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
34 #include "alListener.h"
35 #include "alAuxEffectSlot.h"
40 static __inline ALvoid
aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
42 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
43 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
44 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
47 static __inline ALfloat
aluDotproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
)
49 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
50 inVector1
[2]*inVector2
[2];
53 static __inline ALvoid
aluNormalize(ALfloat
*inVector
)
55 ALfloat length
, inverse_length
;
57 length
= aluSqrt(aluDotproduct(inVector
, inVector
));
60 inverse_length
= 1.0f
/length
;
61 inVector
[0] *= inverse_length
;
62 inVector
[1] *= inverse_length
;
63 inVector
[2] *= inverse_length
;
67 static __inline ALvoid
aluMatrixVector(ALfloat
*vector
,ALfloat w
,ALfloat matrix
[4][4])
70 vector
[0], vector
[1], vector
[2], w
73 vector
[0] = temp
[0]*matrix
[0][0] + temp
[1]*matrix
[1][0] + temp
[2]*matrix
[2][0] + temp
[3]*matrix
[3][0];
74 vector
[1] = temp
[0]*matrix
[0][1] + temp
[1]*matrix
[1][1] + temp
[2]*matrix
[2][1] + temp
[3]*matrix
[3][1];
75 vector
[2] = temp
[0]*matrix
[0][2] + temp
[1]*matrix
[1][2] + temp
[2]*matrix
[2][2] + temp
[3]*matrix
[3][2];
79 ALvoid
CalcNonAttnSourceParams(ALsource
*ALSource
, const ALCcontext
*ALContext
)
81 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
82 ALbufferlistitem
*BufferListItem
;
83 ALfloat DryGain
, DryGainHF
;
84 ALfloat WetGain
[MAX_SENDS
];
85 ALfloat WetGainHF
[MAX_SENDS
];
86 ALint NumSends
, Frequency
;
94 //Get context properties
95 Format
= ALContext
->Device
->Format
;
96 DupStereo
= ALContext
->Device
->DuplicateStereo
;
97 NumSends
= ALContext
->Device
->NumAuxSends
;
98 Frequency
= ALContext
->Device
->Frequency
;
100 //Get listener properties
101 ListenerGain
= ALContext
->Listener
.Gain
;
103 //Get source properties
104 SourceVolume
= ALSource
->flGain
;
105 MinVolume
= ALSource
->flMinGain
;
106 MaxVolume
= ALSource
->flMaxGain
;
108 //1. Multi-channel buffers always play "normal"
110 Pitch
= ALSource
->flPitch
;
111 BufferListItem
= ALSource
->queue
;
112 while(BufferListItem
!= NULL
)
115 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
117 Channels
= aluChannelsFromFormat(ALBuffer
->format
);
118 Pitch
= Pitch
* ALBuffer
->frequency
/ Frequency
;
121 BufferListItem
= BufferListItem
->next
;
124 if(Pitch
> (float)MAX_PITCH
)
125 ALSource
->Params
.Step
= MAX_PITCH
<<FRACTIONBITS
;
126 else if(!(Pitch
> 0.0f
))
127 ALSource
->Params
.Step
= 1<<FRACTIONBITS
;
130 ALSource
->Params
.Step
= Pitch
*(1<<FRACTIONBITS
);
131 if(ALSource
->Params
.Step
== 0)
132 ALSource
->Params
.Step
= 1;
135 DryGain
= SourceVolume
;
136 DryGain
= __min(DryGain
,MaxVolume
);
137 DryGain
= __max(DryGain
,MinVolume
);
140 switch(ALSource
->DirectFilter
.type
)
142 case AL_FILTER_LOWPASS
:
143 DryGain
*= ALSource
->DirectFilter
.Gain
;
144 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
150 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
151 ALSource
->Params
.DryGains
[i
] = 0.0f
;
153 if(DupStereo
== AL_FALSE
)
155 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
156 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
162 case AL_FORMAT_MONO8
:
163 case AL_FORMAT_MONO16
:
164 case AL_FORMAT_MONO_FLOAT32
:
165 case AL_FORMAT_STEREO8
:
166 case AL_FORMAT_STEREO16
:
167 case AL_FORMAT_STEREO_FLOAT32
:
168 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
169 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
172 case AL_FORMAT_QUAD8
:
173 case AL_FORMAT_QUAD16
:
