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
35 #include "alListener.h"
36 #include "alAuxEffectSlot.h"
40 #if defined(HAVE_STDINT_H)
42 typedef int64_t ALint64
;
43 #elif defined(HAVE___INT64)
44 typedef __int64 ALint64
;
45 #elif (SIZEOF_LONG == 8)
47 #elif (SIZEOF_LONG_LONG == 8)
48 typedef long long ALint64
;
51 #define FRACTIONBITS 14
52 #define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
53 #define MAX_PITCH 65536
55 /* Minimum ramp length in milliseconds. The value below was chosen to
56 * adequately reduce clicks and pops from harsh gain changes. */
57 #define MIN_RAMP_LENGTH 16
59 ALboolean DuplicateStereo
= AL_FALSE
;
62 static __inline ALfloat
aluF2F(ALfloat Value
)
64 if(Value
< 0.f
) return Value
/32768.f
;
65 if(Value
> 0.f
) return Value
/32767.f
;
69 static __inline ALshort
aluF2S(ALfloat Value
)
79 static __inline ALubyte
aluF2UB(ALfloat Value
)
81 ALshort i
= aluF2S(Value
);
86 static __inline ALvoid
aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
88 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
89 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
90 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
93 static __inline ALfloat
aluDotproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
)
95 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
96 inVector1
[2]*inVector2
[2];
99 static __inline ALvoid
aluNormalize(ALfloat
*inVector
)
101 ALfloat length
, inverse_length
;
103 length
= aluSqrt(aluDotproduct(inVector
, inVector
));
106 inverse_length
= 1.0f
/length
;
107 inVector
[0] *= inverse_length
;
108 inVector
[1] *= inverse_length
;
109 inVector
[2] *= inverse_length
;
113 static __inline ALvoid
aluMatrixVector(ALfloat
*vector
,ALfloat w
,ALfloat matrix
[4][4])
116 vector
[0], vector
[1], vector
[2], w
119 vector
[0] = temp
[0]*matrix
[0][0] + temp
[1]*matrix
[1][0] + temp
[2]*matrix
[2][0] + temp
[3]*matrix
[3][0];
120 vector
[1] = temp
[0]*matrix
[0][1] + temp
[1]*matrix
[1][1] + temp
[2]*matrix
[2][1] + temp
[3]*matrix
[3][1];
121 vector
[2] = temp
[0]*matrix
[0][2] + temp
[1]*matrix
[1][2] + temp
[2]*matrix
[2][2] + temp
[3]*matrix
[3][2];
124 static ALvoid
SetSpeakerArrangement(const char *name
, ALfloat SpeakerAngle
[OUTPUTCHANNELS
],
125 ALint Speaker2Chan
[OUTPUTCHANNELS
], ALint chans
)
133 confkey
= GetConfigValue(NULL
, name
, "");
138 next
= strchr(confkey
, ',');
143 } while(isspace(*next
));
146 sep
= strchr(confkey
, '=');
147 if(!sep
|| confkey
== sep
)
151 while(isspace(*end
) && end
!= confkey
)
155 if(strncmp(confkey
, "fl", end
-confkey
) == 0)
157 else if(strncmp(confkey
, "fr", end
-confkey
) == 0)
159 else if(strncmp(confkey
, "fc", end
-confkey
) == 0)
161 else if(strncmp(confkey
, "bl", end
-confkey
) == 0)
163 else if(strncmp(confkey
, "br", end
-confkey
) == 0)
165 else if(strncmp(confkey
, "bc", end
-confkey
) == 0)
167 else if(strncmp(confkey
, "sl", end
-confkey
) == 0)
169 else if(strncmp(confkey
, "sr", end
-confkey
) == 0)
173 AL_PRINT("Unknown speaker for %s: \"%c%c\"\n", name
, confkey
[0], confkey
[1]);
181 for(i
= 0;i
< chans
;i
++)
183 if(Speaker2Chan
[i
] == val
)
185 val
= strtol(sep
, NULL
, 10);
186 if(val
>= -180 && val
<= 180)
187 SpeakerAngle
[i
] = val
* M_PI
/180.0f
;
189 AL_PRINT("Invalid angle for speaker \"%c%c\": %d\n", confkey
[0], confkey
[1], val
);
195 for(i
= 1;i
< chans
;i
++)
197 if(SpeakerAngle
[i
] <= SpeakerAngle
[i
-1])
199 AL_PRINT("Speaker %d of %d does not follow previous: %f > %f\n", i
, chans
,
200 SpeakerAngle
[i
-1] * 180.0f
/M_PI
, SpeakerAngle
[i
] * 180.0f
/M_PI
);
201 SpeakerAngle
[i
] = SpeakerAngle
[i
-1] + 1 * 180.0f
/M_PI
;
206 static __inline ALfloat
aluLUTpos2Angle(ALint pos
)
208 if(pos
< QUADRANT_NUM
)
209 return aluAtan((ALfloat
)pos
/ (ALfloat
)(QUADRANT_NUM
- pos
));
210 if(pos
< 2 * QUADRANT_NUM
)
211 return M_PI_2
+ aluAtan((ALfloat
)(pos
- QUADRANT_NUM
) / (ALfloat
)(2 * QUADRANT_NUM
- pos
));
212 if(pos
< 3 * QUADRANT_NUM
)
213 return aluAtan((ALfloat
)(pos
- 2 * QUADRANT_NUM
) / (ALfloat
)(3 * QUADRANT_NUM
- pos
)) - M_PI
;
214 return aluAtan((ALfloat
)(pos
- 3 * QUADRANT_NUM
) / (ALfloat
)(4 * QUADRANT_NUM
- pos
)) - M_PI_2
;
217 ALvoid
aluInitPanning(ALCcontext
*Context
)
219 ALint pos
, offset
, s
;
220 ALfloat Alpha
, Theta
;
221 ALfloat SpeakerAngle
[OUTPUTCHANNELS
];
222 ALint Speaker2Chan
[OUTPUTCHANNELS
];
224 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
227 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
228 Context
->ChannelMatrix
[s
][s2
] = ((s
==s2
) ? 1.0f
: 0.0f
);
231 switch(Context
->Device
->Format
)
233 /* Mono is rendered as stereo, then downmixed during post-process */
234 case AL_FORMAT_MONO8
:
235 case AL_FORMAT_MONO16
:
236 case AL_FORMAT_MONO_FLOAT32
:
237 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
238 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
239 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
240 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
241 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
242 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
243 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
244 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
245 Context
->NumChan
= 2;
246 Speaker2Chan
[0] = FRONT_LEFT
;
247 Speaker2Chan
[1] = FRONT_RIGHT
;
248 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
249 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
252 case AL_FORMAT_STEREO8
:
253 case AL_FORMAT_STEREO16
:
