2 * OpenAL cross platform audio library
3 * Copyright (C) 1999-2007 by authors.
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
35 #include "alListener.h"
36 #include "alAuxEffectSlot.h"
40 #if defined(HAVE_STDINT_H)
42 typedef int64_t ALint64
;
43 #elif defined(HAVE___INT64)
44 typedef __int64 ALint64
;
45 #elif (SIZEOF_LONG == 8)
47 #elif (SIZEOF_LONG_LONG == 8)
48 typedef long long ALint64
;
51 #define FRACTIONBITS 14
52 #define FRACTIONMASK ((1L<<FRACTIONBITS)-1)
53 #define MAX_PITCH 65536
55 /* Minimum ramp length in milliseconds. The value below was chosen to
56 * adequately reduce clicks and pops from harsh gain changes. */
57 #define MIN_RAMP_LENGTH 16
59 ALboolean DuplicateStereo
= AL_FALSE
;
62 static __inline ALfloat
aluF2F(ALfloat Value
)
64 if(Value
< 0.f
) return Value
/32768.f
;
65 if(Value
> 0.f
) return Value
/32767.f
;
69 static __inline ALshort
aluF2S(ALfloat Value
)
79 static __inline ALubyte
aluF2UB(ALfloat Value
)
81 ALshort i
= aluF2S(Value
);
86 static __inline ALvoid
aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
88 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
89 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
90 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
93 static __inline ALfloat
aluDotproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
)
95 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
96 inVector1
[2]*inVector2
[2];
99 static __inline ALvoid
aluNormalize(ALfloat
*inVector
)
101 ALfloat length
, inverse_length
;
103 length
= aluSqrt(aluDotproduct(inVector
, inVector
));
106 inverse_length
= 1.0f
/length
;
107 inVector
[0] *= inverse_length
;
108 inVector
[1] *= inverse_length
;
109 inVector
[2] *= inverse_length
;
113 static __inline ALvoid
aluMatrixVector(ALfloat
*vector
,ALfloat matrix
[3][3])
117 result
[0] = vector
[0]*matrix
[0][0] + vector
[1]*matrix
[1][0] + vector
[2]*matrix
[2][0];
118 result
[1] = vector
[0]*matrix
[0][1] + vector
[1]*matrix
[1][1] + vector
[2]*matrix
[2][1];
119 result
[2] = vector
[0]*matrix
[0][2] + vector
[1]*matrix
[1][2] + vector
[2]*matrix
[2][2];
120 memcpy(vector
, result
, sizeof(result
));
123 static ALvoid
SetSpeakerArrangement(const char *name
, ALfloat SpeakerAngle
[OUTPUTCHANNELS
],
124 ALint Speaker2Chan
[OUTPUTCHANNELS
], ALint chans
)
132 confkey
= GetConfigValue(NULL
, name
, "");
137 next
= strchr(confkey
, ',');
142 } while(isspace(*next
));
145 sep
= strchr(confkey
, '=');
146 if(!sep
|| confkey
== sep
)
150 while(isspace(*end
) && end
!= confkey
)
154 if(strncmp(confkey
, "fl", end
-confkey
) == 0)
156 else if(strncmp(confkey
, "fr", end
-confkey
) == 0)
158 else if(strncmp(confkey
, "fc", end
-confkey
) == 0)
160 else if(strncmp(confkey
, "bl", end
-confkey
) == 0)
162 else if(strncmp(confkey
, "br", end
-confkey
) == 0)
164 else if(strncmp(confkey
, "bc", end
-confkey
) == 0)
166 else if(strncmp(confkey
, "sl", end
-confkey
) == 0)
168 else if(strncmp(confkey
, "sr", end
-confkey
) == 0)
172 AL_PRINT("Unknown speaker for %s: \"%c%c\"\n", name
, confkey
[0], confkey
[1]);
180 for(i
= 0;i
< chans
;i
++)
182 if(Speaker2Chan
[i
] == val
)
184 val
= strtol(sep
, NULL
, 10);
185 if(val
>= -180 && val
<= 180)
186 SpeakerAngle
[i
] = val
* M_PI
/180.0f
;
188 AL_PRINT("Invalid angle for speaker \"%c%c\": %d\n", confkey
[0], confkey
[1], val
);
194 for(i
= 1;i
< chans
;i
++)
196 if(SpeakerAngle
[i
] <= SpeakerAngle
[i
-1])
198 AL_PRINT("Speaker %d of %d does not follow previous: %f > %f\n", i
, chans
,
199 SpeakerAngle
[i
-1] * 180.0f
/M_PI
, SpeakerAngle
[i
] * 180.0f
/M_PI
);
200 SpeakerAngle
[i
] = SpeakerAngle
[i
-1] + 1 * 180.0f
/M_PI
;
205 static __inline ALfloat
aluLUTpos2Angle(ALint pos
)
207 if(pos
< QUADRANT_NUM
)
208 return aluAtan((ALfloat
)pos
/ (ALfloat
)(QUADRANT_NUM
- pos
));
209 if(pos
< 2 * QUADRANT_NUM
)
210 return M_PI_2
+ aluAtan((ALfloat
)(pos
- QUADRANT_NUM
) / (ALfloat
)(2 * QUADRANT_NUM
- pos
));
211 if(pos
< 3 * QUADRANT_NUM
)
212 return aluAtan((ALfloat
)(pos
- 2 * QUADRANT_NUM
) / (ALfloat
)(3 * QUADRANT_NUM
- pos
)) - M_PI
;
213 return aluAtan((ALfloat
)(pos
- 3 * QUADRANT_NUM
) / (ALfloat
)(4 * QUADRANT_NUM
- pos
)) - M_PI_2
;
216 ALvoid
aluInitPanning(ALCcontext
*Context
)
218 ALint pos
, offset
, s
;
219 ALfloat Alpha
, Theta
;
220 ALfloat SpeakerAngle
[OUTPUTCHANNELS
];
221 ALint Speaker2Chan
[OUTPUTCHANNELS
];
223 for(s
= 0;s
< OUTPUTCHANNELS
;s
++)
226 for(s2
= 0;s2
< OUTPUTCHANNELS
;s2
++)
227 Context
->ChannelMatrix
[s
][s2
] = ((s
==s2
) ? 1.0f
: 0.0f
);
230 switch(Context
->Device
->Format
)
232 /* Mono is rendered as stereo, then downmixed during post-process */
233 case AL_FORMAT_MONO8
:
234 case AL_FORMAT_MONO16
:
235 case AL_FORMAT_MONO_FLOAT32
:
236 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
237 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
238 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
239 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
240 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
241 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
242 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
