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.,
17 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
18 * Or go to http://www.gnu.org/copyleft/lgpl.html
32 #include "alListener.h"
33 #include "alAuxEffectSlot.h"
37 #include "static_assert.h"
39 #include "mixer_defs.h"
41 #include "backends/base.h"
42 #include "midi/base.h"
45 static_assert((INT_MAX
>>FRACTIONBITS
)/MAX_PITCH
> BUFFERSIZE
,
46 "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!");
55 ALfloat ConeScale
= 1.0f
;
57 /* Localized Z scalar for mono sources */
58 ALfloat ZScale
= 1.0f
;
60 extern inline ALfloat
minf(ALfloat a
, ALfloat b
);
61 extern inline ALfloat
maxf(ALfloat a
, ALfloat b
);
62 extern inline ALfloat
clampf(ALfloat val
, ALfloat min
, ALfloat max
);
64 extern inline ALdouble
mind(ALdouble a
, ALdouble b
);
65 extern inline ALdouble
maxd(ALdouble a
, ALdouble b
);
66 extern inline ALdouble
clampd(ALdouble val
, ALdouble min
, ALdouble max
);
68 extern inline ALuint
minu(ALuint a
, ALuint b
);
69 extern inline ALuint
maxu(ALuint a
, ALuint b
);
70 extern inline ALuint
clampu(ALuint val
, ALuint min
, ALuint max
);
72 extern inline ALint
mini(ALint a
, ALint b
);
73 extern inline ALint
maxi(ALint a
, ALint b
);
74 extern inline ALint
clampi(ALint val
, ALint min
, ALint max
);
76 extern inline ALint64
mini64(ALint64 a
, ALint64 b
);
77 extern inline ALint64
maxi64(ALint64 a
, ALint64 b
);
78 extern inline ALint64
clampi64(ALint64 val
, ALint64 min
, ALint64 max
);
80 extern inline ALuint64
minu64(ALuint64 a
, ALuint64 b
);
81 extern inline ALuint64
maxu64(ALuint64 a
, ALuint64 b
);
82 extern inline ALuint64
clampu64(ALuint64 val
, ALuint64 min
, ALuint64 max
);
84 extern inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALfloat mu
);
85 extern inline ALfloat
cubic(ALfloat val0
, ALfloat val1
, ALfloat val2
, ALfloat val3
, ALuint frac
);
87 extern inline void aluVectorSet(aluVector
*restrict vector
, ALfloat x
, ALfloat y
, ALfloat z
, ALfloat w
);
89 extern inline void aluMatrixSetRow(aluMatrix
*restrict matrix
, ALuint row
,
90 ALfloat m0
, ALfloat m1
, ALfloat m2
, ALfloat m3
);
91 extern inline void aluMatrixSet(aluMatrix
*restrict matrix
, ALfloat m00
, ALfloat m01
, ALfloat m02
, ALfloat m03
,
92 ALfloat m10
, ALfloat m11
, ALfloat m12
, ALfloat m13
,
93 ALfloat m20
, ALfloat m21
, ALfloat m22
, ALfloat m23
,
94 ALfloat m30
, ALfloat m31
, ALfloat m32
, ALfloat m33
);
97 static inline HrtfMixerFunc
SelectHrtfMixer(void)
100 if((CPUCapFlags
&CPU_CAP_SSE
))
104 if((CPUCapFlags
&CPU_CAP_NEON
))
112 static inline void aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
114 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
115 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
116 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
119 static inline ALfloat
aluDotproduct(const aluVector
*vec1
, const aluVector
*vec2
)
121 return vec1
->v
[0]*vec2
->v
[0] + vec1
->v
[1]*vec2
->v
[1] + vec1
->v
[2]*vec2
->v
[2];
124 static inline void aluNormalize(ALfloat
*vec
)
126 ALfloat lengthsqr
= vec
[0]*vec
[0] + vec
[1]*vec
[1] + vec
[2]*vec
[2];
129 ALfloat inv_length
= 1.0f
/sqrtf(lengthsqr
);
130 vec
[0] *= inv_length
;
131 vec
[1] *= inv_length
;
132 vec
[2] *= inv_length
;
136 static inline ALvoid
aluMatrixVector(aluVector
*vec
, const aluMatrix
*mtx
)
140 vec
->v
[0] = v
.v
[0]*mtx
->m
[0][0] + v
.v
[1]*mtx
->m
[1][0] + v
.v
[2]*mtx
->m
[2][0] + v
.v
[3]*mtx
->m
[3][0];
141 vec
->v
[1] = v
.v
[0]*mtx
->m
[0][1] + v
.v
[1]*mtx
->m
[1][1] + v
.v
[2]*mtx
->m
[2][1] + v
.v
[3]*mtx
->m
[3][1];
142 vec
->v
[2] = v
.v
[0]*mtx
->m
[0][2] + v
.v
[1]*mtx
->m
[1][2] + v
.v
[2]*mtx
->m
[2][2] + v
.v
[3]*mtx
->m
[3][2];
143 vec
->v
[3] = v
.v
[0]*mtx
->m
[0][3] + v
.v
[1]*mtx
->m
[1][3] + v
.v
[2]*mtx
->m
[2][3] + v
.v
[3]*mtx
->m
[3][3];
147 /* Calculates the fade time from the changes in gain and listener to source
148 * angle between updates. The result is a the time, in seconds, for the
149 * transition to complete.
151 static ALfloat
CalcFadeTime(ALfloat oldGain
, ALfloat newGain
, const aluVector
*olddir
, const aluVector
*newdir
)
153 ALfloat gainChange
, angleChange
, change
;
155 /* Calculate the normalized dB gain change. */
156 newGain
= maxf(newGain
, 0.0001f
);
157 oldGain
= maxf(oldGain
, 0.0001f
);
158 gainChange
= fabsf(log10f(newGain
/ oldGain
) / log10f(0.0001f
));
160 /* Calculate the angle change only when there is enough gain to notice it. */
162 if(gainChange
> 0.0001f
|| newGain
> 0.0001f
)
164 /* No angle change when the directions are equal or degenerate (when
165 * both have zero length).