254 case AL_FORMAT_STEREO_FLOAT32
:
255 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
256 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
257 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
258 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
259 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
260 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
261 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
262 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
263 Context
->NumChan
= 2;
264 Speaker2Chan
[0] = FRONT_LEFT
;
265 Speaker2Chan
[1] = FRONT_RIGHT
;
266 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
267 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
268 SetSpeakerArrangement("layout_STEREO", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
271 case AL_FORMAT_QUAD8
:
272 case AL_FORMAT_QUAD16
:
273 case AL_FORMAT_QUAD32
:
274 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
275 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
276 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
277 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
278 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
279 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
280 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
281 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
282 Context
->NumChan
= 4;
283 Speaker2Chan
[0] = BACK_LEFT
;
284 Speaker2Chan
[1] = FRONT_LEFT
;
285 Speaker2Chan
[2] = FRONT_RIGHT
;
286 Speaker2Chan
[3] = BACK_RIGHT
;
287 SpeakerAngle
[0] = -135.0f
* M_PI
/180.0f
;
288 SpeakerAngle
[1] = -45.0f
* M_PI
/180.0f
;
289 SpeakerAngle
[2] = 45.0f
* M_PI
/180.0f
;
290 SpeakerAngle
[3] = 135.0f
* M_PI
/180.0f
;
291 SetSpeakerArrangement("layout_QUAD", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
294 case AL_FORMAT_51CHN8
:
295 case AL_FORMAT_51CHN16
:
296 case AL_FORMAT_51CHN32
:
297 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
298 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
299 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
300 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
301 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
302 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
303 Context
->NumChan
= 5;
304 Speaker2Chan
[0] = BACK_LEFT
;
305 Speaker2Chan
[1] = FRONT_LEFT
;
306 Speaker2Chan
[2] = FRONT_CENTER
;
307 Speaker2Chan
[3] = FRONT_RIGHT
;
308 Speaker2Chan
[4] = BACK_RIGHT
;
309 SpeakerAngle
[0] = -110.0f
* M_PI
/180.0f
;
310 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
311 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
312 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
313 SpeakerAngle
[4] = 110.0f
* M_PI
/180.0f
;
314 SetSpeakerArrangement("layout_51CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
317 case AL_FORMAT_61CHN8
:
318 case AL_FORMAT_61CHN16
:
319 case AL_FORMAT_61CHN32
:
320 Context
->ChannelMatrix
[BACK_LEFT
][BACK_CENTER
] = aluSqrt(0.5);
321 Context
->ChannelMatrix
[BACK_LEFT
][SIDE_LEFT
] = aluSqrt(0.5);
322 Context
->ChannelMatrix
[BACK_RIGHT
][BACK_CENTER
] = aluSqrt(0.5);
323 Context
->ChannelMatrix
[BACK_RIGHT
][SIDE_RIGHT
] = aluSqrt(0.5);
324 Context
->NumChan
= 6;
325 Speaker2Chan
[0] = SIDE_LEFT
;
326 Speaker2Chan
[1] = FRONT_LEFT
;
327 Speaker2Chan
[2] = FRONT_CENTER
;
328 Speaker2Chan
[3] = FRONT_RIGHT
;
329 Speaker2Chan
[4] = SIDE_RIGHT
;
330 Speaker2Chan
[5] = BACK_CENTER
;
331 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
332 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
333 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
334 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
335 SpeakerAngle
[4] = 90.0f
* M_PI
/180.0f
;
336 SpeakerAngle
[5] = 180.0f
* M_PI
/180.0f
;
337 SetSpeakerArrangement("layout_61CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
340 case AL_FORMAT_71CHN8
:
341 case AL_FORMAT_71CHN16
:
342 case AL_FORMAT_71CHN32
:
343 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
344 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
345 Context
->NumChan
= 7;
346 Speaker2Chan
[0] = BACK_LEFT
;
347 Speaker2Chan
[1] = SIDE_LEFT
;
348 Speaker2Chan
[2] = FRONT_LEFT
;
349 Speaker2Chan
[3] = FRONT_CENTER
;
350 Speaker2Chan
[4] = FRONT_RIGHT
;
351 Speaker2Chan
[5] = SIDE_RIGHT
;
352 Speaker2Chan
[6] = BACK_RIGHT
;
353 SpeakerAngle
[0] = -150.0f
* M_PI
/180.0f
;
354 SpeakerAngle
[1] = -90.0f
* M_PI
/180.0f
;
355 SpeakerAngle
[2] = -30.0f
* M_PI
/180.0f
;
356 SpeakerAngle
[3] = 0.0f
* M_PI
/180.0f
;
357 SpeakerAngle
[4] = 30.0f
* M_PI
/180.0f
;
358 SpeakerAngle
[5] = 90.0f
* M_PI
/180.0f
;
359 SpeakerAngle
[6] = 150.0f
* M_PI
/180.0f
;
360 SetSpeakerArrangement("layout_71CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
367 for(pos
= 0; pos
< LUT_NUM
; pos
++)
370 Theta
= aluLUTpos2Angle(pos
);
372 /* clear all values */
373 offset
= OUTPUTCHANNELS
* pos
;
374 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
375 Context
->PanningLUT
[offset
+s
] = 0.0f
;
377 /* set panning values */
378 for(s
= 0; s
< Context
->NumChan
- 1; s
++)
380 if(Theta
>= SpeakerAngle
[s
] && Theta
< SpeakerAngle
[s
+1])
382 /* source between speaker s and speaker s+1 */
383 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
384 (SpeakerAngle
[s
+1]-SpeakerAngle
[s
]);
385 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
386 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
+1]] = sin(Alpha
);
390 if(s
== Context
->NumChan
- 1)
392 /* source between last and first speaker */