243 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
244 Context
->NumChan
= 2;
245 Speaker2Chan
[0] = FRONT_LEFT
;
246 Speaker2Chan
[1] = FRONT_RIGHT
;
247 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
248 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
251 case AL_FORMAT_STEREO8
:
252 case AL_FORMAT_STEREO16
:
253 case AL_FORMAT_STEREO_FLOAT32
:
254 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
255 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
256 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = 1.0f
;
257 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = 1.0f
;
258 Context
->ChannelMatrix
[BACK_LEFT
][FRONT_LEFT
] = 1.0f
;
259 Context
->ChannelMatrix
[BACK_RIGHT
][FRONT_RIGHT
] = 1.0f
;
260 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
261 Context
->ChannelMatrix
[BACK_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
262 Context
->NumChan
= 2;
263 Speaker2Chan
[0] = FRONT_LEFT
;
264 Speaker2Chan
[1] = FRONT_RIGHT
;
265 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
266 SpeakerAngle
[1] = 90.0f
* M_PI
/180.0f
;
267 SetSpeakerArrangement("layout_STEREO", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
270 case AL_FORMAT_QUAD8
:
271 case AL_FORMAT_QUAD16
:
272 case AL_FORMAT_QUAD32
:
273 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_LEFT
] = aluSqrt(0.5);
274 Context
->ChannelMatrix
[FRONT_CENTER
][FRONT_RIGHT
] = aluSqrt(0.5);
275 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
276 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
277 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
278 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
279 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
280 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
281 Context
->NumChan
= 4;
282 Speaker2Chan
[0] = BACK_LEFT
;
283 Speaker2Chan
[1] = FRONT_LEFT
;
284 Speaker2Chan
[2] = FRONT_RIGHT
;
285 Speaker2Chan
[3] = BACK_RIGHT
;
286 SpeakerAngle
[0] = -135.0f
* M_PI
/180.0f
;
287 SpeakerAngle
[1] = -45.0f
* M_PI
/180.0f
;
288 SpeakerAngle
[2] = 45.0f
* M_PI
/180.0f
;
289 SpeakerAngle
[3] = 135.0f
* M_PI
/180.0f
;
290 SetSpeakerArrangement("layout_QUAD", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
293 case AL_FORMAT_51CHN8
:
294 case AL_FORMAT_51CHN16
:
295 case AL_FORMAT_51CHN32
:
296 Context
->ChannelMatrix
[SIDE_LEFT
][FRONT_LEFT
] = aluSqrt(0.5);
297 Context
->ChannelMatrix
[SIDE_LEFT
][BACK_LEFT
] = aluSqrt(0.5);
298 Context
->ChannelMatrix
[SIDE_RIGHT
][FRONT_RIGHT
] = aluSqrt(0.5);
299 Context
->ChannelMatrix
[SIDE_RIGHT
][BACK_RIGHT
] = aluSqrt(0.5);
300 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
301 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
302 Context
->NumChan
= 5;
303 Speaker2Chan
[0] = BACK_LEFT
;
304 Speaker2Chan
[1] = FRONT_LEFT
;
305 Speaker2Chan
[2] = FRONT_CENTER
;
306 Speaker2Chan
[3] = FRONT_RIGHT
;
307 Speaker2Chan
[4] = BACK_RIGHT
;
308 SpeakerAngle
[0] = -110.0f
* M_PI
/180.0f
;
309 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
310 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
311 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
312 SpeakerAngle
[4] = 110.0f
* M_PI
/180.0f
;
313 SetSpeakerArrangement("layout_51CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
316 case AL_FORMAT_61CHN8
:
317 case AL_FORMAT_61CHN16
:
318 case AL_FORMAT_61CHN32
:
319 Context
->ChannelMatrix
[BACK_LEFT
][BACK_CENTER
] = aluSqrt(0.5);
320 Context
->ChannelMatrix
[BACK_LEFT
][SIDE_LEFT
] = aluSqrt(0.5);
321 Context
->ChannelMatrix
[BACK_RIGHT
][BACK_CENTER
] = aluSqrt(0.5);
322 Context
->ChannelMatrix
[BACK_RIGHT
][SIDE_RIGHT
] = aluSqrt(0.5);
323 Context
->NumChan
= 6;
324 Speaker2Chan
[0] = SIDE_LEFT
;
325 Speaker2Chan
[1] = FRONT_LEFT
;
326 Speaker2Chan
[2] = FRONT_CENTER
;
327 Speaker2Chan
[3] = FRONT_RIGHT
;
328 Speaker2Chan
[4] = SIDE_RIGHT
;
329 Speaker2Chan
[5] = BACK_CENTER
;
330 SpeakerAngle
[0] = -90.0f
* M_PI
/180.0f
;
331 SpeakerAngle
[1] = -30.0f
* M_PI
/180.0f
;
332 SpeakerAngle
[2] = 0.0f
* M_PI
/180.0f
;
333 SpeakerAngle
[3] = 30.0f
* M_PI
/180.0f
;
334 SpeakerAngle
[4] = 90.0f
* M_PI
/180.0f
;
335 SpeakerAngle
[5] = 180.0f
* M_PI
/180.0f
;
336 SetSpeakerArrangement("layout_61CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
339 case AL_FORMAT_71CHN8
:
340 case AL_FORMAT_71CHN16
:
341 case AL_FORMAT_71CHN32
:
342 Context
->ChannelMatrix
[BACK_CENTER
][BACK_LEFT
] = aluSqrt(0.5);
343 Context
->ChannelMatrix
[BACK_CENTER
][BACK_RIGHT
] = aluSqrt(0.5);
344 Context
->NumChan
= 7;
345 Speaker2Chan
[0] = BACK_LEFT
;
346 Speaker2Chan
[1] = SIDE_LEFT
;
347 Speaker2Chan
[2] = FRONT_LEFT
;
348 Speaker2Chan
[3] = FRONT_CENTER
;
349 Speaker2Chan
[4] = FRONT_RIGHT
;
350 Speaker2Chan
[5] = SIDE_RIGHT
;
351 Speaker2Chan
[6] = BACK_RIGHT
;
352 SpeakerAngle
[0] = -150.0f
* M_PI
/180.0f
;
353 SpeakerAngle
[1] = -90.0f
* M_PI
/180.0f
;
354 SpeakerAngle
[2] = -30.0f
* M_PI
/180.0f
;
355 SpeakerAngle
[3] = 0.0f
* M_PI
/180.0f
;
356 SpeakerAngle
[4] = 30.0f
* M_PI
/180.0f
;
357 SpeakerAngle
[5] = 90.0f
* M_PI
/180.0f
;
358 SpeakerAngle
[6] = 150.0f
* M_PI
/180.0f
;
