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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
32 #include "alListener.h"
33 #include "alAuxEffectSlot.h"
38 #include "mixer_defs.h"
40 #include "midi/base.h"
49 ALfloat ConeScale
= 1.0f
;
51 /* Localized Z scalar for mono sources */
52 ALfloat ZScale
= 1.0f
;
54 extern inline ALfloat
minf(ALfloat a
, ALfloat b
);
55 extern inline ALfloat
maxf(ALfloat a
, ALfloat b
);
56 extern inline ALfloat
clampf(ALfloat val
, ALfloat min
, ALfloat max
);
58 extern inline ALdouble
mind(ALdouble a
, ALdouble b
);
59 extern inline ALdouble
maxd(ALdouble a
, ALdouble b
);
60 extern inline ALdouble
clampd(ALdouble val
, ALdouble min
, ALdouble max
);
62 extern inline ALuint
minu(ALuint a
, ALuint b
);
63 extern inline ALuint
maxu(ALuint a
, ALuint b
);
64 extern inline ALuint
clampu(ALuint val
, ALuint min
, ALuint max
);
66 extern inline ALint
mini(ALint a
, ALint b
);
67 extern inline ALint
maxi(ALint a
, ALint b
);
68 extern inline ALint
clampi(ALint val
, ALint min
, ALint max
);
70 extern inline ALint64
mini64(ALint64 a
, ALint64 b
);
71 extern inline ALint64
maxi64(ALint64 a
, ALint64 b
);
72 extern inline ALint64
clampi64(ALint64 val
, ALint64 min
, ALint64 max
);
74 extern inline ALuint64
minu64(ALuint64 a
, ALuint64 b
);
75 extern inline ALuint64
maxu64(ALuint64 a
, ALuint64 b
);
76 extern inline ALuint64
clampu64(ALuint64 val
, ALuint64 min
, ALuint64 max
);
78 extern inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALfloat mu
);
79 extern inline ALfloat
cubic(ALfloat val0
, ALfloat val1
, ALfloat val2
, ALfloat val3
, ALfloat mu
);
81 static ResamplerFunc
SelectResampler(enum Resampler Resampler
, ALuint increment
)
83 if(increment
== FRACTIONONE
)
84 return Resample_copy32_C
;
88 return Resample_point32_C
;
90 return Resample_lerp32_C
;
92 return Resample_cubic32_C
;
94 /* Shouldn't happen */
98 return Resample_point32_C
;
102 static DryMixerFunc
SelectHrtfMixer(void)
105 if((CPUCapFlags
&CPU_CAP_SSE
))
106 return MixDirect_Hrtf_SSE
;
109 if((CPUCapFlags
&CPU_CAP_NEON
))
110 return MixDirect_Hrtf_Neon
;
113 return MixDirect_Hrtf_C
;
116 static DryMixerFunc
SelectDirectMixer(void)
119 if((CPUCapFlags
&CPU_CAP_SSE
))
120 return MixDirect_SSE
;
123 if((CPUCapFlags
&CPU_CAP_NEON
))
124 return MixDirect_Neon
;
130 static WetMixerFunc
SelectSendMixer(void)
133 if((CPUCapFlags
&CPU_CAP_SSE
))
137 if((CPUCapFlags
&CPU_CAP_NEON
))
145 static inline void aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
147 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
148 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
149 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
152 static inline ALfloat
aluDotproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
)
154 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
155 inVector1
[2]*inVector2
[2];
158 static inline void aluNormalize(ALfloat
*inVector
)
160 ALfloat lengthsqr
= aluDotproduct(inVector
, inVector
);
163 ALfloat inv_length
= 1.0f
/sqrtf(lengthsqr
);
164 inVector
[0] *= inv_length
;
165 inVector
[1] *= inv_length
;
166 inVector
[2] *= inv_length
;
170 static inline ALvoid
aluMatrixVector(ALfloat
*vector
, ALfloat w
, ALfloat (*restrict matrix
)[4])
173 vector
[0], vector
[1], vector
[2], w
176 vector
[0] = temp
[0]*matrix
[0][0] + temp
[1]*matrix
[1][0] + temp
[2]*matrix
[2][0] + temp
[3]*matrix
[3][0];
177 vector
[1] = temp
[0]*matrix
[0][1] + temp
[1]*matrix
[1][1] + temp
[2]*matrix
[2][1] + temp
[3]*matrix
[3][1];
178 vector
[2] = temp
[0]*matrix
[0][2] + temp
[1]*matrix
[1][2] + temp
[2]*matrix
[2][2] + temp
[3]*matrix
[3][2];
182 static ALvoid
CalcListenerParams(ALlistener
*Listener
)
184 ALfloat N
[3], V
[3], U
[3], P
[3];
187 N
[0] = Listener
->Forward
[0];
188 N
[1] = Listener
->Forward
[1];
189 N
[2] = Listener
->Forward
[2];
191 V
[0] = Listener
->Up
[0];
192 V
[1] = Listener
->Up
[1];
193 V
[2] = Listener
->Up
[2];
195 /* Build and normalize right-vector */
196 aluCrossproduct(N
, V
, U
);
199 Listener
->Params
.Matrix
[0][0] = U
[0];
200 Listener
->Params
.Matrix
[0][1] = V
[0];
201 Listener
->Params
