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
)
67 static __inline ALshort
aluF2S(ALfloat Value
)
73 i
= (ALint
)(Value
*32768.0f
);
78 i
= (ALint
)(Value
*32767.0f
);
84 static __inline ALubyte
aluF2UB(ALfloat Value
)
86 ALshort i
= aluF2S(Value
);
91 static __inline ALvoid
aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
93 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
94 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
95 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
98 static __inline ALfloat
aluDotproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
)
100 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
101 inVector1
[2]*inVector2
[2];
104 static __inline ALvoid
aluNormalize(ALfloat
*inVector
)
106 ALfloat length
, inverse_length
;
108 length
= aluSqrt(aluDotproduct(inVector
, inVector
));
111 inverse_length
= 1.0f
/length
;
112 inVector
[0] *= inverse_length
;
113 inVector
[1] *= inverse_length
;
114 inVector
[2] *= inverse_length
;
118 static __inline ALvoid
aluMatrixVector(ALfloat
*vector
,ALfloat w
,ALfloat matrix
[4][4])
121 vector
[0], vector
[1], vector
[2], w
124 vector
[0] = temp
[0]*matrix
[0][0] + temp
[1]*matrix
[1][0] + temp
[2]*matrix
[2][0] + temp
[3]*matrix
[3][0];
125 vector
[1] = temp
[0]*matrix
[0][1] + temp
[1]*matrix
[1][1] + temp
[2]*matrix
[2][1] + temp
[3]*matrix
[3][1];
126 vector
[2] = temp
[0]*matrix
[0][2] + temp
[1]*matrix
[1][2] + temp
[2]*matrix
[2][2] + temp
[3]*matrix
[3][2];
129 static ALvoid
SetSpeakerArrangement(const char *name
, ALfloat SpeakerAngle
[OUTPUTCHANNELS
],
130 ALint Speaker2Chan
[OUTPUTCHANNELS
], ALint chans
)
138 confkey
= GetConfigValue(NULL
, name
, "");
143 next
= strchr(confkey
, ',');
148 } while(isspace(*next
));
151 sep
= strchr(confkey
, '=');
152 if(!sep
|| confkey
== sep
)
156 while(isspace(*end
) && end
!= confkey
)
160 if(strncmp(confkey
, "fl", end
-confkey
) == 0)
162 else if(strncmp(confkey
, "fr", end
-confkey
) == 0)
164 else if(strncmp(confkey
, "fc", end
-confkey
) == 0)
166 else if(strncmp(confkey
, "bl", end
-confkey
) == 0)
168 else if(strncmp(confkey
, "br", end
-confkey
) == 0)
170 else if(strncmp(confkey
, "bc", end
-confkey
) == 0)
172 else if(strncmp(confkey
, "sl", end
-confkey
) == 0)
174 else if(strncmp(confkey
, "sr", end
-confkey
) == 0)
178 AL_PRINT("Unknown speaker for %s: \"%c%c\"\n", name
, confkey
[0], confkey
[1]);
186 for(i
= 0;i
< chans
;i
++)
188 if(Speaker2Chan
[i
] == val
)
190 val
= strtol(sep
, NULL
, 10);
191 if(val
>= -180 && val
<= 180)
192 SpeakerAngle
[i
] = val
* M_PI
/180.0f
;
194 AL_PRINT("Invalid angle for speaker \"%c%c\": %d\n", confkey
[0], confkey
[1], val
);
200 for(i
= 1;i
< chans
;i
++)
202 if(SpeakerAngle
[i
] <= SpeakerAngle
[i
-1])
204 AL_PRINT("Speaker %d of %d does not follow previous: %f > %f\n", i
, chans
,
205 SpeakerAngle
[i
-1] * 180.0f
/M_PI
, SpeakerAngle
[i
] * 180.0f
/M_PI
);
206 SpeakerAngle
[i
] = SpeakerAngle
[i
-1] + 1 * 180.0f
/M_PI
;
211 static __inline ALfloat
aluLUTpos2Angle(ALint pos
)
213 if(pos
< QUADRANT_NUM
)
214 return aluAtan((ALfloat
)pos
/ (ALfloat
)(QUADRANT_NUM
- pos
));
215 if(pos
< 2 * QUADRANT_NUM
)
216 return M_PI_2
+ aluAtan((ALfloat
)(pos
- QUADRANT_NUM
) / (ALfloat
)(2 * QUADRANT_NUM
- pos
));
217 if(pos
< 3 * QUADRANT_NUM
)
218 return aluAtan((ALfloat
)(pos
- 2 * QUADRANT_NUM
) / (ALfloat
)(3 * QUADRANT_NUM
- pos
)) - M_PI
;
219 return aluAtan((ALfloat
)(pos
- 3 * QUADRANT_NUM
) / (ALfloat
)(4 * QUADRANT_NUM
- pos
)) - M_PI_2
;
222 ALvoid
aluInitPanning(ALCcontext
*Context
)
224 ALint pos
, offset
, s
;
225 ALfloat Alpha
, Theta
;
226 ALfloat SpeakerAngle
[OUTPUTCHANNELS
];
227 ALint Speaker2Chan
[OUTPUTCHANNELS
];
229 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
232 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
233 Context
->ChannelMatrix
[s
][s2
] = ((s
==s2
) ? 1.0f
: 0.0f
);
236 switch(Context
->Device
->Format
)
238 /* Mono is rendered as stereo, then downmixed during post-process */
239 case AL_FORMAT_MONO8
:
240 case AL_FORMAT_MONO16
:
241 case AL_FORMAT_MONO_FLOAT32
:
242 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
243 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
244 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
245 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
246 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
247 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
248 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
249 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
250 Context
->NumChan
= 2;
251 Speaker2Chan
[0] = FRONT_LEFT
;
252 Speaker2Chan
[1] = FRONT_RIGHT
;
253 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
254 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
257 case AL_FORMAT_STEREO8
:
258 case AL_FORMAT_STEREO16
:
259 case AL_FORMAT_STEREO_FLOAT32
:
260 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
261 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
262 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
263 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
264 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
265 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
266 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
267 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
