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 #define FRACTIONBITS 14
41 #define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
42 #define MAX_PITCH 65536
44 /* Minimum ramp length in milliseconds. The value below was chosen to
45 * adequately reduce clicks and pops from harsh gain changes. */
46 #define MIN_RAMP_LENGTH 16
48 ALboolean DuplicateStereo
= AL_FALSE
;
51 static __inline ALfloat
aluF2F(ALfloat Value
)
56 static __inline ALshort
aluF2S(ALfloat Value
)
62 i
= (ALint
)(Value
*32768.0f
);
67 i
= (ALint
)(Value
*32767.0f
);
73 static __inline ALubyte
aluF2UB(ALfloat Value
)
75 ALshort i
= aluF2S(Value
);
80 static __inline ALvoid
aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
82 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
83 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
84 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
87 static __inline ALfloat
aluDotproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
)
89 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
90 inVector1
[2]*inVector2
[2];
93 static __inline ALvoid
aluNormalize(ALfloat
*inVector
)
95 ALfloat length
, inverse_length
;
97 length
= aluSqrt(aluDotproduct(inVector
, inVector
));
100 inverse_length
= 1.0f
/length
;
101 inVector
[0] *= inverse_length
;
102 inVector
[1] *= inverse_length
;
103 inVector
[2] *= inverse_length
;
107 static __inline ALvoid
aluMatrixVector(ALfloat
*vector
,ALfloat w
,ALfloat matrix
[4][4])
110 vector
[0], vector
[1], vector
[2], w
113 vector
[0] = temp
[0]*matrix
[0][0] + temp
[1]*matrix
[1][0] + temp
[2]*matrix
[2][0] + temp
[3]*matrix
[3][0];
114 vector
[1] = temp
[0]*matrix
[0][1] + temp
[1]*matrix
[1][1] + temp
[2]*matrix
[2][1] + temp
[3]*matrix
[3][1];
115 vector
[2] = temp
[0]*matrix
[0][2] + temp
[1]*matrix
[1][2] + temp
[2]*matrix
[2][2] + temp
[3]*matrix
[3][2];
118 static ALvoid
SetSpeakerArrangement(const char *name
, ALfloat SpeakerAngle
[OUTPUTCHANNELS
],
119 ALint Speaker2Chan
[OUTPUTCHANNELS
], ALint chans
)
121 char layout_str
[256];
122 char *confkey
, *next
;
126 strncpy(layout_str
, GetConfigValue(NULL
, name
, ""), sizeof(layout_str
));
129 next
= confkey
= layout_str
;
133 next
= strchr(confkey
, ',');
139 } while(isspace(*next
) || *next
== ',');
142 sep
= strchr(confkey
, '=');
143 if(!sep
|| confkey
== sep
)
147 while(isspace(*end
) && end
!= confkey
)
151 if(strcmp(confkey
, "fl") == 0 || strcmp(confkey
, "front-left") == 0)
153 else if(strcmp(confkey
, "fr") == 0 || strcmp(confkey
, "front-right") == 0)
155 else if(strcmp(confkey
, "fc") == 0 || strcmp(confkey
, "front-center") == 0)
157 else if(strcmp(confkey
, "bl") == 0 || strcmp(confkey
, "back-left") == 0)
159 else if(strcmp(confkey
, "br") == 0 || strcmp(confkey
, "back-right") == 0)
161 else if(strcmp(confkey
, "bc") == 0 || strcmp(confkey
, "back-center") == 0)
163 else if(strcmp(confkey
, "sl") == 0 || strcmp(confkey
, "side-left") == 0)
165 else if(strcmp(confkey
, "sr") == 0 || strcmp(confkey
, "side-right") == 0)
169 AL_PRINT("Unknown speaker for %s: \"%s\"\n", name
, confkey
);
177 for(i
= 0;i
< chans
;i
++)
179 if(Speaker2Chan
[i
] == val
)
181 val
= strtol(sep
, NULL
, 10);
182 if(val
>= -180 && val
<= 180)
183 SpeakerAngle
[i
] = val
* M_PI
/180.0f
;
185 AL_PRINT("Invalid angle for speaker \"%s\": %d\n", confkey
, val
);
191 for(i
= 0;i
< chans
;i
++)
196 for(i2
= i
+1;i2
< chans
;i2
++)
198 if(SpeakerAngle
[i2
] < SpeakerAngle
[min
])
207 tmpf
= SpeakerAngle
[i
];
208 SpeakerAngle
[i
] = SpeakerAngle
[min
];
209 SpeakerAngle
[min
] = tmpf
;
211 tmpi
= Speaker2Chan
[i
];
212 Speaker2Chan
[i
] = Speaker2Chan
[min
];
213 Speaker2Chan
[min
] = tmpi
;
218 static __inline ALfloat
aluLUTpos2Angle(ALint pos
)
220 if(pos
< QUADRANT_NUM
)
221 return aluAtan((ALfloat
)pos
/ (ALfloat
)(QUADRANT_NUM
- pos
));
222 if(pos
< 2 * QUADRANT_NUM
)
223 return M_PI_2
+ aluAtan((ALfloat
)(pos
- QUADRANT_NUM
) / (ALfloat
)(2 * QUADRANT_NUM
- pos
));
224 if(pos
< 3 * QUADRANT_NUM
)
225 return aluAtan((ALfloat
)(pos
- 2 * QUADRANT_NUM
) / (ALfloat
)(3 * QUADRANT_NUM
- pos
)) - M_PI
;
226 return aluAtan((ALfloat
)(pos
- 3 * QUADRANT_NUM
) / (ALfloat
)(4 * QUADRANT_NUM
- pos
)) - M_PI_2
;
229 ALvoid
aluInitPanning(ALCcontext
*Context
)
231 ALint pos
, offset
, s
;
232 ALfloat Alpha
, Theta
;
233 ALfloat SpeakerAngle
[OUTPUTCHANNELS
];
234 ALint Speaker2Chan
[OUTPUTCHANNELS
];
236 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
239 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
240 Context
->ChannelMatrix
[s
][s2
] = ((s
==s2
) ? 1.0f
: 0.0f
);
243 switch(Context
->Device
->Format
)
245 case AL_FORMAT_MONO8
:
246 case AL_FORMAT_MONO16
:
247 case AL_FORMAT_MONO_FLOAT32
:
248 Context
->ChannelMatrix
[FRONT_LEFT
][FRONT_CENTER
] = aluSqrt(0.5);
249 Context
->ChannelMatrix
[FRONT_RIGHT
][FRONT_CENTER
] = aluSqrt(0.5);
250 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_CENTER
] = aluSqrt(0.5);
251 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_CENTER
] = aluSqrt(0.5);
252 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_CENTER
] = aluSqrt(0.5);
253 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_CENTER
] = aluSqrt(0.5);
254 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_CENTER
] = 1.0f
;
255 Context
->NumChan
= 1;
256 Speaker2Chan
[0] = FRONT_CENTER
;
257 SpeakerAngle
[0] = 0.0f
* M_PI
/180.0f
;
260 case AL_FORMAT_STEREO8
:
261 case AL_FORMAT_STEREO16
:
262 case AL_FORMAT_STEREO_FLOAT32
:
263 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
264 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
265 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