.Matrix
[0][2] = -N
[0];
202 Listener
->Params
.Matrix
[0][3] = 0.0f
;
203 Listener
->Params
.Matrix
[1][0] = U
[1];
204 Listener
->Params
.Matrix
[1][1] = V
[1];
205 Listener
->Params
.Matrix
[1][2] = -N
[1];
206 Listener
->Params
.Matrix
[1][3] = 0.0f
;
207 Listener
->Params
.Matrix
[2][0] = U
[2];
208 Listener
->Params
.Matrix
[2][1] = V
[2];
209 Listener
->Params
.Matrix
[2][2] = -N
[2];
210 Listener
->Params
.Matrix
[2][3] = 0.0f
;
211 Listener
->Params
.Matrix
[3][0] = 0.0f
;
212 Listener
->Params
.Matrix
[3][1] = 0.0f
;
213 Listener
->Params
.Matrix
[3][2] = 0.0f
;
214 Listener
->Params
.Matrix
[3][3] = 1.0f
;
216 P
[0] = Listener
->Position
[0];
217 P
[1] = Listener
->Position
[1];
218 P
[2] = Listener
->Position
[2];
219 aluMatrixVector(P
, 1.0f
, Listener
->Params
.Matrix
);
220 Listener
->Params
.Matrix
[3][0] = -P
[0];
221 Listener
->Params
.Matrix
[3][1] = -P
[1];
222 Listener
->Params
.Matrix
[3][2] = -P
[2];
224 Listener
->Params
.Velocity
[0] = Listener
->Velocity
[0];
225 Listener
->Params
.Velocity
[1] = Listener
->Velocity
[1];
226 Listener
->Params
.Velocity
[2] = Listener
->Velocity
[2];
227 aluMatrixVector(Listener
->Params
.Velocity
, 0.0f
, Listener
->Params
.Matrix
);
230 ALvoid
CalcNonAttnSourceParams(ALactivesource
*src
, const ALCcontext
*ALContext
)
232 static const struct ChanMap MonoMap
[1] = { { FrontCenter
, 0.0f
} };
233 static const struct ChanMap StereoMap
[2] = {
234 { FrontLeft
, DEG2RAD(-30.0f
) },
235 { FrontRight
, DEG2RAD( 30.0f
) }
237 static const struct ChanMap StereoWideMap
[2] = {
238 { FrontLeft
, DEG2RAD(-90.0f
) },
239 { FrontRight
, DEG2RAD( 90.0f
) }
241 static const struct ChanMap RearMap
[2] = {
242 { BackLeft
, DEG2RAD(-150.0f
) },
243 { BackRight
, DEG2RAD( 150.0f
) }
245 static const struct ChanMap QuadMap
[4] = {
246 { FrontLeft
, DEG2RAD( -45.0f
) },
247 { FrontRight
, DEG2RAD( 45.0f
) },
248 { BackLeft
, DEG2RAD(-135.0f
) },
249 { BackRight
, DEG2RAD( 135.0f
) }
251 static const struct ChanMap X51Map
[6] = {
252 { FrontLeft
, DEG2RAD( -30.0f
) },
253 { FrontRight
, DEG2RAD( 30.0f
) },
254 { FrontCenter
, DEG2RAD( 0.0f
) },
256 { BackLeft
, DEG2RAD(-110.0f
) },
257 { BackRight
, DEG2RAD( 110.0f
) }
259 static const struct ChanMap X61Map
[7] = {
260 { FrontLeft
, DEG2RAD(-30.0f
) },
261 { FrontRight
, DEG2RAD( 30.0f
) },
262 { FrontCenter
, DEG2RAD( 0.0f
) },
264 { BackCenter
, DEG2RAD(180.0f
) },
265 { SideLeft
, DEG2RAD(-90.0f
) },
266 { SideRight
, DEG2RAD( 90.0f
) }
268 static const struct ChanMap X71Map
[8] = {
269 { FrontLeft
, DEG2RAD( -30.0f
) },
270 { FrontRight
, DEG2RAD( 30.0f
) },
271 { FrontCenter
, DEG2RAD( 0.0f
) },
273 { BackLeft
, DEG2RAD(-150.0f
) },
274 { BackRight
, DEG2RAD( 150.0f
) },
275 { SideLeft
, DEG2RAD( -90.0f
) },
276 { SideRight
, DEG2RAD( 90.0f
) }
279 ALCdevice
*Device
= ALContext
->Device
;
280 ALsource
*ALSource
= src
->Source
;
281 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
282 ALbufferlistitem
*BufferListItem
;
283 enum FmtChannels Channels
;
284 ALfloat DryGain
, DryGainHF
;
285 ALfloat WetGain
[MAX_SENDS
];
286 ALfloat WetGainHF
[MAX_SENDS
];
287 ALint NumSends
, Frequency
;
288 const struct ChanMap
*chans
= NULL
;
289 enum Resampler Resampler
;
290 ALint num_channels
= 0;
291 ALboolean DirectChannels
;
292 ALfloat hwidth
= 0.0f
;
296 /* Get device properties */
297 NumSends
= Device
->NumAuxSends
;
298 Frequency
= Device
->Frequency
;
300 /* Get listener properties */
301 ListenerGain
= ALContext
->Listener
->Gain
;
303 /* Get source properties */
304 SourceVolume
= ALSource
->Gain
;
305 MinVolume
= ALSource
->MinGain
;
306 MaxVolume
= ALSource
->MaxGain
;
307 Pitch
= ALSource
->Pitch
;
308 Resampler
= ALSource
->Resampler
;
309 DirectChannels
= ALSource
->DirectChannels
;
311 src
->Direct
.OutBuffer
= Device
->DryBuffer
;
312 src
->Direct
.ClickRemoval
= Device
->ClickRemoval
;
313 src
->Direct
.PendingClicks
= Device
->PendingClicks
;
314 for(i
= 0;i
< NumSends
;i
++)
316 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
318 Slot
= Device
->DefaultSlot
;
319 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
321 src