268 Context
->NumChan
= 2;
269 Speaker2Chan
[0] = FRONT_LEFT
;
270 Speaker2Chan
[1] = FRONT_RIGHT
;
271 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
272 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
273 SetSpeakerArrangement("layout_STEREO", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
276 case AL_FORMAT_QUAD8
:
277 case AL_FORMAT_QUAD16
:
278 case AL_FORMAT_QUAD32
:
279 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
280 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
281 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
282 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
283 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
284 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
285 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
286 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
287 Context
->NumChan
= 4;
288 Speaker2Chan
[0] = BACK_LEFT
;
289 Speaker2Chan
[1] = FRONT_LEFT
;
290 Speaker2Chan
[2] = FRONT_RIGHT
;
291 Speaker2Chan
[3] = BACK_RIGHT
;
292 SpeakerAngle
[0] = -135.0f
* M_PI
/180.0f
;
293 SpeakerAngle
[1] = -45.0f
* M_PI
/180.0f
;
294 SpeakerAngle
[2] = 45.0f
* M_PI
/180.0f
;
295 SpeakerAngle
[3] = 135.0f
* M_PI
/180.0f
;
296 SetSpeakerArrangement("layout_QUAD", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
299 case AL_FORMAT_51CHN8
:
300 case AL_FORMAT_51CHN16
:
301 case AL_FORMAT_51CHN32
:
302 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
303 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
304 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
305 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
306 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
307 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
308 Context
->NumChan
= 5;
309 Speaker2Chan
[0] = BACK_LEFT
;
310 Speaker2Chan
[1] = FRONT_LEFT
;
311 Speaker2Chan
[2] = FRONT_CENTER
;
312 Speaker2Chan
[3] = FRONT_RIGHT
;
313 Speaker2Chan
[4] = BACK_RIGHT
;
314 SpeakerAngle
[0] = -110.0f
* M_PI
/180.0f
;
315 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
316 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
317 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
318 SpeakerAngle
[4] = 110.0f
* M_PI
/180.0f
;
319 SetSpeakerArrangement("layout_51CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
322 case AL_FORMAT_61CHN8
:
323 case AL_FORMAT_61CHN16
:
324 case AL_FORMAT_61CHN32
:
325 Context
->ChannelMatrix
[BACK_LEFT
][BACK_CENTER
] = aluSqrt(0.5);
326 Context
->ChannelMatrix
[BACK_LEFT
][SIDE_LEFT
] = aluSqrt(0.5);
327 Context
->ChannelMatrix
[BACK_RIGHT
][BACK_CENTER
] = aluSqrt(0.5);
328 Context
->ChannelMatrix
[BACK_RIGHT
][SIDE_RIGHT
] = aluSqrt(0.5);
329 Context
->NumChan
= 6;
330 Speaker2Chan
[0] = SIDE_LEFT
;
331 Speaker2Chan
[1] = FRONT_LEFT
;
332 Speaker2Chan
[2] = FRONT_CENTER
;
333 Speaker2Chan
[3] = FRONT_RIGHT
;
334 Speaker2Chan
[4] = SIDE_RIGHT
;
335 Speaker2Chan
[5] = BACK_CENTER
;
336 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
337 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
338 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
339 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
340 SpeakerAngle
[4] = 90.0f
* M_PI
/180.0f
;
341 SpeakerAngle
[5] = 180.0f
* M_PI
/180.0f
;
342 SetSpeakerArrangement("layout_61CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
345 case AL_FORMAT_71CHN8
:
346 case AL_FORMAT_71CHN16
:
347 case AL_FORMAT_71CHN32
:
348 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
349 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
350 Context
->NumChan
= 7;
351 Speaker2Chan
[0] = BACK_LEFT
;
352 Speaker2Chan
[1] = SIDE_LEFT
;
353 Speaker2Chan
[2] = FRONT_LEFT
;
354 Speaker2Chan
[3] = FRONT_CENTER
;
355 Speaker2Chan
[4] = FRONT_RIGHT
;
356 Speaker2Chan
[5] = SIDE_RIGHT
;
357 Speaker2Chan
[6] = BACK_RIGHT
;
358 SpeakerAngle
[0] = -150.0f
* M_PI
/180.0f
;
359 SpeakerAngle
[1] = -90.0f
* M_PI
/180.0f
;
360 SpeakerAngle
[2] = -30.0f
* M_PI
/180.0f
;
361 SpeakerAngle
[3] = 0.0f
* M_PI
/180.0f
;
362 SpeakerAngle
[4] = 30.0f
* M_PI
/180.0f
;
363 SpeakerAngle
[5] = 90.0f
* M_PI
/180.0f
;
364 SpeakerAngle
[6] = 150.0f
* M_PI
/180.0f
;
365 SetSpeakerArrangement("layout_71CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
372 for(pos
= 0; pos
< LUT_NUM
; pos
++)
375 Theta
= aluLUTpos2Angle(pos
);
377 /* clear all values */
378 offset
= OUTPUTCHANNELS
* pos
;
379 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
380 Context
->PanningLUT
[offset
+s
] = 0.0f
;
382 /* set panning values */
383 for(s
= 0; s
< Context
->NumChan
- 1; s
++)
385 if(Theta
>= SpeakerAngle
[s
] && Theta
< SpeakerAngle
[s
+1])
387 /* source between speaker s and speaker s+1 */
388 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
389 (SpeakerAngle
[s
+1]-SpeakerAngle
[s
]);
390 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
391 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
+1]] = sin(Alpha
);
395 if(s
== Context
->NumChan
- 1)
397 /* source between last and first speaker */
398 if(Theta
< SpeakerAngle
[0])
399 Theta
+= 2.0f
* M_PI
;
400 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
401 (2.0f
* M_PI
+ SpeakerAngle
[0]-SpeakerAngle
[s
]);
402 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
403 Context
->PanningLUT
[offset
+ Speaker2Chan
[0]] = sin(Alpha
);