359 SetSpeakerArrangement("layout_71CHN", SpeakerAngle
, Speaker2Chan
, Context
->NumChan
);
366 for(pos
= 0; pos
< LUT_NUM
; pos
++)
369 Theta
= aluLUTpos2Angle(pos
);
371 /* clear all values */
372 offset
= OUTPUTCHANNELS
* pos
;
373 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
374 Context
->PanningLUT
[offset
+s
] = 0.0f
;
376 /* set panning values */
377 for(s
= 0; s
< Context
->NumChan
- 1; s
++)
379 if(Theta
>= SpeakerAngle
[s
] && Theta
< SpeakerAngle
[s
+1])
381 /* source between speaker s and speaker s+1 */
382 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
383 (SpeakerAngle
[s
+1]-SpeakerAngle
[s
]);
384 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
385 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
+1]] = sin(Alpha
);
389 if(s
== Context
->NumChan
- 1)
391 /* source between last and first speaker */
392 if(Theta
< SpeakerAngle
[0])
393 Theta
+= 2.0f
* M_PI
;
394 Alpha
= M_PI_2
* (Theta
-SpeakerAngle
[s
]) /
395 (2.0f
* M_PI
+ SpeakerAngle
[0]-SpeakerAngle
[s
]);
396 Context
->PanningLUT
[offset
+ Speaker2Chan
[s
]] = cos(Alpha
);
397 Context
->PanningLUT
[offset
+ Speaker2Chan
[0]] = sin(Alpha
);
402 static __inline ALint
aluCart2LUTpos(ALfloat re
, ALfloat im
)
405 ALfloat denom
= aluFabs(re
) + aluFabs(im
);
407 pos
= (ALint
)(QUADRANT_NUM
*aluFabs(im
) / denom
+ 0.5);
410 pos
= 2 * QUADRANT_NUM
- pos
;
416 static ALvoid
CalcSourceParams(const ALCcontext
*ALContext
, ALsource
*ALSource
,
419 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,DryMix
;
420 ALfloat Direction
[3],Position
[3],SourceToListener
[3];
421 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
,OuterGainHF
;
422 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
423 ALfloat U
[3],V
[3],N
[3];
424 ALfloat DopplerFactor
, DopplerVelocity
, flSpeedOfSound
;
425 ALfloat Matrix
[3][3];
426 ALfloat flAttenuation
;
427 ALfloat RoomAttenuation
[MAX_SENDS
];
428 ALfloat MetersPerUnit
;
429 ALfloat RoomRolloff
[MAX_SENDS
];
430 ALfloat DryGainHF
= 1.0f
;
431 ALfloat WetGain
[MAX_SENDS
];
432 ALfloat WetGainHF
[MAX_SENDS
];
433 ALfloat DirGain
, AmbientGain
;
435 const ALfloat
*SpeakerGain
;
441 //Get context properties
442 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
443 DopplerVelocity
= ALContext
->DopplerVelocity
;
444 flSpeedOfSound
= ALContext
->flSpeedOfSound
;
445 NumSends
= ALContext
->Device
->NumAuxSends
;
446 Frequency
= ALContext
->Device
->Frequency
;
448 //Get listener properties
449 ListenerGain
= ALContext
->Listener
.Gain
;
450 MetersPerUnit
= ALContext
->Listener
.MetersPerUnit
;
452 //Get source properties
453 SourceVolume
= ALSource
->flGain
;
454 memcpy(Position
, ALSource
->vPosition
, sizeof(ALSource
->vPosition
));
455 memcpy(Direction
, ALSource
->vOrientation
, sizeof(ALSource
->vOrientation
));
456 MinVolume
= ALSource
->flMinGain
;
457 MaxVolume
= ALSource
->flMaxGain
;
458 MinDist
= ALSource
->flRefDistance
;
459 MaxDist
= ALSource
->flMaxDistance
;
460 Rolloff
= ALSource
->flRollOffFactor
;
461 InnerAngle
= ALSource
->flInnerAngle
;
462 OuterAngle
= ALSource
->flOuterAngle
;
463 OuterGainHF
= ALSource
->OuterGainHF
;
465 //Only apply 3D calculations for mono buffers
466 if(isMono
== AL_FALSE
)
468 //1. Multi-channel buffers always play "normal"
469 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
471 DryMix
= SourceVolume
;
472 DryMix
= __min(DryMix
,MaxVolume
);
473 DryMix
= __max(DryMix
,MinVolume
);
475 switch(ALSource
->DirectFilter
.type
)
477 case AL_FILTER_LOWPASS
:
478 DryMix
*= ALSource
->DirectFilter
.Gain
;
479 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
483 ALSource
->Params
.DryGains
[FRONT_LEFT
] = DryMix
* ListenerGain
;
484 ALSource
->Params
.DryGains
[FRONT_RIGHT
] = DryMix
* ListenerGain
;
485 ALSource
->Params
.DryGains
[SIDE_LEFT
] = DryMix
* ListenerGain
;
486 ALSource
->Params
.DryGains
[SIDE_RIGHT
] = DryMix
* ListenerGain
;
487 ALSource
->Params
.DryGains
[BACK_LEFT
] = DryMix
* ListenerGain
;
488 ALSource
->Params
.DryGains
[BACK_RIGHT
] = DryMix
* ListenerGain
;
489 ALSource
->Params
.DryGains
[FRONT_CENTER
] = DryMix
* ListenerGain
;
490 ALSource
->Params
.DryGains
[BACK_CENTER
] = DryMix
* ListenerGain
;
491 ALSource
->Params
.DryGains
[LFE
] = DryMix
* ListenerGain
;
492 for(i
= 0;i
< MAX_SENDS
;i
++)
493 ALSource
->Params
.WetGains
[i
] = 0.0f
;
495 /* Update filter coefficients. Calculations based on the I3DL2
497 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
498 /* We use two chained one-pole filters, so we need to take the
499 * square root of the squared gain, which is the same as the base
501 g
= __max(DryGainHF
, 0.01f
);
503 /* Be careful with gains < 0.0001, as that causes the coefficient
504 * head towards 1, which will flatten the signal */
505 if(g
< 0.9999f
) /* 1-epsilon */
506 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
508 ALSource
->Params
.iirFilter
.coeff
= a
;
509 for(i
= 0;i
< MAX_SENDS
;i
++)
510 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= 0.0f
;
515 //1. Translate Listener to origin (convert to head relative)
516 // Note that Direction and SourceToListener are *not* transformed.