->Send
[i
].OutBuffer
= NULL
;
322 src
->Send
[i
].ClickRemoval
= NULL
;
323 src
->Send
[i
].PendingClicks
= NULL
;
327 src
->Send
[i
].OutBuffer
= Slot
->WetBuffer
;
328 src
->Send
[i
].ClickRemoval
= Slot
->ClickRemoval
;
329 src
->Send
[i
].PendingClicks
= Slot
->PendingClicks
;
333 /* Calculate the stepping value */
335 BufferListItem
= ALSource
->queue
;
336 while(BufferListItem
!= NULL
)
339 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
341 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
343 src
->Step
= 10<<FRACTIONBITS
;
346 src
->Step
= fastf2i(Pitch
*FRACTIONONE
);
350 src
->Resample
= SelectResampler(Resampler
, src
->Step
);
352 Channels
= ALBuffer
->FmtChannels
;
355 BufferListItem
= BufferListItem
->next
;
358 /* Calculate gains */
359 DryGain
= clampf(SourceVolume
, MinVolume
, MaxVolume
);
360 DryGain
*= ALSource
->DirectGain
* ListenerGain
;
361 DryGainHF
= ALSource
->DirectGainHF
;
362 for(i
= 0;i
< NumSends
;i
++)
364 WetGain
[i
] = clampf(SourceVolume
, MinVolume
, MaxVolume
);
365 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
366 WetGainHF
[i
] = ALSource
->Send
[i
].GainHF
;
377 if(!(Device
->Flags
&DEVICE_WIDE_STEREO
))
379 /* HACK: Place the stereo channels at +/-90 degrees when using non-
380 * HRTF stereo output. This helps reduce the "monoization" caused
381 * by them panning towards the center. */
382 if(Device
->FmtChans
== DevFmtStereo
&& !Device
->Hrtf
)
383 chans
= StereoWideMap
;
389 chans
= StereoWideMap
;
390 hwidth
= DEG2RAD(60.0f
);
421 if(DirectChannels
!= AL_FALSE
)
423 ALfloat (*SrcMatrix
)[MaxChannels
] = src
->Direct
.Mix
.Gains
;
424 for(i
= 0;i
< MAX_INPUT_CHANNELS
;i
++)
426 for(c
= 0;c
< MaxChannels
;c
++)
427 SrcMatrix
[i
][c
] = 0.0f
;
429 for(c
= 0;c
< num_channels
;c
++)
431 for(i
= 0;i
< (ALint
)Device
->NumChan
;i
++)
433 enum Channel chan
= Device
->Speaker2Chan
[i
];
434 if(chan
== chans
[c
].channel
)
436 SrcMatrix
[c
][chan
] = DryGain
;
441 src
->DryMix
= SelectDirectMixer();
443 else if(Device
->Hrtf
)
445 for(c
= 0;c
< num_channels
;c
++)
447 if(chans
[c
].channel
== LFE
)
450 src
->Direct
.Mix
.Hrtf
.Params
.Delay
[c
][0] = 0;
451 src
->Direct
.Mix
.Hrtf
.Params
.Delay
[c
][1] = 0;
452 for(i
= 0;i
< HRIR_LENGTH
;i
++)
454 src
->Direct
.Mix
.Hrtf
.Params
.Coeffs
[c
][i
][0] = 0.0f
;
455 src
->Direct
.Mix
.Hrtf
.Params
.Coeffs
[c
][i
][1] = 0.0f
;
460 /* Get the static HRIR coefficients and delays for this
462 GetLerpedHrtfCoeffs(Device
->Hrtf
,
463 0.0f
, chans
[c
].angle
, DryGain
,
464 src
->Direct
.Mix
.Hrtf
.Params
.Coeffs
[c
],
465 src
->Direct
.Mix
.Hrtf
.Params
.Delay
[c
]);
468 ALSource
->Hrtf
.Counter
= 0;
469 src
->Direct
.Mix
.Hrtf
.Params
.IrSize
= GetHrtfIrSize(Device
->Hrtf
);
471 src
->Direct
.Mix
.Hrtf
.State
= &ALSource
->Hrtf
;
472 src
->DryMix
= SelectHrtfMixer();
476 ALfloat (*SrcMatrix
)[MaxChannels
] = src
->Direct
.Mix
.Gains
;
477 for(i
= 0;i
< MAX_INPUT_CHANNELS
;i
++)
479 for(c
= 0;c
< MaxChannels
;c
++)
480 SrcMatrix
[i
][c
] = 0.0f
;
483 DryGain
*= lerp(1.0f
, 1.0f
/sqrtf((float)Device
->NumChan
), hwidth
/F_PI
);
484 for(c
= 0;c
< num_channels
;c
++)
486 /* Special-case LFE */
487 if(chans
[c
].channel
== LFE
)
489 SrcMatrix
[c
][chans
[c
].channel
] = DryGain
;
492 ComputeAngleGains(Device
, chans
[c
].angle
, hwidth
, DryGain
,
495 src
->DryMix
= SelectDirectMixer();
497 for(i
= 0;i
< NumSends
;i
++)
498 src
->Send
[i
].Gain
= WetGain
[i
];
499 src
->WetMix
= SelectSendMixer();
502 ALfloat gain
= maxf(0.01f
, DryGainHF
);
503 for(c
= 0;c
< num_channels
;c
++)
504 ALfilterState_setParams(&src
->Direct
.LpFilter
[c
],
505 ALfilterType_HighShelf
, gain
,
506 (ALfloat
)LOWPASSFREQREF
/Frequency
, 0.0f
);
508 for(i
= 0;i
< NumSends
;i
++)
510 ALfloat gain
= maxf(0.01f
, WetGainHF
[i
]);
511 for(c
= 0;c
< num_channels
;c
++)
512 ALfilterState_setParams(&src
->Send
[i
].LpFilter
[c
],
513 ALfilterType_HighShelf
, gain
,
514 (ALfloat
)LOWPASSFREQREF
/Frequency
, 0.0f
);
518 ALvoid
CalcSourceParams(ALactivesource
*src
, const ALCcontext
*ALContext
)
520 ALCdevice
*Device
= ALContext
->Device
;
521 ALsource
*ALSource
= src
->Source
;