408 static ALvoid
CalcSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
,
411 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,DryMix
;
412 ALfloat Direction
[3],Position
[3],SourceToListener
[3];
413 ALfloat Velocity
[3],ListenerVel
[3];
414 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
,OuterGainHF
;
415 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
416 ALfloat DopplerFactor
, DopplerVelocity
, flSpeedOfSound
;
417 ALfloat Matrix
[4][4];
418 ALfloat flAttenuation
;
419 ALfloat RoomAttenuation
[MAX_SENDS
];
420 ALfloat MetersPerUnit
;
421 ALfloat RoomRolloff
[MAX_SENDS
];
422 ALfloat DryGainHF
= 1.0f
;
423 ALfloat WetGain
[MAX_SENDS
];
424 ALfloat WetGainHF
[MAX_SENDS
];
425 ALfloat DirGain
, AmbientGain
;
427 const ALfloat
*SpeakerGain
;
433 for(i
= 0;i
< MAX_SENDS
;i
++)
436 //Get context properties
437 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
438 DopplerVelocity
= ALContext
->DopplerVelocity
;
439 flSpeedOfSound
= ALContext
->flSpeedOfSound
;
440 NumSends
= ALContext
->Device
->NumAuxSends
;
441 Frequency
= ALContext
->Device
->Frequency
;
443 //Get listener properties
444 ListenerGain
= ALContext
->Listener
.Gain
;
445 MetersPerUnit
= ALContext
->Listener
.MetersPerUnit
;
446 memcpy(ListenerVel
, ALContext
->Listener
.Velocity
, sizeof(ALContext
->Listener
.Velocity
));
448 //Get source properties
449 SourceVolume
= ALSource
->flGain
;
450 memcpy(Position
, ALSource
->vPosition
, sizeof(ALSource
->vPosition
));
451 memcpy(Direction
, ALSource
->vOrientation
, sizeof(ALSource
->vOrientation
));
452 memcpy(Velocity
, ALSource
->vVelocity
, sizeof(ALSource
->vVelocity
));
453 MinVolume
= ALSource
->flMinGain
;
454 MaxVolume
= ALSource
->flMaxGain
;
455 MinDist
= ALSource
->flRefDistance
;
456 MaxDist
= ALSource
->flMaxDistance
;
457 Rolloff
= ALSource
->flRollOffFactor
;
458 InnerAngle
= ALSource
->flInnerAngle
;
459 OuterAngle
= ALSource
->flOuterAngle
;
460 OuterGainHF
= ALSource
->OuterGainHF
;
462 //Only apply 3D calculations for mono buffers
463 if(isMono
== AL_FALSE
)
465 //1. Multi-channel buffers always play "normal"
466 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
468 DryMix
= SourceVolume
;
469 DryMix
= __min(DryMix
,MaxVolume
);
470 DryMix
= __max(DryMix
,MinVolume
);
472 switch(ALSource
->DirectFilter
.type
)
474 case AL_FILTER_LOWPASS
:
475 DryMix
*= ALSource
->DirectFilter
.Gain
;
476 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
480 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryMix
* ListenerGain
;
481 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryMix
* ListenerGain
;
482 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryMix
* ListenerGain
;
483 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryMix
* ListenerGain
;
484 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryMix
* ListenerGain
;
485 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryMix
* ListenerGain
;
486 ALSource
->Params
.DryGains
[FRONT_CENTER
] = DryMix
* ListenerGain
;
487 ALSource
->Params
.DryGains
[BACK_CENTER
] = DryMix
* ListenerGain
;
488 ALSource
->Params
.DryGains
[LFE
] = DryMix
* ListenerGain
;
490 for(i
= 0;i
< NumSends
;i
++)
492 WetGain
[i
] = SourceVolume
;
493 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
494 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
497 switch(ALSource
->Send
[i
].WetFilter
.type
)
499 case AL_FILTER_LOWPASS
:
500 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
501 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
505 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
507 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
509 ALSource
->Params
.WetGains
[i
] = 0.0f
;
513 /* Update filter coefficients. Calculations based on the I3DL2
515 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
516 /* We use two chained one-pole filters, so we need to take the
517 * square root of the squared gain, which is the same as the base
519 g
= __max(DryGainHF
, 0.01f
);
521 /* Be careful with gains < 0.0001, as that causes the coefficient
522 * head towards 1, which will flatten the signal */
523 if(g
< 0.9999f
) /* 1-epsilon */
524 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
526 ALSource
->Params
.iirFilter
.coeff
= a
;
528 for(i
= 0;i
< NumSends
;i
++)
530 /* We use a one-pole filter, so we need to take the squared gain */
531 g
= __max(WetGainHF
[i
], 0.1f
);
534 if(g
< 0.9999f
) /* 1-epsilon */
535 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
537 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
543 //1. Translate Listener to origin (convert to head relative)
544 if(ALSource
->bHeadRelative
==AL_FALSE
)
546 ALfloat U
[3],V
[3],N
[3],P
[3];
548 // Build transform matrix
549 memcpy(N
, ALContext
->Listener
.Forward
, sizeof(N
)); // At-vector
550 aluNormalize(N
); // Normalized At-vector
551 memcpy(V
, ALContext
->Listener
.Up
, sizeof(V
)); // Up-vector
552 aluNormalize(V
); // Normalized Up-vector
553 aluCrossproduct(N
, V
, U
); // Right-vector
554 aluNormalize(U
); // Normalized Right-vector
555 P
[0] = -(ALContext
->Listener
.Position
[0]*U
[0] + // Translation
556 ALContext
->Listener
.Position
[1]*U
[1] +
557 ALContext
->Listener
.Position
[2]*U
[2]);
558 P
[1] = -(ALContext
->Listener
.Position
[0]*V
[0] +
559 ALContext
->Listener
.Position
[1]*V
[1] +
