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 Channel Speaker2Chan
[OUTPUTCHANNELS
], ALint chans
)
121 char layout_str
[256];
122 char *confkey
, *next
;
127 strncpy(layout_str
, GetConfigValue(NULL
, name
, ""), sizeof(layout_str
));
130 next
= confkey
= layout_str
;
134 next
= strchr(confkey
, ',');
140 } while(isspace(*next
) || *next
== ',');
143 sep
= strchr(confkey
, '=');
144 if(!sep
|| confkey
== sep
)
148 while(isspace(*end
) && end
!= confkey
)
152 if(strcmp(confkey
, "fl") == 0 || strcmp(confkey
, "front-left") == 0)
154 else if(strcmp(confkey
, "fr") == 0 || strcmp(confkey
, "front-right") == 0)
156 else if(strcmp(confkey
, "fc") == 0 || strcmp(confkey
, "front-center") == 0)
158 else if(strcmp(confkey
, "bl") == 0 || strcmp(confkey
, "back-left") == 0)
160 else if(strcmp(confkey
, "br") == 0 || strcmp(confkey
, "back-right") == 0)
162 else if(strcmp(confkey
, "bc") == 0 || strcmp(confkey
, "back-center") == 0)
164 else if(strcmp(confkey
, "sl") == 0 || strcmp(confkey
, "side-left") == 0)
166 else if(strcmp(confkey
, "sr") == 0 || strcmp(confkey
, "side-right") == 0)
170 AL_PRINT("Unknown speaker for %s: \"%s\"\n", name
, confkey
);
178 for(i
= 0;i
< chans
;i
++)
180 if(Speaker2Chan
[i
] == val
)
182 long angle
= strtol(sep
, NULL
, 10);
183 if(angle
>= -180 && angle
<= 180)
184 SpeakerAngle
[i
] = angle
* M_PI
/180.0f
;
186 AL_PRINT("Invalid angle for speaker \"%s\": %ld\n", confkey
, angle
);
192 for(i
= 0;i
< chans
;i
++)
197 for(i2
= i
+1;i2
< chans
;i2
++)
199 if(SpeakerAngle
[i2
] < SpeakerAngle
[min
])
208 tmpf
= SpeakerAngle
[i
];
209 SpeakerAngle
[i
] = SpeakerAngle
[min
];
210 SpeakerAngle
[min
] = tmpf
;
212 tmpi
= Speaker2Chan
[i
];
213 Speaker2Chan
[i
] = Speaker2Chan
[min
];
214 Speaker2Chan
[min
] = tmpi
;
219 static __inline ALfloat
aluLUTpos2Angle(ALint pos
)
221 if(pos
< QUADRANT_NUM
)
222 return aluAtan((ALfloat
)pos
/ (ALfloat
)(QUADRANT_NUM
- pos
));
223 if(pos
< 2 * QUADRANT_NUM
)
224 return M_PI_2
+ aluAtan((ALfloat
)(pos
- QUADRANT_NUM
) / (ALfloat
)(2 * QUADRANT_NUM
- pos
));
225 if(pos
< 3 * QUADRANT_NUM
)
226 return aluAtan((ALfloat
)(pos
- 2 * QUADRANT_NUM
) / (ALfloat
)(3 * QUADRANT_NUM
- pos
)) - M_PI
;
227 return aluAtan((ALfloat
)(pos
- 3 * QUADRANT_NUM
) / (ALfloat
)(4 * QUADRANT_NUM
- pos
)) - M_PI_2
;
230 ALvoid
aluInitPanning(ALCdevice
*Device
)
232 ALfloat SpeakerAngle
[OUTPUTCHANNELS
];
233 Channel Speaker2Chan
[OUTPUTCHANNELS
];
234 ALfloat Alpha
, Theta
;
239 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
241 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
242 Device
->ChannelMatrix
[s
][s2
] = ((s
==s2
) ? 1.0f
: 0.0f
);
245 switch(Device
->Format
)
247 case AL_FORMAT_MONO8
:
248 case AL_FORMAT_MONO16
:
249 case AL_FORMAT_MONO_FLOAT32
:
250 Device
->ChannelMatrix
[FRONT_LEFT
][FRONT_CENTER
] = aluSqrt(0.5);
251 Device
->ChannelMatrix
[FRONT_RIGHT
][FRONT_CENTER
] = aluSqrt(0.5);
252 Device
->ChannelMatrix
[SIDE_LEFT
][FRONT_CENTER
] = aluSqrt(0.5);
253 Device
->ChannelMatrix
[SIDE_RIGHT
][FRONT_CENTER
] = aluSqrt(0.5);
254 Device
->ChannelMatrix
[BACK_LEFT
][FRONT_CENTER
] = aluSqrt(0.5);
255 Device
->ChannelMatrix
[BACK_RIGHT
][FRONT_CENTER
] = aluSqrt(0.5);
256 Device
->ChannelMatrix
[BACK_CENTER
][FRONT_CENTER
] = 1.0f
;
258 Speaker2Chan
[0] = FRONT_CENTER
;
259 SpeakerAngle
[0] = 0.0f
* M_PI
/180.0f
;
262 case AL_FORMAT_STEREO8
:
263 case AL_FORMAT_STEREO16
:
264 case AL_FORMAT_STEREO_FLOAT32
:
265 Device
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
266 Device
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
267 Device
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
268 Device
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
269 Device
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
270 Device
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
271 Device
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
272 Device
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
274 Speaker2Chan
[0] = FRONT_LEFT
;
275 Speaker2Chan
[1] = FRONT_RIGHT
;
276 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
277 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
278 SetSpeakerArrangement("layout", SpeakerAngle
, Speaker2Chan
, Device
->NumChan
);
281 case AL_FORMAT_QUAD8
:
282 case AL_FORMAT_QUAD16
:
283 case AL_FORMAT_QUAD32
:
284 Device
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
285 Device
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
286 Device
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
287 Device
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
288 Device
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
289 Device
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
290 Device
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
291 Device
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
293 Speaker2Chan
[0] = BACK_LEFT
;
294 Speaker2Chan
[1] = FRONT_LEFT
;
295 Speaker2Chan
[2] = FRONT_RIGHT
;
296 Speaker2Chan
[3] = BACK_RIGHT
;
297 SpeakerAngle
[0] = -135.0f
* M_PI
/180.0f
;
298 SpeakerAngle
[1] = -45.0f
* M_PI
/180.0f
;
299 SpeakerAngle
[2] = 45.0f
* M_PI
/180.0f
;
300 SpeakerAngle
[3] = 135.0f
* M_PI
/180.0f
;
301 SetSpeakerArrangement("layout", SpeakerAngle
, Speaker2Chan
, Device
->NumChan
);