167 if(newdir
->v
[0] != olddir
->v
[0] || newdir
->v
[1] != olddir
->v
[1] || newdir
->v
[2] != olddir
->v
[2])
169 ALfloat dotp
= aluDotproduct(olddir
, newdir
);
170 angleChange
= acosf(clampf(dotp
, -1.0f
, 1.0f
)) / F_PI
;
174 /* Use the largest of the two changes, and apply a significance shaping
175 * function to it. The result is then scaled to cover a 15ms transition
178 change
= maxf(angleChange
* 25.0f
, gainChange
) * 2.0f
;
179 return minf(change
, 1.0f
) * 0.015f
;
183 static void UpdateDryStepping(DirectParams
*params
, ALuint num_chans
, ALuint steps
)
190 for(i
= 0;i
< num_chans
;i
++)
192 MixGains
*gains
= params
->Gains
[i
];
193 for(j
= 0;j
< params
->OutChannels
;j
++)
195 gains
[j
].Current
= gains
[j
].Target
;
196 gains
[j
].Step
= 0.0f
;
203 delta
= 1.0f
/ (ALfloat
)steps
;
204 for(i
= 0;i
< num_chans
;i
++)
206 MixGains
*gains
= params
->Gains
[i
];
207 for(j
= 0;j
< params
->OutChannels
;j
++)
209 ALfloat diff
= gains
[j
].Target
- gains
[j
].Current
;
210 if(fabs(diff
) >= GAIN_SILENCE_THRESHOLD
)
211 gains
[j
].Step
= diff
* delta
;
213 gains
[j
].Step
= 0.0f
;
216 params
->Counter
= steps
;
219 static void UpdateWetStepping(SendParams
*params
, ALuint steps
)
225 params
->Gain
.Current
= params
->Gain
.Target
;
226 params
->Gain
.Step
= 0.0f
;
232 delta
= 1.0f
/ (ALfloat
)steps
;
234 ALfloat diff
= params
->Gain
.Target
- params
->Gain
.Current
;
235 if(fabs(diff
) >= GAIN_SILENCE_THRESHOLD
)
236 params
->Gain
.Step
= diff
* delta
;
238 params
->Gain
.Step
= 0.0f
;
240 params
->Counter
= steps
;
244 static ALvoid
CalcListenerParams(ALlistener
*Listener
)
246 ALfloat N
[3], V
[3], U
[3];
250 N
[0] = Listener
->Forward
[0];
251 N
[1] = Listener
->Forward
[1];
252 N
[2] = Listener
->Forward
[2];
254 V
[0] = Listener
->Up
[0];
255 V
[1] = Listener
->Up
[1];
256 V
[2] = Listener
->Up
[2];
258 /* Build and normalize right-vector */
259 aluCrossproduct(N
, V
, U
);
262 P
= Listener
->Position
;
264 aluMatrixSet(&Listener
->Params
.Matrix
,
265 U
[0], V
[0], -N
[0], 0.0f
,
266 U
[1], V
[1], -N
[1], 0.0f
,
267 U
[2], V
[2], -N
[2], 0.0f
,
268 0.0f
, 0.0f
, 0.0f
, 1.0f
270 aluMatrixVector(&P
, &Listener
->Params
.Matrix
);
271 aluMatrixSetRow(&Listener
->Params
.Matrix
, 3, -P
.v
[0], -P
.v
[1], -P
.v
[2], 1.0f
);
273 Listener
->Params
.Velocity
= Listener
->Velocity
;
274 aluMatrixVector(&Listener
->Params
.Velocity
, &Listener
->Params
.Matrix
);
277 ALvoid
CalcNonAttnSourceParams(ALvoice
*voice
, const ALsource
*ALSource
, const ALCcontext
*ALContext
)
279 static const struct ChanMap MonoMap
[1] = { { FrontCenter
, 0.0f
, 0.0f
} };
280 static const struct ChanMap StereoMap
[2] = {
281 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
282 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) }
284 static const struct ChanMap StereoWideMap
[2] = {
285 { FrontLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
286 { FrontRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
288 static const struct ChanMap RearMap
[2] = {
289 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
290 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) }
292 static const struct ChanMap QuadMap
[4] = {
293 { FrontLeft
, DEG2RAD( -45.0f
), DEG2RAD(0.0f
) },
294 { FrontRight
, DEG2RAD( 45.0f
), DEG2RAD(0.0f
) },
295 { BackLeft
, DEG2RAD(-135.0f
), DEG2RAD(0.0f
) },
296 { BackRight
, DEG2RAD( 135.0f
), DEG2RAD(0.0f
) }
298 static const struct ChanMap X51Map
[6] = {
299 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
300 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
301 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
303 { SideLeft
, DEG2RAD(-110.0f
), DEG2RAD(0.0f
) },
304 { SideRight
, DEG2RAD( 110.0f
), DEG2RAD(0.0f
) }
306 static const struct ChanMap X61Map
[7] = {
307 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
308 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
309 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
311 { BackCenter
, DEG2RAD(180.0f
), DEG2RAD(0.0f
) },
312 { SideLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
313 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
315 static const struct ChanMap X71Map
[8] = {
316 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
317 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
318 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
320 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
321 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) },
322 { SideLeft
, DEG2RAD( -90.0f
), DEG2RAD(0.0f
) },
323 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
326 ALCdevice
*Device
= ALContext
->Device
;
327 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
328 ALbufferlistitem
*BufferListItem
;
329 enum FmtChannels Channels
;
330 ALfloat DryGain
, DryGainHF
, DryGainLF
;
331 ALfloat WetGain