560 ALContext
->Listener
.Position
[2]*V
[2]);
561 P
[2] = -(ALContext
->Listener
.Position
[0]*-N
[0] +
562 ALContext
->Listener
.Position
[1]*-N
[1] +
563 ALContext
->Listener
.Position
[2]*-N
[2]);
564 Matrix
[0][0] = U
[0]; Matrix
[0][1] = V
[0]; Matrix
[0][2] = -N
[0]; Matrix
[0][3] = 0.0f
;
565 Matrix
[1][0] = U
[1]; Matrix
[1][1] = V
[1]; Matrix
[1][2] = -N
[1]; Matrix
[1][3] = 0.0f
;
566 Matrix
[2][0] = U
[2]; Matrix
[2][1] = V
[2]; Matrix
[2][2] = -N
[2]; Matrix
[2][3] = 0.0f
;
567 Matrix
[3][0] = P
[0]; Matrix
[3][1] = P
[1]; Matrix
[3][2] = P
[2]; Matrix
[3][3] = 1.0f
;
569 // Transform source position and direction into listener space
570 aluMatrixVector(Position
, 1.0f
, Matrix
);
571 aluMatrixVector(Direction
, 0.0f
, Matrix
);
572 // Transform source and listener velocity into listener space
573 aluMatrixVector(Velocity
, 0.0f
, Matrix
);
574 aluMatrixVector(ListenerVel
, 0.0f
, Matrix
);
577 ListenerVel
[0] = ListenerVel
[1] = ListenerVel
[2] = 0.0f
;
579 SourceToListener
[0] = -Position
[0];
580 SourceToListener
[1] = -Position
[1];
581 SourceToListener
[2] = -Position
[2];
582 aluNormalize(SourceToListener
);
583 aluNormalize(Direction
);
585 //2. Calculate distance attenuation
586 Distance
= aluSqrt(aluDotproduct(Position
, Position
));
588 flAttenuation
= 1.0f
;
589 for(i
= 0;i
< NumSends
;i
++)
591 RoomAttenuation
[i
] = 1.0f
;
593 RoomRolloff
[i
] = ALSource
->RoomRolloffFactor
;
594 if(ALSource
->Send
[i
].Slot
&&
595 (ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
596 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
))
597 RoomRolloff
[i
] += ALSource
->Send
[i
].Slot
->effect
.Reverb
.RoomRolloffFactor
;
600 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
601 ALContext
->DistanceModel
)
603 case AL_INVERSE_DISTANCE_CLAMPED
:
604 Distance
=__max(Distance
,MinDist
);
605 Distance
=__min(Distance
,MaxDist
);
606 if(MaxDist
< MinDist
)
609 case AL_INVERSE_DISTANCE
:
612 if((MinDist
+ (Rolloff
* (Distance
- MinDist
))) > 0.0f
)
613 flAttenuation
= MinDist
/ (MinDist
+ (Rolloff
* (Distance
- MinDist
)));
614 for(i
= 0;i
< NumSends
;i
++)
616 if((MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
))) > 0.0f
)
617 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
)));
622 case AL_LINEAR_DISTANCE_CLAMPED
:
623 Distance
=__max(Distance
,MinDist
);
624 Distance
=__min(Distance
,MaxDist
);
625 if(MaxDist
< MinDist
)
628 case AL_LINEAR_DISTANCE
:
629 Distance
=__min(Distance
,MaxDist
);
630 if(MaxDist
!= MinDist
)
632 flAttenuation
= 1.0f
- (Rolloff
*(Distance
-MinDist
)/(MaxDist
- MinDist
));
633 for(i
= 0;i
< NumSends
;i
++)
634 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(Distance
-MinDist
)/(MaxDist
- MinDist
));
638 case AL_EXPONENT_DISTANCE_CLAMPED
:
639 Distance
=__max(Distance
,MinDist
);
640 Distance
=__min(Distance
,MaxDist
);
641 if(MaxDist
< MinDist
)
644 case AL_EXPONENT_DISTANCE
:
645 if(Distance
> 0.0f
&& MinDist
> 0.0f
)
647 flAttenuation
= (ALfloat
)pow(Distance
/MinDist
, -Rolloff
);
648 for(i
= 0;i
< NumSends
;i
++)
649 RoomAttenuation
[i
] = (ALfloat
)pow(Distance
/MinDist
, -RoomRolloff
[i
]);
657 // Source Gain + Attenuation
658 DryMix
= SourceVolume
* flAttenuation
;
659 for(i
= 0;i
< NumSends
;i
++)
660 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
662 // Distance-based air absorption
663 if(ALSource
->AirAbsorptionFactor
> 0.0f
&& flAttenuation
< 1.0f
)
665 ALfloat absorb
= 0.0f
;
667 // Absorption calculation is done in dB
668 if(flAttenuation
> 0.0f
)
670 absorb
= (MinDist
/flAttenuation
- MinDist
)*MetersPerUnit
*
671 (ALSource
->AirAbsorptionFactor
*AIRABSORBGAINDBHF
);
672 // Convert dB to linear gain before applying
673 absorb
= pow(10.0, absorb
/20.0);
676 for(i
= 0;i
< NumSends
;i
++)
677 WetGainHF
[i
] *= absorb
;
680 //3. Apply directional soundcones
681 Angle
= aluAcos(aluDotproduct(Direction
,SourceToListener
)) * 180.0f
/M_PI
;
682 if(Angle
>= InnerAngle
&& Angle
<= OuterAngle
)
684 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
685 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
)*scale
);
686 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
)*scale
);
687 DryMix
*= ConeVolume
;
688 if(ALSource
->DryGainHFAuto
)
691 else if(Angle
> OuterAngle
)
693 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
));
694 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
));
695 DryMix
*= ConeVolume
;
696 if(ALSource
->DryGainHFAuto
)
705 // Clamp to Min/Max Gain
706 DryMix
= __min(DryMix
,MaxVolume
);
707 DryMix
= __max(DryMix
,MinVolume
);
709 for(i
= 0;i
< NumSends
;i
++)
711 if(ALSource
->Send
[i
].Slot
&&
712 ALSource
->Send
[i
].Slot
->effect
.type
!= AL_EFFECT_NULL
)
714 if(ALSource
->Send
[i
].Slot
->AuxSendAuto
)
716 if(ALSource
->WetGainAuto
)
717 WetGain
[i
] *= ConeVolume
;
718 if(ALSource
->WetGainHFAuto
)
719 WetGainHF
[i
] *= ConeHF
;
721 // Clamp to Min/Max Gain
722 WetGain
[i
] = __min(WetGain
[i
],MaxVolume
);
723 WetGain
[i
] = __max(WetGain
[i
],MinVolume
);
725 if(ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
||
726 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_EAXREVERB
)
728 /* Apply a decay-time transformation to the wet path,
729 * based on the attenuation of the dry path. This should
730 * better approximate the statistical attenuation model
731 * for the reverb effect.
