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 ALfloat
*inVector1
, const ALfloat
*inVector2
)
121 return inVector1
[0]*inVector2
[0] + inVector1
[1]*inVector2
[1] +
122 inVector1
[2]*inVector2
[2];
125 static inline void aluNormalize(ALfloat
*inVector
)
127 ALfloat lengthsqr
= aluDotproduct(inVector
, inVector
);
130 ALfloat inv_length
= 1.0f
/sqrtf(lengthsqr
);
131 inVector
[0] *= inv_length
;
132 inVector
[1] *= inv_length
;
133 inVector
[2] *= inv_length
;
137 static inline ALvoid
aluMatrixVector(ALfloat
*vec
, ALfloat w
, const aluMatrix
*mtx
)
140 vec
[0], vec
[1], vec
[2], w
143 vec
[0] = v
[0]*mtx
->m
[0][0] + v
[1]*mtx
->m
[1][0] + v
[2]*mtx
->m
[2][0] + v
[3]*mtx
->m
[3][0];
144 vec
[1] = v
[0]*mtx
->m
[0][1] + v
[1]*mtx
->m
[1][1] + v
[2]*mtx
->m
[2][1] + v
[3]*mtx
->m
[3][1];
145 vec
[2] = v
[0]*mtx
->m
[0][2] + v
[1]*mtx
->m
[1][2] + v
[2]*mtx
->m
[2][2] + v
[3]*mtx
->m
[3][2];
149 /* Calculates the fade time from the changes in gain and listener to source
150 * angle between updates. The result is a the time, in seconds, for the
151 * transition to complete.
153 static ALfloat
CalcFadeTime(ALfloat oldGain
, ALfloat newGain
, const ALfloat olddir
[3], const ALfloat newdir
[3])
155 ALfloat gainChange
, angleChange
, change
;
157 /* Calculate the normalized dB gain change. */
158 newGain
= maxf(newGain
, 0.0001f
);
159 oldGain
= maxf(oldGain
, 0.0001f
);
160 gainChange
= fabsf(log10f(newGain
/ oldGain
) / log10f(0.0001f
));
162 /* Calculate the angle change only when there is enough gain to notice it. */
164 if(gainChange
> 0.0001f
|| newGain
> 0.0001f
)
166 /* No angle change when the directions are equal or degenerate (when
167 * both have zero length).
169 if(newdir
[0] != olddir
[0] || newdir
[1] != olddir
[1] || newdir
[2] != olddir
[2])
171 ALfloat dotp
= aluDotproduct(olddir
, newdir
);
172 angleChange
= acosf(clampf(dotp
, -1.0f
, 1.0f
)) / F_PI
;
176 /* Use the largest of the two changes, and apply a significance shaping
177 * function to it. The result is then scaled to cover a 15ms transition
180 change
= maxf(angleChange
* 25.0f
, gainChange
) * 2.0f
;
181 return minf(change
, 1.0f
) * 0.015f
;
185 static void UpdateDryStepping(DirectParams
*params
, ALuint num_chans
, ALuint steps
)
192 for(i
= 0;i
< num_chans
;i
++)
194 MixGains
*gains
= params
->Gains
[i
];
195 for(j
= 0;j
< params
->OutChannels
;j
++)
197 gains
[j
].Current
= gains
[j
].Target
;
198 gains
[j
].Step
= 0.0f
;
205 delta
= 1.0f
/ (ALfloat
)steps
;
206 for(i
= 0;i
< num_chans
;i
++)
208 MixGains
*gains
= params
->Gains
[i
];
209 for(j
= 0;j
< params
->OutChannels
;j
++)
211 ALfloat diff
= gains
[j
].Target
- gains
[j
].Current
;
212 if(fabs(diff
) >= GAIN_SILENCE_THRESHOLD
)
213 gains
[j
].Step
= diff
* delta
;
215 gains
[j
].Step
= 0.0f
;
218 params
->Counter
= steps
;
221 static void UpdateWetStepping(SendParams
*params
, ALuint steps
)
227 params
->Gain
.Current
= params
->Gain
.Target
;
228 params
->Gain
.Step
= 0.0f
;
234 delta
= 1.0f
/ (ALfloat
)steps
;
236 ALfloat diff
= params
->Gain
.Target
- params
->Gain
.Current
;
237 if(fabs(diff
) >= GAIN_SILENCE_THRESHOLD
)
238 params
->Gain
.Step
= diff
* delta
;
240 params
->Gain
.Step
= 0.0f
;
242 params
->Counter
= steps
;
246 static ALvoid
CalcListenerParams(ALlistener
*Listener
)
248 ALfloat N
[3], V
[3], U
[3], P
[3];
251 N
[0] = Listener
->Forward
[0];
252 N
[1] = Listener
->Forward
[1];
253 N
[2] = Listener
->Forward
[2];
255 V
[0] = Listener
->Up
[0];
256 V
[1] = Listener
->Up
[1];
257 V
[2] = Listener
->Up
[2];
259 /* Build and normalize right-vector */
260 aluCrossproduct(N
, V
, U
);
263 P
[0] = Listener
->Position
.v
[0];
264 P
[1] = Listener
->Position
.v
[1];
265 P
[2] = Listener
->Position
.v
[2];
267 aluMatrixSet(&Listener
->Params
.Matrix
,
268 U
[0], V
[0], -N
[0], 0.0f
,
269 U
[1], V
[1], -N
[1], 0.0f
,
270 U
[2], V
[2], -N
[2], 0.0f
,
271 0.0f
, 0.0f
, 0.0f
, 1.0f
273 aluMatrixVector(P
, 1.0f
, &Listener
->Params
.Matrix
);
274 aluMatrixSetRow(&Listener
->Params
.Matrix
, 3, -P
[0], -P
[1], -P
[2], 1.0f
);
276 Listener
->Params
.Velocity
= Listener
->Velocity
;
277 aluMatrixVector(Listener
->Params
.Velocity
.v
, 0.0f
, &Listener
->Params