522 ALfloat Velocity
[3],Direction
[3],Position
[3],SourceToListener
[3];
523 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,ClampedDist
;
524 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
;
525 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
526 ALfloat DopplerFactor
, SpeedOfSound
;
527 ALfloat AirAbsorptionFactor
;
528 ALfloat RoomAirAbsorption
[MAX_SENDS
];
529 ALbufferlistitem
*BufferListItem
;
531 ALfloat RoomAttenuation
[MAX_SENDS
];
532 ALfloat MetersPerUnit
;
533 ALfloat RoomRolloffBase
;
534 ALfloat RoomRolloff
[MAX_SENDS
];
535 ALfloat DecayDistance
[MAX_SENDS
];
538 ALboolean DryGainHFAuto
;
539 ALfloat WetGain
[MAX_SENDS
];
540 ALfloat WetGainHF
[MAX_SENDS
];
541 ALboolean WetGainAuto
;
542 ALboolean WetGainHFAuto
;
543 enum Resampler Resampler
;
550 for(i
= 0;i
< MAX_SENDS
;i
++)
553 /* Get context/device properties */
554 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
555 SpeedOfSound
= ALContext
->SpeedOfSound
* ALContext
->DopplerVelocity
;
556 NumSends
= Device
->NumAuxSends
;
557 Frequency
= Device
->Frequency
;
559 /* Get listener properties */
560 ListenerGain
= ALContext
->Listener
->Gain
;
561 MetersPerUnit
= ALContext
->Listener
->MetersPerUnit
;
563 /* Get source properties */
564 SourceVolume
= ALSource
->Gain
;
565 MinVolume
= ALSource
->MinGain
;
566 MaxVolume
= ALSource
->MaxGain
;
567 Pitch
= ALSource
->Pitch
;
568 Resampler
= ALSource
->Resampler
;
569 Position
[0] = ALSource
->Position
[0];
570 Position
[1] = ALSource
->Position
[1];
571 Position
[2] = ALSource
->Position
[2];
572 Direction
[0] = ALSource
->Orientation
[0];
573 Direction
[1] = ALSource
->Orientation
[1];
574 Direction
[2] = ALSource
->Orientation
[2];
575 Velocity
[0] = ALSource
->Velocity
[0];
576 Velocity
[1] = ALSource
->Velocity
[1];
577 Velocity
[2] = ALSource
->Velocity
[2];
578 MinDist
= ALSource
->RefDistance
;
579 MaxDist
= ALSource
->MaxDistance
;
580 Rolloff
= ALSource
->RollOffFactor
;
581 InnerAngle
= ALSource
->InnerAngle
;
582 OuterAngle
= ALSource
->OuterAngle
;
583 AirAbsorptionFactor
= ALSource
->AirAbsorptionFactor
;
584 DryGainHFAuto
= ALSource
->DryGainHFAuto
;
585 WetGainAuto
= ALSource
->WetGainAuto
;
586 WetGainHFAuto
= ALSource
->WetGainHFAuto
;
587 RoomRolloffBase
= ALSource
->RoomRolloffFactor
;
589 src
->Direct
.OutBuffer
= Device
->DryBuffer
;
590 src
->Direct
.ClickRemoval
= Device
->ClickRemoval
;
591 src
->Direct
.PendingClicks
= Device
->PendingClicks
;
592 for(i
= 0;i
< NumSends
;i
++)
594 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
597 Slot
= Device
->DefaultSlot
;
598 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
601 RoomRolloff
[i
] = 0.0f
;
602 DecayDistance
[i
] = 0.0f
;
603 RoomAirAbsorption
[i
] = 1.0f
;
605 else if(Slot
->AuxSendAuto
)
607 RoomRolloff
[i
] = RoomRolloffBase
;
608 if(IsReverbEffect(Slot
->EffectType
))
610 RoomRolloff
[i
] += Slot
->EffectProps
.Reverb
.RoomRolloffFactor
;
611 DecayDistance
[i
] = Slot
->EffectProps
.Reverb
.DecayTime
*
612 SPEEDOFSOUNDMETRESPERSEC
;
613 RoomAirAbsorption
[i
] = Slot
->EffectProps
.Reverb
.AirAbsorptionGainHF
;
617 DecayDistance
[i
] = 0.0f
;
618 RoomAirAbsorption
[i
] = 1.0f
;
623 /* If the slot's auxiliary send auto is off, the data sent to the
624 * effect slot is the same as the dry path, sans filter effects */
625 RoomRolloff
[i
] = Rolloff
;
626 DecayDistance
[i
] = 0.0f
;
627 RoomAirAbsorption
[i
] = AIRABSORBGAINHF
;
630 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
632 src
->Send
[i
].OutBuffer
= NULL
;
633 src
->Send
[i
].ClickRemoval
= NULL
;
634 src
->Send
[i
].PendingClicks
= NULL
;
638 src
->Send
[i
].OutBuffer
= Slot
->WetBuffer
;
639 src
->Send
[i
].ClickRemoval
= Slot
->ClickRemoval
;
640 src
->Send
[i
].PendingClicks
= Slot
->PendingClicks
;
644 /* Transform source to listener space (convert to head relative) */
645 if(ALSource
->HeadRelative
== AL_FALSE
)
647 ALfloat (*restrict Matrix
)[4] = ALContext
->Listener
->Params
.Matrix
;
648 /* Transform source vectors */
649 aluMatrixVector(Position
, 1.0f
, Matrix
);
650 aluMatrixVector(Direction
, 0.0f
, Matrix
);