266 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
267 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
268 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
269 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
270 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
271 Context
->NumChan
= 2;
272 Speaker2Chan
[0] = FRONT_LEFT
;
273 Speaker2Chan
[1] = FRONT_RIGHT
;
274 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
275 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
276 SetSpeakerArrangement("layout_STEREO", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
279 case AL_FORMAT_QUAD8
:
280 case AL_FORMAT_QUAD16
:
281 case AL_FORMAT_QUAD32
:
282 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
283 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
284 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
285 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
286 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
287 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
288 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
289 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
290 Context
->NumChan
= 4;
291 Speaker2Chan
[0] = BACK_LEFT
;
292 Speaker2Chan
[1] = FRONT_LEFT
;
293 Speaker2Chan
[2] = FRONT_RIGHT
;
294 Speaker2Chan
[3] = BACK_RIGHT
;
295 SpeakerAngle
[0] = -135.0f
* M_PI
/180.0f
;
296 SpeakerAngle
[1] = -45.0f
* M_PI
/180.0f
;
297 SpeakerAngle
[2] = 45.0f
* M_PI
/180.0f
;
298 SpeakerAngle
[3] = 135.0f
* M_PI
/180.0f
;
299 SetSpeakerArrangement("layout_QUAD", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
302 case AL_FORMAT_51CHN8
:
303 case AL_FORMAT_51CHN16
:
304 case AL_FORMAT_51CHN32
:
305 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
306 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
307 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
308 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
309 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
310 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
311 Context
->NumChan
= 5;
312 Speaker2Chan
[0] = BACK_LEFT
;
313 Speaker2Chan
[1] = FRONT_LEFT
;
314 Speaker2Chan
[2] = FRONT_CENTER
;
315 Speaker2Chan
[3] = FRONT_RIGHT
;
316 Speaker2Chan
[4] = BACK_RIGHT
;
317 SpeakerAngle
[0] = -110.0f
* M_PI
/180.0f
;
318 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
319 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
320 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
321 SpeakerAngle
[4] = 110.0f
* M_PI
/180.0f
;
322 SetSpeakerArrangement("layout_51CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
325 case AL_FORMAT_61CHN8
:
326 case AL_FORMAT_61CHN16
:
327 case AL_FORMAT_61CHN32
:
328 Context
->ChannelMatrix
[BACK_LEFT
][BACK_CENTER
] = aluSqrt(0.5);
329 Context
->ChannelMatrix
[BACK_LEFT
][SIDE_LEFT
] = aluSqrt(0.5);
330 Context
->ChannelMatrix
[BACK_RIGHT
][BACK_CENTER
] = aluSqrt(0.5);
331 Context
->ChannelMatrix
[BACK_RIGHT
][SIDE_RIGHT
] = aluSqrt(0.5);
332 Context
->NumChan
= 6;
333 Speaker2Chan
[0] = SIDE_LEFT
;
334 Speaker2Chan
[1] = FRONT_LEFT
;
335 Speaker2Chan
[2] = FRONT_CENTER
;
336 Speaker2Chan
[3] = FRONT_RIGHT
;
337 Speaker2Chan
[4] = SIDE_RIGHT
;
338 Speaker2Chan
[5] = BACK_CENTER
;
339 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
340 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
341 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
342 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
343 SpeakerAngle
[4] = 90.0f
* M_PI
/180.0f
;
344 SpeakerAngle
[5] = 180.0f
* M_PI
/180.0f
;
345 SetSpeakerArrangement("layout_61CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
348 case AL_FORMAT_71CHN8
:
349 case AL_FORMAT_71CHN16
:
350 case AL_FORMAT_71CHN32
:
351 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
352 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
353 Context
->NumChan
= 7;
354 Speaker2Chan
[0] = BACK_LEFT
;
355 Speaker2Chan
[1] = SIDE_LEFT
;
356 Speaker2Chan
[2] = FRONT_LEFT
;
357 Speaker2Chan
[3] = FRONT_CENTER
;
358 Speaker2Chan
[4] = FRONT_RIGHT
;
359 Speaker2Chan
[5] = SIDE_RIGHT
;
360 Speaker2Chan
[6] = BACK_RIGHT
;
361 SpeakerAngle
[0] = -150.0f
* M_PI
/180.0f
;
362 SpeakerAngle
[1] = -90.0f
* M_PI
/180.0f
;
363 SpeakerAngle
[2] = -30.0f
* M_PI
/180.0f
;
364 SpeakerAngle
[3] = 0.0f
* M_PI
/180.0f
;
365 SpeakerAngle
[4] = 30.0f
* M_PI
/180.0f
;
366 SpeakerAngle
[5] = 90.0f
* M_PI
/180.0f
;
367 SpeakerAngle
[6] = 150.0f
* M_PI
/180.0f
;
368 SetSpeakerArrangement("layout_71CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
375 for(pos
= 0; pos
< LUT_NUM
; pos
++)
377 /* clear all values */
378 offset
= OUTPUTCHANNELS
* pos
;
379 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
380 Context
->PanningLUT
[offset
+s
] = 0.0f
;
382 if(Context
->NumChan
== 1)
384 Context
->PanningLUT
[offset
+ Speaker2Chan
[0]] = 1.0f
;
389 Theta
= aluLUTpos2Angle(pos
);
391 /* set panning values */
392 for(s
= 0; s
< Context
->NumChan
- 1; s
++)
394 if(Theta
>= SpeakerAngle
[s
] && Theta
< SpeakerAngle
[s
+1])
396 /* source between speaker s and speaker s+1 */
397 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
398 (SpeakerAngle
[s
+1]-SpeakerAngle
[s
]);
399 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
400 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
+1]] = sin(Alpha
);
404 if(s
== Context
->NumChan
- 1)
406 /* source between last and first speaker */
407 if(Theta
< SpeakerAngle
[0])
408 Theta
+= 2.0f
* M_PI
;
409 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
410 (2.0f
* M_PI
+ SpeakerAngle
[0]-SpeakerAngle
[s
]);
411 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
412 Context
->PanningLUT