174 case AL_FORMAT_QUAD32
:
175 case AL_FORMAT_51CHN8
:
176 case AL_FORMAT_51CHN16
:
177 case AL_FORMAT_51CHN32
:
178 DryGain
*= aluSqrt(2.0f
/4.0f
);
179 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
180 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
181 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryGain
* ListenerGain
;
182 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryGain
* ListenerGain
;
185 case AL_FORMAT_61CHN8
:
186 case AL_FORMAT_61CHN16
:
187 case AL_FORMAT_61CHN32
:
188 DryGain
*= aluSqrt(2.0f
/4.0f
);
189 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
190 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
191 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryGain
* ListenerGain
;
192 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryGain
* ListenerGain
;
195 case AL_FORMAT_71CHN8
:
196 case AL_FORMAT_71CHN16
:
197 case AL_FORMAT_71CHN32
:
198 DryGain
*= aluSqrt(2.0f
/6.0f
);
199 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
200 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
201 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryGain
* ListenerGain
;
202 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryGain
* ListenerGain
;
203 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryGain
* ListenerGain
;
204 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryGain
* ListenerGain
;
214 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
215 ALSource
->Params
.DryGains
[i
] = DryGain
* ListenerGain
;
218 for(i
= 0;i
< NumSends
;i
++)
220 WetGain
[i
] = SourceVolume
;
221 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
222 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
225 switch(ALSource
->Send
[i
].WetFilter
.type
)
227 case AL_FILTER_LOWPASS
:
228 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
229 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
233 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
235 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
237 ALSource
->Params
.WetGains
[i
] = 0.0f
;
241 /* Update filter coefficients. Calculations based on the I3DL2
243 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
245 /* We use two chained one-pole filters, so we need to take the
246 * square root of the squared gain, which is the same as the base
248 ALSource
->Params
.iirFilter
.coeff
= lpCoeffCalc(DryGainHF
, cw
);
250 for(i
= 0;i
< NumSends
;i
++)
252 /* We use a one-pole filter, so we need to take the squared gain */
253 ALfloat a
= lpCoeffCalc(WetGainHF
[i
]*WetGainHF
[i
], cw
);
254 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
258 ALvoid
CalcSourceParams(ALsource
*ALSource
, const ALCcontext
*ALContext
)
260 const ALCdevice
*Device
= ALContext
->Device
;
261 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,DryMix
,OrigDist
;
262 ALfloat Direction
[3],Position
[3],SourceToListener
[3];
263 ALfloat Velocity
[3],ListenerVel
[3];
264 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
,OuterGainHF
;
265 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
266 ALfloat DopplerFactor
, DopplerVelocity
, flSpeedOfSound
;
267 ALbufferlistitem
*BufferListItem
;
268 ALfloat Matrix
[4][4];
269 ALfloat flAttenuation
, effectiveDist
;
270 ALfloat RoomAttenuation
[MAX_SENDS
];
271 ALfloat MetersPerUnit
;
272 ALfloat RoomRolloff
[MAX_SENDS
];
273 ALfloat DryGainHF
= 1.0f
;
274 ALfloat WetGain
[MAX_SENDS
];
275 ALfloat WetGainHF
[MAX_SENDS
];
276 ALfloat DirGain
, AmbientGain
;
277 const ALfloat
*SpeakerGain
;
285 for(i
= 0;i
< MAX_SENDS
;i
++)
288 //Get context properties
289 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
290 DopplerVelocity
= ALContext