651 aluMatrixVector(Velocity
, 0.0f
, Matrix
);
655 const ALfloat
*ListenerVel
= ALContext
->Listener
->Params
.Velocity
;
656 /* Offset the source velocity to be relative of the listener velocity */
657 Velocity
[0] += ListenerVel
[0];
658 Velocity
[1] += ListenerVel
[1];
659 Velocity
[2] += ListenerVel
[2];
662 SourceToListener
[0] = -Position
[0];
663 SourceToListener
[1] = -Position
[1];
664 SourceToListener
[2] = -Position
[2];
665 aluNormalize(SourceToListener
);
666 aluNormalize(Direction
);
668 /* Calculate distance attenuation */
669 Distance
= sqrtf(aluDotproduct(Position
, Position
));
670 ClampedDist
= Distance
;
673 for(i
= 0;i
< NumSends
;i
++)
674 RoomAttenuation
[i
] = 1.0f
;
675 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
676 ALContext
->DistanceModel
)
678 case InverseDistanceClamped
:
679 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
680 if(MaxDist
< MinDist
)
683 case InverseDistance
:
686 if((MinDist
+ (Rolloff
* (ClampedDist
- MinDist
))) > 0.0f
)
687 Attenuation
= MinDist
/ (MinDist
+ (Rolloff
* (ClampedDist
- MinDist
)));
688 for(i
= 0;i
< NumSends
;i
++)
690 if((MinDist
+ (RoomRolloff
[i
] * (ClampedDist
- MinDist
))) > 0.0f
)
691 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (ClampedDist
- MinDist
)));
696 case LinearDistanceClamped
:
697 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
698 if(MaxDist
< MinDist
)
702 if(MaxDist
!= MinDist
)
704 Attenuation
= 1.0f
- (Rolloff
*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
705 Attenuation
= maxf(Attenuation
, 0.0f
);
706 for(i
= 0;i
< NumSends
;i
++)
708 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
709 RoomAttenuation
[i
] = maxf(RoomAttenuation
[i
], 0.0f
);
714 case ExponentDistanceClamped
:
715 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
716 if(MaxDist
< MinDist
)
719 case ExponentDistance
:
720 if(ClampedDist
> 0.0f
&& MinDist
> 0.0f
)
722 Attenuation
= powf(ClampedDist
/MinDist
, -Rolloff
);
723 for(i
= 0;i
< NumSends
;i
++)
724 RoomAttenuation
[i
] = powf(ClampedDist
/MinDist
, -RoomRolloff
[i
]);
728 case DisableDistance
:
729 ClampedDist
= MinDist
;
733 /* Source Gain + Attenuation */
734 DryGain
= SourceVolume
* Attenuation
;
735 for(i
= 0;i
< NumSends
;i
++)
736 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
738 /* Distance-based air absorption */
739 if(AirAbsorptionFactor
> 0.0f
&& ClampedDist
> MinDist
)
741 ALfloat meters
= maxf(ClampedDist
-MinDist
, 0.0f
) * MetersPerUnit
;
742 DryGainHF
*= powf(AIRABSORBGAINHF
, AirAbsorptionFactor
*meters
);
743 for(i
= 0;i
< NumSends
;i
++)
744 WetGainHF
[i
] *= powf(RoomAirAbsorption
[i
], AirAbsorptionFactor
*meters
);
749 ALfloat ApparentDist
= 1.0f
/maxf(Attenuation
, 0.00001f
) - 1.0f
;
751 /* Apply a decay-time transformation to the wet path, based on the
752 * attenuation of the dry path.
754 * Using the apparent distance, based on the distance attenuation, the
755 * initial decay of the reverb effect is calculated and applied to the
758 for(i
= 0;i
< NumSends
;i
++)
760 if(DecayDistance
[i
] > 0.0f
)
761 WetGain
[i
] *= powf(0.001f
/*-60dB*/, ApparentDist
/DecayDistance
[i
]);
765 /* Calculate directional soundcones */
766 Angle
= RAD2DEG(acosf(aluDotproduct(Direction
,SourceToListener
)) * ConeScale
) * 2.0f
;
767 if(Angle
> InnerAngle
&& Angle
<= OuterAngle
)
769 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
770 ConeVolume
= lerp(1.0f
, ALSource
->OuterGain
, scale
);
771 ConeHF
= lerp(1.0f
, ALSource
->OuterGainHF
, scale
);
773 else if(Angle
> OuterAngle
)
775 ConeVolume
= ALSource
->OuterGain
;
776 ConeHF
= ALSource
->OuterGainHF
;
784 DryGain
*= ConeVolume
;
787 for(i
= 0;i
< NumSends
;i
++)
788 WetGain
[i
] *= ConeVolume
;
794 for(i
= 0;i
< NumSends
;i
++)
795 WetGainHF
[i
] *= ConeHF
;
798 /* Clamp to Min/Max Gain */
799 DryGain
= clampf(DryGain
, MinVolume
, MaxVolume
);
800 for(i
= 0;i
< NumSends
;i
++)
801 WetGain
[i
] = clampf(WetGain
[i
], MinVolume
, MaxVolume
);
803 /* Apply gain and frequency filters */
804 DryGain
*= ALSource
->DirectGain
* ListenerGain
;
805 DryGainHF
*= ALSource
->DirectGainHF
;
806 for(i
= 0;i
< NumSends