393 if(Theta
< SpeakerAngle
[0])
394 Theta
+= 2.0f
* M_PI
;
395 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
396 (2.0f
* M_PI
+ SpeakerAngle
[0]-SpeakerAngle
[s
]);
397 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
398 Context
->PanningLUT
[offset
+ Speaker2Chan
[0]] = sin(Alpha
);
403 static ALvoid
CalcSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
,
406 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,DryMix
;
407 ALfloat Direction
[3],Position
[3],SourceToListener
[3];
408 ALfloat Velocity
[3],ListenerVel
[3];
409 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
,OuterGainHF
;
410 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
411 ALfloat DopplerFactor
, DopplerVelocity
, flSpeedOfSound
;
412 ALfloat Matrix
[4][4];
413 ALfloat flAttenuation
;
414 ALfloat RoomAttenuation
[MAX_SENDS
];
415 ALfloat MetersPerUnit
;
416 ALfloat RoomRolloff
[MAX_SENDS
];
417 ALfloat DryGainHF
= 1.0f
;
418 ALfloat WetGain
[MAX_SENDS
];
419 ALfloat WetGainHF
[MAX_SENDS
];
420 ALfloat DirGain
, AmbientGain
;
422 const ALfloat
*SpeakerGain
;
428 for(i
= 0;i
< MAX_SENDS
;i
++)
431 //Get context properties
432 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
433 DopplerVelocity
= ALContext
->DopplerVelocity
;
434 flSpeedOfSound
= ALContext
->flSpeedOfSound
;
435 NumSends
= ALContext
->Device
->NumAuxSends
;
436 Frequency
= ALContext
->Device
->Frequency
;
438 //Get listener properties
439 ListenerGain
= ALContext
->Listener
.Gain
;
440 MetersPerUnit
= ALContext
->Listener
.MetersPerUnit
;
441 memcpy(ListenerVel
, ALContext
->Listener
.Velocity
, sizeof(ALContext
->Listener
.Velocity
));
443 //Get source properties
444 SourceVolume
= ALSource
->flGain
;
445 memcpy(Position
, ALSource
->vPosition
, sizeof(ALSource
->vPosition
));
446 memcpy(Direction
, ALSource
->vOrientation
, sizeof(ALSource
->vOrientation
));
447 memcpy(Velocity
, ALSource
->vVelocity
, sizeof(ALSource
->vVelocity
));
448 MinVolume
= ALSource
->flMinGain
;
449 MaxVolume
= ALSource
->flMaxGain
;
450 MinDist
= ALSource
->flRefDistance
;
451 MaxDist
= ALSource
->flMaxDistance
;
452 Rolloff
= ALSource
->flRollOffFactor
;
453 InnerAngle
= ALSource
->flInnerAngle
;
454 OuterAngle
= ALSource
->flOuterAngle
;
455 OuterGainHF
= ALSource
->OuterGainHF
;
457 //Only apply 3D calculations for mono buffers
458 if(isMono
== AL_FALSE
)
460 //1. Multi-channel buffers always play "normal"
461 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
463 DryMix
= SourceVolume
;
464 DryMix
= __min(DryMix
,MaxVolume
);
465 DryMix
= __max(DryMix
,MinVolume
);
467 switch(ALSource
->DirectFilter
.type
)
469 case AL_FILTER_LOWPASS
:
470 DryMix
*= ALSource
->DirectFilter
.Gain
;
471 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
475 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryMix
* ListenerGain
;
476 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryMix
* ListenerGain
;
477 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryMix
* ListenerGain
;
478 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryMix
* ListenerGain
;
479 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryMix
* ListenerGain
;
480 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryMix
* ListenerGain
;
481 ALSource
->Params
.DryGains
[FRONT_CENTER
] = DryMix
* ListenerGain
;
482 ALSource
->Params
.DryGains
[BACK_CENTER
] = DryMix
* ListenerGain
;
483 ALSource
->Params
.DryGains
[LFE
] = DryMix
* ListenerGain
;
485 for(i
= 0;i
< NumSends
;i
++)
487 WetGain
[i
] = SourceVolume
;
488 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
489 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
492 switch(ALSource
->Send
[i
].WetFilter
.type
)
494 case AL_FILTER_LOWPASS
:
495 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
496 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
500 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
502 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
504 ALSource
->Params
.WetGains
[i
] = 0.0f
;
508 /* Update filter coefficients. Calculations based on the I3DL2
510 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
511 /* We use two chained one-pole filters, so we need to take the
512 * square root of the squared gain, which is the same as the base
514 g
= __max(DryGainHF
, 0.01f
);
516 /* Be careful with gains < 0.0001, as that causes the coefficient
517 * head towards 1, which will flatten the signal */
518 if(g
< 0.9999f
) /* 1-epsilon */
519 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
521 ALSource
->Params
.iirFilter
.coeff
= a
;
523 for(i
= 0;i
< NumSends
;i
++)
525 /* We use a one-pole filter, so we need to take the squared gain */
526 g
= __max(WetGainHF
[i
], 0.1f
);
529 if(g
< 0.9999f
) /* 1-epsilon */
530 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
532 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
538 //1. Translate Listener to origin (convert to head relative)
539 if(ALSource
->bHeadRelative
==AL_FALSE
)
541 ALfloat U
[3],V
[3],N
[3],P
[3];
543 // Build transform matrix
544 memcpy(N
, ALContext
->Listener
.Forward
, sizeof(N
)); // At-vector
545 aluNormalize(N
); // Normalized At-vector
546 memcpy(V
, ALContext
->Listener
.Up
, sizeof(V
)); // Up-vector
547 aluNormalize(V
); // Normalized Up-vector
548 aluCrossproduct(N
, V
, U
); // Right-vector
549 aluNormalize(U
); // Normalized Right-vector
550 P
[0] = -(ALContext
->Listener
.Position
[0]*U
[0] + // Translation
551 ALContext
->Listener
.Position