[MAX_SENDS
];
332 ALfloat WetGainHF
[MAX_SENDS
];
333 ALfloat WetGainLF
[MAX_SENDS
];
334 ALuint NumSends
, Frequency
;
336 const struct ChanMap
*chans
= NULL
;
337 ALuint num_channels
= 0;
338 ALboolean DirectChannels
;
339 ALboolean isbformat
= AL_FALSE
;
343 /* Get device properties */
344 NumSends
= Device
->NumAuxSends
;
345 Frequency
= Device
->Frequency
;
347 /* Get listener properties */
348 ListenerGain
= ALContext
->Listener
->Gain
;
350 /* Get source properties */
351 SourceVolume
= ALSource
->Gain
;
352 MinVolume
= ALSource
->MinGain
;
353 MaxVolume
= ALSource
->MaxGain
;
354 Pitch
= ALSource
->Pitch
;
355 Relative
= ALSource
->HeadRelative
;
356 DirectChannels
= ALSource
->DirectChannels
;
358 voice
->Direct
.OutBuffer
= Device
->DryBuffer
;
359 voice
->Direct
.OutChannels
= Device
->NumChannels
;
360 for(i
= 0;i
< NumSends
;i
++)
362 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
364 Slot
= Device
->DefaultSlot
;
365 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
366 voice
->Send
[i
].OutBuffer
= NULL
;
368 voice
->Send
[i
].OutBuffer
= Slot
->WetBuffer
;
371 /* Calculate the stepping value */
373 BufferListItem
= ATOMIC_LOAD(&ALSource
->queue
);
374 while(BufferListItem
!= NULL
)
377 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
379 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
380 if(Pitch
> (ALfloat
)MAX_PITCH
)
381 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
384 voice
->Step
= fastf2i(Pitch
*FRACTIONONE
);
389 Channels
= ALBuffer
->FmtChannels
;
392 BufferListItem
= BufferListItem
->next
;
395 /* Calculate gains */
396 DryGain
= clampf(SourceVolume
, MinVolume
, MaxVolume
);
397 DryGain
*= ALSource
->Direct
.Gain
* ListenerGain
;
398 DryGainHF
= ALSource
->Direct
.GainHF
;
399 DryGainLF
= ALSource
->Direct
.GainLF
;
400 for(i
= 0;i
< NumSends
;i
++)
402 WetGain
[i
] = clampf(SourceVolume
, MinVolume
, MaxVolume
);
403 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
404 WetGainHF
[i
] = ALSource
->Send
[i
].GainHF
;
405 WetGainLF
[i
] = ALSource
->Send
[i
].GainLF
;
416 /* HACK: Place the stereo channels at +/-90 degrees when using non-
417 * HRTF stereo output. This helps reduce the "monoization" caused
418 * by them panning towards the center. */
419 if(Device
->FmtChans
== DevFmtStereo
&& !Device
->Hrtf
)
420 chans
= StereoWideMap
;
454 DirectChannels
= AL_FALSE
;
460 DirectChannels
= AL_FALSE
;
466 ALfloat N
[3], V
[3], U
[3];
470 N
[0] = ALSource
->Orientation
[0][0];
471 N
[1] = ALSource
->Orientation
[0][1];
472 N
[2] = ALSource
->Orientation
[0][2];
474 V
[0] = ALSource
->Orientation
[1][0];
475 V
[1] = ALSource
->Orientation
[1][1];
476 V
[2] = ALSource
->Orientation
[1][2];
480 const aluMatrix
*lmatrix
= &ALContext
->Listener
->Params
.Matrix
;
482 aluVectorSet(&at
, N
[0], N
[1], N
[2], 0.0f
);
483 aluVectorSet(&up
, V
[0], V
[1], V
[2], 0.0f
);
484 aluMatrixVector(&at
, lmatrix
);
485 aluMatrixVector(&up
, lmatrix
);
486 N
[0] = at
.v
[0]; N
[1] = at
.v
[1]; N
[2] = at
.v
[2];
487 V
[0] = up
.v
[0]; V
[1] = up
.v
[1]; V
[2] = up
.v
[2];
489 /* Build and normalize right-vector */
490 aluCrossproduct(N
, V
, U
);
493 aluMatrixSet(&matrix
,
494 1.0f
, 0.0f
, 0.0f
, 0.0f
,
495 0.0f
, -N
[2], -N
[0], N
[1],
496 0.0f
, U
[2], U
[0], -U
[1],
497 0.0f
, -V
[2], -V
[0], V
[1]
500 for(c
= 0;c
< num_channels
;c
++)
502 MixGains
*gains
= voice
->Direct
.Gains
[c
];
503 ALfloat Target
[MAX_OUTPUT_CHANNELS
];
505 ComputeBFormatGains(Device
, matrix
.m
[c
], DryGain
, Target
);
506 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
507 gains
[i
].Target
= Target
[i
];
509 UpdateDryStepping(&voice
->Direct
, num_channels
, (voice
->Direct
.Moving
? 64 : 0));
510 voice
->Direct
.Moving
= AL_TRUE
;
512 voice
->IsHrtf
= AL_FALSE
;
513 for(i
= 0;i
< NumSends
;i
++)
514 WetGain
[i
] *= 1.4142f
;
516 else if(DirectChannels
!= AL_FALSE
)
520 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
521 voice
->Direct
.OutChannels
= 2;
522 for(c
= 0;c
< num_channels
;c
++)
524 MixGains
*gains
= voice
->Direct
.Gains
[c
];
526 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
527 gains
[j
].Target
= 0.0f
;
529 if(chans
[c
].channel
== FrontLeft
)
530 gains
[0].Target
= DryGain
;
531 else if(chans
[c
].channel
== FrontRight
)
532 gains
[1].Target
= DryGain
;
535 else for(c
= 0;c
< num_channels
;c
++)
537 MixGains
*gains
= voice
->Direct
.Gains
[c
];
540 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
541 gains
[j
].Target
= 0.0f
;
542 if((idx
=GetChannelIdxByName(Device
, chans
[c
].channel
)) != -1)
543 gains
[idx
].Target
= DryGain
;
545 UpdateDryStepping(&voice
->Direct
, num_channels
, (voice
->Direct
.Moving
? 64 : 0));
546 voice
->Direct
.Moving
= AL_TRUE
;
548 voice
->IsHrtf
= AL_FALSE
;
550 else if(Device
->Hrtf
)
552 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