733 * This simple equation converts the distance attenuation
734 * into the time it would take to reach -60 dB. From
735 * there it establishes an origin (0.333s; the decay time
736 * that will produce equal attenuation) and applies the
737 * current decay time. Finally, it converts the result
738 * back to an attenuation for the reverb path.
740 WetGain
[i
] *= pow(10.0f
, log10(flAttenuation
) * 0.333f
/
741 ALSource
->Send
[i
].Slot
->effect
.Reverb
.DecayTime
);
746 // If the slot's auxiliary send auto is off, the data sent to
747 // the effect slot is the same as the dry path, sans filter
750 WetGainHF
[i
] = DryGainHF
;
753 switch(ALSource
->Send
[i
].WetFilter
.type
)
755 case AL_FILTER_LOWPASS
:
756 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
757 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
760 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
764 ALSource
->Params
.WetGains
[i
] = 0.0f
;
768 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
770 ALSource
->Params
.WetGains
[i
] = 0.0f
;
774 // Apply filter gains and filters
775 switch(ALSource
->DirectFilter
.type
)
777 case AL_FILTER_LOWPASS
:
778 DryMix
*= ALSource
->DirectFilter
.Gain
;
779 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
782 DryMix
*= ListenerGain
;
784 // Calculate Velocity
785 if(DopplerFactor
!= 0.0f
)
787 ALfloat flVSS
, flVLS
;
788 ALfloat flMaxVelocity
= (DopplerVelocity
* flSpeedOfSound
) /
791 flVSS
= aluDotproduct(Velocity
, SourceToListener
);
792 if(flVSS
>= flMaxVelocity
)
793 flVSS
= (flMaxVelocity
- 1.0f
);
794 else if(flVSS
<= -flMaxVelocity
)
795 flVSS
= -flMaxVelocity
+ 1.0f
;
797 flVLS
= aluDotproduct(ListenerVel
, SourceToListener
);
798 if(flVLS
>= flMaxVelocity
)
799 flVLS
= (flMaxVelocity
- 1.0f
);
800 else if(flVLS
<= -flMaxVelocity
)
801 flVLS
= -flMaxVelocity
+ 1.0f
;
803 ALSource
->Params
.Pitch
= ALSource
->flPitch
*
804 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVLS
)) /
805 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVSS
));
808 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
810 // Use energy-preserving panning algorithm for multi-speaker playback
811 length
= aluSqrt(Position
[0]*Position
[0] + Position
[1]*Position
[1] +
812 Position
[2]*Position
[2]);
813 length
= __max(length
, MinDist
);
816 ALfloat invlen
= 1.0f
/length
;
817 Position
[0] *= invlen
;
818 Position
[1] *= invlen
;
819 Position
[2] *= invlen
;
822 pos
= aluCart2LUTpos(-Position
[2], Position
[0]);
823 SpeakerGain
= &ALContext
->PanningLUT
[OUTPUTCHANNELS
* pos
];
825 DirGain
= aluSqrt(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
826 // elevation adjustment for directional gain. this sucks, but
827 // has low complexity
828 AmbientGain
= 1.0/aluSqrt(ALContext
->NumChan
) * (1.0-DirGain
);
829 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
831 ALfloat gain
= SpeakerGain
[s
]*DirGain
+ AmbientGain
;
832 ALSource
->Params
.DryGains
[s
] = DryMix
* gain
;
835 /* Update filter coefficients. */
836 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
837 /* Spatialized sources use four chained one-pole filters, so we need to
838 * take the fourth root of the squared gain, which is the same as the
839 * square root of the base gain. */
840 g
= aluSqrt(__max(DryGainHF
, 0.0001f
));
842 if(g
< 0.9999f
) /* 1-epsilon */
843 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
845 ALSource
->Params
.iirFilter
.coeff
= a
;
847 for(i
= 0;i
< NumSends
;i
++)
849 /* The wet path uses two chained one-pole filters, so take the
850 * base gain (square root of the squared gain) */
851 g
= __max(WetGainHF
[i
], 0.01f
);
853 if(g
< 0.9999f
) /* 1-epsilon */
854 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
856 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
860 static __inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALint frac
)
862 return val1
+ ((val2
-val1
)*(frac
* (1.0f
/(1<<FRACTIONBITS
))));
865 static void MixSomeSources(ALCcontext
*ALContext
, float (*DryBuffer
)[OUTPUTCHANNELS
], ALuint SamplesToDo
)
867 static float DummyBuffer
[BUFFERSIZE
];
868 ALfloat
*WetBuffer
[MAX_SENDS
];
869 ALfloat (*Matrix
)[OUTPUTCHANNELS
] = ALContext
->ChannelMatrix
;
870 ALfloat DrySend
[OUTPUTCHANNELS
];
871 ALfloat dryGainStep
[OUTPUTCHANNELS
];
872 ALfloat wetGainStep
[MAX_SENDS
];
875 ALfloat value
, outsamp
;
876 ALbufferlistitem
*BufferListItem
;
877 ALint64 DataSize64
,DataPos64
;
878 FILTER
*DryFilter
, *WetFilter
[MAX_SENDS
];
879 ALfloat WetSend
[MAX_SENDS
];
883 ALuint DataPosInt
, DataPosFrac
;
884 ALuint Channels
, Bytes
;
886 ALuint BuffersPlayed
;
890 if(!(ALSource
=ALContext