[1]*U
[1] +
552 ALContext
->Listener
.Position
[2]*U
[2]);
553 P
[1] = -(ALContext
->Listener
.Position
[0]*V
[0] +
554 ALContext
->Listener
.Position
[1]*V
[1] +
555 ALContext
->Listener
.Position
[2]*V
[2]);
556 P
[2] = -(ALContext
->Listener
.Position
[0]*-N
[0] +
557 ALContext
->Listener
.Position
[1]*-N
[1] +
558 ALContext
->Listener
.Position
[2]*-N
[2]);
559 Matrix
[0][0] = U
[0]; Matrix
[0][1] = V
[0]; Matrix
[0][2] = -N
[0]; Matrix
[0][3] = 0.0f
;
560 Matrix
[1][0] = U
[1]; Matrix
[1][1] = V
[1]; Matrix
[1][2] = -N
[1]; Matrix
[1][3] = 0.0f
;
561 Matrix
[2][0] = U
[2]; Matrix
[2][1] = V
[2]; Matrix
[2][2] = -N
[2]; Matrix
[2][3] = 0.0f
;
562 Matrix
[3][0] = P
[0]; Matrix
[3][1] = P
[1]; Matrix
[3][2] = P
[2]; Matrix
[3][3] = 1.0f
;
564 // Transform source position and direction into listener space
565 aluMatrixVector(Position
, 1.0f
, Matrix
);
566 aluMatrixVector(Direction
, 0.0f
, Matrix
);
567 // Transform source and listener velocity into listener space
568 aluMatrixVector(Velocity
, 0.0f
, Matrix
);
569 aluMatrixVector(ListenerVel
, 0.0f
, Matrix
);
572 ListenerVel
[0] = ListenerVel
[1] = ListenerVel
[2] = 0.0f
;
574 SourceToListener
[0] = -Position
[0];
575 SourceToListener
[1] = -Position
[1];
576 SourceToListener
[2] = -Position
[2];
577 aluNormalize(SourceToListener
);
578 aluNormalize(Direction
);
580 //2. Calculate distance attenuation
581 Distance
= aluSqrt(aluDotproduct(Position
, Position
));
583 flAttenuation
= 1.0f
;
584 for(i
= 0;i
< MAX_SENDS
;i
++)
586 RoomAttenuation
[i
] = 1.0f
;
588 RoomRolloff
[i
] = ALSource
->RoomRolloffFactor
;
589 if(ALSource
->Send
[i
].Slot
&&
590 (ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
591 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
))
592 RoomRolloff
[i
] += ALSource
->Send
[i
].Slot
->effect
.Reverb
.RoomRolloffFactor
;
595 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
596 ALContext
->DistanceModel
)
598 case AL_INVERSE_DISTANCE_CLAMPED
:
599 Distance
=__max(Distance
,MinDist
);
600 Distance
=__min(Distance
,MaxDist
);
601 if(MaxDist
< MinDist
)
604 case AL_INVERSE_DISTANCE
:
607 if((MinDist
+ (Rolloff
* (Distance
- MinDist
))) > 0.0f
)
608 flAttenuation
= MinDist
/ (MinDist
+ (Rolloff
* (Distance
- MinDist
)));
609 for(i
= 0;i
< NumSends
;i
++)
611 if((MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
))) > 0.0f
)
612 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
)));
617 case AL_LINEAR_DISTANCE_CLAMPED
:
618 Distance
=__max(Distance
,MinDist
);
619 Distance
=__min(Distance
,MaxDist
);
620 if(MaxDist
< MinDist
)
623 case AL_LINEAR_DISTANCE
:
624 Distance
=__min(Distance
,MaxDist
);
625 if(MaxDist
!= MinDist
)
627 flAttenuation
= 1.0f
- (Rolloff
*(Distance
-MinDist
)/(MaxDist
- MinDist
));
628 for(i
= 0;i
< NumSends
;i
++)
629 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(Distance
-MinDist
)/(MaxDist
- MinDist
));
633 case AL_EXPONENT_DISTANCE_CLAMPED
:
634 Distance
=__max(Distance
,MinDist
);
635 Distance
=__min(Distance
,MaxDist
);
636 if(MaxDist
< MinDist
)
639 case AL_EXPONENT_DISTANCE
:
640 if(Distance
> 0.0f
&& MinDist
> 0.0f
)
642 flAttenuation
= (ALfloat
)pow(Distance
/MinDist
, -Rolloff
);
643 for(i
= 0;i
< NumSends
;i
++)
644 RoomAttenuation
[i
] = (ALfloat
)pow(Distance
/MinDist
, -RoomRolloff
[i
]);
652 // Source Gain + Attenuation
653 DryMix
= SourceVolume
* flAttenuation
;
654 for(i
= 0;i
< NumSends
;i
++)
655 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
657 // Distance-based air absorption
658 if(ALSource
->AirAbsorptionFactor
> 0.0f
&& flAttenuation
< 1.0f
)
660 ALfloat absorb
= 0.0f
;
662 // Absorption calculation is done in dB
663 if(flAttenuation
> 0.0f
)
665 absorb
= (MinDist
/flAttenuation
- MinDist
)*MetersPerUnit
*
666 (ALSource
->AirAbsorptionFactor
*AIRABSORBGAINDBHF
);
667 // Convert dB to linear gain before applying
668 absorb
= pow(10.0, absorb
/20.0);
671 for(i
= 0;i
< MAX_SENDS
;i
++)
672 WetGainHF
[i
] *= absorb
;
675 //3. Apply directional soundcones
676 Angle
= aluAcos(aluDotproduct(Direction
,SourceToListener
)) * 180.0f
/M_PI
;
677 if(Angle
>= InnerAngle
&& Angle
<= OuterAngle
)
679 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
680 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
)*scale
);
681 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
)*scale
);
682 DryMix
*= ConeVolume
;
683 if(ALSource
->DryGainHFAuto
)
686 else if(Angle
> OuterAngle
)
688 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
));
689 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
));
690 DryMix
*= ConeVolume
;
691 if(ALSource
->DryGainHFAuto
)
700 // Clamp to Min/Max Gain
701 DryMix
= __min(DryMix
,MaxVolume
);
702 DryMix
= __max(DryMix
,MinVolume
);
704 for(i
= 0;i
< NumSends
;i
++)
706 if(ALSource
->Send
[i
].Slot
&&
707 ALSource
->Send
[i
].Slot
->effect
.type
!= AL_EFFECT_NULL
)
709 if(ALSource
->Send
[i
].Slot
->AuxSendAuto
)
711 if(ALSource
->WetGainAuto
)
712 WetGain
[i
] *= ConeVolume
;
713 if(ALSource
->WetGainHFAuto
)
714 WetGainHF
[i
] *= ConeHF
;
716 // Clamp to Min/Max Gain
717 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
718 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
720 if(ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
721 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
)
723 /* Apply a decay-time transformation to the wet path,
724 * based on the attenuation of the dry path. This should
725 * better approximate the statistical attenuation model
726 * for the reverb effect.