->DopplerVelocity
;
291 flSpeedOfSound
= ALContext
->flSpeedOfSound
;
292 NumSends
= Device
->NumAuxSends
;
293 Frequency
= Device
->Frequency
;
295 //Get listener properties
296 ListenerGain
= ALContext
->Listener
.Gain
;
297 MetersPerUnit
= ALContext
->Listener
.MetersPerUnit
;
298 memcpy(ListenerVel
, ALContext
->Listener
.Velocity
, sizeof(ALContext
->Listener
.Velocity
));
300 //Get source properties
301 SourceVolume
= ALSource
->flGain
;
302 memcpy(Position
, ALSource
->vPosition
, sizeof(ALSource
->vPosition
));
303 memcpy(Direction
, ALSource
->vOrientation
, sizeof(ALSource
->vOrientation
));
304 memcpy(Velocity
, ALSource
->vVelocity
, sizeof(ALSource
->vVelocity
));
305 MinVolume
= ALSource
->flMinGain
;
306 MaxVolume
= ALSource
->flMaxGain
;
307 MinDist
= ALSource
->flRefDistance
;
308 MaxDist
= ALSource
->flMaxDistance
;
309 Rolloff
= ALSource
->flRollOffFactor
;
310 InnerAngle
= ALSource
->flInnerAngle
;
311 OuterAngle
= ALSource
->flOuterAngle
;
312 OuterGainHF
= ALSource
->OuterGainHF
;
314 //1. Translate Listener to origin (convert to head relative)
315 if(ALSource
->bHeadRelative
==AL_FALSE
)
317 ALfloat U
[3],V
[3],N
[3];
319 // Build transform matrix
320 memcpy(N
, ALContext
->Listener
.Forward
, sizeof(N
)); // At-vector
321 aluNormalize(N
); // Normalized At-vector
322 memcpy(V
, ALContext
->Listener
.Up
, sizeof(V
)); // Up-vector
323 aluNormalize(V
); // Normalized Up-vector
324 aluCrossproduct(N
, V
, U
); // Right-vector
325 aluNormalize(U
); // Normalized Right-vector
326 Matrix
[0][0] = U
[0]; Matrix
[0][1] = V
[0]; Matrix
[0][2] = -N
[0]; Matrix
[0][3] = 0.0f
;
327 Matrix
[1][0] = U
[1]; Matrix
[1][1] = V
[1]; Matrix
[1][2] = -N
[1]; Matrix
[1][3] = 0.0f
;
328 Matrix
[2][0] = U
[2]; Matrix
[2][1] = V
[2]; Matrix
[2][2] = -N
[2]; Matrix
[2][3] = 0.0f
;
329 Matrix
[3][0] = 0.0f
; Matrix
[3][1] = 0.0f
; Matrix
[3][2] = 0.0f
; Matrix
[3][3] = 1.0f
;
331 // Translate position
332 Position
[0] -= ALContext
->Listener
.Position
[0];
333 Position
[1] -= ALContext
->Listener
.Position
[1];
334 Position
[2] -= ALContext
->Listener
.Position
[2];
336 // Transform source position and direction into listener space
337 aluMatrixVector(Position
, 1.0f
, Matrix
);
338 aluMatrixVector(Direction
, 0.0f
, Matrix
);
339 // Transform source and listener velocity into listener space
340 aluMatrixVector(Velocity
, 0.0f
, Matrix
);
341 aluMatrixVector(ListenerVel
, 0.0f
, Matrix
);
344 ListenerVel
[0] = ListenerVel
[1] = ListenerVel
[2] = 0.0f
;
346 SourceToListener
[0] = -Position
[0];
347 SourceToListener
[1] = -Position
[1];
348 SourceToListener
[2] = -Position
[2];
349 aluNormalize(SourceToListener
);
350 aluNormalize(Direction
);
352 //2. Calculate distance attenuation
353 Distance
= aluSqrt(aluDotproduct(Position
, Position
));
356 flAttenuation
= 1.0f
;
357 for(i
= 0;i
< NumSends
;i
++)
359 RoomAttenuation
[i
] = 1.0f
;
361 RoomRolloff
[i
] = ALSource
->RoomRolloffFactor
;
362 if(ALSource
->Send
[i
].Slot
&&
363 (ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
364 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
))
365 RoomRolloff
[i
] += ALSource
->Send
[i
].Slot
->effect
.Reverb
.RoomRolloffFactor
;
368 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
369 ALContext
->DistanceModel
)
371 case AL_INVERSE_DISTANCE_CLAMPED
:
372 Distance
=__max(Distance
,MinDist
);
373 Distance
=__min(Distance
,MaxDist
);
374 if(MaxDist
< MinDist
)
377 case AL_INVERSE_DISTANCE
:
380 if((MinDist
+ (Rolloff
* (Distance
- MinDist
))) > 0.0f
)
381 flAttenuation
= MinDist
/ (MinDist
+ (Rolloff
* (Distance
- MinDist
)));
382 for(i
= 0;i
< NumSends
;i
++)
384 if((MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