->Source
))
893 DeviceFreq
= ALContext
->Device
->Frequency
;
895 rampLength
= DeviceFreq
* MIN_RAMP_LENGTH
/ 1000;
896 rampLength
= max(rampLength
, SamplesToDo
);
899 State
= ALSource
->state
;
900 if(State
!= AL_PLAYING
)
902 if((ALSource
=ALSource
->next
) != NULL
)
908 /* Find buffer format */
912 BufferListItem
= ALSource
->queue
;
913 while(BufferListItem
!= NULL
)
916 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
918 Channels
= aluChannelsFromFormat(ALBuffer
->format
);
919 Bytes
= aluBytesFromFormat(ALBuffer
->format
);
920 Frequency
= ALBuffer
->frequency
;
923 BufferListItem
= BufferListItem
->next
;
926 /* Get source info */
927 BuffersPlayed
= ALSource
->BuffersPlayed
;
928 DataPosInt
= ALSource
->position
;
929 DataPosFrac
= ALSource
->position_fraction
;
931 if(ALSource
->NeedsUpdate
)
933 CalcSourceParams(ALContext
, ALSource
, (Channels
==1)?AL_TRUE
:AL_FALSE
);
934 ALSource
->NeedsUpdate
= AL_FALSE
;
937 /* Compute 18.14 fixed point step */
938 Pitch
= (ALSource
->Params
.Pitch
*Frequency
) / DeviceFreq
;
939 if(Pitch
> (float)MAX_PITCH
) Pitch
= (float)MAX_PITCH
;
940 increment
= (ALint
)(Pitch
*(ALfloat
)(1L<<FRACTIONBITS
));
941 if(increment
<= 0) increment
= (1<<FRACTIONBITS
);
943 /* Compute the gain steps for each output channel */
944 if(ALSource
->FirstStart
)
946 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
947 DrySend
[i
] = ALSource
->Params
.DryGains
[i
];
948 for(i
= 0;i
< MAX_SENDS
;i
++)
949 WetSend
[i
] = ALSource
->Params
.WetGains
[i
];
953 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
954 DrySend
[i
] = ALSource
->DryGains
[i
];
955 for(i
= 0;i
< MAX_SENDS
;i
++)
956 WetSend
[i
] = ALSource
->WetGains
[i
];
959 DryFilter
= &ALSource
->Params
.iirFilter
;
960 for(i
= 0;i
< MAX_SENDS
;i
++)
962 WetFilter
[i
] = &ALSource
->Params
.Send
[i
].iirFilter
;
963 WetBuffer
[i
] = (ALSource
->Send
[i
].Slot
?
964 ALSource
->Send
[i
].Slot
->WetBuffer
:
968 if(DuplicateStereo
&& Channels
== 2)
970 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 1.0f
;
971 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 1.0f
;
972 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 1.0f
;
973 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 1.0f
;
975 else if(DuplicateStereo
)
977 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 0.0f
;
978 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 0.0f
;
979 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 0.0f
;
980 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 0.0f
;
983 /* Get current buffer queue item */
984 BufferListItem
= ALSource
->queue
;
985 for(i
= 0;i
< BuffersPlayed
&& BufferListItem
;i
++)
986 BufferListItem
= BufferListItem
->next
;
988 while(State
== AL_PLAYING
&& j
< SamplesToDo
)
995 /* Get buffer info */
996 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
998 Data
= ALBuffer
->data
;
999 DataSize
= ALBuffer
->size
;
1000 DataSize
/= Channels
* Bytes
;
1002 if(DataPosInt
>= DataSize
)
1005 if(BufferListItem
->next
)
1007 ALbuffer
*NextBuf
= BufferListItem
->next
->buffer
;
1008 if(NextBuf
&& NextBuf
->data
)
1010 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1011 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1012 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1015 else if(ALSource
->bLooping
)
1017 ALbuffer
*NextBuf
= ALSource
->queue
->buffer
;
1018 if(NextBuf
&& NextBuf
->data
)
1020 ALint ulExtraSamples
= BUFFER_PADDING
*Channels
*Bytes
;
1021 ulExtraSamples
= min(NextBuf
->size
, ulExtraSamples
);
1022 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
1026 memset(&Data
[DataSize
*Channels
], 0, (BUFFER_PADDING
*Channels
*Bytes
));
1028 /* Compute the gain steps for each output channel */
1029 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1030 dryGainStep
[i
] = (ALSource
->Params
.DryGains
[i
]-
1031 DrySend
[i
]) / rampLength
;
1032 for(i
= 0;i
< MAX_SENDS
;i
++)
1033 wetGainStep
[i
] = (ALSource
->Params
.WetGains
[i
]-
1034 WetSend
[i
]) / rampLength
;
1036 /* Figure out how many samples we can mix. */
1037 DataSize64
= DataSize
;
1038 DataSize64
<<= FRACTIONBITS
;
1039 DataPos64
= DataPosInt
;
1040 DataPos64
<<= FRACTIONBITS
;
1041 DataPos64
+= DataPosFrac
;
1042 BufferSize
= (ALuint
)((DataSize64
-DataPos64
+(increment
-1)) / increment
);
1044 BufferSize
= min(BufferSize
, (SamplesToDo
-j
));
1046 /* Actual sample mixing loop */
1048 Data
+= DataPosInt
*Channels
;
1050 if(Channels
== 1) /* Mono */
1054 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1055 DrySend
[i
] += dryGainStep
[i
];
1056 for(i
= 0;i
< MAX_SENDS
;i
++)
1057 WetSend
[i
] += wetGainStep
[i