304 case AL_FORMAT_51CHN8
:
305 case AL_FORMAT_51CHN16
:
306 case AL_FORMAT_51CHN32
:
307 Device
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
308 Device
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
309 Device
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
310 Device
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
311 Device
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
312 Device
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
314 Speaker2Chan
[0] = BACK_LEFT
;
315 Speaker2Chan
[1] = FRONT_LEFT
;
316 Speaker2Chan
[2] = FRONT_CENTER
;
317 Speaker2Chan
[3] = FRONT_RIGHT
;
318 Speaker2Chan
[4] = BACK_RIGHT
;
319 SpeakerAngle
[0] = -110.0f
* M_PI
/180.0f
;
320 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
321 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
322 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
323 SpeakerAngle
[4] = 110.0f
* M_PI
/180.0f
;
324 SetSpeakerArrangement("layout", SpeakerAngle
, Speaker2Chan
, Device
->NumChan
);
327 case AL_FORMAT_61CHN8
:
328 case AL_FORMAT_61CHN16
:
329 case AL_FORMAT_61CHN32
:
330 Device
->ChannelMatrix
[BACK_LEFT
][BACK_CENTER
] = aluSqrt(0.5);
331 Device
->ChannelMatrix
[BACK_LEFT
][SIDE_LEFT
] = aluSqrt(0.5);
332 Device
->ChannelMatrix
[BACK_RIGHT
][BACK_CENTER
] = aluSqrt(0.5);
333 Device
->ChannelMatrix
[BACK_RIGHT
][SIDE_RIGHT
] = aluSqrt(0.5);
335 Speaker2Chan
[0] = SIDE_LEFT
;
336 Speaker2Chan
[1] = FRONT_LEFT
;
337 Speaker2Chan
[2] = FRONT_CENTER
;
338 Speaker2Chan
[3] = FRONT_RIGHT
;
339 Speaker2Chan
[4] = SIDE_RIGHT
;
340 Speaker2Chan
[5] = BACK_CENTER
;
341 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
342 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
343 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
344 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
345 SpeakerAngle
[4] = 90.0f
* M_PI
/180.0f
;
346 SpeakerAngle
[5] = 180.0f
* M_PI
/180.0f
;
347 SetSpeakerArrangement("layout", SpeakerAngle
, Speaker2Chan
, Device
->NumChan
);
350 case AL_FORMAT_71CHN8
:
351 case AL_FORMAT_71CHN16
:
352 case AL_FORMAT_71CHN32
:
353 Device
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
354 Device
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
356 Speaker2Chan
[0] = BACK_LEFT
;
357 Speaker2Chan
[1] = SIDE_LEFT
;
358 Speaker2Chan
[2] = FRONT_LEFT
;
359 Speaker2Chan
[3] = FRONT_CENTER
;
360 Speaker2Chan
[4] = FRONT_RIGHT
;
361 Speaker2Chan
[5] = SIDE_RIGHT
;
362 Speaker2Chan
[6] = BACK_RIGHT
;
363 SpeakerAngle
[0] = -150.0f
* M_PI
/180.0f
;
364 SpeakerAngle
[1] = -90.0f
* M_PI
/180.0f
;
365 SpeakerAngle
[2] = -30.0f
* M_PI
/180.0f
;
366 SpeakerAngle
[3] = 0.0f
* M_PI
/180.0f
;
367 SpeakerAngle
[4] = 30.0f
* M_PI
/180.0f
;
368 SpeakerAngle
[5] = 90.0f
* M_PI
/180.0f
;
369 SpeakerAngle
[6] = 150.0f
* M_PI
/180.0f
;
370 SetSpeakerArrangement("layout", SpeakerAngle
, Speaker2Chan
, Device
->NumChan
);
378 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
381 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
382 out
+= Device
->ChannelMatrix
[s2
][s
];
383 maxout
= __max(maxout
, out
);
386 maxout
= 1.0f
/maxout
;
387 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
389 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
390 Device
->ChannelMatrix
[s2
][s
] *= maxout
;
394 for(pos
= 0; pos
< LUT_NUM
; pos
++)
396 /* clear all values */
397 offset
= OUTPUTCHANNELS
* pos
;
398 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
399 Device
->PanningLUT
[offset
+s
] = 0.0f
;
401 if(Device
->NumChan
== 1)
403 Device
->PanningLUT
[offset
+ Speaker2Chan
[0]] = 1.0f
;
408 Theta
= aluLUTpos2Angle(pos
);
410 /* set panning values */
411 for(s
= 0; s
< Device
->NumChan
- 1; s
++)
413 if(Theta
>= SpeakerAngle
[s
] && Theta
< SpeakerAngle
[s
+1])
415 /* source between speaker s and speaker s+1 */
416 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
417 (SpeakerAngle
[s
+1]-SpeakerAngle
[s
]);
418 Device
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
419 Device
->PanningLUT
[offset
+ Speaker2Chan
[s
+1]] = sin(Alpha
);
423 if(s
== Device
->NumChan
- 1)
425 /* source between last and first speaker */
426 if(Theta
< SpeakerAngle
[0])
427 Theta
+= 2.0f
* M_PI
;
428 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
429 (2.0f
* M_PI
+ SpeakerAngle
[0]-SpeakerAngle
[s
]);
430 Device
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
431 Device
->PanningLUT
[offset
+ Speaker2Chan
[0]] = sin(Alpha
);
436 static ALvoid
CalcNonAttnSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
)
438 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
439 ALfloat DryGain
, DryGainHF
;
440 ALfloat WetGain
[MAX_SENDS
];
441 ALfloat WetGainHF
[MAX_SENDS
];
442 ALint NumSends
, Frequency
;
446 //Get context properties
447 NumSends
= ALContext
->Device
->NumAuxSends
;
448 Frequency
= ALContext
->Device
->Frequency
;
450 //Get listener properties
451 ListenerGain
= ALContext
->Listener
.Gain
;
453 //Get source properties
454 SourceVolume
= ALSource
->flGain
;
455 MinVolume
= ALSource
->flMinGain
;
456 MaxVolume
= ALSource
->flMaxGain
;
458 //1. Multi-channel buffers always play "normal"
459 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
461 DryGain
= SourceVolume
;
462 DryGain
= __min(DryGain
,MaxVolume
);
463 DryGain