[offset
+ Speaker2Chan
[0]] = sin(Alpha
);
417 static ALvoid
CalcNonAttnSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
)
419 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
420 ALfloat DryGain
, DryGainHF
;
421 ALfloat WetGain
[MAX_SENDS
];
422 ALfloat WetGainHF
[MAX_SENDS
];
423 ALint NumSends
, Frequency
;
427 //Get context properties
428 NumSends
= ALContext
->Device
->NumAuxSends
;
429 Frequency
= ALContext
->Device
->Frequency
;
431 //Get listener properties
432 ListenerGain
= ALContext
->Listener
.Gain
;
434 //Get source properties
435 SourceVolume
= ALSource
->flGain
;
436 MinVolume
= ALSource
->flMinGain
;
437 MaxVolume
= ALSource
->flMaxGain
;
439 //1. Multi-channel buffers always play "normal"
440 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
442 DryGain
= SourceVolume
;
443 DryGain
= __min(DryGain
,MaxVolume
);
444 DryGain
= __max(DryGain
,MinVolume
);
447 switch(ALSource
->DirectFilter
.type
)
449 case AL_FILTER_LOWPASS
:
450 DryGain
*= ALSource
->DirectFilter
.Gain
;
451 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
455 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
456 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
457 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryGain
* ListenerGain
;
458 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryGain
* ListenerGain
;
459 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryGain
* ListenerGain
;
460 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryGain
* ListenerGain
;
461 ALSource
->Params
.DryGains
[FRONT_CENTER
] = DryGain
* ListenerGain
;
462 ALSource
->Params
.DryGains
[BACK_CENTER
] = DryGain
* ListenerGain
;
463 ALSource
->Params
.DryGains
[LFE
] = DryGain
* ListenerGain
;
465 for(i
= 0;i
< NumSends
;i
++)
467 WetGain
[i
] = SourceVolume
;
468 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
469 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
472 switch(ALSource
->Send
[i
].WetFilter
.type
)
474 case AL_FILTER_LOWPASS
:
475 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
476 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
480 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
482 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
484 ALSource
->Params
.WetGains
[i
] = 0.0f
;
488 /* Update filter coefficients. Calculations based on the I3DL2
490 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
492 /* We use two chained one-pole filters, so we need to take the
493 * square root of the squared gain, which is the same as the base
495 ALSource
->Params
.iirFilter
.coeff
= lpCoeffCalc(DryGainHF
, cw
);
497 for(i
= 0;i
< NumSends
;i
++)
499 /* We use a one-pole filter, so we need to take the squared gain */
500 ALfloat a
= lpCoeffCalc(WetGainHF
[i
]*WetGainHF
[i
], cw
);
501 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
505 static ALvoid
CalcSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
)
507 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,DryMix
,OrigDist
;
508 ALfloat Direction
[3],Position
[3],SourceToListener
[3];
509 ALfloat Velocity
[3],ListenerVel
[3];
510 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
,OuterGainHF
;
511 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
512 ALfloat DopplerFactor
, DopplerVelocity
, flSpeedOfSound
;
513 ALfloat Matrix
[4][4];
514 ALfloat flAttenuation
, effectiveDist
;
515 ALfloat RoomAttenuation
[MAX_SENDS
];
516 ALfloat MetersPerUnit
;
517 ALfloat RoomRolloff
[MAX_SENDS
];
518 ALfloat DryGainHF
= 1.0f
;
519 ALfloat WetGain
[MAX_SENDS
];
520 ALfloat WetGainHF
[MAX_SENDS
];
521 ALfloat DirGain
, AmbientGain
;
523 const ALfloat
*SpeakerGain
;
529 for(i
= 0;i
< MAX_SENDS
;i
++)
532 //Get context properties
533 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
534 DopplerVelocity
= ALContext
->DopplerVelocity
;
535 flSpeedOfSound
= ALContext
->flSpeedOfSound
;
536 NumSends
= ALContext
->Device
->NumAuxSends
;
537 Frequency
= ALContext
->Device
->Frequency
;
539 //Get listener properties
540 ListenerGain
= ALContext
->Listener
.Gain
;
541 MetersPerUnit
= ALContext
->Listener
.MetersPerUnit
;
542 memcpy(ListenerVel
, ALContext
->Listener
.Velocity
, sizeof(ALContext
->Listener
.Velocity
));
544 //Get source properties
545 SourceVolume
= ALSource
->flGain
;
546 memcpy(Position
, ALSource
->vPosition
, sizeof(ALSource
->vPosition
));
547 memcpy(Direction
, ALSource
->vOrientation
, sizeof(ALSource
->vOrientation
));
548 memcpy(Velocity
, ALSource
->vVelocity
, sizeof(ALSource
->vVelocity
));
549 MinVolume
= ALSource
->flMinGain
;
550 MaxVolume
= ALSource
->flMaxGain
;
551 MinDist
= ALSource
->flRefDistance
;
552 MaxDist
= ALSource
->flMaxDistance
;
553 Rolloff
= ALSource
->flRollOffFactor
;
554 InnerAngle
= ALSource
->flInnerAngle
;
555 OuterAngle
= ALSource
->flOuterAngle
;
556 OuterGainHF
= ALSource
->OuterGainHF
;
558 //1. Translate Listener to origin (convert to head relative)
559 if(ALSource
->bHeadRelative
==AL_FALSE
)
561 ALfloat U
[3],V
[3],N
[3],P
[3];
563 // Build transform matrix
564 memcpy(N
, ALContext
->Listener
.Forward
, sizeof(N
)); // At-vector
565 aluNormalize(N
); // Normalized At-vector
566 memcpy(V
, ALContext
->Listener
.Up
, sizeof(V
)); // Up-vector
567 aluNormalize(V
); // Normalized Up-vector
568 aluCrossproduct(N
, V
, U
); // Right-vector
569 aluNormalize(U
); // Normalized Right-vector
570 P
[0] = -(ALContext
->Listener
.Position
[0]*U
[0] + // Translation
571 ALContext
->Listener
.Position
[1]*U
[1] +
572 ALContext
->Listener
.Position
[2]*U
[2]);
573 P
[1] = -(ALContext
->Listener
.Position
[0]*V
[0] +
574 ALContext
->Listener
.Position
[1]*V
[1] +
575 ALContext
->Listener