553 voice
->Direct
.OutChannels
= 2;
554 for(c
= 0;c
< num_channels
;c
++)
556 if(chans
[c
].channel
== LFE
)
559 voice
->Direct
.Hrtf
.Params
[c
].Delay
[0] = 0;
560 voice
->Direct
.Hrtf
.Params
[c
].Delay
[1] = 0;
561 for(i
= 0;i
< HRIR_LENGTH
;i
++)
563 voice
->Direct
.Hrtf
.Params
[c
].Coeffs
[i
][0] = 0.0f
;
564 voice
->Direct
.Hrtf
.Params
[c
].Coeffs
[i
][1] = 0.0f
;
569 /* Get the static HRIR coefficients and delays for this
571 GetLerpedHrtfCoeffs(Device
->Hrtf
,
572 chans
[c
].elevation
, chans
[c
].angle
, 1.0f
, DryGain
,
573 voice
->Direct
.Hrtf
.Params
[c
].Coeffs
,
574 voice
->Direct
.Hrtf
.Params
[c
].Delay
);
577 voice
->Direct
.Counter
= 0;
578 voice
->Direct
.Moving
= AL_TRUE
;
580 voice
->IsHrtf
= AL_TRUE
;
584 for(c
= 0;c
< num_channels
;c
++)
586 MixGains
*gains
= voice
->Direct
.Gains
[c
];
587 ALfloat Target
[MAX_OUTPUT_CHANNELS
];
589 /* Special-case LFE */
590 if(chans
[c
].channel
== LFE
)
593 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
594 gains
[i
].Target
= 0.0f
;
595 if((idx
=GetChannelIdxByName(Device
, chans
[c
].channel
)) != -1)
596 gains
[idx
].Target
= DryGain
;
600 ComputeAngleGains(Device
, chans
[c
].angle
, chans
[c
].elevation
, DryGain
, Target
);
601 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
602 gains
[i
].Target
= Target
[i
];
604 UpdateDryStepping(&voice
->Direct
, num_channels
, (voice
->Direct
.Moving
? 64 : 0));
605 voice
->Direct
.Moving
= AL_TRUE
;
607 voice
->IsHrtf
= AL_FALSE
;
609 for(i
= 0;i
< NumSends
;i
++)
611 voice
->Send
[i
].Gain
.Target
= WetGain
[i
];
612 UpdateWetStepping(&voice
->Send
[i
], (voice
->Send
[i
].Moving
? 64 : 0));
613 voice
->Send
[i
].Moving
= AL_TRUE
;
617 ALfloat gainhf
= maxf(0.01f
, DryGainHF
);
618 ALfloat gainlf
= maxf(0.01f
, DryGainLF
);
619 ALfloat hfscale
= ALSource
->Direct
.HFReference
/ Frequency
;
620 ALfloat lfscale
= ALSource
->Direct
.LFReference
/ Frequency
;
621 for(c
= 0;c
< num_channels
;c
++)
623 voice
->Direct
.Filters
[c
].ActiveType
= AF_None
;
624 if(gainhf
!= 1.0f
) voice
->Direct
.Filters
[c
].ActiveType
|= AF_LowPass
;
625 if(gainlf
!= 1.0f
) voice
->Direct
.Filters
[c
].ActiveType
|= AF_HighPass
;
626 ALfilterState_setParams(
627 &voice
->Direct
.Filters
[c
].LowPass
, ALfilterType_HighShelf
, gainhf
,
630 ALfilterState_setParams(
631 &voice
->Direct
.Filters
[c
].HighPass
, ALfilterType_LowShelf
, gainlf
,
636 for(i
= 0;i
< NumSends
;i
++)
638 ALfloat gainhf
= maxf(0.01f
, WetGainHF
[i
]);
639 ALfloat gainlf
= maxf(0.01f
, WetGainLF
[i
]);
640 ALfloat hfscale
= ALSource
->Send
[i
].HFReference
/ Frequency
;
641 ALfloat lfscale
= ALSource
->Send
[i
].LFReference
/ Frequency
;
642 for(c
= 0;c
< num_channels
;c
++)
644 voice
->Send
[i
].Filters
[c
].ActiveType
= AF_None
;
645 if(gainhf
!= 1.0f
) voice
->Send
[i
].Filters
[c
].ActiveType
|= AF_LowPass
;
646 if(gainlf
!= 1.0f
) voice
->Send
[i
].Filters
[c
].ActiveType
|= AF_HighPass
;
647 ALfilterState_setParams(
648 &voice
->Send
[i
].Filters
[c
].LowPass
, ALfilterType_HighShelf
, gainhf
,
651 ALfilterState_setParams(
652 &voice
->Send
[i
].Filters
[c
].HighPass
, ALfilterType_LowShelf
, gainlf
,
659 ALvoid
CalcSourceParams(ALvoice
*voice
, const ALsource
*ALSource
, const ALCcontext
*ALContext
)
661 ALCdevice
*Device
= ALContext
->Device
;
662 aluVector Position
, Velocity
, Direction
, SourceToListener
;
663 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,ClampedDist
;
664 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
;
665 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
666 ALfloat DopplerFactor
, SpeedOfSound
;
667 ALfloat AirAbsorptionFactor
;
668 ALfloat RoomAirAbsorption
[MAX_SENDS
];
669 ALbufferlistitem
*BufferListItem
;
671 ALfloat RoomAttenuation
[MAX_SENDS
];
672 ALfloat MetersPerUnit
;
673 ALfloat RoomRolloffBase
;
674 ALfloat RoomRolloff
[MAX_SENDS
];
675 ALfloat DecayDistance
[MAX_SENDS
];
679 ALboolean DryGainHFAuto
;
680 ALfloat WetGain
[MAX_SENDS
];
681 ALfloat WetGainHF
[MAX_SENDS
];
682 ALfloat WetGainLF
[MAX_SENDS
];
683 ALboolean WetGainAuto
;
684 ALboolean WetGainHFAuto
;
692 for(i
= 0;i
< MAX_SENDS
;i
++)
698 /* Get context/device properties */
699 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
700 SpeedOfSound
= ALContext
->SpeedOfSound
* ALContext
->DopplerVelocity
;
701 NumSends
= Device
->NumAuxSends
;
702 Frequency
= Device
->Frequency
;
704 /* Get listener properties */
705 ListenerGain
= ALContext
->Listener
->Gain
;
706 MetersPerUnit
= ALContext
->Listener
->MetersPerUnit
;
708 /* Get source properties */
709 SourceVolume
= ALSource
->Gain
;
710 MinVolume
= ALSource
->MinGain
;
711 MaxVolume
= ALSource
->MaxGain
;
712 Pitch
= ALSource
->Pitch
;
713 Position
= ALSource
->Position
;
714 Direction
= ALSource
->Direction
;
715 Velocity
= ALSource
->Velocity
;