517 // SourceToListener is used with the source and listener velocities,
518 // which are untransformed, and Direction is used with SourceToListener
519 // for the sound cone
520 if(ALSource
->bHeadRelative
==AL_FALSE
)
522 // Build transform matrix
523 aluCrossproduct(ALContext
->Listener
.Forward
, ALContext
->Listener
.Up
, U
); // Right-vector
524 aluNormalize(U
); // Normalized Right-vector
525 memcpy(V
, ALContext
->Listener
.Up
, sizeof(V
)); // Up-vector
526 aluNormalize(V
); // Normalized Up-vector
527 memcpy(N
, ALContext
->Listener
.Forward
, sizeof(N
)); // At-vector
528 aluNormalize(N
); // Normalized At-vector
529 Matrix
[0][0] = U
[0]; Matrix
[0][1] = V
[0]; Matrix
[0][2] = -N
[0];
530 Matrix
[1][0] = U
[1]; Matrix
[1][1] = V
[1]; Matrix
[1][2] = -N
[1];
531 Matrix
[2][0] = U
[2]; Matrix
[2][1] = V
[2]; Matrix
[2][2] = -N
[2];
533 // Translate source position into listener space
534 Position
[0] -= ALContext
->Listener
.Position
[0];
535 Position
[1] -= ALContext
->Listener
.Position
[1];
536 Position
[2] -= ALContext
->Listener
.Position
[2];
538 SourceToListener
[0] = -Position
[0];
539 SourceToListener
[1] = -Position
[1];
540 SourceToListener
[2] = -Position
[2];
542 // Transform source position into listener space
543 aluMatrixVector(Position
, Matrix
);
547 SourceToListener
[0] = -Position
[0];
548 SourceToListener
[1] = -Position
[1];
549 SourceToListener
[2] = -Position
[2];
551 aluNormalize(SourceToListener
);
552 aluNormalize(Direction
);
554 //2. Calculate distance attenuation
555 Distance
= aluSqrt(aluDotproduct(Position
, Position
));
557 flAttenuation
= 1.0f
;
558 for(i
= 0;i
< MAX_SENDS
;i
++)
560 RoomAttenuation
[i
] = 1.0f
;
562 RoomRolloff
[i
] = ALSource
->RoomRolloffFactor
;
563 if(ALSource
->Send
[i
].Slot
&&
564 ALSource
->Send
[i
].Slot
->effect
.type
== AL_EFFECT_REVERB
)
565 RoomRolloff
[i
] += ALSource
->Send
[i
].Slot
->effect
.Reverb
.RoomRolloffFactor
;
568 switch(ALSource
->DistanceModel
)
570 case AL_INVERSE_DISTANCE_CLAMPED
:
571 Distance
=__max(Distance
,MinDist
);
572 Distance
=__min(Distance
,MaxDist
);
573 if(MaxDist
< MinDist
)
576 case AL_INVERSE_DISTANCE
:
579 if((MinDist
+ (Rolloff
* (Distance
- MinDist
))) > 0.0f
)
580 flAttenuation
= MinDist
/ (MinDist
+ (Rolloff
* (Distance
- MinDist
)));
581 for(i
= 0;i
< NumSends
;i
++)
583 if((MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
))) > 0.0f
)
584 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (Distance
- MinDist
)));
589 case AL_LINEAR_DISTANCE_CLAMPED
:
590 Distance
=__max(Distance
,MinDist
);
591 Distance
=__min(Distance
,MaxDist
);
592 if(MaxDist
< MinDist
)
595 case AL_LINEAR_DISTANCE
:
596 Distance
=__min(Distance
,MaxDist
);
597 if(MaxDist
!= MinDist
)
599 flAttenuation
= 1.0f
- (Rolloff
*(Distance
-MinDist
)/(MaxDist
- MinDist
));
600 for(i
= 0;i
< NumSends
;i
++)
601 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(Distance
-MinDist
)/(MaxDist
- MinDist
));
605 case AL_EXPONENT_DISTANCE_CLAMPED
:
606 Distance
=__max(Distance
,MinDist
);
607 Distance
=__min(Distance
,MaxDist
);
608 if(MaxDist
< MinDist
)
611 case AL_EXPONENT_DISTANCE
:
612 if(Distance
> 0.0f
&& MinDist
> 0.0f
)
614 flAttenuation
= (ALfloat
)pow(Distance
/MinDist
, -Rolloff
);
615 for(i
= 0;i
< NumSends
;i
++)
616 RoomAttenuation
[i
] = (ALfloat
)pow(Distance
/MinDist
, -RoomRolloff
[i
]);
624 // Source Gain + Attenuation and clamp to Min/Max Gain
625 DryMix
= SourceVolume
* flAttenuation
;
626 DryMix
= __min(DryMix
,MaxVolume
);
627 DryMix
= __max(DryMix
,MinVolume
);
629 for(i
= 0;i
< NumSends
;i
++)
631 ALfloat WetMix
= SourceVolume
* RoomAttenuation
[i
];
632 WetMix
= __min(WetMix
,MaxVolume
);
633 WetGain
[i
] = __max(WetMix
,MinVolume
);
637 // Distance-based air absorption
638 if(ALSource
->AirAbsorptionFactor
> 0.0f
&& ALSource
->DistanceModel
!= AL_NONE
)
640 ALfloat dist
= Distance
-MinDist
;
643 if(dist
< 0.0f
) dist
= 0.0f
;
644 // Absorption calculation is done in dB
645 absorb
= (ALSource
->AirAbsorptionFactor
*AIRABSORBGAINDBHF
) *
646 (dist
*MetersPerUnit
);
647 // Convert dB to linear gain before applying
648 absorb
= pow(10.0, absorb
/20.0);
650 for(i
= 0;i
< MAX_SENDS
;i
++)
651 WetGainHF
[i
] *= absorb
;
654 //3. Apply directional soundcones
655 Angle
= aluAcos(aluDotproduct(Direction
,SourceToListener
)) * 180.0f
/M_PI
;
656 if(Angle
>= InnerAngle
&& Angle
<= OuterAngle
)
658 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
659 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
)*scale
);
660 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
)*scale
);
661 DryMix
*= ConeVolume
;
662 if(ALSource