.Matrix
);
280 ALvoid
CalcNonAttnSourceParams(ALvoice
*voice
, const ALsource
*ALSource
, const ALCcontext
*ALContext
)
282 static const struct ChanMap MonoMap
[1] = { { FrontCenter
, 0.0f
, 0.0f
} };
283 static const struct ChanMap StereoMap
[2] = {
284 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
285 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) }
287 static const struct ChanMap StereoWideMap
[2] = {
288 { FrontLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
289 { FrontRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
291 static const struct ChanMap RearMap
[2] = {
292 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
293 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) }
295 static const struct ChanMap QuadMap
[4] = {
296 { FrontLeft
, DEG2RAD( -45.0f
), DEG2RAD(0.0f
) },
297 { FrontRight
, DEG2RAD( 45.0f
), DEG2RAD(0.0f
) },
298 { BackLeft
, DEG2RAD(-135.0f
), DEG2RAD(0.0f
) },
299 { BackRight
, DEG2RAD( 135.0f
), DEG2RAD(0.0f
) }
301 static const struct ChanMap X51Map
[6] = {
302 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
303 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
304 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
306 { SideLeft
, DEG2RAD(-110.0f
), DEG2RAD(0.0f
) },
307 { SideRight
, DEG2RAD( 110.0f
), DEG2RAD(0.0f
) }
309 static const struct ChanMap X61Map
[7] = {
310 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
311 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
312 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
314 { BackCenter
, DEG2RAD(180.0f
), DEG2RAD(0.0f
) },
315 { SideLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
316 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
318 static const struct ChanMap X71Map
[8] = {
319 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
320 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
321 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
323 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
324 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) },
325 { SideLeft
, DEG2RAD( -90.0f
), DEG2RAD(0.0f
) },
326 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
329 ALCdevice
*Device
= ALContext
->Device
;
330 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
331 ALbufferlistitem
*BufferListItem
;
332 enum FmtChannels Channels
;
333 ALfloat DryGain
, DryGainHF
, DryGainLF
;
334 ALfloat WetGain
[MAX_SENDS
];
335 ALfloat WetGainHF
[MAX_SENDS
];
336 ALfloat WetGainLF
[MAX_SENDS
];
337 ALuint NumSends
, Frequency
;
339 const struct ChanMap
*chans
= NULL
;
340 ALuint num_channels
= 0;
341 ALboolean DirectChannels
;
342 ALboolean isbformat
= AL_FALSE
;
346 /* Get device properties */
347 NumSends
= Device
->NumAuxSends
;
348 Frequency
= Device
->Frequency
;
350 /* Get listener properties */
351 ListenerGain
= ALContext
->Listener
->Gain
;
353 /* Get source properties */
354 SourceVolume
= ALSource
->Gain
;
355 MinVolume
= ALSource
->MinGain
;
356 MaxVolume
= ALSource
->MaxGain
;
357 Pitch
= ALSource
->Pitch
;
358 Relative
= ALSource
->HeadRelative
;
359 DirectChannels
= ALSource
->DirectChannels
;
361 voice
->Direct
.OutBuffer
= Device
->DryBuffer
;
362 voice
->Direct
.OutChannels
= Device
->NumChannels
;
363 for(i
= 0;i
< NumSends
;i
++)
365 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
367 Slot
= Device
->DefaultSlot
;
368 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
369 voice
->Send
[i
].OutBuffer
= NULL
;
371 voice
->Send
[i
].OutBuffer
= Slot
->WetBuffer
;
374 /* Calculate the stepping value */
376 BufferListItem
= ATOMIC_LOAD(&ALSource
->queue
);
377 while(BufferListItem
!= NULL
)
380 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
382 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
383 if(Pitch
> (ALfloat
)MAX_PITCH
)
384 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
387 voice
->Step
= fastf2i(Pitch
*FRACTIONONE
);
392 Channels
= ALBuffer
->FmtChannels
;
395 BufferListItem
= BufferListItem
->next
;
398 /* Calculate gains */
399 DryGain
= clampf(SourceVolume
, MinVolume
, MaxVolume
);
400 DryGain
*= ALSource
->Direct
.Gain
* ListenerGain
;
401 DryGainHF
= ALSource
->Direct
.GainHF
;