716 MinDist
= ALSource
->RefDistance
;
717 MaxDist
= ALSource
->MaxDistance
;
718 Rolloff
= ALSource
->RollOffFactor
;
719 InnerAngle
= ALSource
->InnerAngle
;
720 OuterAngle
= ALSource
->OuterAngle
;
721 AirAbsorptionFactor
= ALSource
->AirAbsorptionFactor
;
722 DryGainHFAuto
= ALSource
->DryGainHFAuto
;
723 WetGainAuto
= ALSource
->WetGainAuto
;
724 WetGainHFAuto
= ALSource
->WetGainHFAuto
;
725 RoomRolloffBase
= ALSource
->RoomRolloffFactor
;
727 voice
->Direct
.OutBuffer
= Device
->DryBuffer
;
728 voice
->Direct
.OutChannels
= Device
->NumChannels
;
729 for(i
= 0;i
< NumSends
;i
++)
731 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
734 Slot
= Device
->DefaultSlot
;
735 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
738 RoomRolloff
[i
] = 0.0f
;
739 DecayDistance
[i
] = 0.0f
;
740 RoomAirAbsorption
[i
] = 1.0f
;
742 else if(Slot
->AuxSendAuto
)
744 RoomRolloff
[i
] = RoomRolloffBase
;
745 if(IsReverbEffect(Slot
->EffectType
))
747 RoomRolloff
[i
] += Slot
->EffectProps
.Reverb
.RoomRolloffFactor
;
748 DecayDistance
[i
] = Slot
->EffectProps
.Reverb
.DecayTime
*
749 SPEEDOFSOUNDMETRESPERSEC
;
750 RoomAirAbsorption
[i
] = Slot
->EffectProps
.Reverb
.AirAbsorptionGainHF
;
754 DecayDistance
[i
] = 0.0f
;
755 RoomAirAbsorption
[i
] = 1.0f
;
760 /* If the slot's auxiliary send auto is off, the data sent to the
761 * effect slot is the same as the dry path, sans filter effects */
762 RoomRolloff
[i
] = Rolloff
;
763 DecayDistance
[i
] = 0.0f
;
764 RoomAirAbsorption
[i
] = AIRABSORBGAINHF
;
767 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
768 voice
->Send
[i
].OutBuffer
= NULL
;
770 voice
->Send
[i
].OutBuffer
= Slot
->WetBuffer
;
773 /* Transform source to listener space (convert to head relative) */
774 if(ALSource
->HeadRelative
== AL_FALSE
)
776 const aluMatrix
*Matrix
= &ALContext
->Listener
->Params
.Matrix
;
777 /* Transform source vectors */
778 aluMatrixVector(&Position
, Matrix
);
779 aluMatrixVector(&Velocity
, Matrix
);
780 aluMatrixVector(&Direction
, Matrix
);
784 const aluVector
*lvelocity
= &ALContext
->Listener
->Params
.Velocity
;
785 /* Offset the source velocity to be relative of the listener velocity */
786 Velocity
.v
[0] += lvelocity
->v
[0];
787 Velocity
.v
[1] += lvelocity
->v
[1];
788 Velocity
.v
[2] += lvelocity
->v
[2];
791 SourceToListener
.v
[0] = -Position
.v
[0];
792 SourceToListener
.v
[1] = -Position
.v
[1];
793 SourceToListener
.v
[2] = -Position
.v
[2];
794 SourceToListener
.v
[3] = 0.0f
;
795 aluNormalize(SourceToListener
.v
);
796 aluNormalize(Direction
.v
);
798 /* Calculate distance attenuation */
799 Distance
= sqrtf(aluDotproduct(&Position
, &Position
));
800 ClampedDist
= Distance
;
803 for(i
= 0;i
< NumSends
;i
++)
804 RoomAttenuation
[i
] = 1.0f
;
805 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
806 ALContext
->DistanceModel
)
808 case InverseDistanceClamped
:
809 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
810 if(MaxDist
< MinDist
)
813 case InverseDistance
:
816 ALfloat dist
= lerp(MinDist
, ClampedDist
, Rolloff
);
817 if(dist
> 0.0f
) Attenuation
= MinDist
/ dist
;
818 for(i
= 0;i
< NumSends
;i
++)
820 dist
= lerp(MinDist
, ClampedDist
, RoomRolloff
[i
]);
821 if(dist
> 0.0f
) RoomAttenuation
[i
] = MinDist
/ dist
;
826 case LinearDistanceClamped
:
827 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
828 if(MaxDist
< MinDist
)
832 if(MaxDist
!= MinDist
)
834 Attenuation
= 1.0f
- (Rolloff
*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
835 Attenuation
= maxf(Attenuation
, 0.0f
);
836 for(i
= 0;i
< NumSends
;i
++)
838 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
839 RoomAttenuation
[i
] = maxf(RoomAttenuation
[i
], 0.0f
);
844 case ExponentDistanceClamped
:
845 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
846 if(MaxDist
< MinDist
)
849 case ExponentDistance
:
850 if(ClampedDist
> 0.0f
&& MinDist
> 0.0f
)
852 Attenuation
= powf(ClampedDist
/MinDist
, -Rolloff
);
853 for(i
= 0;i
< NumSends
;i
++)
854 RoomAttenuation
[i
] = powf(ClampedDist
/MinDist
, -RoomRolloff
[i
]);
858 case DisableDistance
:
859 ClampedDist
= MinDist
;
863 /* Source Gain + Attenuation */
864 DryGain
= SourceVolume
* Attenuation
;
865 for(i
= 0;i
< NumSends
;i
++)
866 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
868 /* Distance-based air absorption */
869 if(AirAbsorptionFactor
> 0.0f
&& ClampedDist
> MinDist
)
871 ALfloat meters
= (ClampedDist
-MinDist
) * MetersPerUnit
;
872 DryGainHF
*= powf(AIRABSORBGAINHF
, AirAbsorptionFactor
*meters
);
873 for(i
= 0;i
< NumSends
;i
++)
874 WetGainHF
[i
] *= powf(RoomAirAbsorption
[i
], AirAbsorptionFactor
*meters
);
879 ALfloat ApparentDist
= 1.0f
/maxf(Attenuation
, 0.00001f
) - 1.0f
;
881 /* Apply a decay-time transformation to the wet path, based on the
882 * attenuation of the dry path.