728 * This simple equation converts the distance attenuation
729 * into the time it would take to reach -60 dB. From
730 * there it establishes an origin (0.333s; the decay time
731 * that will produce equal attenuation) and applies the
732 * current decay time. Finally, it converts the result
733 * back to an attenuation for the reverb path.
735 WetGain
[i
] *= pow(10.0f
, log10(flAttenuation
) * 0.333f
/
736 ALSource
->Send
[i
].Slot
->effect
.Reverb
.DecayTime
);
741 // If the slot's auxiliary send auto is off, the data sent to
742 // the effect slot is the same as the dry path, sans filter
745 WetGainHF
[i
] = DryGainHF
;
748 switch(ALSource
->Send
[i
].WetFilter
.type
)
750 case AL_FILTER_LOWPASS
:
751 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
752 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
755 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
759 ALSource
->Params
.WetGains
[i
] = 0.0f
;
763 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
765 ALSource
->Params
.WetGains
[i
] = 0.0f
;
769 // Apply filter gains and filters
770 switch(ALSource
->DirectFilter
.type
)
772 case AL_FILTER_LOWPASS
:
773 DryMix
*= ALSource
->DirectFilter
.Gain
;
774 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
777 DryMix
*= ListenerGain
;
779 // Calculate Velocity
780 if(DopplerFactor
!= 0.0f
)
782 ALfloat flVSS
, flVLS
;
783 ALfloat flMaxVelocity
= (DopplerVelocity
* flSpeedOfSound
) /
786 flVSS
= aluDotproduct(Velocity
, SourceToListener
);
787 if(flVSS
>= flMaxVelocity
)
788 flVSS
= (flMaxVelocity
- 1.0f
);
789 else if(flVSS
<= -flMaxVelocity
)
790 flVSS
= -flMaxVelocity
+ 1.0f
;
792 flVLS
= aluDotproduct(ListenerVel
, SourceToListener
);
793 if(flVLS
>= flMaxVelocity
)
794 flVLS
= (flMaxVelocity
- 1.0f
);
795 else if(flVLS
<= -flMaxVelocity
)
796 flVLS
= -flMaxVelocity
+ 1.0f
;
798 ALSource
->Params
.Pitch
= ALSource
->flPitch
*
799 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVLS
)) /
800 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVSS
));
803 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
805 // Use energy-preserving panning algorithm for multi-speaker playback
806 length
= aluSqrt(Position
[0]*Position
[0] + Position
[1]*Position
[1] +
807 Position
[2]*Position
[2]);
808 length
= __max(length
, MinDist
);
811 ALfloat invlen
= 1.0f
/length
;
812 Position
[0] *= invlen
;
813 Position
[1] *= invlen
;
814 Position
[2] *= invlen
;
817 pos
= aluCart2LUTpos(-Position
[2], Position
[0]);
818 SpeakerGain
= &ALContext
->PanningLUT
[OUTPUTCHANNELS
* pos
];
820 DirGain
= aluSqrt(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
821 // elevation adjustment for directional gain. this sucks, but
822 // has low complexity
823 AmbientGain
= 1.0/aluSqrt(ALContext
->NumChan
) * (1.0-DirGain
);
824 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
826 ALfloat gain
= SpeakerGain
[s
]*DirGain
+ AmbientGain
;
827 ALSource
->Params
.DryGains
[s
] = DryMix
* gain
;
830 /* Update filter coefficients. */
831 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
832 /* Spatialized sources use four chained one-pole filters, so we need to
833 * take the fourth root of the squared gain, which is the same as the
834 * square root of the base gain. */
835 g
= aluSqrt(__max(DryGainHF
, 0.0001f
));
837 if(g
< 0.9999f
) /* 1-epsilon */
838 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
840 ALSource
->Params
.iirFilter
.coeff
= a
;
842 for(i
= 0;i
< NumSends
;i
++)
844 /* The wet path uses two chained one-pole filters, so take the
845 * base gain (square root of the squared gain) */
846 g
= __max(WetGainHF
[i
], 0.01f
);
848 if(g
< 0.9999f
) /* 1-epsilon */
849 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
851 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
855 static __inline ALshort
lerp(ALshort val1
, ALshort val2
, ALint frac
)
857 return val1
+ (((val2
-val1
)*frac
)>>FRACTIONBITS
);
860 static void MixSomeSources(ALCcontext
*ALContext
, float (*DryBuffer
)[OUTPUTCHANNELS
], ALuint SamplesToDo
)
862 static float DummyBuffer
[BUFFERSIZE
];
863 ALfloat
*WetBuffer
[MAX_SENDS
];
864 ALfloat (*Matrix
)[OUTPUTCHANNELS
] = ALContext
->ChannelMatrix
;
865 ALfloat DrySend
[OUTPUTCHANNELS
];
866 ALfloat dryGainStep
[OUTPUTCHANNELS
];
867 ALfloat wetGainStep
[MAX_SENDS
];
870 ALfloat value
, outsamp
;
871 ALbufferlistitem
*BufferListItem
;
872 ALint64 DataSize64
,DataPos64
;
873 FILTER
*DryFilter
, *WetFilter
[MAX_SENDS
];
874 ALfloat WetSend
[MAX_SENDS
];
878 ALuint DataPosInt
, DataPosFrac
;
879 ALuint Channels
, Bytes
;
881 ALuint BuffersPlayed
;
885 if(!(ALSource
=ALContext
->Source
))
888 DeviceFreq
= ALContext
->Device
->Frequency
;
890 rampLength
= DeviceFreq
* MIN_RAMP_LENGTH
/ 1000;
891 rampLength
= max(rampLength
, SamplesToDo
);
894 State
= ALSource