402 DryGainLF
= ALSource
->Direct
.GainLF
;
403 for(i
= 0;i
< NumSends
;i
++)
405 WetGain
[i
] = clampf(SourceVolume
, MinVolume
, MaxVolume
);
406 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
407 WetGainHF
[i
] = ALSource
->Send
[i
].GainHF
;
408 WetGainLF
[i
] = ALSource
->Send
[i
].GainLF
;
419 /* HACK: Place the stereo channels at +/-90 degrees when using non-
420 * HRTF stereo output. This helps reduce the "monoization" caused
421 * by them panning towards the center. */
422 if(Device
->FmtChans
== DevFmtStereo
&& !Device
->Hrtf
)
423 chans
= StereoWideMap
;
457 DirectChannels
= AL_FALSE
;
463 DirectChannels
= AL_FALSE
;
469 ALfloat N
[3], V
[3], U
[3];
470 ALfloat matrix
[4][4];
473 N
[0] = ALSource
->Orientation
[0][0];
474 N
[1] = ALSource
->Orientation
[0][1];
475 N
[2] = ALSource
->Orientation
[0][2];
477 V
[0] = ALSource
->Orientation
[1][0];
478 V
[1] = ALSource
->Orientation
[1][1];
479 V
[2] = ALSource
->Orientation
[1][2];
483 const aluMatrix
*lmatrix
= &ALContext
->Listener
->Params
.Matrix
;
484 aluMatrixVector(N
, 0.0f
, lmatrix
);
485 aluMatrixVector(V
, 0.0f
, lmatrix
);
487 /* Build and normalize right-vector */
488 aluCrossproduct(N
, V
, U
);
496 matrix
[1][1] = -N
[2];
497 matrix
[1][2] = -N
[0];
502 matrix
[2][3] = -U
[1];
504 matrix
[3][1] = -V
[2];
505 matrix
[3][2] = -V
[0];
508 for(c
= 0;c
< num_channels
;c
++)
510 MixGains
*gains
= voice
->Direct
.Gains
[c
];
511 ALfloat Target
[MAX_OUTPUT_CHANNELS
];
513 ComputeBFormatGains(Device
, matrix
[c
], DryGain
, Target
);
514 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
515 gains
[i
].Target
= Target
[i
];
517 UpdateDryStepping(&voice
->Direct
, num_channels
, (voice
->Direct
.Moving
? 64 : 0));
518 voice
->Direct
.Moving
= AL_TRUE
;
520 voice
->IsHrtf
= AL_FALSE
;
521 for(i
= 0;i
< NumSends
;i
++)
522 WetGain
[i
] *= 1.4142f
;
524 else if(DirectChannels
!= AL_FALSE
)
528 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
529 voice
->Direct
.OutChannels
= 2;
530 for(c
= 0;c
< num_channels
;c
++)
532 MixGains
*gains
= voice
->Direct
.Gains
[c
];
534 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
535 gains
[j
].Target
= 0.0f
;
537 if(chans
[c
].channel
== FrontLeft
)
538 gains
[0].Target
= DryGain
;
539 else if(chans
[c
].channel
== FrontRight
)
540 gains
[1].Target
= DryGain
;
543 else for(c
= 0;c
< num_channels
;c
++)
545 MixGains
*gains
= voice
->Direct
.Gains
[c
];
548 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
549 gains
[j
].Target
= 0.0f
;
550 if((idx
=GetChannelIdxByName(Device
, chans
[c
].channel
)) != -1)
551 gains
[idx
].Target
= DryGain
;
553 UpdateDryStepping(&voice
->Direct
, num_channels
, (voice
->Direct
.Moving
? 64 : 0));
554 voice
->Direct
.Moving
= AL_TRUE
;
556 voice
->IsHrtf
= AL_FALSE
;
558 else if(Device
->Hrtf
)
560 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
561 voice
->Direct
.OutChannels
= 2;
562 for(c
= 0;c
< num_channels
;c
++)
564 if(chans
[c
].channel
== LFE
)
567 voice
->Direct
.Hrtf
.Params
[c
].Delay
[0] = 0;
568 voice
->Direct
.Hrtf
.Params
[c
].Delay
[1] = 0;
569 for(i
= 0;i
< HRIR_LENGTH
;i
++)
571 voice
->Direct
.Hrtf
.Params
[c
].Coeffs
[i
][0] = 0.0f
;
572 voice
->Direct
.Hrtf
.Params
[c
].Coeffs
[i
][1] = 0.0f
;
577 /* Get the static HRIR coefficients and delays for this
579 GetLerpedHrtfCoeffs(Device
->Hrtf
,
580 chans
[c
].elevation
, chans
[c
].angle
, 1.0f
, DryGain
,
581 voice
->Direct
.Hrtf
.Params
[c
].Coeffs
,
582 voice
->Direct
.Hrtf
.Params
[c
].Delay
);
585 voice
->Direct
.Counter
= 0;
586 voice
->Direct
.Moving
= AL_TRUE
;
588 voice
->IsHrtf
= AL_TRUE
;
592 for(c
= 0;c
< num_channels
;c
++)
594 MixGains
*gains
= voice
->Direct
.Gains
[c
];
595 ALfloat Target
[MAX_OUTPUT_CHANNELS
];
597 /* Special-case LFE */
598 if(chans
[c
].channel
== LFE
)
601 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
602 gains
[i
].Target
= 0.0f
;
603 if((idx
=GetChannelIdxByName(Device
, chans
[c
].channel
)) != -1)
604 gains
[idx
].Target
= DryGain
;
608 ComputeAngleGains(Device
, chans
[c
].angle
, chans
[c
].elevation
, DryGain
, Target
);
609 for(i
= 0;i
< MAX_OUTPUT_CHANNELS
;i
++)
610 gains
[i
].Target
= Target
[i
];
612 UpdateDryStepping(&voice