->DryGainHFAuto
)
665 else if(Angle
> OuterAngle
)
667 ConeVolume
= (1.0f
+(ALSource
->flOuterGain
-1.0f
));
668 ConeHF
= (1.0f
+(OuterGainHF
-1.0f
));
669 DryMix
*= ConeVolume
;
670 if(ALSource
->DryGainHFAuto
)
679 //4. Calculate Velocity
680 if(DopplerFactor
!= 0.0f
)
682 ALfloat flVSS
, flVLS
= 0.0f
;
683 ALfloat flMaxVelocity
= (DopplerVelocity
* flSpeedOfSound
) /
686 flVSS
= aluDotproduct(ALSource
->vVelocity
, SourceToListener
);
687 if(flVSS
>= flMaxVelocity
)
688 flVSS
= (flMaxVelocity
- 1.0f
);
689 else if(flVSS
<= -flMaxVelocity
)
690 flVSS
= -flMaxVelocity
+ 1.0f
;
692 if(ALSource
->bHeadRelative
== AL_FALSE
)
694 flVLS
= aluDotproduct(ALContext
->Listener
.Velocity
, SourceToListener
);
695 if(flVLS
>= flMaxVelocity
)
696 flVLS
= (flMaxVelocity
- 1.0f
);
697 else if(flVLS
<= -flMaxVelocity
)
698 flVLS
= -flMaxVelocity
+ 1.0f
;
701 ALSource
->Params
.Pitch
= ALSource
->flPitch
*
702 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVLS
)) /
703 ((flSpeedOfSound
* DopplerVelocity
) - (DopplerFactor
* flVSS
));
706 ALSource
->Params
.Pitch
= ALSource
->flPitch
;
708 for(i
= 0;i
< NumSends
;i
++)
710 if(ALSource
->Send
[i
].Slot
&&
711 ALSource
->Send
[i
].Slot
->effect
.type
!= AL_EFFECT_NULL
)
713 if(ALSource
->Send
[i
].Slot
->AuxSendAuto
)
715 if(ALSource
->WetGainAuto
)
716 WetGain
[i
] *= ConeVolume
;
717 if(ALSource
->WetGainHFAuto
)
718 WetGainHF
[i
] *= ConeHF
;
720 // Apply minimal attenuation in place of missing
721 // statistical reverb model.
722 WetGain
[i
] *= pow(DryMix
, 1.0f
/ 2.0f
);
726 // If the slot's auxiliary send auto is off, the data sent to
727 // the effect slot is the same as the dry path, sans filter
730 WetGainHF
[i
] = DryGainHF
;
733 switch(ALSource
->Send
[i
].WetFilter
.type
)
735 case AL_FILTER_LOWPASS
:
736 WetGain
[i
] *= ALSource
->Send
[i
].WetFilter
.Gain
;
737 WetGainHF
[i
] *= ALSource
->Send
[i
].WetFilter
.GainHF
;
740 ALSource
->Params
.WetGains
[i
] = WetGain
[i
] * ListenerGain
;
744 ALSource
->Params
.WetGains
[i
] = 0.0f
;
748 for(i
= NumSends
;i
< MAX_SENDS
;i
++)
750 ALSource
->Params
.WetGains
[i
] = 0.0f
;
754 //5. Apply filter gains and filters
755 switch(ALSource
->DirectFilter
.type
)
757 case AL_FILTER_LOWPASS
:
758 DryMix
*= ALSource
->DirectFilter
.Gain
;
759 DryGainHF
*= ALSource
->DirectFilter
.GainHF
;
762 DryMix
*= ListenerGain
;
764 // Use energy-preserving panning algorithm for multi-speaker playback
765 length
= aluSqrt(Position
[0]*Position
[0] + Position
[1]*Position
[1] +
766 Position
[2]*Position
[2]);
767 length
= __max(length
, MinDist
);
770 ALfloat invlen
= 1.0f
/length
;
771 Position
[0] *= invlen
;
772 Position
[1] *= invlen
;
773 Position
[2] *= invlen
;
776 pos
= aluCart2LUTpos(-Position
[2], Position
[0]);
777 SpeakerGain
= &ALContext
->PanningLUT
[OUTPUTCHANNELS
* pos
];
779 DirGain
= aluSqrt(Position
[0]*Position
[0] + Position
[2]*Position
[2]);
780 // elevation adjustment for directional gain. this sucks, but
781 // has low complexity
782 AmbientGain
= 1.0/aluSqrt(ALContext
->NumChan
) * (1.0-DirGain
);
783 for(s
= 0; s
< OUTPUTCHANNELS
; s
++)
785 ALfloat gain
= SpeakerGain
[s
]*DirGain
+ AmbientGain
;
786 ALSource
->Params
.DryGains
[s
] = DryMix
* gain
;
789 /* Update filter coefficients. */
790 cw
= cos(2.0*M_PI
* LOWPASSFREQCUTOFF
/ Frequency
);
791 /* Spatialized sources use four chained one-pole filters, so we need to
792 * take the fourth root of the squared gain, which is the same as the
793 * square root of the base gain. */
794 g
= aluSqrt(__max(DryGainHF
, 0.0001f
));
796 if(g
< 0.9999f
) /* 1-epsilon */
797 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
799 ALSource
->Params
.iirFilter
.coeff
= a
;
801 for(i
= 0;i
< NumSends
;i
++)
803 /* The wet path uses two chained one-pole filters, so take the
804 * base gain (square root of the squared gain) */
805 g
= __max(WetGainHF
[i
], 0.01f
);
807 if(g
< 0.9999f
) /* 1-epsilon */
808 a
= (1 - g
*cw
- aluSqrt(2*g
*(1-cw
) - g
*g
*(1 - cw
*cw
))) /
810 ALSource
->Params
.Send
[i
].iirFilter
.coeff
= a
;
814 static __inline ALshort
lerp(ALshort val1
, ALshort val2
, ALint frac
)
816 return val1
+ (((val2
-val1
)*frac
)>>FRACTIONBITS
);
819 static void MixSomeSources(ALCcontext
*ALContext
, float (*DryBuffer
)[OUTPUTCHANNELS
], ALuint SamplesToDo
)
821 static float DummyBuffer
[BUFFERSIZE
];
822 ALfloat
*WetBuffer
[MAX_SENDS
];
823 ALfloat (*Matrix
)[OUTPUTCHANNELS
] = ALContext
->ChannelMatrix
;
824 ALfloat DrySend
[OUTPUTCHANNELS
];
825 ALfloat dryGainStep
[OUTPUTCHANNELS
];
826 ALfloat wetGainStep
[MAX_SENDS
];
830 ALbufferlistitem
*BufferListItem
;
831 ALint64 DataSize64