;i
++)
808 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
809 WetGainHF
[i
] *= ALSource
->Send
[i
].GainHF
;
812 /* Calculate velocity-based doppler effect */
813 if(DopplerFactor
> 0.0f
)
815 const ALfloat
*ListenerVel
= ALContext
->Listener
->Params
.Velocity
;
818 if(SpeedOfSound
< 1.0f
)
820 DopplerFactor
*= 1.0f
/SpeedOfSound
;
824 VSS
= aluDotproduct(Velocity
, SourceToListener
) * DopplerFactor
;
825 VLS
= aluDotproduct(ListenerVel
, SourceToListener
) * DopplerFactor
;
827 Pitch
*= clampf(SpeedOfSound
-VLS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
) /
828 clampf(SpeedOfSound
-VSS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
);
831 BufferListItem
= ALSource
->queue
;
832 while(BufferListItem
!= NULL
)
835 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
837 /* Calculate fixed-point stepping value, based on the pitch, buffer
838 * frequency, and output frequency. */
839 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
841 src
->Step
= 10<<FRACTIONBITS
;
844 src
->Step
= fastf2i(Pitch
*FRACTIONONE
);
848 src
->Resample
= SelectResampler(Resampler
, src
->Step
);
852 BufferListItem
= BufferListItem
->next
;
857 /* Use a binaural HRTF algorithm for stereo headphone playback */
858 ALfloat delta
, ev
= 0.0f
, az
= 0.0f
;
860 if(Distance
> FLT_EPSILON
)
862 ALfloat invlen
= 1.0f
/Distance
;
863 Position
[0] *= invlen
;
864 Position
[1] *= invlen
;
865 Position
[2] *= invlen
;
867 /* Calculate elevation and azimuth only when the source is not at
868 * the listener. This prevents +0 and -0 Z from producing
869 * inconsistent panning. Also, clamp Y in case FP precision errors
870 * cause it to land outside of -1..+1. */
871 ev
= asinf(clampf(Position
[1], -1.0f
, 1.0f
));
872 az
= atan2f(Position
[0], -Position
[2]*ZScale
);
875 /* Check to see if the HRIR is already moving. */
876 if(ALSource
->Hrtf
.Moving
)
878 /* Calculate the normalized HRTF transition factor (delta). */
879 delta
= CalcHrtfDelta(src
->Direct
.Mix
.Hrtf
.Params
.Gain
, DryGain
,
880 src
->Direct
.Mix
.Hrtf
.Params
.Dir
, Position
);
881 /* If the delta is large enough, get the moving HRIR target
882 * coefficients, target delays, steppping values, and counter. */
885 ALSource
->Hrtf
.Counter
= GetMovingHrtfCoeffs(Device
->Hrtf
,
886 ev
, az
, DryGain
, delta
,
887 ALSource
->Hrtf
.Counter
,
888 src
->Direct
.Mix
.Hrtf
.Params
.Coeffs
[0],
889 src
->Direct
.Mix
.Hrtf
.Params
.Delay
[0],
890 src
->Direct
.Mix
.Hrtf
.Params
.CoeffStep
,
891 src
->Direct
.Mix
.Hrtf
.Params
.DelayStep
);
892 src
->Direct
.Mix
.Hrtf
.Params
.Gain
= DryGain
;
893 src
->Direct
.Mix
.Hrtf
.Params
.Dir
[0] = Position
[0];
894 src
->Direct
.Mix
.Hrtf
.Params
.Dir
[1] = Position
[1];
895 src
->Direct
.Mix
.Hrtf
.Params
.Dir
[2] = Position
[2];
900 /* Get the initial (static) HRIR coefficients and delays. */
901 GetLerpedHrtfCoeffs(Device
->Hrtf
, ev
, az
, DryGain
,
902 src
->Direct
.Mix
.Hrtf
.Params
.Coeffs
[0],
903 src
->Direct
.Mix
.Hrtf
.Params
.Delay
[0]);
904 ALSource
->Hrtf
.Counter
= 0;
905 ALSource
->Hrtf
.Moving
= AL_TRUE
;
906 src
->Direct
.Mix
.Hrtf
.Params
.Gain
= DryGain
;
907 src
->Direct
.Mix
.Hrtf
.Params
.Dir
[0] = Position
[0];
908 src
->Direct
.Mix
.Hrtf
.Params
.Dir
[1] = Position
[1];
909 src
->Direct
.Mix
.Hrtf
.Params
.Dir
[2] = Position
[2];
911 src
->Direct
.Mix
.Hrtf
.Params
.IrSize
= GetHrtfIrSize(Device
->Hrtf
);
913 src
->Direct
.Mix
.Hrtf
.State
= &ALSource
->Hrtf
;
914 src
->DryMix
= SelectHrtfMixer();
918 ALfloat (*Matrix
)[MaxChannels
] = src
->Direct
.Mix
.Gains
;
919 ALfloat DirGain
= 0.0f
;
922 for(i
= 0;i
< MAX_INPUT_CHANNELS
;i
++)
924 for(j
= 0;j
< MaxChannels
;j
++)
928 /* Normalize the length, and compute panned gains. */
929 if(Distance
> FLT_EPSILON
)
931 ALfloat invlen
= 1.0f
/Distance
;
932 Position
[0] *= invlen
;
933 Position
[1] *= invlen
;
934 Position
[2] *= invlen
;
936 DirGain
= sqrtf(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
937 ComputeAngleGains(Device
, atan2f(Position
[0], -Position
[2]*ZScale
), 0.0f
,
938 DryGain
*DirGain
, Matrix
[0]);
941 /* Adjustment for vertical offsets. Not the greatest, but simple
