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 "uhjfilter.h"
38 #include "static_assert.h"
40 #include "mixer_defs.h"
42 #include "backends/base.h"
52 ALfloat ConeScale
= 1.0f
;
54 /* Localized Z scalar for mono sources */
55 ALfloat ZScale
= 1.0f
;
57 extern inline ALfloat
minf(ALfloat a
, ALfloat b
);
58 extern inline ALfloat
maxf(ALfloat a
, ALfloat b
);
59 extern inline ALfloat
clampf(ALfloat val
, ALfloat min
, ALfloat max
);
61 extern inline ALdouble
mind(ALdouble a
, ALdouble b
);
62 extern inline ALdouble
maxd(ALdouble a
, ALdouble b
);
63 extern inline ALdouble
clampd(ALdouble val
, ALdouble min
, ALdouble max
);
65 extern inline ALuint
minu(ALuint a
, ALuint b
);
66 extern inline ALuint
maxu(ALuint a
, ALuint b
);
67 extern inline ALuint
clampu(ALuint val
, ALuint min
, ALuint max
);
69 extern inline ALint
mini(ALint a
, ALint b
);
70 extern inline ALint
maxi(ALint a
, ALint b
);
71 extern inline ALint
clampi(ALint val
, ALint min
, ALint max
);
73 extern inline ALint64
mini64(ALint64 a
, ALint64 b
);
74 extern inline ALint64
maxi64(ALint64 a
, ALint64 b
);
75 extern inline ALint64
clampi64(ALint64 val
, ALint64 min
, ALint64 max
);
77 extern inline ALuint64
minu64(ALuint64 a
, ALuint64 b
);
78 extern inline ALuint64
maxu64(ALuint64 a
, ALuint64 b
);
79 extern inline ALuint64
clampu64(ALuint64 val
, ALuint64 min
, ALuint64 max
);
81 extern inline ALfloat
lerp(ALfloat val1
, ALfloat val2
, ALfloat mu
);
82 extern inline ALfloat
resample_fir4(ALfloat val0
, ALfloat val1
, ALfloat val2
, ALfloat val3
, ALuint frac
);
83 extern inline ALfloat
resample_fir8(ALfloat val0
, ALfloat val1
, ALfloat val2
, ALfloat val3
, ALfloat val4
, ALfloat val5
, ALfloat val6
, ALfloat val7
, ALuint frac
);
85 extern inline void aluVectorSet(aluVector
*restrict vector
, ALfloat x
, ALfloat y
, ALfloat z
, ALfloat w
);
87 extern inline void aluMatrixfSetRow(aluMatrixf
*matrix
, ALuint row
,
88 ALfloat m0
, ALfloat m1
, ALfloat m2
, ALfloat m3
);
89 extern inline void aluMatrixfSet(aluMatrixf
*matrix
,
90 ALfloat m00
, ALfloat m01
, ALfloat m02
, ALfloat m03
,
91 ALfloat m10
, ALfloat m11
, ALfloat m12
, ALfloat m13
,
92 ALfloat m20
, ALfloat m21
, ALfloat m22
, ALfloat m23
,
93 ALfloat m30
, ALfloat m31
, ALfloat m32
, ALfloat m33
);
95 extern inline void aluMatrixdSetRow(aluMatrixd
*matrix
, ALuint row
,
96 ALdouble m0
, ALdouble m1
, ALdouble m2
, ALdouble m3
);
97 extern inline void aluMatrixdSet(aluMatrixd
*matrix
,
98 ALdouble m00
, ALdouble m01
, ALdouble m02
, ALdouble m03
,
99 ALdouble m10
, ALdouble m11
, ALdouble m12
, ALdouble m13
,
100 ALdouble m20
, ALdouble m21
, ALdouble m22
, ALdouble m23
,
101 ALdouble m30
, ALdouble m31
, ALdouble m32
, ALdouble m33
);
104 /* NOTE: HRTF and UHJ are set up a bit special in the device. Normally the
105 * device's DryBuffer, NumChannels, ChannelName, and Channel fields correspond
106 * to the output format, and the DryBuffer is then converted and written to the
107 * backend's audio buffer.