.Position
[2]*V
[2]);
576 P
[2] = -(ALContext
->Listener
.Position
[0]*-N
[0] +
577 ALContext
->Listener
.Position
[1]*-N
[1] +
578 ALContext
->Listener
.Position
[2]*-N
[2]);
579 Matrix
[0][0] = U
[0]; Matrix
[0][1] = V
[0]; Matrix
[0][2] = -N
[0]; Matrix
[0][3] = 0.0f
;
580 Matrix
[1][0] = U
[1]; Matrix
[1][1] = V
[1]; Matrix
[1][2] = -N
[1]; Matrix
[1][3] = 0.0f
;
581 Matrix
[2][0] = U
[2]; Matrix
[2][1] = V
[2]; Matrix
[2][2] = -N
[2]; Matrix
[2][3] = 0.0f
;
582 Matrix
[3][0] = P
[0]; Matrix
[3][1] = P
[1]; Matrix
[3][2] = P
[2]; Matrix
[3][3] = 1.0f
;
584 // Transform source position and direction into listener space
585 aluMatrixVector(Position
, 1.0f
, Matrix
);
586 aluMatrixVector(Direction
, 0.0f
, Matrix
);
587 // Transform source and listener velocity into listener space
588 aluMatrixVector(Velocity
, 0.0f
, Matrix
);
589 aluMatrixVector(ListenerVel
, 0.0f
, Matrix
);
592 ListenerVel
[0] = ListenerVel
[1] = ListenerVel
[2] = 0.0f
;
594 SourceToListener
[0] = -Position
[0];
595 SourceToListener
[1] = -Position
[1];
596 SourceToListener
[2] = -Position
[2];
597 aluNormalize(SourceToListener
);
598 aluNormalize(Direction
);
600 //2. Calculate distance attenuation
601 Distance
= aluSqrt(aluDotproduct(Position
, Position
));
604 flAttenuation
= 1.0f
;
605 for(i
= 0;i
< NumSends
;i
++)
607 RoomAttenuation
[i
] = 1.0f
;
609 RoomRolloff
[i
] = ALSource
->RoomRolloffFactor
;
610 if(ALSource
->Send
[i
].Slot
&&
611 (ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
612 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
))
613 RoomRolloff
[i
] += ALSource
->Send
[i
].Slot
->effect
.Reverb
.RoomRolloffFactor
;
616 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
617 ALContext
->DistanceModel
)
619 case AL_INVERSE_DISTANCE_CLAMPED
:
620 Distance
=__max(Distance
,MinDist
);
621 Distance
=__min(Distance
,MaxDist
);
622 if(MaxDist
< MinDist
)
625 case AL_INVERSE_DISTANCE
:
628 if((MinDist
+ (Rolloff
* (Distance
- MinDist
))) > 0.0f
)
629 flAttenuation
= MinDist
/ (MinDist
+ (Rolloff
* (Distance
- MinDist
)));
630 for(i
= 0;i
< NumSends
;i
++)
632 if((MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
))) > 0.0f
)
633 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
)));
638 case AL_LINEAR_DISTANCE_CLAMPED
:
639 Distance
=__max(Distance
,MinDist
);
640 Distance
=__min(Distance
,MaxDist
);
641 if(MaxDist
< MinDist
)
644 case AL_LINEAR_DISTANCE
:
645 Distance
=__min(Distance
,MaxDist
);
646 if(MaxDist
!= MinDist
)
648 flAttenuation
= 1.0f
- (Rolloff
*(Distance
-MinDist
)/(MaxDist
- MinDist
));
649 for(i
= 0;i
< NumSends
;i
++)
650 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(Distance
-MinDist
)/(MaxDist
- MinDist
));
654 case AL_EXPONENT_DISTANCE_CLAMPED
:
655 Distance
=__max(Distance
,MinDist
);
656 Distance
=__min(Distance
,MaxDist
);
657 if(MaxDist
< MinDist
)
660 case AL_EXPONENT_DISTANCE
:
661 if(Distance
> 0.0f
&& MinDist
> 0.0f
)
663 flAttenuation
= aluPow(Distance
/MinDist
, -Rolloff
);
664 for(i
= 0;i
< NumSends
;i
++)
665 RoomAttenuation
[i
] = aluPow(Distance
/MinDist
, -RoomRolloff
[i
]);
673 // Source Gain + Attenuation
674 DryMix
= SourceVolume
* flAttenuation
;
675 for(i
= 0;i
< NumSends
;i
++)
676 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
678 effectiveDist
= 0.0f
;
680 effectiveDist
= (MinDist
/flAttenuation
- MinDist
)*MetersPerUnit
;
682 // Distance-based air absorption
683 if(ALSource
->AirAbsorptionFactor
> 0.0f
&& effectiveDist
> 0.0f
)
687 // Absorption calculation is done in dB
688 absorb
= (ALSource
->AirAbsorptionFactor
*AIRABSORBGAINDBHF
) *
690 // Convert dB to linear gain before applying
691 absorb
= aluPow(10.0f
, absorb
/20.0f
);
696 //3. Apply directional soundcones
697 Angle
= aluAcos(aluDotproduct(Direction
,SourceToListener
)) * 180.0f
/M_PI
;
698 if(Angle
>= InnerAngle
&& Angle
<= OuterAngle
)
700 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
701 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
)*scale
);
702 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
)*scale
);
704 else if(Angle
> OuterAngle
)
706 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
));
707 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
));
715 // Apply some high-frequency attenuation for sources behind the listener
716 // NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
717 // that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
718 // the same as SourceToListener[2]
719 Angle
= aluAcos(SourceToListener
[2]) * 180.0f
/M_PI
;
720 // Sources within the minimum distance attenuate less
721 if(OrigDist
< MinDist
)
722 Angle
*= OrigDist
/MinDist
;
725 ALfloat scale
= (Angle
-90.0f
) / (180.1f
-90.0f
); // .1 to account for fp errors
726 ConeHF
*= 1.0f
- (ALContext
->Device
->HeadDampen
*scale
);
729 DryMix
*= ConeVolume
;
730 if(ALSource
->DryGainHFAuto
)
733 // Clamp to Min/Max Gain
734 DryMix
= __min(DryMix
,MaxVolume
);
735 DryMix
= __max(DryMix
,MinVolume
);
737 for(i
= 0;i
< NumSends
;i
++)
739 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
741 if(!Slot
|| Slot
->effect
.type
== AL_EFFECT_NULL
)
743 ALSource
->Params
.WetGains
[i
] = 0.0f
;
748 if(Slot
->AuxSendAuto
)
750 if(ALSource
->WetGainAuto
)
751 WetGain
[i
] *= ConeVolume
;
752 if(ALSource
->WetGainHFAuto
)
753 WetGainHF
[i
] *= ConeHF
;
755 // Clamp to Min/Max Gain
756 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
757 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
759 if(Slot
->effect
.type
== AL_EFFECT_REVERB
||
760 Slot
->effect
.type
== AL_EFFECT_EAXREVERB
)
762 /* Apply a decay-time transformation to the wet path, based on
763 * the attenuation of the dry path.