943 AmbientGain
= DryGain
* sqrtf(1.0f
/Device
->NumChan
) * (1.0f
-DirGain
);
944 for(i
= 0;i
< (ALint
)Device
->NumChan
;i
++)
946 enum Channel chan
= Device
->Speaker2Chan
[i
];
947 Matrix
[0][chan
] = maxf(Matrix
[0][chan
], AmbientGain
);
949 src
->DryMix
= SelectDirectMixer();
951 for(i
= 0;i
< NumSends
;i
++)
952 src
->Send
[i
].Gain
= WetGain
[i
];
953 src
->WetMix
= SelectSendMixer();
956 ALfloat gain
= maxf(0.01f
, DryGainHF
);
957 ALfilterState_setParams(&src
->Direct
.LpFilter
[0],
958 ALfilterType_HighShelf
, gain
,
959 (ALfloat
)LOWPASSFREQREF
/Frequency
, 0.0f
);
961 for(i
= 0;i
< NumSends
;i
++)
963 ALfloat gain
= maxf(0.01f
, WetGainHF
[i
]);
964 ALfilterState_setParams(&src
->Send
[i
].LpFilter
[0],
965 ALfilterType_HighShelf
, gain
,
966 (ALfloat
)LOWPASSFREQREF
/Frequency
, 0.0f
);
971 static inline ALint
aluF2I25(ALfloat val
)
973 /* Clamp the value between -1 and +1. This handles that with only a single branch. */
974 if(fabsf(val
) > 1.0f
)
975 val
= (ALfloat
)((0.0f
< val
) - (val
< 0.0f
));
976 /* Convert to a signed integer, between -16777215 and +16777215. */
977 return fastf2i(val
*16777215.0f
);
980 static inline ALfloat
aluF2F(ALfloat val
)
982 static inline ALint
aluF2I(ALfloat val
)
983 { return aluF2I25(val
)<<7; }
984 static inline ALuint
aluF2UI(ALfloat val
)
985 { return aluF2I(val
)+2147483648u; }
986 static inline ALshort
aluF2S(ALfloat val
)
987 { return aluF2I25(val
)>>9; }
988 static inline ALushort
aluF2US(ALfloat val
)
989 { return aluF2S(val
)+32768; }
990 static inline ALbyte
aluF2B(ALfloat val
)
991 { return aluF2I25(val
)>>17; }
992 static inline ALubyte
aluF2UB(ALfloat val
)
993 { return aluF2B(val
)+128; }
995 #define DECL_TEMPLATE(T, func) \
996 static int Write_##T(ALCdevice *device, T *restrict buffer, \
997 ALuint SamplesToDo) \
999 ALfloat (*restrict DryBuffer)[BUFFERSIZE] = device->DryBuffer; \
1000 ALuint numchans = ChannelsFromDevFmt(device->FmtChans); \
1001 const ALuint *offsets = device->ChannelOffsets; \
1004 for(j = 0;j < MaxChannels;j++) \
1008 if(offsets[j] == INVALID_OFFSET) \
1011 out = buffer + offsets[j]; \
1012 for(i = 0;i < SamplesToDo;i++) \
1013 out[i*numchans] = func(DryBuffer[j][i]); \
1015 return SamplesToDo*numchans*sizeof(T); \
1018 DECL_TEMPLATE(ALfloat
, aluF2F
)
1019 DECL_TEMPLATE(ALuint
, aluF2UI
)
1020 DECL_TEMPLATE(ALint
, aluF2I
)
1021 DECL_TEMPLATE(ALushort
, aluF2US
)
1022 DECL_TEMPLATE(ALshort
, aluF2S
)
1023 DECL_TEMPLATE(ALubyte
, aluF2UB
)
1024 DECL_TEMPLATE(ALbyte
, aluF2B
)
1026 #undef DECL_TEMPLATE
1029 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1032 ALeffectslot
**slot
, **slot_end
;
1033 ALactivesource
**src
, **src_end
;
1038 SetMixerFPUMode(&oldMode
);
1042 IncrementRef(&device
->MixCount
);
1044 SamplesToDo
= minu(size
, BUFFERSIZE
);
1045 for(c
= 0;c
< MaxChannels
;c
++)
1046 memset(device
->DryBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1048 ALCdevice_Lock(device
);
1049 V(device
->Synth
,process
)(SamplesToDo
, device
->DryBuffer
);
1051 ctx
= device
->ContextList
;
1054 ALenum DeferUpdates
= ctx
->DeferUpdates
;
1055 ALenum UpdateSources
= AL_FALSE
;
1058 UpdateSources
= ExchangeInt(&ctx
->UpdateSources
, AL_FALSE
);
1061 CalcListenerParams(ctx
->Listener
);
1063 /* source processing */
1064 src
= ctx
->ActiveSources
;
1065 src_end
= src
+ ctx
->ActiveSourceCount
;
1066 while(src
!= src_end
)
1068 ALsource
*source
= (*src
)->Source
;
1070 if(source
->state
!= AL_PLAYING
&& source
->state
!= AL_PAUSED
)
1072 ALactivesource
*temp
= *(--src_end
);
1075 --(ctx
->ActiveSourceCount
);
1079 if(!DeferUpdates
&& (ExchangeInt(&source
->NeedsUpdate
, AL_FALSE
) ||
1081 (*src
)->Update(*src
, ctx
);
1083 if(source
->state
!= AL_PAUSED
)
1084 MixSource(*src
, device
, SamplesToDo
);
1088 /* effect slot processing */
1089 slot
= VECTOR_ITER_BEGIN(ctx
->ActiveAuxSlots
);
1090 slot_end
= VECTOR_ITER_END(ctx
->ActiveAuxSlots
);
1091 while(slot
!= slot_end
)
1093 ALfloat offset
= (*slot
)->ClickRemoval
[0];
1094 if(offset
< (1.0f
/32768.0f
))
1096 else for(i
= 0;i
< SamplesToDo
;i
++)
1098 (*slot
)->WetBuffer