109 * With HRTF or UHJ, these fields correspond to a virtual format, and the
110 * actual output is stored in DryBuffer[NumChannels] for the left channel and
111 * DryBuffer[NumChannels+1] for the right. As a final output step,
112 * the virtual channels will have HRTF filters or UHJ encoding applied and
113 * written to the actual output.
115 * Sources that get mixed using HRTF directly (or that want to skip HRTF or UHJ
116 * completely) will need to offset the output buffer so that they skip the
117 * virtual output and write to the actual output channels. This is the reason
120 * voice->Direct.OutBuffer += voice->Direct.OutChannels;
121 * voice->Direct.OutChannels = 2;
123 * at various points in the code where HRTF is explicitly used or bypassed.
126 static inline HrtfMixerFunc
SelectHrtfMixer(void)
129 if((CPUCapFlags
&CPU_CAP_SSE
))
133 if((CPUCapFlags
&CPU_CAP_NEON
))
141 static inline void aluCrossproduct(const ALfloat
*inVector1
, const ALfloat
*inVector2
, ALfloat
*outVector
)
143 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
144 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
145 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
148 static inline ALfloat
aluDotproduct(const aluVector
*vec1
, const aluVector
*vec2
)
150 return vec1
->v
[0]*vec2
->v
[0] + vec1
->v
[1]*vec2
->v
[1] + vec1
->v
[2]*vec2
->v
[2];
153 static inline ALfloat
aluNormalize(ALfloat
*vec
)
155 ALfloat length
= sqrtf(vec
[0]*vec
[0] + vec
[1]*vec
[1] + vec
[2]*vec
[2]);
158 ALfloat inv_length
= 1.0f
/length
;
159 vec
[0] *= inv_length
;
160 vec
[1] *= inv_length
;
161 vec
[2] *= inv_length
;
167 static inline void aluCrossproductd(const ALdouble
*inVector1
, const ALdouble
*inVector2
, ALdouble
*outVector
)
169 outVector
[0] = inVector1
[1]*inVector2
[2] - inVector1
[2]*inVector2
[1];
170 outVector
[1] = inVector1
[2]*inVector2
[0] - inVector1
[0]*inVector2
[2];
171 outVector
[2] = inVector1
[0]*inVector2
[1] - inVector1
[1]*inVector2
[0];
174 static inline ALdouble
aluNormalized(ALdouble
*vec
)
176 ALdouble length
= sqrt(vec
[0]*vec
[0] + vec
[1]*vec
[1] + vec
[2]*vec
[2]);
179 ALdouble inv_length
= 1.0/length
;
180 vec
[0] *= inv_length
;
181 vec
[1] *= inv_length
;
182 vec
[2] *= inv_length
;
187 static inline ALvoid
aluMatrixdFloat3(ALfloat
*vec
, ALfloat w
, const aluMatrixd
*mtx
)
189 ALdouble v
[4] = { vec
[0], vec
[1], vec
[2], w
};
191 vec
[0] = (ALfloat
)(v
[0]*mtx
->m
[0][0] + v
[1]*mtx
->m
[1][0] + v
[2]*mtx
->m
[2][0] + v
[3]*mtx
->m
[3][0]);
192 vec
[1] = (ALfloat
)(v
[0]*mtx
->m
[0][1] + v
[1]*mtx
->m
[1][1] + v
[2]*mtx
->m
[2][1] + v
[3]*mtx
->m
[3][1]);
193 vec
[2] = (ALfloat
)(v
[0]*mtx
->m
[0][2] + v
[1]*mtx
->m
[1][2] + v
[2]*mtx
->m
[2][2] + v
[3]*mtx
->m
[3][2]);
196 static inline ALvoid
aluMatrixdDouble3(ALdouble
*vec
, ALdouble w
, const aluMatrixd
*mtx
)
198 ALdouble v
[4] = { vec
[0], vec
[1], vec
[2], w
};
200 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];
201 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];
202 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];
205 static inline aluVector
aluMatrixdVector(const aluMatrixd