765 * Using the approximate (effective) source to listener
766 * distance, the initial decay of the reverb effect is
767 * calculated and applied to the wet path.
769 WetGain
[i
] *= aluPow(10.0f
, effectiveDist
/
770 (SPEEDOFSOUNDMETRESPERSEC
*
771 Slot
->effect
.Reverb
.DecayTime
) *
774 WetGainHF
[i
] *= aluPow(10.0f
,
775 log10(Slot
->effect
.Reverb
.AirAbsorptionGainHF
) *
776 ALSource
->AirAbsorptionFactor
* effectiveDist
);
781 /* If the slot's auxiliary send auto is off, the data sent to the
782 * effect slot is the same as the dry path, sans filter effects */
784 WetGainHF
[i
] = DryGainHF
;
787 switch(ALSource
->Send
[i
].WetFilter
.type
)
789 case AL_FILTER_LOWPASS
:
790 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
791 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
794 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
796 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
798 ALSource
->Params
.WetGains
[i
] = 0.0f
;
802 // Apply filter gains and filters
803 switch(ALSource
->DirectFilter
.type
)
805 case AL_FILTER_LOWPASS
:
806 DryMix
*= ALSource
->DirectFilter
.Gain
;
807 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
810 DryMix
*= ListenerGain
;
812 // Calculate Velocity
813 if(DopplerFactor
!= 0.0f
)
815 ALfloat flVSS
, flVLS
;
816 ALfloat flMaxVelocity
= (DopplerVelocity
* flSpeedOfSound
) /
819 flVSS
= aluDotproduct(Velocity
, SourceToListener
);
820 if(flVSS
>= flMaxVelocity
)
821 flVSS
= (flMaxVelocity
- 1.0f
);
822 else if(flVSS
<= -flMaxVelocity
)
823 flVSS
= -flMaxVelocity
+ 1.0f
;
825 flVLS
= aluDotproduct(ListenerVel
, SourceToListener
);
826 if(flVLS
>= flMaxVelocity
)
827 flVLS
= (flMaxVelocity
- 1.0f
);
828 else if(flVLS
<= -flMaxVelocity
)
829 flVLS
= -flMaxVelocity
+ 1.0f
;
831 ALSource
->Params
.Pitch
= ALSource
->flPitch
*
832 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVLS
)) /
833 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVSS
));
836 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
838 // Use energy-preserving panning algorithm for multi-speaker playback
839 length
= __max(OrigDist
, MinDist
);
842 ALfloat invlen
= 1.0f
/length
;
843 Position
[0] *= invlen
;
844 Position
[1] *= invlen
;
845 Position
[2] *= invlen
;
848 pos
= aluCart2LUTpos(-Position
[2], Position
[0]);
849 SpeakerGain
= &ALContext
->PanningLUT
[OUTPUTCHANNELS
* pos
];
851 DirGain
= aluSqrt(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
852 // elevation adjustment for directional gain. this sucks, but
853 // has low complexity
854 AmbientGain
= 1.0/aluSqrt(ALContext
->NumChan
) * (1.0-DirGain
);
855 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
857 ALfloat gain
= SpeakerGain
[s
]*DirGain
+ AmbientGain
;
858 ALSource
->Params
.DryGains
[s
] = DryMix
* gain
;
861 /* Update filter coefficients. */
862 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
864 /* Spatialized sources use four chained one-pole filters, so we need to
865 * take the fourth root of the squared gain, which is the same as the
866 * square root of the base gain. */
867 ALSource
->Params
.iirFilter
.coeff
= lpCoeffCalc(aluSqrt(DryGainHF
), cw
);
869 for(i
= 0;i
< NumSends
;i
++)
871 /* The wet path uses two chained one-pole filters, so take the
872 * base gain (square root of the squared gain) */
873 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= lpCoeffCalc(WetGainHF
[i
], cw
);
877 static __inline ALfloat
point(ALfloat val1
, ALfloat val2
, ALint frac
)
883 static __inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALint frac
)
885 return val1
+ ((val2
-val1
)*(frac
* (1.0f
/(1<<FRACTIONBITS
))));
887 static __inline ALfloat
cos_lerp(ALfloat val1
, ALfloat val2
, ALint frac
)
889 ALfloat mult
= (1.0f
-cos(frac
* (1.0f
/(1<<FRACTIONBITS
)) * M_PI
)) * 0.5f
;
890 return val1
+ ((val2
-val1
)*mult
);
893 static void MixSomeSources(ALCcontext
*ALContext
, float (*DryBuffer
)[OUTPUTCHANNELS
], ALuint SamplesToDo
)
895 static float DummyBuffer
[BUFFERSIZE
];
896 ALfloat
*WetBuffer
[MAX_SENDS
];
897 ALfloat (*Matrix
)[OUTPUTCHANNELS
] = ALContext
->ChannelMatrix
;
898 ALfloat DrySend
[OUTPUTCHANNELS
];
899 ALfloat dryGainStep
[OUTPUTCHANNELS
];
900 ALfloat wetGainStep
[MAX_SENDS
];
903 ALfloat value
, outsamp
;
904 ALbufferlistitem
*BufferListItem
;
905 ALint64 DataSize64
,DataPos64
;
906 FILTER
*DryFilter
, *WetFilter
[MAX_SENDS
];
907 ALfloat WetSend
[MAX_SENDS
];
911 ALuint DataPosInt
, DataPosFrac
;
912 ALuint Channels
, Bytes
;
914 resampler_t Resampler
;
915 ALuint BuffersPlayed
;
919 if(!(ALSource
=ALContext
->SourceList
))
922 DeviceFreq
= ALContext
->Device
->Frequency
;
924 rampLength
= DeviceFreq
* MIN_RAMP_LENGTH
/ 1000;
925 rampLength
= max(rampLength
, SamplesToDo
);
928 if(ALSource
->state
!= AL_PLAYING
)
930 if((ALSource
=ALSource
->next
) != NULL
)
936 /* Find buffer format */
940 BufferListItem
= ALSource
->queue
;