];
1059 /* First order interpolator */
1060 value
= lerp(Data
[k
], Data
[k
+1], DataPosFrac
);
1062 /* Direct path final mix buffer and panning */
1063 outsamp
= lpFilter4P(DryFilter
, 0, value
);
1064 DryBuffer
[j
][FRONT_LEFT
] += outsamp
*DrySend
[FRONT_LEFT
];
1065 DryBuffer
[j
][FRONT_RIGHT
] += outsamp
*DrySend
[FRONT_RIGHT
];
1066 DryBuffer
[j
][SIDE_LEFT
] += outsamp
*DrySend
[SIDE_LEFT
];
1067 DryBuffer
[j
][SIDE_RIGHT
] += outsamp
*DrySend
[SIDE_RIGHT
];
1068 DryBuffer
[j
][BACK_LEFT
] += outsamp
*DrySend
[BACK_LEFT
];
1069 DryBuffer
[j
][BACK_RIGHT
] += outsamp
*DrySend
[BACK_RIGHT
];
1070 DryBuffer
[j
][FRONT_CENTER
] += outsamp
*DrySend
[FRONT_CENTER
];
1071 DryBuffer
[j
][BACK_CENTER
] += outsamp
*DrySend
[BACK_CENTER
];
1073 /* Room path final mix buffer and panning */
1074 for(i
= 0;i
< MAX_SENDS
;i
++)
1076 outsamp
= lpFilter2P(WetFilter
[i
], 0, value
);
1077 WetBuffer
[i
][j
] += outsamp
*WetSend
[i
];
1080 DataPosFrac
+= increment
;
1081 k
+= DataPosFrac
>>FRACTIONBITS
;
1082 DataPosFrac
&= FRACTIONMASK
;
1086 else if(Channels
== 2) /* Stereo */
1088 const int chans
[] = {
1089 FRONT_LEFT
, FRONT_RIGHT
1091 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1093 #define DO_MIX() do { \
1094 while(BufferSize--) \
1096 for(i = 0;i < OUTPUTCHANNELS;i++) \
1097 DrySend[i] += dryGainStep[i]; \
1098 for(i = 0;i < MAX_SENDS;i++) \
1099 WetSend[i] += wetGainStep[i]; \
1101 for(i = 0;i < Channels;i++) \
1103 value = lerp(Data[k*Channels + i], Data[(k+1)*Channels + i], DataPosFrac); \
1104 outsamp = lpFilter2P(DryFilter, chans[i]*2, value)*DrySend[chans[i]]; \
1105 for(out = 0;out < OUTPUTCHANNELS;out++) \
1106 DryBuffer[j][out] += outsamp*Matrix[chans[i]][out]; \
1107 for(out = 0;out < MAX_SENDS;out++) \
1109 outsamp = lpFilter1P(WetFilter[out], chans[i], value); \
1110 WetBuffer[out][j] += outsamp*WetSend[out]*scaler; \
1114 DataPosFrac += increment; \
1115 k += DataPosFrac>>FRACTIONBITS; \
1116 DataPosFrac &= FRACTIONMASK; \
1123 else if(Channels
== 4) /* Quad */
1125 const int chans
[] = {
1126 FRONT_LEFT
, FRONT_RIGHT
,
1127 BACK_LEFT
, BACK_RIGHT
1129 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1133 else if(Channels
== 6) /* 5.1 */
1135 const int chans
[] = {
1136 FRONT_LEFT
, FRONT_RIGHT
,
1138 BACK_LEFT
, BACK_RIGHT
1140 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1144 else if(Channels
== 7) /* 6.1 */
1146 const int chans
[] = {
1147 FRONT_LEFT
, FRONT_RIGHT
,
1150 SIDE_LEFT
, SIDE_RIGHT
1152 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1156 else if(Channels
== 8) /* 7.1 */
1158 const int chans
[] = {
1159 FRONT_LEFT
, FRONT_RIGHT
,
1161 BACK_LEFT
, BACK_RIGHT
,
1162 SIDE_LEFT
, SIDE_RIGHT
1164 const ALfloat scaler
= aluSqrt(1.0f
/Channels
);
1171 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1172 DrySend
[i
] += dryGainStep
[i
]*BufferSize
;
1173 for(i
= 0;i
< MAX_SENDS
;i
++)
1174 WetSend
[i
] += wetGainStep
[i
]*BufferSize
;
1177 DataPosFrac
+= increment
;
1178 k
+= DataPosFrac
>>FRACTIONBITS
;
1179 DataPosFrac
&= FRACTIONMASK
;
1186 /* Handle looping sources */
1187 if(DataPosInt
>= DataSize
)
1189 if(BuffersPlayed
< (ALSource
->BuffersInQueue
-1))
1191 BufferListItem
= BufferListItem
->next
;
1193 DataPosInt
-= DataSize
;
1197 if(!ALSource
->bLooping
)
1200 BufferListItem
= ALSource
->queue
;
1201 BuffersPlayed
= ALSource
->BuffersInQueue
;
1207 BufferListItem
= ALSource
->queue
;
1209 if(ALSource
->BuffersInQueue
== 1)
1210 DataPosInt
%= DataSize
;
1212 DataPosInt
-= DataSize
;
1218 /* Update source info */
1219 ALSource
->state
= State
;
1220 ALSource
->BuffersPlayed
= BuffersPlayed
;
1221 ALSource
->position
= DataPosInt
;
1222 ALSource
->position_fraction
= DataPosFrac
;
1223 ALSource
->Buffer
= BufferListItem
->buffer
;
1225 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1226 ALSource
->DryGains
[i
] = DrySend
[i
];
1227 for(i
= 0;i
< MAX_SENDS
;i
++)
1228 ALSource
->WetGains
[i
] = WetSend
[i
];
1230 ALSource
->FirstStart
= AL_FALSE
;
1232 if((ALSource
=ALSource
->next
) != NULL
)
1233 goto another_source
;
1236 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1238 float (*DryBuffer
)[OUTPUTCHANNELS
];
1239 const Channel
*ChanMap
;
1241 ALeffectslot
*ALEffectSlot
;
1242 ALCcontext
*ALContext
;
1246 SuspendContext(NULL
);
1248 #if defined(HAVE_FESETROUND)
1249 fpuState
= fegetround();
1250 fesetround(FE_TOWARDZERO
);
1251 #elif defined(HAVE__CONTROLFP)
1252 fpuState
= _controlfp(0, 0);
1253 _controlfp(_RC_CHOP
, _MCW_RC
);
1258 DryBuffer
= device
->DryBuffer
;
1261 /* Setup variables */
1262 SamplesToDo
= min(size
, BUFFERSIZE
);
1264 /* Clear mixing buffer */