884 * Using the apparent distance, based on the distance attenuation, the
885 * initial decay of the reverb effect is calculated and applied to the
888 for(i
= 0;i
< NumSends
;i
++)
890 if(DecayDistance
[i
] > 0.0f
)
891 WetGain
[i
] *= powf(0.001f
/*-60dB*/, ApparentDist
/DecayDistance
[i
]);
895 /* Calculate directional soundcones */
896 Angle
= RAD2DEG(acosf(aluDotproduct(&Direction
, &SourceToListener
)) * ConeScale
) * 2.0f
;
897 if(Angle
> InnerAngle
&& Angle
<= OuterAngle
)
899 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
900 ConeVolume
= lerp(1.0f
, ALSource
->OuterGain
, scale
);
901 ConeHF
= lerp(1.0f
, ALSource
->OuterGainHF
, scale
);
903 else if(Angle
> OuterAngle
)
905 ConeVolume
= ALSource
->OuterGain
;
906 ConeHF
= ALSource
->OuterGainHF
;
914 DryGain
*= ConeVolume
;
917 for(i
= 0;i
< NumSends
;i
++)
918 WetGain
[i
] *= ConeVolume
;
924 for(i
= 0;i
< NumSends
;i
++)
925 WetGainHF
[i
] *= ConeHF
;
928 /* Clamp to Min/Max Gain */
929 DryGain
= clampf(DryGain
, MinVolume
, MaxVolume
);
930 for(i
= 0;i
< NumSends
;i
++)
931 WetGain
[i
] = clampf(WetGain
[i
], MinVolume
, MaxVolume
);
933 /* Apply gain and frequency filters */
934 DryGain
*= ALSource
->Direct
.Gain
* ListenerGain
;
935 DryGainHF
*= ALSource
->Direct
.GainHF
;
936 DryGainLF
*= ALSource
->Direct
.GainLF
;
937 for(i
= 0;i
< NumSends
;i
++)
939 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
940 WetGainHF
[i
] *= ALSource
->Send
[i
].GainHF
;
941 WetGainLF
[i
] *= ALSource
->Send
[i
].GainLF
;
944 /* Calculate velocity-based doppler effect */
945 if(DopplerFactor
> 0.0f
)
947 const aluVector
*lvelocity
= &ALContext
->Listener
->Params
.Velocity
;
950 if(SpeedOfSound
< 1.0f
)
952 DopplerFactor
*= 1.0f
/SpeedOfSound
;
956 VSS
= aluDotproduct(&Velocity
, &SourceToListener
) * DopplerFactor
;
957 VLS
= aluDotproduct(lvelocity
, &SourceToListener
) * DopplerFactor
;
959 Pitch
*= clampf(SpeedOfSound
-VLS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
) /
960 clampf(SpeedOfSound
-VSS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
);
963 BufferListItem
= ATOMIC_LOAD(&ALSource
->queue
);
964 while(BufferListItem
!= NULL
)
967 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
969 /* Calculate fixed-point stepping value, based on the pitch, buffer
970 * frequency, and output frequency. */
971 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
972 if(Pitch
> (ALfloat
)MAX_PITCH
)
973 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
976 voice
->Step
= fastf2i(Pitch
*FRACTIONONE
);
983 BufferListItem
= BufferListItem
->next
;
988 /* Use a binaural HRTF algorithm for stereo headphone playback */
989 aluVector dir
= {{ 0.0f
, 0.0f
, -1.0f
, 0.0f
}};
990 ALfloat ev
= 0.0f
, az
= 0.0f
;
991 ALfloat radius
= ALSource
->Radius
;
992 ALfloat dirfact
= 1.0f
;
994 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
995 voice
->Direct
.OutChannels
= 2;
997 if(Distance
> FLT_EPSILON
)
999 ALfloat invlen
= 1.0f
/Distance
;
1000 dir
.v
[0] = Position
.v
[0] * invlen
;
1001 dir
.v
[1] = Position
.v
[1] * invlen
;
1002 dir
.v
[2] = Position
.v
[2] * invlen
* ZScale
;
1004 /* Calculate elevation and azimuth only when the source is not at
1005 * the listener. This prevents +0 and -0 Z from producing
1006 * inconsistent panning. Also, clamp Y in case FP precision errors
1007 * cause it to land outside of -1..+1. */
1008 ev
= asinf(clampf(dir
.v
[1], -1.0f
, 1.0f
));
1009 az
= atan2f(dir
.v
[0], -dir
.v
[2]);
1011 if(radius
> Distance
)
1012 dirfact
*= Distance
/ radius
;
1014 /* Check to see if the HRIR is already moving. */
1015 if(voice
->Direct
.Moving
)
1018 delta
= CalcFadeTime(voice
->Direct
.LastGain
, DryGain
,
1019 &voice
->Direct
.LastDir
, &dir
);
1020 /* If the delta is large enough, get the moving HRIR target
1021 * coefficients, target delays, steppping values, and counter. */
1022 if(delta
> 0.000015f
)
1024 ALuint counter
= GetMovingHrtfCoeffs(Device
->Hrtf
,
1025 ev
, az
, dirfact
, DryGain
, delta
, voice
->Direct
.Counter
,
1026 voice
->Direct
.Hrtf
.Params
[0].Coeffs
, voice
->Direct
.Hrtf
.Params
[0].Delay
,
1027 voice
->Direct
.Hrtf
.Params
[0].CoeffStep
, voice
->Direct
.Hrtf
.Params
[0].DelayStep
1029 voice
->Direct
.Counter
= counter
;
1030 voice
->Direct
.LastGain
= DryGain
;
1031 voice
->Direct
.LastDir
= dir
;
1036 /* Get the initial (static) HRIR coefficients and delays. */
1037 GetLerpedHrtfCoeffs(Device
->Hrtf
, ev
, az
, dirfact
, DryGain
,
1038 voice
->Direct
.Hrtf
.Params
[0].Coeffs
,
1039 voice
->Direct
.Hrtf
.Params
[0].Delay
);
1040 voice
->Direct
.Counter