941 while(BufferListItem
!= NULL
)
944 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
946 Channels
= aluChannelsFromFormat(ALBuffer
->format
);
947 Bytes
= aluBytesFromFormat(ALBuffer
->format
);
948 Frequency
= ALBuffer
->frequency
;
951 BufferListItem
= BufferListItem
->next
;
954 if(ALSource
->NeedsUpdate
)
956 //Only apply 3D calculations for mono buffers
958 CalcSourceParams(ALContext
, ALSource
);
960 CalcNonAttnSourceParams(ALContext
, ALSource
);
961 ALSource
->NeedsUpdate
= AL_FALSE
;
964 /* Get source info */
965 Resampler
= ALSource
->Resampler
;
966 State
= ALSource
->state
;
967 BuffersPlayed
= ALSource
->BuffersPlayed
;
968 DataPosInt
= ALSource
->position
;
969 DataPosFrac
= ALSource
->position_fraction
;
971 /* Compute 18.14 fixed point step */
972 Pitch
= (ALSource
->Params
.Pitch
*Frequency
) / DeviceFreq
;
973 if(Pitch
> (float)MAX_PITCH
) Pitch
= (float)MAX_PITCH
;
974 increment
= (ALint
)(Pitch
*(ALfloat
)(1L<<FRACTIONBITS
));
975 if(increment
<= 0) increment
= (1<<FRACTIONBITS
);
977 if(ALSource
->FirstStart
)
979 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
980 DrySend
[i
] = ALSource
->Params
.DryGains
[i
];
981 for(i
= 0;i
< MAX_SENDS
;i
++)
982 WetSend
[i
] = ALSource
->Params
.WetGains
[i
];
986 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
987 DrySend
[i
] = ALSource
->DryGains
[i
];
988 for(i
= 0;i
< MAX_SENDS
;i
++)
989 WetSend
[i
] = ALSource
->WetGains
[i
];
992 DryFilter
= &ALSource
->Params
.iirFilter
;
993 for(i
= 0;i
< MAX_SENDS
;i
++)
995 WetFilter
[i
] = &ALSource
->Params
.Send
[i
].iirFilter
;
996 WetBuffer
[i
] = (ALSource
->Send
[i
].Slot
?
997 ALSource
->Send
[i
].Slot
->WetBuffer
:
1001 if(DuplicateStereo
&& Channels
== 2)
1003 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 1.0f
;
1004 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 1.0f
;
1005 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 1.0f
;
1006 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 1.0f
;
1008 else if(DuplicateStereo
)
1010 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 0.0f
;
1011 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 0.0f
;
1012 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 0.0f
;
1013 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 0.0f
;
1016 /* Get current buffer queue item */
1017 BufferListItem
= ALSource
->queue
;
1018 for(i
= 0;i
< BuffersPlayed
&& BufferListItem
;i
++)
1019 BufferListItem
= BufferListItem
->next
;
1021 while(State
== AL_PLAYING
&& j
< SamplesToDo
)
1023 ALuint DataSize
= 0;
1028 /* Get buffer info */
1029 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
1031 Data
= ALBuffer
->data
;
1032 DataSize
= ALBuffer
->size
;
1033 DataSize
/= Channels
* Bytes
;
1035 if(DataPosInt
>= DataSize
)
1038 if(BufferListItem
->next
)
1040 ALbuffer
*NextBuf
= BufferListItem
->next
->buffer
;
1041 if(NextBuf
&& NextBuf
->size
)
1043 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1044 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1045 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1048 else if(ALSource
->bLooping
)
1050 ALbuffer
*NextBuf
= ALSource
->queue
->buffer
;
1051 if(NextBuf
&& NextBuf
->size
)
1053 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1054 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1055 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1059 memset(&Data
[DataSize
*Channels
], 0, (BUFFER_PADDING
*Channels
*Bytes
));
1061 /* Compute the gain steps for each output channel */
1062 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1063 dryGainStep
[i
] = (ALSource
->Params
.DryGains
[i
]-DrySend
[i
]) /
1065 for(i
= 0;i
< MAX_SENDS
;i
++)
1066 wetGainStep
[i
] = (ALSource
->Params
.WetGains
[i
]-WetSend
[i
]) /
1069 /* Figure out how many samples we can mix. */
1070 DataSize64
= DataSize
;
1071 DataSize64
<<= FRACTIONBITS
;
1072 DataPos64
= DataPosInt
;
1073 DataPos64
<<= FRACTIONBITS
;
1074 DataPos64
+= DataPosFrac
;
1075 BufferSize
= (ALuint
)((DataSize64
-DataPos64
+(increment
-1)) / increment
);
1077 BufferSize
= min(BufferSize
, (SamplesToDo
-j
));
1079 /* Actual sample mixing loop */
1081 Data
+= DataPosInt
*Channels
;
1083 if(Channels
== 1) /* Mono */
1085 #define DO_MIX(resampler) do { \
1086 while(BufferSize--) \
1088 for(i = 0;i < OUTPUTCHANNELS;i++) \
1089 DrySend[i] += dryGainStep[i]; \
1090 for(i = 0;i < MAX_SENDS;i++) \
1091 WetSend[i] += wetGainStep[i]; \
1093 /* First order interpolator */ \
1094 value = (resampler)(Data[k], Data[k+1], DataPosFrac); \
1096 /* Direct path final mix buffer and panning */ \
1097 outsamp = lpFilter4P(DryFilter, 0, value); \
1098 DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT]; \
1099 DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT]; \
1100 DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT]; \
1101 DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT]; \