->Direct
, num_channels
, (voice
->Direct
.Moving
? 64 : 0));
613 voice
->Direct
.Moving
= AL_TRUE
;
615 voice
->IsHrtf
= AL_FALSE
;
617 for(i
= 0;i
< NumSends
;i
++)
619 voice
->Send
[i
].Gain
.Target
= WetGain
[i
];
620 UpdateWetStepping(&voice
->Send
[i
], (voice
->Send
[i
].Moving
? 64 : 0));
621 voice
->Send
[i
].Moving
= AL_TRUE
;
625 ALfloat gainhf
= maxf(0.01f
, DryGainHF
);
626 ALfloat gainlf
= maxf(0.01f
, DryGainLF
);
627 ALfloat hfscale
= ALSource
->Direct
.HFReference
/ Frequency
;
628 ALfloat lfscale
= ALSource
->Direct
.LFReference
/ Frequency
;
629 for(c
= 0;c
< num_channels
;c
++)
631 voice
->Direct
.Filters
[c
].ActiveType
= AF_None
;
632 if(gainhf
!= 1.0f
) voice
->Direct
.Filters
[c
].ActiveType
|= AF_LowPass
;
633 if(gainlf
!= 1.0f
) voice
->Direct
.Filters
[c
].ActiveType
|= AF_HighPass
;
634 ALfilterState_setParams(
635 &voice
->Direct
.Filters
[c
].LowPass
, ALfilterType_HighShelf
, gainhf
,
638 ALfilterState_setParams(
639 &voice
->Direct
.Filters
[c
].HighPass
, ALfilterType_LowShelf
, gainlf
,
644 for(i
= 0;i
< NumSends
;i
++)
646 ALfloat gainhf
= maxf(0.01f
, WetGainHF
[i
]);
647 ALfloat gainlf
= maxf(0.01f
, WetGainLF
[i
]);
648 ALfloat hfscale
= ALSource
->Send
[i
].HFReference
/ Frequency
;
649 ALfloat lfscale
= ALSource
->Send
[i
].LFReference
/ Frequency
;
650 for(c
= 0;c
< num_channels
;c
++)
652 voice
->Send
[i
].Filters
[c
].ActiveType
= AF_None
;
653 if(gainhf
!= 1.0f
) voice
->Send
[i
].Filters
[c
].ActiveType
|= AF_LowPass
;
654 if(gainlf
!= 1.0f
) voice
->Send
[i
].Filters
[c
].ActiveType
|= AF_HighPass
;
655 ALfilterState_setParams(
656 &voice
->Send
[i
].Filters
[c
].LowPass
, ALfilterType_HighShelf
, gainhf
,
659 ALfilterState_setParams(
660 &voice
->Send
[i
].Filters
[c
].HighPass
, ALfilterType_LowShelf
, gainlf
,
667 ALvoid
CalcSourceParams(ALvoice
*voice
, const ALsource
*ALSource
, const ALCcontext
*ALContext
)
669 ALCdevice
*Device
= ALContext
->Device
;
670 aluVector Position
, Velocity
, Direction
, SourceToListener
;
671 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,ClampedDist
;
672 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
;
673 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
674 ALfloat DopplerFactor
, SpeedOfSound
;
675 ALfloat AirAbsorptionFactor
;
676 ALfloat RoomAirAbsorption
[MAX_SENDS
];
677 ALbufferlistitem
*BufferListItem
;
679 ALfloat RoomAttenuation
[MAX_SENDS
];
680 ALfloat MetersPerUnit
;
681 ALfloat RoomRolloffBase
;
682 ALfloat RoomRolloff
[MAX_SENDS
];
683 ALfloat DecayDistance
[MAX_SENDS
];
687 ALboolean DryGainHFAuto
;
688 ALfloat WetGain
[MAX_SENDS
];
689 ALfloat WetGainHF
[MAX_SENDS
];
690 ALfloat WetGainLF
[MAX_SENDS
];
691 ALboolean WetGainAuto
;
692 ALboolean WetGainHFAuto
;
700 for(i
= 0;i
< MAX_SENDS
;i
++)
706 /* Get context/device properties */
707 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
708 SpeedOfSound
= ALContext
->SpeedOfSound
* ALContext
->DopplerVelocity
;
709 NumSends
= Device
->NumAuxSends
;
710 Frequency
= Device
->Frequency
;
712 /* Get listener properties */
713 ListenerGain
= ALContext
->Listener
->Gain
;
714 MetersPerUnit
= ALContext
->Listener
->MetersPerUnit
;
716 /* Get source properties */
717 SourceVolume
= ALSource
->Gain
;
718 MinVolume
= ALSource
->MinGain
;
719 MaxVolume
= ALSource
->MaxGain
;
720 Pitch
= ALSource
->Pitch
;
721 Position
= ALSource
->Position
;
722 Direction
= ALSource
->Direction
;
723 Velocity
= ALSource
->Velocity
;
724 MinDist
= ALSource
->RefDistance
;
725 MaxDist
= ALSource
->MaxDistance
;
726 Rolloff
= ALSource
->RollOffFactor
;
727 InnerAngle
= ALSource
->InnerAngle
;
728 OuterAngle
= ALSource
->OuterAngle
;
729 AirAbsorptionFactor
= ALSource
->AirAbsorptionFactor
;
730 DryGainHFAuto
= ALSource
->DryGainHFAuto
;
731 WetGainAuto
= ALSource
->WetGainAuto
;
732 WetGainHFAuto
= ALSource
->WetGainHFAuto
;
733 RoomRolloffBase
= ALSource
->RoomRolloffFactor
;
735 voice
->Direct
.OutBuffer
= Device
->DryBuffer
;
736 voice
->Direct
.OutChannels
= Device
->NumChannels
;
737 for(i
= 0;i
< NumSends
;i
++)
739 ALeffectslot
*Slot
= ALSource
->Send
[i
].Slot
;
742 Slot
= Device
->DefaultSlot
;
743 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