= __max(DryGain
,MinVolume
);
466 switch(ALSource
->DirectFilter
.type
)
468 case AL_FILTER_LOWPASS
:
469 DryGain
*= ALSource
->DirectFilter
.Gain
;
470 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
474 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryGain
* ListenerGain
;
475 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryGain
* ListenerGain
;
476 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryGain
* ListenerGain
;
477 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryGain
* ListenerGain
;
478 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryGain
* ListenerGain
;
479 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryGain
* ListenerGain
;
480 ALSource
->Params
.DryGains
[FRONT_CENTER
] = DryGain
* ListenerGain
;
481 ALSource
->Params
.DryGains
[BACK_CENTER
] = DryGain
* ListenerGain
;
482 ALSource
->Params
.DryGains
[LFE
] = DryGain
* ListenerGain
;
484 for(i
= 0;i
< NumSends
;i
++)
486 WetGain
[i
] = SourceVolume
;
487 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
488 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
491 switch(ALSource
->Send
[i
].WetFilter
.type
)
493 case AL_FILTER_LOWPASS
:
494 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
495 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
499 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
501 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
503 ALSource
->Params
.WetGains
[i
] = 0.0f
;
507 /* Update filter coefficients. Calculations based on the I3DL2
509 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 ALSource
->Params
.iirFilter
.coeff
= lpCoeffCalc(DryGainHF
, cw
);
516 for(i
= 0;i
< NumSends
;i
++)
518 /* We use a one-pole filter, so we need to take the squared gain */
519 ALfloat a
= lpCoeffCalc(WetGainHF
[i
]*WetGainHF
[i
], cw
);
520 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
524 static ALvoid
CalcSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
)
526 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,DryMix
,OrigDist
;
527 ALfloat Direction
[3],Position
[3],SourceToListener
[3];
528 ALfloat Velocity
[3],ListenerVel
[3];
529 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
,OuterGainHF
;
530 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
531 ALfloat DopplerFactor
, DopplerVelocity
, flSpeedOfSound
;
532 ALfloat Matrix
[4][4];
533 ALfloat flAttenuation
, effectiveDist
;
534 ALfloat RoomAttenuation
[MAX_SENDS
];
535 ALfloat MetersPerUnit
;
536 ALfloat RoomRolloff
[MAX_SENDS
];
537 ALfloat DryGainHF
= 1.0f
;
538 ALfloat WetGain
[MAX_SENDS
];
539 ALfloat WetGainHF
[MAX_SENDS
];
540 ALfloat DirGain
, AmbientGain
;
542 const ALfloat
*SpeakerGain
;
548 for(i
= 0;i
< MAX_SENDS
;i
++)
551 //Get context properties
552 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
553 DopplerVelocity
= ALContext
->DopplerVelocity
;
554 flSpeedOfSound
= ALContext
->flSpeedOfSound
;
555 NumSends
= ALContext
->Device
->NumAuxSends
;
556 Frequency
= ALContext
->Device
->Frequency
;
558 //Get listener properties
559 ListenerGain
= ALContext
->Listener
.Gain
;
560 MetersPerUnit
= ALContext
->Listener
.MetersPerUnit
;
561 memcpy(ListenerVel
, ALContext
->Listener
.Velocity
, sizeof(ALContext
->Listener
.Velocity
));
563 //Get source properties
564 SourceVolume
= ALSource
->flGain
;
565 memcpy(Position
, ALSource
->vPosition
, sizeof(ALSource
->vPosition
));
566 memcpy(Direction
, ALSource
->vOrientation
, sizeof(ALSource
->vOrientation
));
567 memcpy(Velocity
, ALSource
->vVelocity
, sizeof(ALSource
->vVelocity
));
568 MinVolume
= ALSource
->flMinGain
;
569 MaxVolume
= ALSource
->flMaxGain
;
570 MinDist
= ALSource
->flRefDistance
;
571 MaxDist
= ALSource
->flMaxDistance
;
572 Rolloff
= ALSource
->flRollOffFactor
;
573 InnerAngle
= ALSource
->flInnerAngle
;
574 OuterAngle
= ALSource
->flOuterAngle
;
575 OuterGainHF
= ALSource
->OuterGainHF
;
577 //1. Translate Listener to origin (convert to head relative)
578 if(ALSource
->bHeadRelative
==AL_FALSE
)
580 ALfloat U
[3],V
[3],N
[3],P
[3];
582 // Build transform matrix
583 memcpy(N
, ALContext
->Listener
.Forward
, sizeof(N
)); // At-vector
584 aluNormalize(N
); // Normalized At-vector
585 memcpy(V
, ALContext
->Listener
.Up
, sizeof(V
)); // Up-vector
586 aluNormalize(V
); // Normalized Up-vector
587 aluCrossproduct(N
, V
, U
); // Right-vector
588 aluNormalize(U
); // Normalized Right-vector
589 P
[0] = -(ALContext
->Listener
.Position
[0]*U
[0] + // Translation
590 ALContext
->Listener
.Position
[1]*U
[1] +
591 ALContext
->Listener
.Position
[2]*U
[2]);
592 P
[1] = -(ALContext
->Listener
.Position
[0]*V
[0] +
593 ALContext
->Listener
.Position
[1]*V
[1] +
594 ALContext
->Listener
.Position
[2]*V
[2]);
595 P
[2] = -(ALContext
->Listener
.Position
[0]*-N
[0] +
596 ALContext
->Listener
.Position
[1]*-N
[1] +
597 ALContext
->Listener
.Position
[2]*-N
[2]);
598 Matrix
[0][0] = U
[0]; Matrix
[0][1] = V
[0]; Matrix
[0][2] = -N
[0]; Matrix
[0][3] = 0.0f
;
599 Matrix
[1][0] = U
[1]; Matrix
[1][1] = V
[1]; Matrix
[1][2] = -N
[1]; Matrix
[1][3] = 0.0f
;
600 Matrix
[2][0] = U
[2]; Matrix
[2][1] = V
[2]; Matrix
[2][2] = -N
[2]; Matrix
[2][3] = 0.0f
;
601 Matrix
[3][0] = P
[0]; Matrix
[3][1] = P
[1]; Matrix
[3][2] = P
[2]; Matrix
[3][3] = 1.0f
;