->state
;
895 if(State
!= AL_PLAYING
)
897 if((ALSource
=ALSource
->next
) != NULL
)
903 /* Find buffer format */
907 BufferListItem
= ALSource
->queue
;
908 while(BufferListItem
!= NULL
)
911 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
913 Channels
= aluChannelsFromFormat(ALBuffer
->format
);
914 Bytes
= aluBytesFromFormat(ALBuffer
->format
);
915 Frequency
= ALBuffer
->frequency
;
918 BufferListItem
= BufferListItem
->next
;
921 /* Get source info */
922 BuffersPlayed
= ALSource
->BuffersPlayed
;
923 DataPosInt
= ALSource
->position
;
924 DataPosFrac
= ALSource
->position_fraction
;
926 if(ALSource
->NeedsUpdate
)
928 CalcSourceParams(ALContext
, ALSource
, (Channels
==1)?AL_TRUE
:AL_FALSE
);
929 ALSource
->NeedsUpdate
= AL_FALSE
;
932 /* Compute 18.14 fixed point step */
933 Pitch
= (ALSource
->Params
.Pitch
*Frequency
) / DeviceFreq
;
934 if(Pitch
> (float)MAX_PITCH
) Pitch
= (float)MAX_PITCH
;
935 increment
= (ALint
)(Pitch
*(ALfloat
)(1L<<FRACTIONBITS
));
936 if(increment
<= 0) increment
= (1<<FRACTIONBITS
);
938 /* Compute the gain steps for each output channel */
939 if(ALSource
->FirstStart
)
941 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
942 DrySend
[i
] = ALSource
->Params
.DryGains
[i
];
943 for(i
= 0;i
< MAX_SENDS
;i
++)
944 WetSend
[i
] = ALSource
->Params
.WetGains
[i
];
948 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
949 DrySend
[i
] = ALSource
->DryGains
[i
];
950 for(i
= 0;i
< MAX_SENDS
;i
++)
951 WetSend
[i
] = ALSource
->WetGains
[i
];
954 DryFilter
= &ALSource
->Params
.iirFilter
;
955 for(i
= 0;i
< MAX_SENDS
;i
++)
957 WetFilter
[i
] = &ALSource
->Params
.Send
[i
].iirFilter
;
958 WetBuffer
[i
] = (ALSource
->Send
[i
].Slot
?
959 ALSource
->Send
[i
].Slot
->WetBuffer
:
963 if(DuplicateStereo
&& Channels
== 2)
965 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 1.0f
;
966 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 1.0f
;
967 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 1.0f
;
968 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 1.0f
;
970 else if(DuplicateStereo
)
972 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 0.0f
;
973 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 0.0f
;
974 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 0.0f
;
975 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 0.0f
;
978 /* Get current buffer queue item */
979 BufferListItem
= ALSource
->queue
;
980 for(i
= 0;i
< BuffersPlayed
&& BufferListItem
;i
++)
981 BufferListItem
= BufferListItem
->next
;
983 while(State
== AL_PLAYING
&& j
< SamplesToDo
)
990 /* Get buffer info */
991 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
993 Data
= ALBuffer
->data
;
994 DataSize
= ALBuffer
->size
;
995 DataSize
/= Channels
* Bytes
;
997 if(DataPosInt
>= DataSize
)
1000 if(BufferListItem
->next
)
1002 ALbuffer
*NextBuf
= BufferListItem
->next
->buffer
;
1003 if(NextBuf
&& NextBuf
->data
)
1005 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1006 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1007 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1010 else if(ALSource
->bLooping
)
1012 ALbuffer
*NextBuf
= ALSource
->queue
->buffer
;
1013 if(NextBuf
&& NextBuf
->data
)
1015 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1016 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1017 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1021 memset(&Data
[DataSize
*Channels
], 0, (BUFFER_PADDING
*Channels
*Bytes
));
1023 /* Compute the gain steps for each output channel */
1024 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1025 dryGainStep
[i
] = (ALSource
->Params
.DryGains
[i
]-
1026 DrySend
[i
]) / rampLength
;
1027 for(i
= 0;i
< MAX_SENDS
;i
++)
1028 wetGainStep
[i
] = (ALSource
->Params
.WetGains
[i
]-
1029 WetSend
[i
]) / rampLength
;
1031 /* Figure out how many samples we can mix. */
1032 DataSize64
= DataSize
;
1033 DataSize64
<<= FRACTIONBITS
;
1034 DataPos64
= DataPosInt
;
1035 DataPos64
<<= FRACTIONBITS
;
1036 DataPos64
+= DataPosFrac
;
1037 BufferSize
= (ALuint
)((DataSize64
-DataPos64
+(increment
-1)) / increment
);
1039 BufferSize
= min(BufferSize
, (SamplesToDo
-j
));
1041 /* Actual sample mixing loop */
1043 Data
+= DataPosInt
*Channels
;
1045 if(Channels
== 1) /* Mono */
1049 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1050 DrySend
[i
] += dryGainStep
[i
];
1051 for(i
= 0;i
< MAX_SENDS
;i
++)
1052 WetSend
[i
] += wetGainStep
[i
];
1054 /* First order interpolator */
1055 value
= lerp(Data
[k
], Data
[k
+1], DataPosFrac
);
1057 /* Direct path final mix buffer and panning */
1058 outsamp
= lpFilter4P(DryFilter
, 0, value
);
1059 DryBuffer
[j
][FRONT_LEFT
] += outsamp
*DrySend
[FRONT_LEFT
];
1060 DryBuffer
[j
][FRONT_RIGHT
] += outsamp
*DrySend
[FRONT_RIGHT
];
1061 DryBuffer
[j
][SIDE_LEFT