,DataPos64
;
832 FILTER
*DryFilter
, *WetFilter
[MAX_SENDS
];
833 ALfloat WetSend
[MAX_SENDS
];
837 ALuint DataPosInt
, DataPosFrac
;
838 ALboolean FirstStart
;
839 ALuint BuffersPlayed
;
842 if(!(ALSource
=ALContext
->Source
))
845 frequency
= ALContext
->Device
->Frequency
;
847 rampLength
= frequency
* MIN_RAMP_LENGTH
/ 1000;
848 rampLength
= max(rampLength
, SamplesToDo
);
851 State
= ALSource
->state
;
852 if(State
!= AL_PLAYING
)
854 if((ALSource
=ALSource
->next
) != NULL
)
860 /* Get source info */
861 BuffersPlayed
= ALSource
->BuffersPlayed
;
862 DataPosInt
= ALSource
->position
;
863 DataPosFrac
= ALSource
->position_fraction
;
864 FirstStart
= ALSource
->FirstStart
;
866 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
867 DrySend
[i
] = ALSource
->DryGains
[i
];
868 for(i
= 0;i
< MAX_SENDS
;i
++)
869 WetSend
[i
] = ALSource
->WetGains
[i
];
871 DryFilter
= &ALSource
->Params
.iirFilter
;
872 for(i
= 0;i
< MAX_SENDS
;i
++)
874 WetFilter
[i
] = &ALSource
->Params
.Send
[i
].iirFilter
;
875 WetBuffer
[i
] = (ALSource
->Send
[i
].Slot
?
876 ALSource
->Send
[i
].Slot
->WetBuffer
:
880 BufferListItem
= ALSource
->queue
;
881 for(i
= 0;i
< BuffersPlayed
&& BufferListItem
;i
++)
882 BufferListItem
= BufferListItem
->next
;
884 while(State
== AL_PLAYING
&& j
< SamplesToDo
)
889 ALuint Channels
, Bytes
;
893 /* Get buffer info */
894 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
896 Data
= ALBuffer
->data
;
897 Channels
= aluChannelsFromFormat(ALBuffer
->format
);
898 Bytes
= aluBytesFromFormat(ALBuffer
->format
);
899 DataSize
= ALBuffer
->size
;
900 DataSize
/= Channels
* Bytes
;
902 if(DataPosInt
>= DataSize
)
905 CalcSourceParams(ALContext
, ALSource
, (Channels
==1)?AL_TRUE
:AL_FALSE
);
906 Pitch
= (ALSource
->Params
.Pitch
*ALBuffer
->frequency
) / frequency
;
908 /* Compute the gain steps for each output channel */
911 FirstStart
= AL_FALSE
;
912 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
914 DrySend
[i
] = ALSource
->Params
.DryGains
[i
];
915 dryGainStep
[i
] = 0.0f
;
917 for(i
= 0;i
< MAX_SENDS
;i
++)
919 WetSend
[i
] = ALSource
->Params
.WetGains
[i
];
920 wetGainStep
[i
] = 0.0f
;
925 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
926 dryGainStep
[i
] = (ALSource
->Params
.DryGains
[i
]-
927 DrySend
[i
]) / rampLength
;
928 for(i
= 0;i
< MAX_SENDS
;i
++)
929 wetGainStep
[i
] = (ALSource
->Params
.WetGains
[i
]-
930 WetSend
[i
]) / rampLength
;
933 /* Compute 18.14 fixed point step */
934 if(Pitch
> (float)MAX_PITCH
)
935 Pitch
= (float)MAX_PITCH
;
936 increment
= (ALint
)(Pitch
*(ALfloat
)(1L<<FRACTIONBITS
));
938 increment
= (1<<FRACTIONBITS
);
940 /* Figure out how many samples we can mix. */
941 DataSize64
= DataSize
;
942 DataSize64
<<= FRACTIONBITS
;
943 DataPos64
= DataPosInt
;
944 DataPos64
<<= FRACTIONBITS
;
945 DataPos64
+= DataPosFrac
;
946 BufferSize
= (ALuint
)((DataSize64
-DataPos64
+(increment
-1)) / increment
);
948 if(BufferListItem
->next
)
950 ALbuffer
*NextBuf
= BufferListItem
->next
->buffer
;
951 if(NextBuf
&& NextBuf
->data
)
953 ALuint ulExtraSamples
= min(NextBuf
->size
, (ALint
)(ALBuffer
->padding
*Channels
*Bytes
));
954 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
957 else if(ALSource
->bLooping
)
959 ALbuffer
*NextBuf
= ALSource
->queue
->buffer
;
960 if(NextBuf
&& NextBuf
->data
)
962 ALuint ulExtraSamples
= min(NextBuf
->size
, (ALint
)(ALBuffer
->padding
*Channels
*Bytes
));
963 memcpy(&Data
[DataSize
*Channels
], NextBuf
->data
, ulExtraSamples
);
967 memset(&Data
[DataSize
*Channels
], 0, (ALBuffer
->padding
*Channels
*Bytes
));
968 BufferSize
= min(BufferSize
, (SamplesToDo
-j
));
970 if(DuplicateStereo
&& Channels
== 2)
972 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 1.0f
;
973 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 1.0f
;
974 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 1.0f
;
975 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 1.0f
;
977 else if(DuplicateStereo
)
979 Matrix
[FRONT_LEFT
][SIDE_LEFT
] = 0.0f
;
980 Matrix
[FRONT_RIGHT
][SIDE_RIGHT
] = 0.0f
;
981 Matrix
[FRONT_LEFT
][BACK_LEFT
] = 0.0f
;
982 Matrix
[FRONT_RIGHT
][BACK_RIGHT
] = 0.0f
;
985 /* Actual sample mixing loop */
987 Data
+= DataPosInt
*Channels
;
989 if(Channels
== 1) /* Mono */
995 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
996 DrySend
[i
] += dryGainStep
[i
];
997 for(i
= 0;i
< MAX_SENDS
;i
++)
998 WetSend
[i
] += wetGainStep
[i
];
1000 /* First order interpolator */
1001 value
= lerp(Data
[k
], Data
[k
+1], DataPosFrac
);
1003 /* Direct path final mix buffer and panning */
1004 outsamp
= lpFilter4P(DryFilter