603 // Transform source position and direction into listener space
604 aluMatrixVector(Position
, 1.0f
, Matrix
);
605 aluMatrixVector(Direction
, 0.0f
, Matrix
);
606 // Transform source and listener velocity into listener space
607 aluMatrixVector(Velocity
, 0.0f
, Matrix
);
608 aluMatrixVector(ListenerVel
, 0.0f
, Matrix
);
611 ListenerVel
[0] = ListenerVel
[1] = ListenerVel
[2] = 0.0f
;
613 SourceToListener
[0] = -Position
[0];
614 SourceToListener
[1] = -Position
[1];
615 SourceToListener
[2] = -Position
[2];
616 aluNormalize(SourceToListener
);
617 aluNormalize(Direction
);
619 //2. Calculate distance attenuation
620 Distance
= aluSqrt(aluDotproduct(Position
, Position
));
623 flAttenuation
= 1.0f
;
624 for(i
= 0;i
< NumSends
;i
++)
626 RoomAttenuation
[i
] = 1.0f
;
628 RoomRolloff
[i
] = ALSource
->RoomRolloffFactor
;
629 if(ALSource
->Send
[i
].Slot
&&
630 (ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
631 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
))
632 RoomRolloff
[i
] += ALSource
->Send
[i
].Slot
->effect
.Reverb
.RoomRolloffFactor
;
635 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
636 ALContext
->DistanceModel
)
638 case AL_INVERSE_DISTANCE_CLAMPED
:
639 Distance
=__max(Distance
,MinDist
);
640 Distance
=__min(Distance
,MaxDist
);
641 if(MaxDist
< MinDist
)
644 case AL_INVERSE_DISTANCE
:
647 if((MinDist
+ (Rolloff
* (Distance
- MinDist
))) > 0.0f
)
648 flAttenuation
= MinDist
/ (MinDist
+ (Rolloff
* (Distance
- MinDist
)));
649 for(i
= 0;i
< NumSends
;i
++)
651 if((MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
))) > 0.0f
)
652 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
)));
657 case AL_LINEAR_DISTANCE_CLAMPED
:
658 Distance
=__max(Distance
,MinDist
);
659 Distance
=__min(Distance
,MaxDist
);
660 if(MaxDist
< MinDist
)
663 case AL_LINEAR_DISTANCE
:
664 Distance
=__min(Distance
,MaxDist
);
665 if(MaxDist
!= MinDist
)
667 flAttenuation
= 1.0f
- (Rolloff
*(Distance
-MinDist
)/(MaxDist
- MinDist
));
668 for(i
= 0;i
< NumSends
;i
++)
669 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(Distance
-MinDist
)/(MaxDist
- MinDist
));
673 case AL_EXPONENT_DISTANCE_CLAMPED
:
674 Distance
=__max(Distance
,MinDist
);
675 Distance
=__min(Distance
,MaxDist
);
676 if(MaxDist
< MinDist
)
679 case AL_EXPONENT_DISTANCE
:
680 if(Distance
> 0.0f
&& MinDist
> 0.0f
)
682 flAttenuation
= aluPow(Distance
/MinDist
, -Rolloff
);
683 for(i
= 0;i
< NumSends
;i
++)
684 RoomAttenuation
[i
] = aluPow(Distance
/MinDist
, -RoomRolloff
[i
]);
692 // Source Gain + Attenuation
693 DryMix
= SourceVolume
* flAttenuation
;
694 for(i
= 0;i
< NumSends
;i
++)
695 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
697 effectiveDist
= 0.0f
;
699 effectiveDist
= (MinDist
/flAttenuation
- MinDist
)*MetersPerUnit
;
701 // Distance-based air absorption
702 if(ALSource
->AirAbsorptionFactor
> 0.0f
&& effectiveDist
> 0.0f
)
706 // Absorption calculation is done in dB
707 absorb
= (ALSource
->AirAbsorptionFactor
*AIRABSORBGAINDBHF
) *
709 // Convert dB to linear gain before applying
710 absorb
= aluPow(10.0f
, absorb
/20.0f
);
715 //3. Apply directional soundcones
716 Angle
= aluAcos(aluDotproduct(Direction
,SourceToListener
)) * 180.0f
/M_PI
;
717 if(Angle
>= InnerAngle
&& Angle
<= OuterAngle
)
719 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
720 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
)*scale
);
721 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
)*scale
);
723 else if(Angle
> OuterAngle
)
725 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
));
726 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
));
734 // Apply some high-frequency attenuation for sources behind the listener
735 // NOTE: This should be aluDotproduct({0,0,-1}, ListenerToSource), however
736 // that is equivalent to aluDotproduct({0,0,1}, SourceToListener), which is
737 // the same as SourceToListener[2]
738 Angle
= aluAcos(SourceToListener
[2]) * 180.0f
/M_PI
;
739 // Sources within the minimum distance attenuate less
740 if(OrigDist
< MinDist
)
741 Angle
*= OrigDist
/MinDist
;
744 ALfloat scale
= (Angle
-90.0f
) / (180.1f
-90.0f
); // .1 to account for fp errors
745 ConeHF
*= 1.0f
- (ALContext
->Device
->HeadDampen
*scale
);
748 DryMix
*= ConeVolume
;
749 if(ALSource
->DryGainHFAuto
)
752 // Clamp to Min/Max Gain
753 DryMix
= __min(DryMix
,MaxVolume
);
754 DryMix
= __max(DryMix
,MinVolume
);
756 for(i
= 0;i
< NumSends
;i
++)
758 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
760 if(!Slot
|| Slot
->effect
.type
== AL_EFFECT_NULL
)
762 ALSource
->Params
.WetGains
[i
] = 0.0f
;
767 if(Slot
->AuxSendAuto
)
769 if(ALSource
->WetGainAuto
)
770 WetGain
[i
] *= ConeVolume
;
771 if(ALSource
->WetGainHFAuto
)
772 WetGainHF
[i
] *= ConeHF
;
774 // Clamp to Min/Max Gain
775 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
776 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
778 if(Slot
->effect
.type
== AL_EFFECT_REVERB
||
779 Slot
->effect
.type
== AL_EFFECT_EAXREVERB
)
781 /* Apply a decay-time transformation to the wet path, based on
782 * the attenuation of the dry path.
784 * Using the approximate (effective) source to listener
785 * distance, the initial decay of the reverb effect is
786 * calculated and applied to the wet path.