[0][i
] += offset
;
1099 offset
-= offset
* (1.0f
/256.0f
);
1101 (*slot
)->ClickRemoval
[0] = offset
+ (*slot
)->PendingClicks
[0];
1102 (*slot
)->PendingClicks
[0] = 0.0f
;
1104 if(!DeferUpdates
&& ExchangeInt(&(*slot
)->NeedsUpdate
, AL_FALSE
))
1105 V((*slot
)->EffectState
,update
)(device
, *slot
);
1107 V((*slot
)->EffectState
,process
)(SamplesToDo
, (*slot
)->WetBuffer
[0],
1110 for(i
= 0;i
< SamplesToDo
;i
++)
1111 (*slot
)->WetBuffer
[0][i
] = 0.0f
;
1119 slot
= &device
->DefaultSlot
;
1122 ALfloat offset
= (*slot
)->ClickRemoval
[0];
1123 if(offset
< (1.0f
/32768.0f
))
1125 else for(i
= 0;i
< SamplesToDo
;i
++)
1127 (*slot
)->WetBuffer
[0][i
] += offset
;
1128 offset
-= offset
* (1.0f
/256.0f
);
1130 (*slot
)->ClickRemoval
[0] = offset
+ (*slot
)->PendingClicks
[0];
1131 (*slot
)->PendingClicks
[0] = 0.0f
;
1133 if(ExchangeInt(&(*slot
)->NeedsUpdate
, AL_FALSE
))
1134 V((*slot
)->EffectState
,update
)(device
, *slot
);
1136 V((*slot
)->EffectState
,process
)(SamplesToDo
, (*slot
)->WetBuffer
[0],
1139 for(i
= 0;i
< SamplesToDo
;i
++)
1140 (*slot
)->WetBuffer
[0][i
] = 0.0f
;
1143 /* Increment the clock time. Every second's worth of samples is
1144 * converted and added to clock base so that large sample counts don't
1145 * overflow during conversion. This also guarantees an exact, stable
1147 device
->SamplesDone
+= SamplesToDo
;
1148 device
->ClockBase
+= (device
->SamplesDone
/device
->Frequency
) * DEVICE_CLOCK_RES
;
1149 device
->SamplesDone
%= device
->Frequency
;
1150 ALCdevice_Unlock(device
);
1152 /* Click-removal. Could do better; this only really handles immediate
1153 * changes between updates where a predictive sample could be
1154 * generated. Delays caused by effects and HRTF aren't caught. */
1155 if(device
->FmtChans
== DevFmtStereo
)
1157 /* Assumes the first two channels are FrontLeft and FrontRight */
1158 for(c
= 0;c
< 2;c
++)
1160 ALfloat offset
= device
->ClickRemoval
[c
];
1161 if(offset
< (1.0f
/32768.0f
))
1163 else for(i
= 0;i
< SamplesToDo
;i
++)
1165 device
->DryBuffer
[c
][i
] += offset
;
1166 offset
-= offset
* (1.0f
/256.0f
);
1168 device
->ClickRemoval
[c
] = offset
+ device
->PendingClicks
[c
];
1169 device
->PendingClicks
[c
] = 0.0f
;
1174 for(i
= 0;i
< SamplesToDo
;i
++)
1176 samples
[0] = device
->DryBuffer
[FrontLeft
][i
];
1177 samples
[1] = device
->DryBuffer
[FrontRight
][i
];
1178 bs2b_cross_feed(device
->Bs2b
, samples
);
1179 device
->DryBuffer
[FrontLeft
][i
] = samples
[0];
1180 device
->DryBuffer
[FrontRight
][i
] = samples
[1];
1186 for(c
= 0;c
< MaxChannels
;c
++)
1188 ALfloat offset
= device
->ClickRemoval
[c
];
1189 if(offset
< (1.0f
/32768.0f
))
1191 else for(i
= 0;i
< SamplesToDo
;i
++)
1193 device
->DryBuffer
[c
][i
] += offset
;
1194 offset
-= offset
* (1.0f
/256.0f
);
1196 device
->ClickRemoval
[c
] = offset
+ device
->PendingClicks
[c
];
1197 device
->PendingClicks
[c
] = 0.0f
;
1204 switch(device
->FmtType
)
1207 bytes
= Write_ALbyte(device
, buffer
, SamplesToDo
);
1210 bytes
= Write_ALubyte(device
, buffer
, SamplesToDo
);
1213 bytes
= Write_ALshort(device
, buffer
, SamplesToDo
);
1216 bytes
= Write_ALushort(device
, buffer
, SamplesToDo
);
1219 bytes
= Write_ALint(device
, buffer
, SamplesToDo
);
1222 bytes
= Write_ALuint(device
, buffer
, SamplesToDo
);
1225 bytes
= Write_ALfloat(device
, buffer
, SamplesToDo
);
1229 buffer
= (ALubyte
*)buffer
+ bytes
;
1232 size
-= SamplesToDo
;
1233 IncrementRef(&device
->MixCount
);
1236 RestoreFPUMode(&oldMode
);
1240 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1242 ALCcontext
*Context
;
1244 device
->Connected
= ALC_FALSE
;
1246 Context
= device
->ContextList
;
1249 ALactivesource
**src
, **src_end
;
1251 src
= Context
->ActiveSources
;
1252 src_end
= src
+ Context
->ActiveSourceCount
;
1253 while(src
!= src_end
)
1255 ALsource
*source
= (*src
)->Source
;
1256 if(source
->state
== AL_PLAYING
)
1258 source
->state
= AL_STOPPED
;
1259 source
->BuffersPlayed
= source
->BuffersInQueue
;
1260 source
->position
= 0;
1261 source
->position_fraction
= 0;
1265 Context
->ActiveSourceCount
= 0;
1267 Context
= Context
->next
;