))) > 0.0f
)
385 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
)));
390 case AL_LINEAR_DISTANCE_CLAMPED
:
391 Distance
=__max(Distance
,MinDist
);
392 Distance
=__min(Distance
,MaxDist
);
393 if(MaxDist
< MinDist
)
396 case AL_LINEAR_DISTANCE
:
397 Distance
=__min(Distance
,MaxDist
);
398 if(MaxDist
!= MinDist
)
400 flAttenuation
= 1.0f
- (Rolloff
*(Distance
-MinDist
)/(MaxDist
- MinDist
));
401 for(i
= 0;i
< NumSends
;i
++)
402 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(Distance
-MinDist
)/(MaxDist
- MinDist
));
406 case AL_EXPONENT_DISTANCE_CLAMPED
:
407 Distance
=__max(Distance
,MinDist
);
408 Distance
=__min(Distance
,MaxDist
);
409 if(MaxDist
< MinDist
)
412 case AL_EXPONENT_DISTANCE
:
413 if(Distance
> 0.0f
&& MinDist
> 0.0f
)
415 flAttenuation
= aluPow(Distance
/MinDist
, -Rolloff
);
416 for(i
= 0;i
< NumSends
;i
++)
417 RoomAttenuation
[i
] = aluPow(Distance
/MinDist
, -RoomRolloff
[i
]);
425 // Source Gain + Attenuation
426 DryMix
= SourceVolume
* flAttenuation
;
427 for(i
= 0;i
< NumSends
;i
++)
428 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
430 effectiveDist
= 0.0f
;
432 effectiveDist
= (MinDist
/flAttenuation
- MinDist
)*MetersPerUnit
;
434 // Distance-based air absorption
435 if(ALSource
->AirAbsorptionFactor
> 0.0f
&& effectiveDist
> 0.0f
)
439 // Absorption calculation is done in dB
440 absorb
= (ALSource
->AirAbsorptionFactor
*AIRABSORBGAINDBHF
) *
442 // Convert dB to linear gain before applying
443 absorb
= aluPow(10.0f
, absorb
/20.0f
);
448 //3. Apply directional soundcones
449 Angle
= aluAcos(aluDotproduct(Direction
,SourceToListener
)) * 180.0f
/M_PI
;
450 if(Angle
>= InnerAngle
&& Angle
<= OuterAngle
)
452 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
453 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
)*scale
);
454 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
)*scale
);
456 else if(Angle
> OuterAngle
)
458 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
));
459 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
));
467 // Apply some high-frequency attenuation for sources behind the listener
468 // NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
469 // that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
470 // the same as SourceToListener[2]
471 Angle
= aluAcos(SourceToListener
[2]) * 180.0f
/M_PI
;
472 // Sources within the minimum distance attenuate less
473 if(OrigDist
< MinDist
)
474 Angle
*= OrigDist
/MinDist
;
477 ALfloat scale
= (Angle
-90.0f
) / (180.1f
-90.0f
); // .1 to account for fp errors
478 ConeHF
*= 1.0f
- (Device
->HeadDampen
*scale
);
481 DryMix
*= ConeVolume
;
482 if(ALSource
->DryGainHFAuto
)
485 // Clamp to Min/Max Gain
486 DryMix
= __min(DryMix
,MaxVolume
);
487 DryMix
= __max(DryMix
,MinVolume
);
489 for(i
= 0;i
< NumSends
;i
++)
491 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
493 if(!Slot
|| Slot
->effect
.type
== AL_EFFECT_NULL
)
495 ALSource
->Params
.WetGains
[i
] = 0.0f
;
500 if(Slot
->AuxSendAuto
)
502 if(ALSource
->WetGainAuto
)
503 WetGain
[i
] *= ConeVolume
;
504 if(ALSource
->WetGainHFAuto
)
505 WetGainHF
[i
] *= ConeHF
;
507 // Clamp to Min/Max Gain
508 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
509 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
511 if(Slot
->effect
.type
== AL_EFFECT_REVERB
||
512 Slot
->effect
.type
== AL_EFFECT_EAXREVERB
)
514 /* Apply a decay-time transformation to the wet path, based on
515 * the attenuation of the dry path.
517 * Using the approximate (effective) source to listener
518 * distance, the initial decay of the reverb effect is
519 * calculated and applied to the wet path.