788 WetGain
[i
] *= aluPow(10.0f
, effectiveDist
/
789 (SPEEDOFSOUNDMETRESPERSEC
*
790 Slot
->effect
.Reverb
.DecayTime
) *
793 WetGainHF
[i
] *= aluPow(10.0f
,
794 log10(Slot
->effect
.Reverb
.AirAbsorptionGainHF
) *
795 ALSource
->AirAbsorptionFactor
* effectiveDist
);
800 /* If the slot's auxiliary send auto is off, the data sent to the
801 * effect slot is the same as the dry path, sans filter effects */
803 WetGainHF
[i
] = DryGainHF
;
806 switch(ALSource
->Send
[i
].WetFilter
.type
)
808 case AL_FILTER_LOWPASS
:
809 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
810 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
813 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
815 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
817 ALSource
->Params
.WetGains
[i
] = 0.0f
;
821 // Apply filter gains and filters
822 switch(ALSource
->DirectFilter
.type
)
824 case AL_FILTER_LOWPASS
:
825 DryMix
*= ALSource
->DirectFilter
.Gain
;
826 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
829 DryMix
*= ListenerGain
;
831 // Calculate Velocity
832 if(DopplerFactor
!= 0.0f
)
834 ALfloat flVSS
, flVLS
;
835 ALfloat flMaxVelocity
= (DopplerVelocity
* flSpeedOfSound
) /
838 flVSS
= aluDotproduct(Velocity
, SourceToListener
);
839 if(flVSS
>= flMaxVelocity
)
840 flVSS
= (flMaxVelocity
- 1.0f
);
841 else if(flVSS
<= -flMaxVelocity
)
842 flVSS
= -flMaxVelocity
+ 1.0f
;
844 flVLS
= aluDotproduct(ListenerVel
, SourceToListener
);
845 if(flVLS
>= flMaxVelocity
)
846 flVLS
= (flMaxVelocity
- 1.0f
);
847 else if(flVLS
<= -flMaxVelocity
)
848 flVLS
= -flMaxVelocity
+ 1.0f
;
850 ALSource
->Params
.Pitch
= ALSource
->flPitch
*
851 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVLS
)) /
852 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVSS
));
855 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
857 // Use energy-preserving panning algorithm for multi-speaker playback
858 length
= __max(OrigDist
, MinDist
);
861 ALfloat invlen
= 1.0f
/length
;
862 Position
[0] *= invlen
;
863 Position
[1] *= invlen
;
864 Position
[2] *= invlen
;
867 pos
= aluCart2LUTpos(-Position
[2], Position
[0]);
868 SpeakerGain
= &ALContext
->Device
->PanningLUT
[OUTPUTCHANNELS
* pos
];
870 DirGain
= aluSqrt(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
871 // elevation adjustment for directional gain. this sucks, but
872 // has low complexity
873 AmbientGain
= 1.0/aluSqrt(ALContext
->Device
->NumChan
) * (1.0-DirGain
);
874 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
876 ALfloat gain
= SpeakerGain
[s
]*DirGain
+ AmbientGain
;
877 ALSource
->Params
.DryGains
[s
] = DryMix
* gain
;
880 /* Update filter coefficients. */
881 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
883 /* Spatialized sources use four chained one-pole filters, so we need to
884 * take the fourth root of the squared gain, which is the same as the
885 * square root of the base gain. */
886 ALSource
->Params
.iirFilter
.coeff
= lpCoeffCalc(aluSqrt(DryGainHF
), cw
);
888 for(i
= 0;i
< NumSends
;i
++)
890 /* The wet path uses two chained one-pole filters, so take the
891 * base gain (square root of the squared gain) */
892 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= lpCoeffCalc(WetGainHF
[i
], cw
);
896 static __inline ALfloat
point(ALfloat val1
, ALfloat val2
, ALint frac
)
902 static __inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALint frac
)
904 return val1
+ ((val2
-val1
)*(frac
* (1.0f
/(1<<FRACTIONBITS
))));
906 static __inline ALfloat
cos_lerp(ALfloat val1
, ALfloat val2
, ALint frac
)
908 ALfloat mult
= (1.0f
-cos(frac
* (1.0f
/(1<<FRACTIONBITS
)) * M_PI
)) * 0.5f
;
909 return val1
+ ((val2
-val1
)*mult
);
912 static void MixSomeSources(ALCcontext
*ALContext
, float (*DryBuffer
)[OUTPUTCHANNELS
], ALuint SamplesToDo
)
914 static float DummyBuffer
[BUFFERSIZE
];
915 ALfloat
*WetBuffer
[MAX_SENDS
];
916 ALfloat DrySend
[OUTPUTCHANNELS
];
917 ALfloat dryGainStep
[OUTPUTCHANNELS
];
918 ALfloat wetGainStep
[MAX_SENDS
];
921 ALfloat value
, outsamp
;
922 ALbufferlistitem
*BufferListItem
;
923 ALint64 DataSize64
,DataPos64
;
924 FILTER
*DryFilter
, *WetFilter
[MAX_SENDS
];
925 ALfloat WetSend
[MAX_SENDS
];
929 ALuint DataPosInt
, DataPosFrac
;
930 ALuint Channels
, Bytes
;
932 resampler_t Resampler
;
933 ALuint BuffersPlayed
;
937 if(!(ALSource
=ALContext
->SourceList
))
940 DeviceFreq
= ALContext
->Device
->Frequency
;
942 rampLength
= DeviceFreq
* MIN_RAMP_LENGTH
/ 1000;
943 rampLength
= max(rampLength
, SamplesToDo
);
946 if(ALSource
->state
!= AL_PLAYING
)
948 if((ALSource
=ALSource
->next
) != NULL
)
954 /* Find buffer format */
958 BufferListItem
= ALSource
->queue
;
959 while(BufferListItem
!= NULL
)
962 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
964 Channels
= aluChannelsFromFormat(ALBuffer
->format
);
965 Bytes
= aluBytesFromFormat(ALBuffer
->format
);
966 Frequency
= ALBuffer
->frequency
;
969 BufferListItem
= BufferListItem
->next
;
972 if(ALSource
->NeedsUpdate
)
974 //Only apply 3D calculations for mono buffers
976 CalcSourceParams(ALContext
, ALSource
);
978 CalcNonAttnSourceParams(ALContext
, ALSource
);
979 ALSource
->NeedsUpdate
= AL_FALSE
;
982 /* Get source info */
983 Resampler
= ALSource
->Resampler
;
984 State
= ALSource
->state
;
985 BuffersPlayed
= ALSource
->BuffersPlayed
;
986 DataPosInt
= ALSource
->position
;
987 DataPosFrac
= ALSource
->position_fraction
;
989 /* Compute 18.14 fixed point step */
990 Pitch
= (ALSource
->Params
.Pitch
*Frequency
) / DeviceFreq
;
991 if(Pitch
> (float)MAX_PITCH
) Pitch
= (float)MAX_PITCH
;
992 increment
= (ALint
)(Pitch
*(ALfloat
)(1L<<FRACTIONBITS
));
993 if(increment
<= 0) increment
= (1<<FRACTIONBITS
);
995 if(ALSource
->FirstStart
)
997 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
998 DrySend
[i
] = ALSource
->Params
.DryGains
[i
];
999 for(i
= 0;i
< MAX_SENDS
;i
++)
1000 WetSend
[i
] = ALSource
->Params
.WetGains
[i
];
1004 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1005 DrySend
[i
] = ALSource
->DryGains
[i
];
1006 for(i
= 0;i
< MAX_SENDS
;i
++)
1007 WetSend
[i
] = ALSource
->WetGains
[i
];
1010 DryFilter
= &ALSource
->Params
.iirFilter
;
1011 for(i
= 0;i
< MAX_SENDS
;i
++)
1013 WetFilter
[i
] = &ALSource
->Params
.Send
[i
].iirFilter
;
1014 WetBuffer
[i
] = (ALSource
->Send
[i
].Slot
?