1265 memset(DryBuffer
, 0, SamplesToDo
*OUTPUTCHANNELS
*sizeof(ALfloat
));
1267 for(c
= 0;c
< device
->NumContexts
;c
++)
1269 ALContext
= device
->Contexts
[c
];
1270 SuspendContext(ALContext
);
1272 MixSomeSources(ALContext
, DryBuffer
, SamplesToDo
);
1274 /* effect slot processing */
1275 ALEffectSlot
= ALContext
->AuxiliaryEffectSlot
;
1278 if(ALEffectSlot
->EffectState
)
1279 ALEffect_Process(ALEffectSlot
->EffectState
, ALEffectSlot
, SamplesToDo
, ALEffectSlot
->WetBuffer
, DryBuffer
);
1281 for(i
= 0;i
< SamplesToDo
;i
++)
1282 ALEffectSlot
->WetBuffer
[i
] = 0.0f
;
1283 ALEffectSlot
= ALEffectSlot
->next
;
1285 ProcessContext(ALContext
);
1288 //Post processing loop
1289 ChanMap
= device
->DevChannels
;
1290 switch(device
->Format
)
1292 #define CHECK_WRITE_FORMAT(bits, type, func) \
1293 case AL_FORMAT_MONO##bits: \
1294 for(i = 0;i < SamplesToDo;i++) \
1296 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]] + \
1297 DryBuffer[i][ChanMap[1]]); \
1298 buffer = ((type*)buffer) + 1; \
1301 case AL_FORMAT_STEREO##bits: \
1304 for(i = 0;i < SamplesToDo;i++) \
1307 samples[0] = DryBuffer[i][ChanMap[0]]; \
1308 samples[1] = DryBuffer[i][ChanMap[1]]; \
1309 bs2b_cross_feed(device->Bs2b, samples); \
1310 ((type*)buffer)[0] = (func)(samples[0]); \
1311 ((type*)buffer)[1] = (func)(samples[1]); \
1312 buffer = ((type*)buffer) + 2; \
1317 for(i = 0;i < SamplesToDo;i++) \
1319 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1320 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1321 buffer = ((type*)buffer) + 2; \
1325 case AL_FORMAT_QUAD##bits: \
1326 for(i = 0;i < SamplesToDo;i++) \
1328 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1329 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1330 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1331 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1332 buffer = ((type*)buffer) + 4; \
1335 case AL_FORMAT_51CHN##bits: \
1336 for(i = 0;i < SamplesToDo;i++) \
1338 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1339 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1340 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1341 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1342 ((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
1343 ((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
1344 buffer = ((type*)buffer) + 6; \
1347 case AL_FORMAT_61CHN##bits: \
1348 for(i = 0;i < SamplesToDo;i++) \
1350 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1351 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1352 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1353 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1354 ((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
1355 ((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
1356 ((type*)buffer)[6] = (func)(DryBuffer[i][ChanMap[6]]); \
1357 buffer = ((type*)buffer) + 7; \
1360 case AL_FORMAT_71CHN##bits: \
1361 for(i = 0;i < SamplesToDo;i++) \
1363 ((type*)buffer)[0] = (func)(DryBuffer[i][ChanMap[0]]); \
1364 ((type*)buffer)[1] = (func)(DryBuffer[i][ChanMap[1]]); \
1365 ((type*)buffer)[2] = (func)(DryBuffer[i][ChanMap[2]]); \
1366 ((type*)buffer)[3] = (func)(DryBuffer[i][ChanMap[3]]); \
1367 ((type*)buffer)[4] = (func)(DryBuffer[i][ChanMap[4]]); \
1368 ((type*)buffer)[5] = (func)(DryBuffer[i][ChanMap[5]]); \
1369 ((type*)buffer)[6] = (func)(DryBuffer[i][ChanMap[6]]); \
1370 ((type*)buffer)[7] = (func)(DryBuffer[i][ChanMap[7]]); \
1371 buffer = ((type*)buffer) + 8; \
1375 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1376 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1377 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
)
1378 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
)
1379 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
)
1380 #undef AL_FORMAT_STEREO32
1381 #undef AL_FORMAT_MONO32
1382 #undef CHECK_WRITE_FORMAT
1388 size
-= SamplesToDo
;
1391 #if defined(HAVE_FESETROUND)
1392 fesetround(fpuState
);
1393 #elif defined(HAVE__CONTROLFP)
1394 _controlfp(fpuState
, 0xfffff);
1397 ProcessContext(NULL
);
1400 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1404 SuspendContext(NULL
);
1405 for(i
= 0;i
< device
->NumContexts
;i
++)
1409 SuspendContext(device
->Contexts
[i
]);
1411 source
= device
->Contexts
[i
]->Source
;
1414 if(source
->state
== AL_PLAYING
)
1416 source
->state
= AL_STOPPED
;
1417 source
->BuffersPlayed
= source
->BuffersInQueue
;
1418 source
->position
= 0;
1419 source
->position_fraction
= 0;
1421 source
= source
->next
;
1423 ProcessContext(device
->Contexts
[i
]);
1426 device
->Connected
= ALC_FALSE
;
1427 ProcessContext(NULL
);