746 RoomRolloff
[i
] = 0.0f
;
747 DecayDistance
[i
] = 0.0f
;
748 RoomAirAbsorption
[i
] = 1.0f
;
750 else if(Slot
->AuxSendAuto
)
752 RoomRolloff
[i
] = RoomRolloffBase
;
753 if(IsReverbEffect(Slot
->EffectType
))
755 RoomRolloff
[i
] += Slot
->EffectProps
.Reverb
.RoomRolloffFactor
;
756 DecayDistance
[i
] = Slot
->EffectProps
.Reverb
.DecayTime
*
757 SPEEDOFSOUNDMETRESPERSEC
;
758 RoomAirAbsorption
[i
] = Slot
->EffectProps
.Reverb
.AirAbsorptionGainHF
;
762 DecayDistance
[i
] = 0.0f
;
763 RoomAirAbsorption
[i
] = 1.0f
;
768 /* If the slot's auxiliary send auto is off, the data sent to the
769 * effect slot is the same as the dry path, sans filter effects */
770 RoomRolloff
[i
] = Rolloff
;
771 DecayDistance
[i
] = 0.0f
;
772 RoomAirAbsorption
[i
] = AIRABSORBGAINHF
;
775 if(!Slot
|| Slot
->EffectType
== AL_EFFECT_NULL
)
776 voice
->Send
[i
].OutBuffer
= NULL
;
778 voice
->Send
[i
].OutBuffer
= Slot
->WetBuffer
;
781 /* Transform source to listener space (convert to head relative) */
782 if(ALSource
->HeadRelative
== AL_FALSE
)
784 const aluMatrix
*Matrix
= &ALContext
->Listener
->Params
.Matrix
;
785 /* Transform source vectors */
786 aluMatrixVector(Position
.v
, 1.0f
, Matrix
);
787 aluMatrixVector(Direction
.v
, 0.0f
, Matrix
);
788 aluMatrixVector(Velocity
.v
, 0.0f
, Matrix
);
792 const aluVector
*lvelocity
= &ALContext
->Listener
->Params
.Velocity
;
793 /* Offset the source velocity to be relative of the listener velocity */
794 Velocity
.v
[0] += lvelocity
->v
[0];
795 Velocity
.v
[1] += lvelocity
->v
[1];
796 Velocity
.v
[2] += lvelocity
->v
[2];
799 SourceToListener
.v
[0] = -Position
.v
[0];
800 SourceToListener
.v
[1] = -Position
.v
[1];
801 SourceToListener
.v
[2] = -Position
.v
[2];
802 SourceToListener
.v
[2] = 0.0f
;
803 aluNormalize(SourceToListener
.v
);
804 aluNormalize(Direction
.v
);
806 /* Calculate distance attenuation */
807 Distance
= sqrtf(aluDotproduct(Position
.v
, Position
.v
));
808 ClampedDist
= Distance
;
811 for(i
= 0;i
< NumSends
;i
++)
812 RoomAttenuation
[i
] = 1.0f
;
813 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
814 ALContext
->DistanceModel
)
816 case InverseDistanceClamped
:
817 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
818 if(MaxDist
< MinDist
)
821 case InverseDistance
:
824 if((MinDist
+ (Rolloff
* (ClampedDist
- MinDist
))) > 0.0f
)
825 Attenuation
= MinDist
/ (MinDist
+ (Rolloff
* (ClampedDist
- MinDist
)));
826 for(i
= 0;i
< NumSends
;i
++)
828 if((MinDist
+ (RoomRolloff
[i
] * (ClampedDist
- MinDist
))) > 0.0f
)
829 RoomAttenuation
[i
] = MinDist
/ (MinDist
+ (RoomRolloff
[i
] * (ClampedDist
- MinDist
)));
834 case LinearDistanceClamped
:
835 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
836 if(MaxDist
< MinDist
)
840 if(MaxDist
!= MinDist
)
842 Attenuation
= 1.0f
- (Rolloff
*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
843 Attenuation
= maxf(Attenuation
, 0.0f
);
844 for(i
= 0;i
< NumSends
;i
++)
846 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
847 RoomAttenuation
[i
] = maxf(RoomAttenuation
[i
], 0.0f
);
852 case ExponentDistanceClamped
:
853 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
854 if(MaxDist
< MinDist
)
857 case ExponentDistance
:
858 if(ClampedDist
> 0.0f
&& MinDist
> 0.0f
)
860 Attenuation
= powf(ClampedDist
/MinDist
, -Rolloff
);
861 for(i
= 0;i
< NumSends
;i
++)
862 RoomAttenuation
[i
] = powf(ClampedDist
/MinDist
, -RoomRolloff
[i
]);
866 case DisableDistance
:
867 ClampedDist
= MinDist
;
871 /* Source Gain + Attenuation */
872 DryGain
= SourceVolume
* Attenuation
;
873 for(i
= 0;i
< NumSends
;i
++)
874 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
876 /* Distance-based air absorption */
877 if(AirAbsorptionFactor
> 0.0f
&& ClampedDist
> MinDist
)
879 ALfloat meters
= (ClampedDist
-MinDist
) * MetersPerUnit
;
880 DryGainHF
*= powf(AIRABSORBGAINHF
, AirAbsorptionFactor
*meters
);
881 for(i
= 0;i
< NumSends
;i
++)
882 WetGainHF
[i
] *= powf(RoomAirAbsorption
[i
], AirAbsorptionFactor
*meters
);
887 ALfloat ApparentDist
= 1.0f
/maxf(Attenuation
, 0.00001f
) - 1.0f
;
889 /* Apply a decay-time transformation to the wet path, based on the
890 * attenuation of the dry path.