= 0;
1041 voice
->Direct
.Moving
= AL_TRUE
;
1042 voice
->Direct
.LastGain
= DryGain
;
1043 voice
->Direct
.LastDir
= dir
;
1046 voice
->IsHrtf
= AL_TRUE
;
1050 MixGains
*gains
= voice
->Direct
.Gains
[0];
1051 ALfloat dir
[3] = { 0.0f
, 0.0f
, -1.0f
};
1052 ALfloat radius
= ALSource
->Radius
;
1053 ALfloat Target
[MAX_OUTPUT_CHANNELS
];
1055 /* Normalize the length, and compute panned gains. */
1056 if(Distance
> FLT_EPSILON
|| radius
> FLT_EPSILON
)
1058 ALfloat invlen
= 1.0f
/maxf(Distance
, radius
);
1059 dir
[0] = Position
.v
[0] * invlen
;
1060 dir
[1] = Position
.v
[1] * invlen
;
1061 dir
[2] = Position
.v
[2] * invlen
* ZScale
;
1063 ComputeDirectionalGains(Device
, dir
, DryGain
, Target
);
1065 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
1066 gains
[j
].Target
= Target
[j
];
1067 UpdateDryStepping(&voice
->Direct
, 1, (voice
->Direct
.Moving
? 64 : 0));
1068 voice
->Direct
.Moving
= AL_TRUE
;
1070 voice
->IsHrtf
= AL_FALSE
;
1072 for(i
= 0;i
< NumSends
;i
++)
1074 voice
->Send
[i
].Gain
.Target
= WetGain
[i
];
1075 UpdateWetStepping(&voice
->Send
[i
], (voice
->Send
[i
].Moving
? 64 : 0));
1076 voice
->Send
[i
].Moving
= AL_TRUE
;
1080 ALfloat gainhf
= maxf(0.01f
, DryGainHF
);
1081 ALfloat gainlf
= maxf(0.01f
, DryGainLF
);
1082 ALfloat hfscale
= ALSource
->Direct
.HFReference
/ Frequency
;
1083 ALfloat lfscale
= ALSource
->Direct
.LFReference
/ Frequency
;
1084 voice
->Direct
.Filters
[0].ActiveType
= AF_None
;
1085 if(gainhf
!= 1.0f
) voice
->Direct
.Filters
[0].ActiveType
|= AF_LowPass
;
1086 if(gainlf
!= 1.0f
) voice
->Direct
.Filters
[0].ActiveType
|= AF_HighPass
;
1087 ALfilterState_setParams(
1088 &voice
->Direct
.Filters
[0].LowPass
, ALfilterType_HighShelf
, gainhf
,
1091 ALfilterState_setParams(
1092 &voice
->Direct
.Filters
[0].HighPass
, ALfilterType_LowShelf
, gainlf
,
1096 for(i
= 0;i
< NumSends
;i
++)
1098 ALfloat gainhf
= maxf(0.01f
, WetGainHF
[i
]);
1099 ALfloat gainlf
= maxf(0.01f
, WetGainLF
[i
]);
1100 ALfloat hfscale
= ALSource
->Send
[i
].HFReference
/ Frequency
;
1101 ALfloat lfscale
= ALSource
->Send
[i
].LFReference
/ Frequency
;
1102 voice
->Send
[i
].Filters
[0].ActiveType
= AF_None
;
1103 if(gainhf
!= 1.0f
) voice
->Send
[i
].Filters
[0].ActiveType
|= AF_LowPass
;
1104 if(gainlf
!= 1.0f
) voice
->Send
[i
].Filters
[0].ActiveType
|= AF_HighPass
;
1105 ALfilterState_setParams(
1106 &voice
->Send
[i
].Filters
[0].LowPass
, ALfilterType_HighShelf
, gainhf
,
1109 ALfilterState_setParams(
1110 &voice
->Send
[i
].Filters
[0].HighPass
, ALfilterType_LowShelf
, gainlf
,
1117 static inline ALint
aluF2I25(ALfloat val
)
1119 /* Clamp the value between -1 and +1. This handles that with only a single branch. */
1120 if(fabsf(val
) > 1.0f
)
1121 val
= (ALfloat
)((0.0f
< val
) - (val
< 0.0f
));
1122 /* Convert to a signed integer, between -16777215 and +16777215. */
1123 return fastf2i(val
*16777215.0f
);
1126 static inline ALfloat
aluF2F(ALfloat val
)
1128 static inline ALint
aluF2I(ALfloat val
)
1129 { return aluF2I25(val
)<<7; }
1130 static inline ALuint
aluF2UI(ALfloat val
)
1131 { return aluF2I(val
)+2147483648u; }
1132 static inline ALshort
aluF2S(ALfloat val
)
1133 { return aluF2I25(val
)>>9; }
1134 static inline ALushort
aluF2US(ALfloat val
)
1135 { return aluF2S(val
)+32768; }
1136 static inline ALbyte
aluF2B(ALfloat val
)
1137 { return aluF2I25(val
)>>17; }
1138 static inline ALubyte
aluF2UB(ALfloat val
)
1139 { return aluF2B(val
)+128; }
1141 #define DECL_TEMPLATE(T, func) \
1142 static void Write_##T(const ALfloatBUFFERSIZE *InBuffer, ALvoid *OutBuffer, \
1143 ALuint SamplesToDo, ALuint numchans) \
1146 for(j = 0;j < numchans;j++) \
1148 const ALfloat *in = InBuffer[j]; \
1149 T *restrict out = (T*)OutBuffer + j; \
1150 for(i = 0;i < SamplesToDo;i++) \
1151 out[i*numchans] = func(in[i]); \
1155 DECL_TEMPLATE(ALfloat
, aluF2F
)
1156 DECL_TEMPLATE(ALuint
, aluF2UI
)
1157 DECL_TEMPLATE(ALint
, aluF2I
)
1158 DECL_TEMPLATE(ALushort
, aluF2US
)
1159 DECL_TEMPLATE(ALshort
, aluF2S
)
1160 DECL_TEMPLATE(ALubyte
, aluF2UB
)
1161 DECL_TEMPLATE(ALbyte
, aluF2B
)
1163 #undef DECL_TEMPLATE
1166 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1169 ALeffectslot
**slot
, **slot_end
;
1170 ALvoice
*voice
, *voice_end
;
1175 SetMixerFPUMode(&oldMode
);
1179 ALuint outchanoffset
= 0;
1180 ALuint outchancount
= device
->NumChannels
;
1182 IncrementRef(&device
->MixCount
);
1184 SamplesToDo
= minu(size
, BUFFERSIZE
);
1185 for(c
= 0;c
< device
->NumChannels
;c
++)
1186 memset(device
->DryBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1189 outchanoffset