1102 DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT]; \
1103 DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT]; \
1104 DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \
1105 DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER]; \
1107 /* Room path final mix buffer and panning */ \
1108 for(i = 0;i < MAX_SENDS;i++) \
1110 outsamp = lpFilter2P(WetFilter[i], 0, value); \
1111 WetBuffer[i][j] += outsamp*WetSend[i]; \
1114 DataPosFrac += increment; \
1115 k += DataPosFrac>>FRACTIONBITS; \
1116 DataPosFrac &= FRACTIONMASK; \
1123 case POINT_RESAMPLER
:
1124 DO_MIX(point
); break;
1125 case LINEAR_RESAMPLER
:
1126 DO_MIX(lerp
); break;
1127 case COSINE_RESAMPLER
:
1128 DO_MIX(cos_lerp
); break;
1135 else if(Channels
== 2) /* Stereo */
1137 const int chans
[] = {
1138 FRONT_LEFT
, FRONT_RIGHT
1140 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1142 #define DO_MIX(resampler) do { \
1143 while(BufferSize--) \
1145 for(i = 0;i < OUTPUTCHANNELS;i++) \
1146 DrySend[i] += dryGainStep[i]; \
1147 for(i = 0;i < MAX_SENDS;i++) \
1148 WetSend[i] += wetGainStep[i]; \
1150 for(i = 0;i < Channels;i++) \
1152 value = (resampler)(Data[k*Channels + i], Data[(k+1)*Channels + i], \
1154 outsamp = lpFilter2P(DryFilter, chans[i]*2, value)*DrySend[chans[i]]; \
1155 for(out = 0;out < OUTPUTCHANNELS;out++) \
1156 DryBuffer[j][out] += outsamp*Matrix[chans[i]][out]; \
1157 for(out = 0;out < MAX_SENDS;out++) \
1159 outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
1160 WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
1164 DataPosFrac += increment; \
1165 k += DataPosFrac>>FRACTIONBITS; \
1166 DataPosFrac &= FRACTIONMASK; \
1173 case POINT_RESAMPLER
:
1174 DO_MIX(point
); break;
1175 case LINEAR_RESAMPLER
:
1176 DO_MIX(lerp
); break;
1177 case COSINE_RESAMPLER
:
1178 DO_MIX(cos_lerp
); break;
1184 else if(Channels
== 4) /* Quad */
1186 const int chans
[] = {
1187 FRONT_LEFT
, FRONT_RIGHT
,
1188 BACK_LEFT
, BACK_RIGHT
1190 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1194 case POINT_RESAMPLER
:
1195 DO_MIX(point
); break;
1196 case LINEAR_RESAMPLER
:
1197 DO_MIX(lerp
); break;
1198 case COSINE_RESAMPLER
:
1199 DO_MIX(cos_lerp
); break;
1205 else if(Channels
== 6) /* 5.1 */
1207 const int chans
[] = {
1208 FRONT_LEFT
, FRONT_RIGHT
,
1210 BACK_LEFT
, BACK_RIGHT
1212 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1216 case POINT_RESAMPLER
:
1217 DO_MIX(point
); break;
1218 case LINEAR_RESAMPLER
:
1219 DO_MIX(lerp
); break;
1220 case COSINE_RESAMPLER
:
1221 DO_MIX(cos_lerp
); break;
1227 else if(Channels
== 7) /* 6.1 */
1229 const int chans
[] = {
1230 FRONT_LEFT
, FRONT_RIGHT
,
1233 SIDE_LEFT
, SIDE_RIGHT
1235 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1239 case POINT_RESAMPLER
:
1240 DO_MIX(point
); break;
1241 case LINEAR_RESAMPLER
:
1242 DO_MIX(lerp
); break;
1243 case COSINE_RESAMPLER
:
1244 DO_MIX(cos_lerp
); break;
1250 else if(Channels
== 8) /* 7.1 */
1252 const int chans
[] = {
1253 FRONT_LEFT
, FRONT_RIGHT
,
1255 BACK_LEFT
, BACK_RIGHT
,
1256 SIDE_LEFT
, SIDE_RIGHT
1258 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1262 case POINT_RESAMPLER
:
1263 DO_MIX(point
); break;
1264 case LINEAR_RESAMPLER
:
1265 DO_MIX(lerp
); break;
1266 case COSINE_RESAMPLER
:
1267 DO_MIX(cos_lerp
); break;
1276 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1277 DrySend
[i
] += dryGainStep
[i
]*BufferSize
;
1278 for(i
= 0;i
< MAX_SENDS
;i
++)
1279 WetSend
[i
] += wetGainStep
[i
]*BufferSize
;
1282 DataPosFrac
+= increment
;
1283 k
+= DataPosFrac
>>FRACTIONBITS
;
1284 DataPosFrac
&= FRACTIONMASK
;
1291 /* Handle looping sources */
1292 if(DataPosInt
>= DataSize
)
1294 if(BuffersPlayed
< (ALSource
->BuffersInQueue
-1))
1296 BufferListItem
= BufferListItem
->next
;
1298 DataPosInt
-= DataSize
;
1300 else if(ALSource
->bLooping
)
1302 BufferListItem
= ALSource
->queue
;
1304 if(ALSource
->BuffersInQueue
== 1)
1305 DataPosInt
%= DataSize
;
1307 DataPosInt
-= DataSize
;
1312 BufferListItem
= ALSource
->queue
;
1313 BuffersPlayed
= ALSource
->BuffersInQueue
;
1320 /* Update source info */
1321 ALSource
->state
= State
;
1322 ALSource
->BuffersPlayed
= BuffersPlayed
;
1323 ALSource
->position
= DataPosInt
;
1324 ALSource
->position_fraction
= DataPosFrac
;
1325 ALSource
->Buffer
= BufferListItem
->buffer
;
1327 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1328 ALSource
->DryGains
[i
] = DrySend
[i
];
1329 for(i
= 0;i
< MAX_SENDS
;i
++)
1330 ALSource
->WetGains
[i
] = WetSend
[i
];
1332 ALSource
->FirstStart
= AL_FALSE
;
1334 if((ALSource
=ALSource
->next
) != NULL
)
1335 goto another_source
;
1338 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1340 float (*DryBuffer
)[OUTPUTCHANNELS
];
1341 const Channel
*ChanMap
;
1343 ALeffectslot
*ALEffectSlot
;
1344 ALCcontext
*ALContext
;
1348 #if defined(HAVE_FESETROUND)
1349 fpuState
= fegetround();