892 * Using the apparent distance, based on the distance attenuation, the
893 * initial decay of the reverb effect is calculated and applied to the
896 for(i
= 0;i
< NumSends
;i
++)
898 if(DecayDistance
[i
] > 0.0f
)
899 WetGain
[i
] *= powf(0.001f
/*-60dB*/, ApparentDist
/DecayDistance
[i
]);
903 /* Calculate directional soundcones */
904 Angle
= RAD2DEG(acosf(aluDotproduct(Direction
.v
,SourceToListener
.v
)) * ConeScale
) * 2.0f
;
905 if(Angle
> InnerAngle
&& Angle
<= OuterAngle
)
907 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
908 ConeVolume
= lerp(1.0f
, ALSource
->OuterGain
, scale
);
909 ConeHF
= lerp(1.0f
, ALSource
->OuterGainHF
, scale
);
911 else if(Angle
> OuterAngle
)
913 ConeVolume
= ALSource
->OuterGain
;
914 ConeHF
= ALSource
->OuterGainHF
;
922 DryGain
*= ConeVolume
;
925 for(i
= 0;i
< NumSends
;i
++)
926 WetGain
[i
] *= ConeVolume
;
932 for(i
= 0;i
< NumSends
;i
++)
933 WetGainHF
[i
] *= ConeHF
;
936 /* Clamp to Min/Max Gain */
937 DryGain
= clampf(DryGain
, MinVolume
, MaxVolume
);
938 for(i
= 0;i
< NumSends
;i
++)
939 WetGain
[i
] = clampf(WetGain
[i
], MinVolume
, MaxVolume
);
941 /* Apply gain and frequency filters */
942 DryGain
*= ALSource
->Direct
.Gain
* ListenerGain
;
943 DryGainHF
*= ALSource
->Direct
.GainHF
;
944 DryGainLF
*= ALSource
->Direct
.GainLF
;
945 for(i
= 0;i
< NumSends
;i
++)
947 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
948 WetGainHF
[i
] *= ALSource
->Send
[i
].GainHF
;
949 WetGainLF
[i
] *= ALSource
->Send
[i
].GainLF
;
952 /* Calculate velocity-based doppler effect */
953 if(DopplerFactor
> 0.0f
)
955 const aluVector
*lvelocity
= &ALContext
->Listener
->Params
.Velocity
;
958 if(SpeedOfSound
< 1.0f
)
960 DopplerFactor
*= 1.0f
/SpeedOfSound
;
964 VSS
= aluDotproduct(Velocity
.v
, SourceToListener
.v
) * DopplerFactor
;
965 VLS
= aluDotproduct(lvelocity
->v
, SourceToListener
.v
) * DopplerFactor
;
967 Pitch
*= clampf(SpeedOfSound
-VLS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
) /
968 clampf(SpeedOfSound
-VSS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
);
971 BufferListItem
= ATOMIC_LOAD(&ALSource
->queue
);
972 while(BufferListItem
!= NULL
)
975 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
977 /* Calculate fixed-point stepping value, based on the pitch, buffer
978 * frequency, and output frequency. */
979 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
980 if(Pitch
> (ALfloat
)MAX_PITCH
)
981 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
984 voice
->Step
= fastf2i(Pitch
*FRACTIONONE
);
991 BufferListItem
= BufferListItem
->next
;
996 /* Use a binaural HRTF algorithm for stereo headphone playback */
997 ALfloat dir
[3] = { 0.0f
, 0.0f
, -1.0f
};
998 ALfloat ev
= 0.0f
, az
= 0.0f
;
999 ALfloat radius
= ALSource
->Radius
;
1000 ALfloat dirfact
= 1.0f
;
1002 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
1003 voice
->Direct
.OutChannels
= 2;
1005 if(Distance
> FLT_EPSILON
)
1007 ALfloat invlen
= 1.0f
/Distance
;
1008 dir
[0] = Position
.v
[0] * invlen
;
1009 dir
[1] = Position
.v
[1] * invlen
;
1010 dir
[2] = Position
.v
[2] * invlen
* ZScale
;
1012 /* Calculate elevation and azimuth only when the source is not at
1013 * the listener. This prevents +0 and -0 Z from producing
1014 * inconsistent panning. Also, clamp Y in case FP precision errors
1015 * cause it to land outside of -1..+1. */
1016 ev
= asinf(clampf(dir
[1], -1.0f
, 1.0f
));
1017 az
= atan2f(dir
[0], -dir
[2]);
1019 if(radius
> Distance
)
1020 dirfact
*= Distance
/ radius
;
1022 /* Check to see if the HRIR is already moving. */
1023 if(voice
->Direct
.Moving
)
1026 delta
= CalcFadeTime(voice
->Direct
.LastGain
, DryGain
,
1027 voice
->Direct
.LastDir
, dir
);
1028 /* If the delta is large enough, get the moving HRIR target
1029 * coefficients, target delays, steppping values, and counter. */
1030 if(delta
> 0.000015f
)
1032 ALuint counter
= GetMovingHrtfCoeffs(Device
->Hrtf
,
1033 ev
, az
, dirfact
, DryGain
, delta
, voice
->Direct
.Counter
,
1034 voice
->Direct
.Hrtf
.Params
[0].Coeffs
, voice
->Direct
.Hrtf
.Params
[0].Delay
,
1035 voice
->Direct
.Hrtf
.Params
[0].CoeffStep
, voice
->Direct
.Hrtf
.Params
[0].DelayStep
1037 voice
->Direct
.Counter
= counter
;
1038 voice
->Direct
.LastGain
= DryGain
;
1039 voice
->Direct
.LastDir
[0] = dir
[0];
1040 voice
->Direct
.LastDir
[1] = dir
[1];
1041 voice
->Direct
.LastDir
[2] = dir
[2];
1046 /* Get the initial (static) HRIR coefficients and delays. */
1047 GetLerpedHrtfCoeffs(Device
->Hrtf
, ev
, az
, dirfact
, DryGain
,
1048 voice
->Direct
.Hrtf
.Params
[0].Coeffs
,
1049 voice
->Direct
.Hrtf
.Params
[0].Delay
);
1050 voice