1015 ALSource
->Send
[i
].Slot
->WetBuffer
:
1019 /* Get current buffer queue item */
1020 BufferListItem
= ALSource
->queue
;
1021 for(i
= 0;i
< BuffersPlayed
&& BufferListItem
;i
++)
1022 BufferListItem
= BufferListItem
->next
;
1024 while(State
== AL_PLAYING
&& j
< SamplesToDo
)
1026 ALuint DataSize
= 0;
1031 /* Get buffer info */
1032 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
1034 Data
= ALBuffer
->data
;
1035 DataSize
= ALBuffer
->size
;
1036 DataSize
/= Channels
* Bytes
;
1038 if(DataPosInt
>= DataSize
)
1041 if(BufferListItem
->next
)
1043 ALbuffer
*NextBuf
= BufferListItem
->next
->buffer
;
1044 if(NextBuf
&& NextBuf
->size
)
1046 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1047 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1048 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1051 else if(ALSource
->bLooping
)
1053 ALbuffer
*NextBuf
= ALSource
->queue
->buffer
;
1054 if(NextBuf
&& NextBuf
->size
)
1056 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1057 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1058 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1062 memset(&Data
[DataSize
*Channels
], 0, (BUFFER_PADDING
*Channels
*Bytes
));
1064 /* Compute the gain steps for each output channel */
1065 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1066 dryGainStep
[i
] = (ALSource
->Params
.DryGains
[i
]-DrySend
[i
]) /
1068 for(i
= 0;i
< MAX_SENDS
;i
++)
1069 wetGainStep
[i
] = (ALSource
->Params
.WetGains
[i
]-WetSend
[i
]) /
1072 /* Figure out how many samples we can mix. */
1073 DataSize64
= DataSize
;
1074 DataSize64
<<= FRACTIONBITS
;
1075 DataPos64
= DataPosInt
;
1076 DataPos64
<<= FRACTIONBITS
;
1077 DataPos64
+= DataPosFrac
;
1078 BufferSize
= (ALuint
)((DataSize64
-DataPos64
+(increment
-1)) / increment
);
1080 BufferSize
= min(BufferSize
, (SamplesToDo
-j
));
1082 /* Actual sample mixing loop */
1084 Data
+= DataPosInt
*Channels
;
1086 if(Channels
== 1) /* Mono */
1088 #define DO_MIX(resampler) 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 /* First order interpolator */ \
1097 value = (resampler)(Data[k], Data[k+1], DataPosFrac); \
1099 /* Direct path final mix buffer and panning */ \
1100 outsamp = lpFilter4P(DryFilter, 0, value); \
1101 DryBuffer[j][FRONT_LEFT] += outsamp*DrySend[FRONT_LEFT]; \
1102 DryBuffer[j][FRONT_RIGHT] += outsamp*DrySend[FRONT_RIGHT]; \
1103 DryBuffer[j][SIDE_LEFT] += outsamp*DrySend[SIDE_LEFT]; \
1104 DryBuffer[j][SIDE_RIGHT] += outsamp*DrySend[SIDE_RIGHT]; \
1105 DryBuffer[j][BACK_LEFT] += outsamp*DrySend[BACK_LEFT]; \
1106 DryBuffer[j][BACK_RIGHT] += outsamp*DrySend[BACK_RIGHT]; \
1107 DryBuffer[j][FRONT_CENTER] += outsamp*DrySend[FRONT_CENTER]; \
1108 DryBuffer[j][BACK_CENTER] += outsamp*DrySend[BACK_CENTER]; \
1110 /* Room path final mix buffer and panning */ \
1111 for(i = 0;i < MAX_SENDS;i++) \
1113 outsamp = lpFilter2P(WetFilter[i], 0, value); \
1114 WetBuffer[i][j] += outsamp*WetSend[i]; \
1117 DataPosFrac += increment; \
1118 k += DataPosFrac>>FRACTIONBITS; \
1119 DataPosFrac &= FRACTIONMASK; \
1126 case POINT_RESAMPLER
:
1127 DO_MIX(point
); break;
1128 case LINEAR_RESAMPLER
:
1129 DO_MIX(lerp
); break;
1130 case COSINE_RESAMPLER
:
1131 DO_MIX(cos_lerp
); break;
1138 else if(Channels
== 2 && DuplicateStereo
) /* Stereo */
1140 const int chans
[] = {
1141 FRONT_LEFT
, FRONT_RIGHT
1143 const int chans2
[] = {
1144 BACK_LEFT
, SIDE_LEFT
, BACK_RIGHT
, SIDE_RIGHT
1146 const ALfloat scaler
= 1.0f
/Channels
;
1147 const ALfloat dupscaler
= aluSqrt(1.0f
/3.0f
);
1149 #define DO_MIX(resampler) do { \
1150 while(BufferSize--) \
1152 for(i = 0;i < OUTPUTCHANNELS;i++) \
1153 DrySend[i] += dryGainStep[i]; \
1154 for(i = 0;i < MAX_SENDS;i++) \
1155 WetSend[i] += wetGainStep[i]; \
1157 for(i = 0;i < Channels;i++) \
1159 value = (resampler)(Data[k*Channels + i],Data[(k+1)*Channels + i],\
1161 outsamp = lpFilter2P(DryFilter, chans[i]*2, value) * dupscaler; \
1162 DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \
1163 DryBuffer[j][chans2[i*2+0]] += outsamp*DrySend[chans2[i*2+0]]; \
1164 DryBuffer[j][chans2[i*2+1]] += outsamp*DrySend[chans2[i*2+1]]; \
1165 for(out = 0;out < MAX_SENDS;out++) \
1167 outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
1168 WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
1172 DataPosFrac += increment; \
1173 k += DataPosFrac>>FRACTIONBITS; \
1174 DataPosFrac &= FRACTIONMASK; \
1181 case POINT_RESAMPLER
:
1182 DO_MIX(point
); break;
1183 case LINEAR_RESAMPLER
:
1184 DO_MIX(lerp
); break;
1185 case COSINE_RESAMPLER
:
1186 DO_MIX(cos_lerp
); break;