, 0, value
);
1005 DryBuffer
[j
][FRONT_LEFT
] += outsamp
*DrySend
[FRONT_LEFT
];
1006 DryBuffer
[j
][FRONT_RIGHT
] += outsamp
*DrySend
[FRONT_RIGHT
];
1007 DryBuffer
[j
][SIDE_LEFT
] += outsamp
*DrySend
[SIDE_LEFT
];
1008 DryBuffer
[j
][SIDE_RIGHT
] += outsamp
*DrySend
[SIDE_RIGHT
];
1009 DryBuffer
[j
][BACK_LEFT
] += outsamp
*DrySend
[BACK_LEFT
];
1010 DryBuffer
[j
][BACK_RIGHT
] += outsamp
*DrySend
[BACK_RIGHT
];
1011 DryBuffer
[j
][FRONT_CENTER
] += outsamp
*DrySend
[FRONT_CENTER
];
1012 DryBuffer
[j
][BACK_CENTER
] += outsamp
*DrySend
[BACK_CENTER
];
1014 /* Room path final mix buffer and panning */
1015 for(i
= 0;i
< MAX_SENDS
;i
++)
1017 outsamp
= lpFilter2P(WetFilter
[i
], 0, value
);
1018 WetBuffer
[i
][j
] += outsamp
*WetSend
[i
];
1021 DataPosFrac
+= increment
;
1022 k
+= DataPosFrac
>>FRACTIONBITS
;
1023 DataPosFrac
&= FRACTIONMASK
;
1027 else if(Channels
== 2) /* Stereo */
1029 const int chans
[] = {
1030 FRONT_LEFT
, FRONT_RIGHT
1033 #define DO_MIX() do { \
1034 for(i = 0;i < MAX_SENDS;i++) \
1035 WetSend[i] += wetGainStep[i]*BufferSize; \
1036 while(BufferSize--) \
1038 for(i = 0;i < OUTPUTCHANNELS;i++) \
1039 DrySend[i] += dryGainStep[i]; \
1041 for(i = 0;i < Channels;i++) \
1043 value = lerp(Data[k*Channels + i], Data[(k+1)*Channels + i], DataPosFrac); \
1044 value = lpFilter2P(DryFilter, chans[i]*2, value)*DrySend[chans[i]]; \
1045 for(out = 0;out < OUTPUTCHANNELS;out++) \
1046 DryBuffer[j][out] += value*Matrix[chans[i]][out]; \
1049 DataPosFrac += increment; \
1050 k += DataPosFrac>>FRACTIONBITS; \
1051 DataPosFrac &= FRACTIONMASK; \
1058 else if(Channels
== 4) /* Quad */
1060 const int chans
[] = {
1061 FRONT_LEFT
, FRONT_RIGHT
,
1062 BACK_LEFT
, BACK_RIGHT
1067 else if(Channels
== 6) /* 5.1 */
1069 const int chans
[] = {
1070 FRONT_LEFT
, FRONT_RIGHT
,
1072 BACK_LEFT
, BACK_RIGHT
1077 else if(Channels
== 7) /* 6.1 */
1079 const int chans
[] = {
1080 FRONT_LEFT
, FRONT_RIGHT
,
1083 SIDE_LEFT
, SIDE_RIGHT
1088 else if(Channels
== 8) /* 7.1 */
1090 const int chans
[] = {
1091 FRONT_LEFT
, FRONT_RIGHT
,
1093 BACK_LEFT
, BACK_RIGHT
,
1094 SIDE_LEFT
, SIDE_RIGHT
1102 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1103 DrySend
[i
] += dryGainStep
[i
]*BufferSize
;
1104 for(i
= 0;i
< MAX_SENDS
;i
++)
1105 WetSend
[i
] += wetGainStep
[i
]*BufferSize
;
1108 DataPosFrac
+= increment
;
1109 k
+= DataPosFrac
>>FRACTIONBITS
;
1110 DataPosFrac
&= FRACTIONMASK
;
1117 /* Handle looping sources */
1118 if(DataPosInt
>= DataSize
)
1120 if(BuffersPlayed
< (ALSource
->BuffersInQueue
-1))
1122 BufferListItem
= BufferListItem
->next
;
1124 DataPosInt
-= DataSize
;
1128 if(!ALSource
->bLooping
)
1131 BufferListItem
= ALSource
->queue
;
1132 BuffersPlayed
= ALSource
->BuffersInQueue
;
1138 BufferListItem
= ALSource
->queue
;
1140 if(ALSource
->BuffersInQueue
== 1 && DataSize
)
1141 DataPosInt
%= DataSize
;
1143 DataPosInt
-= DataSize
;
1149 /* Update source info */
1150 ALSource
->state
= State
;
1151 ALSource
->BuffersPlayed
= BuffersPlayed
;
1152 ALSource
->position
= DataPosInt
;
1153 ALSource
->position_fraction
= DataPosFrac
;
1154 ALSource
->Buffer
= BufferListItem
->buffer
;
1156 for(i
= 0;i
< OUTPUTCHANNELS
;i
++)
1157 ALSource
->DryGains
[i
] = DrySend
[i
];
1158 for(i
= 0;i
< MAX_SENDS
;i
++)
1159 ALSource
->WetGains
[i
] = WetSend
[i
];
1161 ALSource
->FirstStart
= FirstStart
;
1163 if((ALSource
=ALSource
->next
) != NULL
)
1164 goto another_source
;
1167 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1169 float (*DryBuffer
)[OUTPUTCHANNELS
];
1171 ALeffectslot
*ALEffectSlot
;
1172 ALCcontext
*ALContext
;
1176 SuspendContext(NULL
);
1178 #if defined(HAVE_FESETROUND)
1179 fpuState
= fegetround();
1180 fesetround(FE_TOWARDZERO
);
1181 #elif defined(HAVE__CONTROLFP)
1182 fpuState
= _controlfp(0, 0);
1183 _controlfp(_RC_CHOP
, _MCW_RC
);
1188 DryBuffer
= device
->DryBuffer
;
1191 /* Setup variables */
1192 SamplesToDo
= min(size
, BUFFERSIZE
);
1194 /* Clear mixing buffer */
1195 memset(DryBuffer
, 0, SamplesToDo
*OUTPUTCHANNELS
*sizeof(ALfloat
));
1197 for(c
= 0;c
< device
->NumContexts
;c
++)
1199 ALContext
= device
->Contexts
[c
];
1200 SuspendContext(ALContext
);
1202 MixSomeSources(ALContext
, DryBuffer
, SamplesToDo
);
1204 /* effect slot processing */
1205 ALEffectSlot
= ALContext
->AuxiliaryEffectSlot
;
1208 if(ALEffectSlot
->EffectState
)
1209 ALEffect_Process(ALEffectSlot
->EffectState
, ALEffectSlot
, SamplesToDo
, ALEffectSlot
->WetBuffer
, DryBuffer
);
1211 for(i
= 0;i
< SamplesToDo
;i
++)
1212 ALEffectSlot
->WetBuffer
[i
] = 0.0f
;
1213 ALEffectSlot
= ALEffectSlot
->next
;
1215 ProcessContext(ALContext