= device
->NumChannels
;
1191 for(c
= 0;c
< outchancount
;c
++)
1192 memset(device
->DryBuffer
[outchanoffset
+c
], 0, SamplesToDo
*sizeof(ALfloat
));
1195 V0(device
->Backend
,lock
)();
1196 V(device
->Synth
,process
)(SamplesToDo
, &device
->DryBuffer
[outchanoffset
]);
1198 ctx
= ATOMIC_LOAD(&device
->ContextList
);
1201 ALenum DeferUpdates
= ctx
->DeferUpdates
;
1202 ALenum UpdateSources
= AL_FALSE
;
1205 UpdateSources
= ATOMIC_EXCHANGE(ALenum
, &ctx
->UpdateSources
, AL_FALSE
);
1208 CalcListenerParams(ctx
->Listener
);
1210 /* source processing */
1211 voice
= ctx
->Voices
;
1212 voice_end
= voice
+ ctx
->VoiceCount
;
1213 while(voice
!= voice_end
)
1215 ALsource
*source
= voice
->Source
;
1216 if(!source
) goto next
;
1218 if(source
->state
!= AL_PLAYING
&& source
->state
!= AL_PAUSED
)
1220 voice
->Source
= NULL
;
1224 if(!DeferUpdates
&& (ATOMIC_EXCHANGE(ALenum
, &source
->NeedsUpdate
, AL_FALSE
) ||
1226 voice
->Update(voice
, source
, ctx
);
1228 if(source
->state
!= AL_PAUSED
)
1229 MixSource(voice
, source
, device
, SamplesToDo
);
1234 /* effect slot processing */
1235 slot
= VECTOR_ITER_BEGIN(ctx
->ActiveAuxSlots
);
1236 slot_end
= VECTOR_ITER_END(ctx
->ActiveAuxSlots
);
1237 while(slot
!= slot_end
)
1239 if(!DeferUpdates
&& ATOMIC_EXCHANGE(ALenum
, &(*slot
)->NeedsUpdate
, AL_FALSE
))
1240 V((*slot
)->EffectState
,update
)(device
, *slot
);
1242 V((*slot
)->EffectState
,process
)(SamplesToDo
, (*slot
)->WetBuffer
[0],
1243 device
->DryBuffer
, device
->NumChannels
);
1245 for(i
= 0;i
< SamplesToDo
;i
++)
1246 (*slot
)->WetBuffer
[0][i
] = 0.0f
;
1254 slot
= &device
->DefaultSlot
;
1257 if(ATOMIC_EXCHANGE(ALenum
, &(*slot
)->NeedsUpdate
, AL_FALSE
))
1258 V((*slot
)->EffectState
,update
)(device
, *slot
);
1260 V((*slot
)->EffectState
,process
)(SamplesToDo
, (*slot
)->WetBuffer
[0],
1261 device
->DryBuffer
, device
->NumChannels
);
1263 for(i
= 0;i
< SamplesToDo
;i
++)
1264 (*slot
)->WetBuffer
[0][i
] = 0.0f
;
1267 /* Increment the clock time. Every second's worth of samples is
1268 * converted and added to clock base so that large sample counts don't
1269 * overflow during conversion. This also guarantees an exact, stable
1271 device
->SamplesDone
+= SamplesToDo
;
1272 device
->ClockBase
+= (device
->SamplesDone
/device
->Frequency
) * DEVICE_CLOCK_RES
;
1273 device
->SamplesDone
%= device
->Frequency
;
1274 V0(device
->Backend
,unlock
)();
1278 HrtfMixerFunc HrtfMix
= SelectHrtfMixer();
1279 ALuint irsize
= GetHrtfIrSize(device
->Hrtf
);
1280 for(c
= 0;c
< device
->NumChannels
;c
++)
1281 HrtfMix(&device
->DryBuffer
[outchanoffset
], device
->DryBuffer
[c
], 0,
1282 device
->Hrtf_Offset
, 0, irsize
, &device
->Hrtf_Params
[c
],
1283 &device
->Hrtf_State
[c
], SamplesToDo
1285 device
->Hrtf_Offset
+= SamplesToDo
;
1287 else if(device
->Bs2b
)
1289 /* Apply binaural/crossfeed filter */
1290 for(i
= 0;i
< SamplesToDo
;i
++)
1293 samples
[0] = device
->DryBuffer
[0][i
];
1294 samples
[1] = device
->DryBuffer
[1][i
];
1295 bs2b_cross_feed(device
->Bs2b
, samples
);
1296 device
->DryBuffer
[0][i
] = samples
[0];
1297 device
->DryBuffer
[1][i
] = samples
[1];
1303 #define WRITE(T, a, b, c, d) do { \
1304 Write_##T((a), (b), (c), (d)); \
1305 buffer = (char*)buffer + (c)*(d)*sizeof(T); \
1307 switch(device
->FmtType
)
1310 WRITE(ALbyte
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1313 WRITE(ALubyte
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1316 WRITE(ALshort
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1319 WRITE(ALushort
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1322 WRITE(ALint
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1325 WRITE(ALuint
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1328 WRITE(ALfloat
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1334 size
-= SamplesToDo
;
1335 IncrementRef(&device
->MixCount
);
1338 RestoreFPUMode(&oldMode
);
1342 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1344 ALCcontext
*Context
;
1346 device
->Connected
= ALC_FALSE
;
1348 Context
= ATOMIC_LOAD(&device
->ContextList
);
1351 ALvoice
*voice
, *voice_end
;
1353 voice
= Context
->Voices
;
1354 voice_end
= voice
+ Context
->VoiceCount
;
1355 while(voice
!= voice_end
)
1357 ALsource
*source
= voice
->Source
;
1358 voice
->Source
= NULL
;
1360 if(source
&& source
->state
== AL_PLAYING
)
1362 source
->state
= AL_STOPPED
;
1363 ATOMIC_STORE(&source
->current_buffer
, NULL
);
1364 source
->position
= 0;
1365 source
->position_fraction
= 0;
1370 Context
->VoiceCount
= 0;
1372 Context
= Context
->next
;