521 WetGain
[i
] *= aluPow(10.0f
, effectiveDist
/
522 (SPEEDOFSOUNDMETRESPERSEC
*
523 Slot
->effect
.Reverb
.DecayTime
) *
526 WetGainHF
[i
] *= aluPow(10.0f
,
527 log10(Slot
->effect
.Reverb
.AirAbsorptionGainHF
) *
528 ALSource
->AirAbsorptionFactor
* effectiveDist
);
533 /* If the slot's auxiliary send auto is off, the data sent to the
534 * effect slot is the same as the dry path, sans filter effects */
536 WetGainHF
[i
] = DryGainHF
;
539 switch(ALSource
->Send
[i
].WetFilter
.type
)
541 case AL_FILTER_LOWPASS
:
542 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
543 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
546 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
548 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
550 ALSource
->Params
.WetGains
[i
] = 0.0f
;
554 // Apply filter gains and filters
555 switch(ALSource
->DirectFilter
.type
)
557 case AL_FILTER_LOWPASS
:
558 DryMix
*= ALSource
->DirectFilter
.Gain
;
559 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
562 DryMix
*= ListenerGain
;
564 // Calculate Velocity
565 if(DopplerFactor
!= 0.0f
)
567 ALfloat flVSS
, flVLS
;
568 ALfloat flMaxVelocity
= (DopplerVelocity
* flSpeedOfSound
) /
571 flVSS
= aluDotproduct(Velocity
, SourceToListener
);
572 if(flVSS
>= flMaxVelocity
)
573 flVSS
= (flMaxVelocity
- 1.0f
);
574 else if(flVSS
<= -flMaxVelocity
)
575 flVSS
= -flMaxVelocity
+ 1.0f
;
577 flVLS
= aluDotproduct(ListenerVel
, SourceToListener
);
578 if(flVLS
>= flMaxVelocity
)
579 flVLS
= (flMaxVelocity
- 1.0f
);
580 else if(flVLS
<= -flMaxVelocity
)
581 flVLS
= -flMaxVelocity
+ 1.0f
;
583 Pitch
= ALSource
->flPitch
*
584 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVLS
)) /
585 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVSS
));
588 Pitch
= ALSource
->flPitch
;
590 BufferListItem
= ALSource
->queue
;
591 while(BufferListItem
!= NULL
)
594 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
596 Pitch
= Pitch
* ALBuffer
->frequency
/ Frequency
;
599 BufferListItem
= BufferListItem
->next
;
602 if(Pitch
> (float)MAX_PITCH
)
603 ALSource
->Params
.Step
= MAX_PITCH
<<FRACTIONBITS
;
604 else if(!(Pitch
> 0.0f
))
605 ALSource
->Params
.Step
= 1<<FRACTIONBITS
;
608 ALSource
->Params
.Step
= Pitch
*(1<<FRACTIONBITS
);
609 if(ALSource
->Params
.Step
== 0)
610 ALSource
->Params
.Step
= 1;
613 // Use energy-preserving panning algorithm for multi-speaker playback
614 length
= __max(OrigDist
, MinDist
);
617 ALfloat invlen
= 1.0f
/length
;
618 Position
[0] *= invlen
;
619 Position
[1] *= invlen
;
620 Position
[2] *= invlen
;
623 pos
= aluCart2LUTpos(-Position
[2], Position
[0]);
624 SpeakerGain
= &Device
->PanningLUT
[OUTPUTCHANNELS
* pos
];
626 DirGain
= aluSqrt(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
627 // elevation adjustment for directional gain. this sucks, but
628 // has low complexity
629 AmbientGain
= 1.0/aluSqrt(Device
->NumChan
) * (1.0-DirGain
);
630 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
631 ALSource
->Params
.DryGains
[s
] = 0.0f
;
632 for(s
= 0;s
< (ALsizei
)Device
->NumChan
;s
++)
634 Channel chan
= Device
->Speaker2Chan
[s
];
635 ALfloat gain
= SpeakerGain
[chan
]*DirGain
+ AmbientGain
;
636 ALSource
->Params
.DryGains
[chan
] = DryMix
* gain
;
639 /* Update filter coefficients. */
640 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
642 /* Spatialized sources use four chained one-pole filters, so we need to
643 * take the fourth root of the squared gain, which is the same as the
644 * square root of the base gain. */
645 ALSource
->Params
.iirFilter
.coeff
= lpCoeffCalc(aluSqrt(DryGainHF
), cw
);
647 for(i
= 0;i
< NumSends
;i
++)
649 /* The wet path uses two chained one-pole filters, so take the
650 * base gain (square root of the squared gain) */
651 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= lpCoeffCalc(WetGainHF
[i
], cw
);
656 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
660 SuspendContext(NULL
);
661 for(i
= 0;i
< device
->NumContexts
;i
++)
663 ALCcontext
*Context
= device
->Contexts
[i
];
667 SuspendContext(Context
);
669 for(pos
= 0;pos
< Context
->SourceMap
.size
;pos
++)
671 source
= Context
->SourceMap
.array
[pos
].value
;
672 if(source
->state
== AL_PLAYING
)
674 source
->state
= AL_STOPPED
;
675 source
->BuffersPlayed
= source
->BuffersInQueue
;
676 source
->position
= 0;
677 source
->position_fraction
= 0;
680 ProcessContext(Context
);
683 device
->Connected
= ALC_FALSE
;
684 ProcessContext(NULL
);