);
1218 //Post processing loop
1219 switch(device
->Format
)
1221 #define CHECK_WRITE_FORMAT(bits, type, func, isWin) \
1222 case AL_FORMAT_MONO##bits: \
1223 for(i = 0;i < SamplesToDo;i++) \
1225 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT] + \
1226 DryBuffer[i][FRONT_RIGHT]); \
1227 buffer = ((type*)buffer) + 1; \
1230 case AL_FORMAT_STEREO##bits: \
1233 for(i = 0;i < SamplesToDo;i++) \
1236 samples[0] = DryBuffer[i][FRONT_LEFT]; \
1237 samples[1] = DryBuffer[i][FRONT_RIGHT]; \
1238 bs2b_cross_feed(device->Bs2b, samples); \
1239 ((type*)buffer)[0] = (func)(samples[0]); \
1240 ((type*)buffer)[1] = (func)(samples[1]); \
1241 buffer = ((type*)buffer) + 2; \
1246 for(i = 0;i < SamplesToDo;i++) \
1248 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1249 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1250 buffer = ((type*)buffer) + 2; \
1254 case AL_FORMAT_QUAD##bits: \
1255 for(i = 0;i < SamplesToDo;i++) \
1257 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1258 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1259 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1260 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1261 buffer = ((type*)buffer) + 4; \
1264 case AL_FORMAT_51CHN##bits: \
1265 for(i = 0;i < SamplesToDo;i++) \
1267 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1268 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1270 /* Of course, Windows can't use the same ordering... */ \
1271 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1272 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1273 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_LEFT]); \
1274 ((type*)buffer)[5] = (func)(DryBuffer[i][BACK_RIGHT]); \
1276 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1277 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1278 ((type*)buffer)[4] = (func)(DryBuffer[i][FRONT_CENTER]); \
1279 ((type*)buffer)[5] = (func)(DryBuffer[i][LFE]); \
1281 buffer = ((type*)buffer) + 6; \
1284 case AL_FORMAT_61CHN##bits: \
1285 for(i = 0;i < SamplesToDo;i++) \
1287 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1288 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1289 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1290 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1291 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_CENTER]); \
1292 ((type*)buffer)[5] = (func)(DryBuffer[i][SIDE_LEFT]); \
1293 ((type*)buffer)[6] = (func)(DryBuffer[i][SIDE_RIGHT]); \
1294 buffer = ((type*)buffer) + 7; \
1297 case AL_FORMAT_71CHN##bits: \
1298 for(i = 0;i < SamplesToDo;i++) \
1300 ((type*)buffer)[0] = (func)(DryBuffer[i][FRONT_LEFT]); \
1301 ((type*)buffer)[1] = (func)(DryBuffer[i][FRONT_RIGHT]); \
1303 ((type*)buffer)[2] = (func)(DryBuffer[i][FRONT_CENTER]); \
1304 ((type*)buffer)[3] = (func)(DryBuffer[i][LFE]); \
1305 ((type*)buffer)[4] = (func)(DryBuffer[i][BACK_LEFT]); \
1306 ((type*)buffer)[5] = (func)(DryBuffer[i][BACK_RIGHT]); \
1308 ((type*)buffer)[2] = (func)(DryBuffer[i][BACK_LEFT]); \
1309 ((type*)buffer)[3] = (func)(DryBuffer[i][BACK_RIGHT]); \
1310 ((type*)buffer)[4] = (func)(DryBuffer[i][FRONT_CENTER]); \
1311 ((type*)buffer)[5] = (func)(DryBuffer[i][LFE]); \
1313 ((type*)buffer)[6] = (func)(DryBuffer[i][SIDE_LEFT]); \
1314 ((type*)buffer)[7] = (func)(DryBuffer[i][SIDE_RIGHT]); \
1315 buffer = ((type*)buffer) + 8; \
1319 #define AL_FORMAT_MONO32 AL_FORMAT_MONO_FLOAT32
1320 #define AL_FORMAT_STEREO32 AL_FORMAT_STEREO_FLOAT32
1322 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
, 1)
1323 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
, 1)
1324 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
, 1)
1326 CHECK_WRITE_FORMAT(8, ALubyte
, aluF2UB
, 0)
1327 CHECK_WRITE_FORMAT(16, ALshort
, aluF2S
, 0)
1328 CHECK_WRITE_FORMAT(32, ALfloat
, aluF2F
, 0)
1330 #undef AL_FORMAT_STEREO32
1331 #undef AL_FORMAT_MONO32
1332 #undef CHECK_WRITE_FORMAT
1338 size
-= SamplesToDo
;
1341 #if defined(HAVE_FESETROUND)
1342 fesetround(fpuState
);
1343 #elif defined(HAVE__CONTROLFP)
1344 _controlfp(fpuState
, 0xfffff);
1347 ProcessContext(NULL
);
1350 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1354 for(i
= 0;i
< device
->NumContexts
;i
++)
1358 SuspendContext(device
->Contexts
[i
]);
1360 source
= device
->Contexts
[i
]->Source
;
1363 if(source
->state
== AL_PLAYING
)
1365 source
->state
= AL_STOPPED
;
1366 source
->BuffersPlayed
= source
->BuffersInQueue
;
1367 source
->position
= 0;
1368 source
->position_fraction
= 0;
1370 source
= source
->next
;
1372 ProcessContext(device
->Contexts
[i
]);
1375 device
->Connected
= ALC_FALSE
;