1193 else if(Channels
== 2) /* Stereo */
1195 const int chans
[] = {
1196 FRONT_LEFT
, FRONT_RIGHT
1198 const ALfloat scaler
= 1.0f
/Channels
;
1200 #define DO_MIX(resampler) do { \
1201 while(BufferSize--) \
1203 for(i = 0;i < OUTPUTCHANNELS;i++) \
1204 DrySend[i] += dryGainStep[i]; \
1205 for(i = 0;i < MAX_SENDS;i++) \
1206 WetSend[i] += wetGainStep[i]; \
1208 for(i = 0;i < Channels;i++) \
1210 value = (resampler)(Data[k*Channels + i],Data[(k+1)*Channels + i],\
1212 outsamp = lpFilter2P(DryFilter, chans[i]*2, value); \
1213 DryBuffer[j][chans[i]] += outsamp*DrySend[chans[i]]; \
1214 for(out = 0;out < MAX_SENDS;out++) \
1216 outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
1217 WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
1221 DataPosFrac += increment; \
1222 k += DataPosFrac>>FRACTIONBITS; \
1223 DataPosFrac &= FRACTIONMASK; \
1230 case POINT_RESAMPLER
:
1231 DO_MIX(point
); break;
1232 case LINEAR_RESAMPLER
:
1233 DO_MIX(lerp
); break;
1234 case COSINE_RESAMPLER
:
1235 DO_MIX(cos_lerp
); break;
1241 else if(Channels
== 4) /* Quad */
1243 const int chans
[] = {
1244 FRONT_LEFT
, FRONT_RIGHT
,
1245 BACK_LEFT
, BACK_RIGHT
1247 const ALfloat scaler
= 1.0f
/Channels
;
1251 case POINT_RESAMPLER
:
1252 DO_MIX(point
); break;
1253 case LINEAR_RESAMPLER
:
1254 DO_MIX(lerp
); break;
1255 case COSINE_RESAMPLER
:
1256 DO_MIX(cos_lerp
); break;
1262 else if(Channels
== 6) /* 5.1 */
1264 const int chans
[] = {
1265 FRONT_LEFT
, FRONT_RIGHT
,
1267 BACK_LEFT
, BACK_RIGHT
1269 const ALfloat scaler
= 1.0f
/Channels
;
1273 case POINT_RESAMPLER
:
1274 DO_MIX(point
); break;
1275 case LINEAR_RESAMPLER
:
1276 DO_MIX(lerp
); break;
1277 case COSINE_RESAMPLER
:
1278 DO_MIX(cos_lerp
); break;
1284 else if(Channels
== 7) /* 6.1 */
1286 const int chans
[] = {
1287 FRONT_LEFT
, FRONT_RIGHT
,
1290 SIDE_LEFT
, SIDE_RIGHT
1292 const ALfloat scaler
= 1.0f
/Channels
;
1296 case POINT_RESAMPLER
:
1297 DO_MIX(point
); break;
1298 case LINEAR_RESAMPLER
:
1299 DO_MIX(lerp
); break;
1300 case COSINE_RESAMPLER
:
1301 DO_MIX(cos_lerp
); break;
1307 else if(Channels
== 8) /* 7.1 */
1309 const int chans
[] = {
1310 FRONT_LEFT
, FRONT_RIGHT
,
1312 BACK_LEFT
, BACK_RIGHT
,
1313 SIDE_LEFT
, SIDE_RIGHT
1315 const ALfloat scaler
= 1.0f
/Channels
;
1319 case POINT_RESAMPLER
:
1320 DO_MIX(point
); break;
1321 case LINEAR_RESAMPLER
:
1322 DO_MIX(lerp
); break;
1323 case COSINE_RESAMPLER
:
1324 DO_MIX(cos_lerp
); break;
1333 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1334 DrySend
[i
] += dryGainStep
[i
]*BufferSize
;
1335 for(i
= 0;i
< MAX_SENDS
;i
++)
1336 WetSend
[i
] += wetGainStep
[i
]*BufferSize
;
1339 DataPosFrac
+= increment
;
1340 k
+= DataPosFrac
>>FRACTIONBITS
;
1341 DataPosFrac
&= FRACTIONMASK
;
1348 /* Handle looping sources */
1349 if(DataPosInt
>= DataSize
)
1351 if(BuffersPlayed
< (ALSource
->BuffersInQueue
-1))
1353 BufferListItem
= BufferListItem
->next
;
1355 DataPosInt
-= DataSize
;
1357 else if(ALSource
->bLooping
)
1359 BufferListItem
= ALSource
->queue
;
1361 if(ALSource
->BuffersInQueue
== 1)
1362 DataPosInt
%= DataSize
;
1364 DataPosInt
-= DataSize
;
1369 BufferListItem
= ALSource
->queue
;
1370 BuffersPlayed
= ALSource
->BuffersInQueue
;
1377 /* Update source info */
1378 ALSource
->state
= State
;
1379 ALSource
->BuffersPlayed
= BuffersPlayed
;
1380 ALSource
->position
= DataPosInt
;
1381 ALSource
->position_fraction
= DataPosFrac
;
1382 ALSource
->Buffer
= BufferListItem
->buffer
;
1384 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1385 ALSource
->DryGains
[i
] = DrySend
[i
];
1386 for(i
= 0;i
< MAX_SENDS
;i
++)
1387 ALSource
->WetGains
[i
] = WetSend
[i
];
1389 ALSource
->FirstStart
= AL_FALSE
;
1391 if((ALSource
=ALSource
->next
) != NULL
)
1392 goto another_source
;
1395 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1397 float (*DryBuffer
)[OUTPUTCHANNELS
];
1398 ALfloat (*Matrix
)[OUTPUTCHANNELS
];
1399 const ALuint
*ChanMap
;
1401 ALeffectslot
*ALEffectSlot
;
1402 ALCcontext
*ALContext
;
1407 #if defined(HAVE_FESETROUND)
1408 fpuState
= fegetround();
1409 fesetround(FE_TOWARDZERO
);
1410 #elif defined(HAVE__CONTROLFP)
1411 fpuState
= _controlfp(0, 0);
1412 _controlfp(_RC_CHOP
, _MCW_RC
);
1417 DryBuffer
= device
->DryBuffer
;
1420 /* Setup variables */
1421 SamplesToDo
= min(size
, BUFFERSIZE
);
1423 /* Clear mixing buffer */
1424 memset(DryBuffer
, 0, SamplesToDo
*OUTPUTCHANNELS
*sizeof(ALfloat
));
1426 SuspendContext(NULL
);
1427 for(c
= 0;c
< device
->NumContexts
;c
++)
1429 ALContext
= device
->Contexts
[c
];
1430 SuspendContext(ALContext
);
1432 MixSomeSources(ALContext