1350 fesetround(FE_TOWARDZERO
);
1351 #elif defined(HAVE__CONTROLFP)
1352 fpuState
= _controlfp(0, 0);
1353 _controlfp(_RC_CHOP
, _MCW_RC
);
1358 DryBuffer
= device
->DryBuffer
;
1361 /* Setup variables */
1362 SamplesToDo
= min(size
, BUFFERSIZE
);
1364 /* Clear mixing buffer */
1365 memset(DryBuffer
, 0, SamplesToDo
*OUTPUTCHANNELS
*sizeof(ALfloat
));
1367 SuspendContext(NULL
);
1368 for(c
= 0;c
< device
->NumContexts
;c
++)
1370 ALContext
= device
->Contexts
[c
];
1371 SuspendContext(ALContext
);
1373 MixSomeSources(ALContext
, DryBuffer
, SamplesToDo
);
1375 /* effect slot processing */
1376 ALEffectSlot
= ALContext
->EffectSlotList
;
1379 if(ALEffectSlot
->EffectState
)
1380 ALEffect_Process(ALEffectSlot
->EffectState
, ALEffectSlot
, SamplesToDo
, ALEffectSlot
->WetBuffer
, DryBuffer
);
1382 for(i
= 0;i
< SamplesToDo
;i
++)
1383 ALEffectSlot
->WetBuffer
[i
] = 0.0f
;
1384 ALEffectSlot
= ALEffectSlot
->next
;
1386 ProcessContext(ALContext
);
1388 ProcessContext(NULL
);
1390 //Post processing loop
1391 ChanMap
= device
->DevChannels
;
1392 switch(device
->Format
)
1394 #define CHECK_WRITE_FORMAT(bits, type, func) \
1395 case AL_FORMAT_MONO##bits: \
1396 for(i = 0;i < SamplesToDo;i++) \
1398 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1399 buffer = ((type*)buffer) + 1; \
1402 case AL_FORMAT_STEREO##bits: \
1405 for(i = 0;i < SamplesToDo;i++) \
1408 samples[0] = DryBuffer[i][ChanMap[0]]; \
1409 samples[1] = DryBuffer[i][ChanMap[1]]; \
1410 bs2b_cross_feed(device->Bs2b, samples); \
1411 ((type*)buffer)[0] = (func)(samples[0]); \
1412 ((type*)buffer)[1] = (func)(samples[1]); \
1413 buffer = ((type*)buffer) + 2; \
1418 for(i = 0;i < SamplesToDo;i++) \
1420 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1421 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1422 buffer = ((type*)buffer) + 2; \
1426 case AL_FORMAT_QUAD##bits: \
1427 for(i = 0;i < SamplesToDo;i++) \
1429 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1430 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1431 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1432 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1433 buffer = ((type*)buffer) + 4; \
1436 case AL_FORMAT_51CHN##bits: \
1437 for(i = 0;i < SamplesToDo;i++) \
1439 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1440 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1441 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1442 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1443 ((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
1444 ((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
1445 buffer = ((type*)buffer) + 6; \
1448 case AL_FORMAT_61CHN##bits: \
1449 for(i = 0;i < SamplesToDo;i++) \
1451 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1452 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1453 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1454 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1455 ((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
1456 ((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
1457 ((type*)buffer)[6] = (func)(DryBuffer[i][ChanMap[6]]); \
1458 buffer = ((type*)buffer) + 7; \
1461 case AL_FORMAT_71CHN##bits: \
1462 for(i = 0;i < SamplesToDo;i++) \
1464 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1465 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1466 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1467 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1468 ((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
1469 ((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
1470 ((type*)buffer)[6] = (func)(DryBuffer[i][ChanMap[6]]); \
1471 ((type*)buffer)[7] = (func)(DryBuffer[i][ChanMap[7]]); \
1472 buffer = ((type*)buffer) + 8; \
1476 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1477 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1478 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
)
1479 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
)
1480 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
)
1481 #undef AL_FORMAT_STEREO32
1482 #undef AL_FORMAT_MONO32
1483 #undef CHECK_WRITE_FORMAT
1489 size
-= SamplesToDo
;
1492 #if defined(HAVE_FESETROUND)
1493 fesetround(fpuState
);
1494 #elif defined(HAVE__CONTROLFP)
1495 _controlfp(fpuState
, 0xfffff);
1499 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1503 SuspendContext(NULL
);
1504 for(i
= 0;i
< device
->NumContexts
;i
++)
1508 SuspendContext(device
->Contexts
[i
]);
1510 source
= device
->Contexts
[i
]->SourceList
;
1513 if(source
->state
== AL_PLAYING
)
1515 source
->state
= AL_STOPPED
;
1516 source
->BuffersPlayed
= source
->BuffersInQueue
;
1517 source
->position
= 0;
1518 source
->position_fraction
= 0;
1520 source
= source
->next
;
1522 ProcessContext(device
->Contexts
[i
]);
1525 device
->Connected
= ALC_FALSE
;
1526 ProcessContext(NULL
);