] += outsamp
*DrySend
[SIDE_LEFT
];
1062 DryBuffer
[j
][SIDE_RIGHT
] += outsamp
*DrySend
[SIDE_RIGHT
];
1063 DryBuffer
[j
][BACK_LEFT
] += outsamp
*DrySend
[BACK_LEFT
];
1064 DryBuffer
[j
][BACK_RIGHT
] += outsamp
*DrySend
[BACK_RIGHT
];
1065 DryBuffer
[j
][FRONT_CENTER
] += outsamp
*DrySend
[FRONT_CENTER
];
1066 DryBuffer
[j
][BACK_CENTER
] += outsamp
*DrySend
[BACK_CENTER
];
1068 /* Room path final mix buffer and panning */
1069 for(i
= 0;i
< MAX_SENDS
;i
++)
1071 outsamp
= lpFilter2P(WetFilter
[i
], 0, value
);
1072 WetBuffer
[i
][j
] += outsamp
*WetSend
[i
];
1075 DataPosFrac
+= increment
;
1076 k
+= DataPosFrac
>>FRACTIONBITS
;
1077 DataPosFrac
&= FRACTIONMASK
;
1081 else if(Channels
== 2) /* Stereo */
1083 const int chans
[] = {
1084 FRONT_LEFT
, FRONT_RIGHT
1086 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1088 #define DO_MIX() do { \
1089 while(BufferSize--) \
1091 for(i = 0;i < OUTPUTCHANNELS;i++) \
1092 DrySend[i] += dryGainStep[i]; \
1093 for(i = 0;i < MAX_SENDS;i++) \
1094 WetSend[i] += wetGainStep[i]; \
1096 for(i = 0;i < Channels;i++) \
1098 value = lerp(Data[k*Channels + i], Data[(k+1)*Channels + i], DataPosFrac); \
1099 outsamp = lpFilter2P(DryFilter, chans[i]*2, value)*DrySend[chans[i]]; \
1100 for(out = 0;out < OUTPUTCHANNELS;out++) \
1101 DryBuffer[j][out] += outsamp*Matrix[chans[i]][out]; \
1102 for(out = 0;out < MAX_SENDS;out++) \
1104 outsamp = lpFilter1P(WetFilter[out], chans[out], value); \
1105 WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
1109 DataPosFrac += increment; \
1110 k += DataPosFrac>>FRACTIONBITS; \
1111 DataPosFrac &= FRACTIONMASK; \
1118 else if(Channels
== 4) /* Quad */
1120 const int chans
[] = {
1121 FRONT_LEFT
, FRONT_RIGHT
,
1122 BACK_LEFT
, BACK_RIGHT
1124 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1128 else if(Channels
== 6) /* 5.1 */
1130 const int chans
[] = {
1131 FRONT_LEFT
, FRONT_RIGHT
,
1133 BACK_LEFT
, BACK_RIGHT
1135 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1139 else if(Channels
== 7) /* 6.1 */
1141 const int chans
[] = {
1142 FRONT_LEFT
, FRONT_RIGHT
,
1145 SIDE_LEFT
, SIDE_RIGHT
1147 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1151 else if(Channels
== 8) /* 7.1 */
1153 const int chans
[] = {
1154 FRONT_LEFT
, FRONT_RIGHT
,
1156 BACK_LEFT
, BACK_RIGHT
,
1157 SIDE_LEFT
, SIDE_RIGHT
1159 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1166 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1167 DrySend
[i
] += dryGainStep
[i
]*BufferSize
;
1168 for(i
= 0;i
< MAX_SENDS
;i
++)
1169 WetSend
[i
] += wetGainStep
[i
]*BufferSize
;
1172 DataPosFrac
+= increment
;
1173 k
+= DataPosFrac
>>FRACTIONBITS
;
1174 DataPosFrac
&= FRACTIONMASK
;
1181 /* Handle looping sources */
1182 if(DataPosInt
>= DataSize
)
1184 if(BuffersPlayed
< (ALSource
->BuffersInQueue
-1))
1186 BufferListItem
= BufferListItem
->next
;
1188 DataPosInt
-= DataSize
;
1192 if(!ALSource
->bLooping
)
1195 BufferListItem
= ALSource
->queue
;
1196 BuffersPlayed
= ALSource
->BuffersInQueue
;
1202 BufferListItem
= ALSource
->queue
;
1204 if(ALSource
->BuffersInQueue
== 1)
1205 DataPosInt
%= DataSize
;
1207 DataPosInt
-= DataSize
;
1213 /* Update source info */
1214 ALSource
->state
= State
;
1215 ALSource
->BuffersPlayed
= BuffersPlayed
;
1216 ALSource
->position
= DataPosInt
;
1217 ALSource
->position_fraction
= DataPosFrac
;
1218 ALSource
->Buffer
= BufferListItem
->buffer
;
1220 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1221 ALSource
->DryGains
[i
] = DrySend
[i
];
1222 for(i
= 0;i
< MAX_SENDS
;i
++)
1223 ALSource
->WetGains
[i
] = WetSend
[i
];
1225 ALSource
->FirstStart
= AL_FALSE
;
1227 if((ALSource
=ALSource
->next
) != NULL
)
1228 goto another_source
;
1231 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1233 float (*DryBuffer
)[OUTPUTCHANNELS
];
1235 ALeffectslot
*ALEffectSlot
;
1236 ALCcontext
*ALContext
;
1240 SuspendContext(NULL
);
1242 #if defined(HAVE_FESETROUND)
1243 fpuState
= fegetround();
1244 fesetround(FE_TOWARDZERO
);
1245 #elif defined(HAVE__CONTROLFP)
1246 fpuState
= _controlfp(0, 0);
1247 _controlfp(_RC_CHOP
, _MCW_RC
);
1252 DryBuffer
= device
->DryBuffer
;
1255 /* Setup variables */
1256 SamplesToDo
= min(size
, BUFFERSIZE
);
1258 /* Clear mixing buffer */
1259 memset(DryBuffer
, 0, SamplesToDo
*OUTPUTCHANNELS
*sizeof(ALfloat
));
1261 for(c
= 0;c
< device
->NumContexts
;c
++)
1263 ALContext
= device
->Contexts
[c
];
1264 SuspendContext(ALContext
);
1266 MixSomeSources(ALContext
, DryBuffer
, SamplesToDo
);
1268 /* effect slot processing */
1269 ALEffectSlot
= ALContext
->AuxiliaryEffectSlot
;
1272 if(ALEffectSlot
->EffectState
)
1273 ALEffect_Process(ALEffectSlot
->EffectState
, ALEffectSlot
, SamplesToDo
, ALEffectSlot
->WetBuffer
, DryBuffer
);
1275 for(i
= 0;i
< SamplesToDo
;i
++)
1276 ALEffectSlot