, DryBuffer
, SamplesToDo
);
1434 /* effect slot processing */
1435 ALEffectSlot
= ALContext
->EffectSlotList
;
1438 if(ALEffectSlot
->EffectState
)
1439 ALEffect_Process(ALEffectSlot
->EffectState
, ALEffectSlot
, SamplesToDo
, ALEffectSlot
->WetBuffer
, DryBuffer
);
1441 for(i
= 0;i
< SamplesToDo
;i
++)
1442 ALEffectSlot
->WetBuffer
[i
] = 0.0f
;
1443 ALEffectSlot
= ALEffectSlot
->next
;
1445 ProcessContext(ALContext
);
1447 ProcessContext(NULL
);
1449 //Post processing loop
1450 ChanMap
= device
->DevChannels
;
1451 Matrix
= device
->ChannelMatrix
;
1452 switch(device
->Format
)
1454 #define CHECK_WRITE_FORMAT(bits, type, func) \
1455 case AL_FORMAT_MONO##bits: \
1456 for(i = 0;i < SamplesToDo;i++) \
1459 for(c = 0;c < OUTPUTCHANNELS;c++) \
1460 samp += DryBuffer[i][c] * Matrix[c][FRONT_CENTER]; \
1461 ((type*)buffer)[ChanMap[FRONT_CENTER]] = (func)(samp); \
1462 buffer = ((type*)buffer) + 1; \
1465 case AL_FORMAT_STEREO##bits: \
1468 for(i = 0;i < SamplesToDo;i++) \
1470 float samples[2] = { 0.0f, 0.0f }; \
1471 for(c = 0;c < OUTPUTCHANNELS;c++) \
1473 samples[0] += DryBuffer[i][c]*Matrix[c][FRONT_LEFT]; \
1474 samples[1] += DryBuffer[i][c]*Matrix[c][FRONT_RIGHT]; \
1476 bs2b_cross_feed(device->Bs2b, samples); \
1477 ((type*)buffer)[ChanMap[FRONT_LEFT]] = (func)(samples[0]);\
1478 ((type*)buffer)[ChanMap[FRONT_RIGHT]]= (func)(samples[1]);\
1479 buffer = ((type*)buffer) + 2; \
1484 for(i = 0;i < SamplesToDo;i++) \
1486 static const Channel chans[] = { \
1487 FRONT_LEFT, FRONT_RIGHT \
1489 for(j = 0;j < 2;j++) \
1492 for(c = 0;c < OUTPUTCHANNELS;c++) \
1493 samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
1494 ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
1496 buffer = ((type*)buffer) + 2; \
1500 case AL_FORMAT_QUAD##bits: \
1501 for(i = 0;i < SamplesToDo;i++) \
1503 static const Channel chans[] = { \
1504 FRONT_LEFT, FRONT_RIGHT, \
1505 BACK_LEFT, BACK_RIGHT, \
1507 for(j = 0;j < 4;j++) \
1510 for(c = 0;c < OUTPUTCHANNELS;c++) \
1511 samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
1512 ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
1514 buffer = ((type*)buffer) + 4; \
1517 case AL_FORMAT_51CHN##bits: \
1518 for(i = 0;i < SamplesToDo;i++) \
1520 static const Channel chans[] = { \
1521 FRONT_LEFT, FRONT_RIGHT, \
1522 FRONT_CENTER, LFE, \
1523 BACK_LEFT, BACK_RIGHT, \
1525 for(j = 0;j < 6;j++) \
1528 for(c = 0;c < OUTPUTCHANNELS;c++) \
1529 samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
1530 ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
1532 buffer = ((type*)buffer) + 6; \
1535 case AL_FORMAT_61CHN##bits: \
1536 for(i = 0;i < SamplesToDo;i++) \
1538 static const Channel chans[] = { \
1539 FRONT_LEFT, FRONT_RIGHT, \
1540 FRONT_CENTER, LFE, BACK_CENTER, \
1541 SIDE_LEFT, SIDE_RIGHT, \
1543 for(j = 0;j < 7;j++) \
1546 for(c = 0;c < OUTPUTCHANNELS;c++) \
1547 samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
1548 ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
1550 buffer = ((type*)buffer) + 7; \
1553 case AL_FORMAT_71CHN##bits: \
1554 for(i = 0;i < SamplesToDo;i++) \
1556 static const Channel chans[] = { \
1557 FRONT_LEFT, FRONT_RIGHT, \
1558 FRONT_CENTER, LFE, \
1559 BACK_LEFT, BACK_RIGHT, \
1560 SIDE_LEFT, SIDE_RIGHT \
1562 for(j = 0;j < 8;j++) \
1565 for(c = 0;c < OUTPUTCHANNELS;c++) \
1566 samp += DryBuffer[i][c] * Matrix[c][chans[j]]; \
1567 ((type*)buffer)[ChanMap[chans[j]]] = (func)(samp); \
1569 buffer = ((type*)buffer) + 8; \
1573 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1574 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1575 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
)
1576 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
)
1577 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
)
1578 #undef AL_FORMAT_STEREO32
1579 #undef AL_FORMAT_MONO32
1580 #undef CHECK_WRITE_FORMAT
1586 size
-= SamplesToDo
;
1589 #if defined(HAVE_FESETROUND)
1590 fesetround(fpuState
);
1591 #elif defined(HAVE__CONTROLFP)
1592 _controlfp(fpuState
, 0xfffff);
1596 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1600 SuspendContext(NULL
);
1601 for(i
= 0;i
< device
->NumContexts
;i
++)
1605 SuspendContext(device
->Contexts
[i
]);
1607 source
= device
->Contexts
[i
]->SourceList
;
1610 if(source
->state
== AL_PLAYING
)
1612 source
->state
= AL_STOPPED
;
1613 source
->BuffersPlayed
= source
->BuffersInQueue
;
1614 source
->position
= 0;
1615 source
->position_fraction
= 0;
1617 source
= source
->next
;
1619 ProcessContext(device
->Contexts
[i
]);
1622 device
->Connected
= ALC_FALSE
;
1623 ProcessContext(NULL
);