->Direct
.Counter
= 0;
1051 voice
->Direct
.Moving
= AL_TRUE
;
1052 voice
->Direct
.LastGain
= DryGain
;
1053 voice
->Direct
.LastDir
[0] = dir
[0];
1054 voice
->Direct
.LastDir
[1] = dir
[1];
1055 voice
->Direct
.LastDir
[2] = dir
[2];
1058 voice
->IsHrtf
= AL_TRUE
;
1062 MixGains
*gains
= voice
->Direct
.Gains
[0];
1063 ALfloat dir
[3] = { 0.0f
, 0.0f
, -1.0f
};
1064 ALfloat radius
= ALSource
->Radius
;
1065 ALfloat Target
[MAX_OUTPUT_CHANNELS
];
1067 /* Normalize the length, and compute panned gains. */
1068 if(Distance
> FLT_EPSILON
|| radius
> FLT_EPSILON
)
1070 ALfloat invlen
= 1.0f
/maxf(Distance
, radius
);
1071 dir
[0] = Position
.v
[0] * invlen
;
1072 dir
[1] = Position
.v
[1] * invlen
;
1073 dir
[2] = Position
.v
[2] * invlen
* ZScale
;
1075 ComputeDirectionalGains(Device
, dir
, DryGain
, Target
);
1077 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
1078 gains
[j
].Target
= Target
[j
];
1079 UpdateDryStepping(&voice
->Direct
, 1, (voice
->Direct
.Moving
? 64 : 0));
1080 voice
->Direct
.Moving
= AL_TRUE
;
1082 voice
->IsHrtf
= AL_FALSE
;
1084 for(i
= 0;i
< NumSends
;i
++)
1086 voice
->Send
[i
].Gain
.Target
= WetGain
[i
];
1087 UpdateWetStepping(&voice
->Send
[i
], (voice
->Send
[i
].Moving
? 64 : 0));
1088 voice
->Send
[i
].Moving
= AL_TRUE
;
1092 ALfloat gainhf
= maxf(0.01f
, DryGainHF
);
1093 ALfloat gainlf
= maxf(0.01f
, DryGainLF
);
1094 ALfloat hfscale
= ALSource
->Direct
.HFReference
/ Frequency
;
1095 ALfloat lfscale
= ALSource
->Direct
.LFReference
/ Frequency
;
1096 voice
->Direct
.Filters
[0].ActiveType
= AF_None
;
1097 if(gainhf
!= 1.0f
) voice
->Direct
.Filters
[0].ActiveType
|= AF_LowPass
;
1098 if(gainlf
!= 1.0f
) voice
->Direct
.Filters
[0].ActiveType
|= AF_HighPass
;
1099 ALfilterState_setParams(
1100 &voice
->Direct
.Filters
[0].LowPass
, ALfilterType_HighShelf
, gainhf
,
1103 ALfilterState_setParams(
1104 &voice
->Direct
.Filters
[0].HighPass
, ALfilterType_LowShelf
, gainlf
,
1108 for(i
= 0;i
< NumSends
;i
++)
1110 ALfloat gainhf
= maxf(0.01f
, WetGainHF
[i
]);
1111 ALfloat gainlf
= maxf(0.01f
, WetGainLF
[i
]);
1112 ALfloat hfscale
= ALSource
->Send
[i
].HFReference
/ Frequency
;
1113 ALfloat lfscale
= ALSource
->Send
[i
].LFReference
/ Frequency
;
1114 voice
->Send
[i
].Filters
[0].ActiveType
= AF_None
;
1115 if(gainhf
!= 1.0f
) voice
->Send
[i
].Filters
[0].ActiveType
|= AF_LowPass
;
1116 if(gainlf
!= 1.0f
) voice
->Send
[i
].Filters
[0].ActiveType
|= AF_HighPass
;
1117 ALfilterState_setParams(
1118 &voice
->Send
[i
].Filters
[0].LowPass
, ALfilterType_HighShelf
, gainhf
,
1121 ALfilterState_setParams(
1122 &voice
->Send
[i
].Filters
[0].HighPass
, ALfilterType_LowShelf
, gainlf
,
1129 static inline ALint
aluF2I25(ALfloat val
)
1131 /* Clamp the value between -1 and +1. This handles that with only a single branch. */
1132 if(fabsf(val
) > 1.0f
)
1133 val
= (ALfloat
)((0.0f
< val
) - (val
< 0.0f
));
1134 /* Convert to a signed integer, between -16777215 and +16777215. */
1135 return fastf2i(val
*16777215.0f
);
1138 static inline ALfloat
aluF2F(ALfloat val
)
1140 static inline ALint
aluF2I(ALfloat val
)
1141 { return aluF2I25(val
)<<7; }
1142 static inline ALuint
aluF2UI(ALfloat val
)
1143 { return aluF2I(val
)+2147483648u; }
1144 static inline ALshort
aluF2S(ALfloat val
)
1145 { return aluF2I25(val
)>>9; }
1146 static inline ALushort
aluF2US(ALfloat val
)
1147 { return aluF2S(val
)+32768; }
1148 static inline ALbyte
aluF2B(ALfloat val
)
1149 { return aluF2I25(val
)>>17; }
1150 static inline ALubyte
aluF2UB(ALfloat val
)
1151 { return aluF2B(val
)+128; }
1153 #define DECL_TEMPLATE(T, func) \
1154 static void Write_##T(const ALfloatBUFFERSIZE *InBuffer, ALvoid *OutBuffer, \
1155 ALuint SamplesToDo, ALuint numchans) \
1158 for(j = 0;j < numchans;j++) \
1160 const ALfloat *in = InBuffer[j]; \
1161 T *restrict out = (T*)OutBuffer + j; \
1162 for(i = 0;i < SamplesToDo;i++) \
1163 out[i*numchans] = func(in[i]); \
1167 DECL_TEMPLATE(ALfloat
, aluF2F
)
1168 DECL_TEMPLATE(ALuint
, aluF2UI
)
1169 DECL_TEMPLATE(ALint
, aluF2I
)
1170 DECL_TEMPLATE(ALushort
, aluF2US
)
1171 DECL_TEMPLATE(ALshort
, aluF2S
)
1172 DECL_TEMPLATE(ALubyte
, aluF2UB
)
1173 DECL_TEMPLATE(ALbyte
, aluF2B
)
1175 #undef DECL_TEMPLATE
1178 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1181 ALeffectslot
**slot
, **slot_end
;
1182 ALvoice
*voice
, *voice_end
;
1187 SetMixerFPUMode(&oldMode
);
1191 ALuint outchanoffset
= 0;
1192 ALuint outchancount
= device
->NumChannels
;
1194 IncrementRef(&device
->MixCount
);
1196 SamplesToDo
= minu(size
, BUFFERSIZE
);
1197 for(c
= 0;c
< device
->NumChannels
;c
++)
1198 memset(device
->DryBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1201 outchanoffset
= device
->NumChannels
;
1203 for(c
= 0;c
< outchancount
;c
++)
1204 memset(device
->DryBuffer
[outchanoffset
+c
], 0, SamplesToDo
*sizeof(ALfloat
));
1207 V0(device
->Backend
,lock
)();
1208 V(device
->Synth
,process
)(SamplesToDo
, &device
->DryBuffer
[outchanoffset
]);
1210 ctx
= ATOMIC_LOAD(&device
->ContextList
);
1213 ALenum DeferUpdates
= ctx
->DeferUpdates
;
1214 ALenum UpdateSources
= AL_FALSE
;
1217 UpdateSources
= ATOMIC_EXCHANGE(ALenum
, &ctx
->UpdateSources
, AL_FALSE
);
1220 CalcListenerParams(ctx
->Listener
);
1222 /* source processing */
1223 voice
= ctx
->Voices
;
1224 voice_end
= voice
+ ctx
->VoiceCount
;
1225 while(voice
!= voice_end
)
1227 ALsource
*source
= voice
->Source
;
1228 if(!source
) goto next
;
1230 if(source
->state
!= AL_PLAYING
&& source
->state
!= AL_PAUSED
)
1232 voice
->Source
= NULL
;
1236 if(!DeferUpdates
&& (ATOMIC_EXCHANGE(ALenum
, &source
->NeedsUpdate
, AL_FALSE
) ||
1238 voice
->Update(voice
, source
, ctx
);
1240 if(source
->state
!= AL_PAUSED
)
1241 MixSource(voice
, source
, device
, SamplesToDo
);
1246 /* effect slot processing */
1247 slot
= VECTOR_ITER_BEGIN(ctx
->ActiveAuxSlots
);
1248 slot_end
= VECTOR_ITER_END(ctx
->ActiveAuxSlots
);
1249 while(slot
!= slot_end
)
1251 if(!DeferUpdates
&& ATOMIC_EXCHANGE(ALenum
, &(*slot
)->NeedsUpdate
, AL_FALSE
))
1252 V((*slot
)->EffectState
,update
)(device
, *slot
);
1254 V((*slot
)->EffectState
,process
)(SamplesToDo
, (*slot
)->WetBuffer
[0],
1255 device
->DryBuffer
, device
->NumChannels
);
1257 for(i
= 0;i
< SamplesToDo
;i
++)
1258 (*slot
)->WetBuffer
[0][i
] = 0.0f
;
1266 slot
= &device
->DefaultSlot
;
1269 if(ATOMIC_EXCHANGE(ALenum
, &(*slot
)->NeedsUpdate
, AL_FALSE
))
1270 V((*slot
)->EffectState
,update
)(device
, *slot
);
1272 V((*slot
)->EffectState
,process
)(SamplesToDo
, (*slot
)->WetBuffer
[0],
1273 device
->DryBuffer
, device
->NumChannels
);
1275 for(i
= 0;i
< SamplesToDo
;i
++)
1276 (*slot
)->WetBuffer
[0][i
] = 0.0f
;
1279 /* Increment the clock time. Every second's worth of samples is
1280 * converted and added to clock base so that large sample counts don't
1281 * overflow during conversion. This also guarantees an exact, stable
1283 device
->SamplesDone
+= SamplesToDo
;
1284 device
->ClockBase
+= (device
->SamplesDone
/device
->Frequency
) * DEVICE_CLOCK_RES
;
1285 device
->SamplesDone
%= device
->Frequency
;
1286 V0(device
->Backend
,unlock
)();
1290 HrtfMixerFunc HrtfMix
= SelectHrtfMixer();
1291 ALuint irsize
= GetHrtfIrSize(device
->Hrtf
);
1292 for(c
= 0;c
< device
->NumChannels
;c
++)
1293 HrtfMix(&device
->DryBuffer
[outchanoffset
], device
->DryBuffer
[c
], 0,
1294 device
->Hrtf_Offset
, 0, irsize
, &device
->Hrtf_Params
[c
],
1295 &device
->Hrtf_State
[c
], SamplesToDo
1297 device
->Hrtf_Offset
+= SamplesToDo
;
1299 else if(device
->Bs2b
)
1301 /* Apply binaural/crossfeed filter */
1302 for(i
= 0;i
< SamplesToDo
;i
++)
1305 samples
[0] = device
->DryBuffer
[0][i
];
1306 samples
[1] = device
->DryBuffer
[1][i
];
1307 bs2b_cross_feed(device
->Bs2b
, samples
);
1308 device
->DryBuffer
[0][i
] = samples
[0];
1309 device
->DryBuffer
[1][i
] = samples
[1];
1315 #define WRITE(T, a, b, c, d) do { \
1316 Write_##T((a), (b), (c), (d)); \
1317 buffer = (char*)buffer + (c)*(d)*sizeof(T); \
1319 switch(device
->FmtType
)
1322 WRITE(ALbyte
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1325 WRITE(ALubyte
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1328 WRITE(ALshort
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1331 WRITE(ALushort
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1334 WRITE(ALint
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1337 WRITE(ALuint
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1340 WRITE(ALfloat
, device
->DryBuffer
+outchanoffset
, buffer
, SamplesToDo
, outchancount
);
1346 size
-= SamplesToDo
;
1347 IncrementRef(&device
->MixCount
);
1350 RestoreFPUMode(&oldMode
);
1354 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1356 ALCcontext
*Context
;
1358 device
->Connected
= ALC_FALSE
;
1360 Context
= ATOMIC_LOAD(&device
->ContextList
);
1363 ALvoice
*voice
, *voice_end
;
1365 voice
= Context
->Voices
;
1366 voice_end
= voice
+ Context
->VoiceCount
;
1367 while(voice
!= voice_end
)
1369 ALsource
*source
= voice
->Source
;
1370 voice
->Source
= NULL
;
1372 if(source
&& source
->state
== AL_PLAYING
)
1374 source
->state
= AL_STOPPED
;
1375 ATOMIC_STORE(&source
->current_buffer
, NULL
);
1376 source
->position
= 0;
1377 source
->position_fraction
= 0;
1382 Context
->VoiceCount
= 0;
1384 Context
= Context
->next
;