->WetBuffer
[i
] = 0.0f
;
1277 ALEffectSlot
= ALEffectSlot
->next
;
1279 ProcessContext(ALContext
);
1282 //Post processing loop
1283 switch(device
->Format
)
1285 #define CHECK_WRITE_FORMAT(bits, type, func, isWin) \
1286 case AL_FORMAT_MONO##bits: \
1287 for(i = 0;i < SamplesToDo;i++) \
1289 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT] + \
1290 DryBuffer[i][FRONT_RIGHT]); \
1291 buffer = ((type*)buffer) + 1; \
1294 case AL_FORMAT_STEREO##bits: \
1297 for(i = 0;i < SamplesToDo;i++) \
1300 samples[0] = DryBuffer[i][FRONT_LEFT]; \
1301 samples[1] = DryBuffer[i][FRONT_RIGHT]; \
1302 bs2b_cross_feed(device->Bs2b, samples); \
1303 ((type*)buffer)[0] = (func)(samples[0]); \
1304 ((type*)buffer)[1] = (func)(samples[1]); \
1305 buffer = ((type*)buffer) + 2; \
1310 for(i = 0;i < SamplesToDo;i++) \
1312 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1313 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1314 buffer = ((type*)buffer) + 2; \
1318 case AL_FORMAT_QUAD##bits: \
1319 for(i = 0;i < SamplesToDo;i++) \
1321 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1322 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1323 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1324 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1325 buffer = ((type*)buffer) + 4; \
1328 case AL_FORMAT_51CHN##bits: \
1329 for(i = 0;i < SamplesToDo;i++) \
1331 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1332 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1334 /* Of course, Windows can't use the same ordering... */ \
1335 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1336 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1337 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_LEFT]); \
1338 ((type*)buffer)[5] = (func)(DryBuffer[i][BACK_RIGHT]); \
1340 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1341 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1342 ((type*)buffer)[4] = (func)(DryBuffer[i][FRONT_CENTER]); \
1343 ((type*)buffer)[5] = (func)(DryBuffer[i][LFE]); \
1345 buffer = ((type*)buffer) + 6; \
1348 case AL_FORMAT_61CHN##bits: \
1349 for(i = 0;i < SamplesToDo;i++) \
1351 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1352 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1353 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1354 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1355 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_CENTER]); \
1356 ((type*)buffer)[5] = (func)(DryBuffer[i][SIDE_LEFT]); \
1357 ((type*)buffer)[6] = (func)(DryBuffer[i][SIDE_RIGHT]); \
1358 buffer = ((type*)buffer) + 7; \
1361 case AL_FORMAT_71CHN##bits: \
1362 for(i = 0;i < SamplesToDo;i++) \
1364 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1365 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1367 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1368 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1369 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_LEFT]); \
1370 ((type*)buffer)[5] = (func)(DryBuffer[i][BACK_RIGHT]); \
1372 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1373 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1374 ((type*)buffer)[4] = (func)(DryBuffer[i][FRONT_CENTER]); \
1375 ((type*)buffer)[5] = (func)(DryBuffer[i][LFE]); \
1377 ((type*)buffer)[6] = (func)(DryBuffer[i][SIDE_LEFT]); \
1378 ((type*)buffer)[7] = (func)(DryBuffer[i][SIDE_RIGHT]); \
1379 buffer = ((type*)buffer) + 8; \
1383 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1384 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1386 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
, 1)
1387 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
, 1)
1388 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
, 1)
1390 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
, 0)
1391 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
, 0)
1392 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
, 0)
1394 #undef AL_FORMAT_STEREO32
1395 #undef AL_FORMAT_MONO32
1396 #undef CHECK_WRITE_FORMAT
1402 size
-= SamplesToDo
;
1405 #if defined(HAVE_FESETROUND)
1406 fesetround(fpuState
);
1407 #elif defined(HAVE__CONTROLFP)
1408 _controlfp(fpuState
, 0xfffff);
1411 ProcessContext(NULL
);
1414 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1418 SuspendContext(NULL
);
1419 for(i
= 0;i
< device
->NumContexts
;i
++)
1423 SuspendContext(device
->Contexts
[i
]);
1425 source
= device
->Contexts
[i
]->Source
;
1428 if(source
->state
== AL_PLAYING
)
1430 source
->state
= AL_STOPPED
;
1431 source
->BuffersPlayed
= source
->BuffersInQueue
;
1432 source
->position
= 0;
1433 source
->position_fraction
= 0;
1435 source
= source
->next
;
1437 ProcessContext(device
->Contexts
[i
]);
1440 device
->Connected
= ALC_FALSE
;
1441 ProcessContext(NULL
);