
= GetChannelIdxByName(device
->RealOut
, FrontRight
);
1766 if(lidx
!= -1 && ridx
!= -1)
1768 /* Encode to stereo-compatible 2-channel UHJ output. */
1769 EncodeUhj2(device
->Uhj_Encoder
,
1770 device
->RealOut
.Buffer
[lidx
], device
->RealOut
.Buffer
[ridx
],
1771 device
->Dry
.Buffer
, SamplesToDo
1775 else if(device
->Bs2b
)
1777 int lidx
= GetChannelIdxByName(device
->RealOut
, FrontLeft
);
1778 int ridx
= GetChannelIdxByName(device
->RealOut
, FrontRight
);
1779 if(lidx
!= -1 && ridx
!= -1)
1781 /* Apply binaural/crossfeed filter */
1782 bs2b_cross_feed(device
->Bs2b
, device
->RealOut
.Buffer
[lidx
],
1783 device
->RealOut
.Buffer
[ridx
], SamplesToDo
);
1789 ALfloat (*Buffer
)[BUFFERSIZE
] = device
->RealOut
.Buffer
;
1790 ALsizei Channels
= device
->RealOut
.NumChannels
;
1792 if(device
->Stablizer
)
1794 int lidx
= GetChannelIdxByName(device
->RealOut
, FrontLeft
);
1795 int ridx
= GetChannelIdxByName(device
->RealOut
, FrontRight
);
1796 int cidx
= GetChannelIdxByName(device
->RealOut
, FrontCenter
);
1797 assert(lidx
>= 0 && ridx
>= 0 && cidx
>= 0);
1799 ApplyStablizer(device
->Stablizer
, Buffer
, lidx
, ridx
, cidx
,
1800 SamplesToDo
, Channels
);
1803 /* Use NFCtrlData for temp value storage. */
1804 ApplyDistanceComp(Buffer
, device
->ChannelDelay
, device
->NFCtrlData
,
1805 SamplesToDo
, Channels
);
1808 ApplyCompression(device
->Limiter
, Channels
, SamplesToDo
, Buffer
);
1810 if(device
->DitherDepth
> 0.0f
)
1811 ApplyDither(Buffer
, &device
->DitherSeed
, device
->DitherDepth
, SamplesToDo
,
1814 switch(device
->FmtType
)
1817 WriteI8(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1820 WriteUI8(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1823 WriteI16(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1826 WriteUI16(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1829 WriteI32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1832 WriteUI32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1835 WriteF32(Buffer
, OutBuffer
, SamplesDone
, SamplesToDo
, Channels
);
1840 SamplesDone
+= SamplesToDo
;
1846 void aluHandleDisconnect(ALCdevice
*device
)
1850 device
->Connected
= ALC_FALSE
;
1852 ctx
= ATOMIC_LOAD_SEQ(&device
->ContextList
);
1856 for(i
= 0;i
< ctx
->VoiceCount
;i
++)
1858 ALvoice
*voice
= ctx
->Voices
[i
];
1861 source
= ATOMIC_EXCHANGE_PTR(&voice
->Source
, NULL
, almemory_order_acq_rel
);
1862 ATOMIC_STORE(&voice
->Playing
, false, almemory_order_release
);
1866 ALenum playing
= AL_PLAYING
;
1867 (void)(ATOMIC_COMPARE_EXCHANGE_STRONG_SEQ(&source
->state
, &playing
, AL_STOPPED
));
1870 ctx
->VoiceCount
= 0;