*mtx
, const aluVector
*vec
)
208 v
.v
[0] = (ALfloat
)(vec
->v
[0]*mtx
->m
[0][0] + vec
->v
[1]*mtx
->m
[1][0] + vec
->v
[2]*mtx
->m
[2][0] + vec
->v
[3]*mtx
->m
[3][0]);
209 v
.v
[1] = (ALfloat
)(vec
->v
[0]*mtx
->m
[0][1] + vec
->v
[1]*mtx
->m
[1][1] + vec
->v
[2]*mtx
->m
[2][1] + vec
->v
[3]*mtx
->m
[3][1]);
210 v
.v
[2] = (ALfloat
)(vec
->v
[0]*mtx
->m
[0][2] + vec
->v
[1]*mtx
->m
[1][2] + vec
->v
[2]*mtx
->m
[2][2] + vec
->v
[3]*mtx
->m
[3][2]);
211 v
.v
[3] = (ALfloat
)(vec
->v
[0]*mtx
->m
[0][3] + vec
->v
[1]*mtx
->m
[1][3] + vec
->v
[2]*mtx
->m
[2][3] + vec
->v
[3]*mtx
->m
[3][3]);
216 /* Prepares the interpolator for a given rate (determined by increment). A
217 * result of AL_FALSE indicates that the filter output will completely cut
220 * With a bit of work, and a trade of memory for CPU cost, this could be
221 * modified for use with an interpolated increment for buttery-smooth pitch
224 static ALboolean
BsincPrepare(const ALuint increment
, BsincState
*state
)
226 static const ALfloat scaleBase
= 1.510578918e-01f
, scaleRange
= 1.177936623e+00f
;
227 static const ALuint m
[BSINC_SCALE_COUNT
] = { 24, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 12 };
228 static const ALuint to
[4][BSINC_SCALE_COUNT
] =
230 { 0, 24, 408, 792, 1176, 1560, 1944, 2328, 2648, 2968, 3288, 3544, 3800, 4056, 4248, 4440 },
231 { 4632, 5016, 5400, 5784, 6168, 6552, 6936, 7320, 7640, 7960, 8280, 8536, 8792, 9048, 9240, 0 },
232 { 0, 9432, 9816, 10200, 10584, 10968, 11352, 11736, 12056, 12376, 12696, 12952, 13208, 13464, 13656, 13848 },
233 { 14040, 14424, 14808, 15192, 15576, 15960, 16344, 16728, 17048, 17368, 17688, 17944, 18200, 18456, 18648, 0 }
235 static const ALuint tm
[2][BSINC_SCALE_COUNT
] =
237 { 0, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 12 },
238 { 24, 24, 24, 24, 24, 24, 24, 20, 20, 20, 16, 16, 16, 12, 12, 0 }
242 ALboolean uncut
= AL_TRUE
;
244 if(increment
> FRACTIONONE
)
246 sf
= (ALfloat
)FRACTIONONE
/ increment
;
249 /* Signal has been completely cut. The return result can be used
250 * to skip the filter (and output zeros) as an optimization.
258 sf
= (BSINC_SCALE_COUNT
- 1) * (sf
- scaleBase
) * scaleRange
;
260 /* The interpolation factor is fit to this diagonally-symmetric
261 * curve to reduce the transition ripple caused by interpolating
262 * different scales of the sinc function.
264 sf
= 1.0f
- cosf(asinf(sf
- si
));
270 si
= BSINC_SCALE_COUNT
- 1;
275 state
->l
= -(ALint
)((m
[si
] / 2) - 1);
276 /* The CPU cost of this table re-mapping could be traded for the memory
277 * cost of a complete table map (1024 elements large).
279 for(pi
= 0;pi
< BSINC_PHASE_COUNT
;pi
++)
281 state
->coeffs
[pi
].filter
= &bsincTab
[to
[0][si
] + tm
[0][si
]*pi
];
282 state
->coeffs
[pi
].scDelta
= &bsincTab
[to
[1][si
] + tm
[1][si
]*pi
];
283 state
->coeffs
[pi
].phDelta
= &bsincTab
[to
[2][si
] + tm
[0][si
]*pi
];
284 state
->coeffs
[pi
].spDelta
= &bsincTab
[to
[3][si
] + tm
[1][si
]*pi
];
290 static ALvoid
CalcListenerParams(ALlistener
*Listener
)
292 ALdouble N
[3], V
[3], U
[3], P
[3];
295 N
[0] = Listener
->Forward
[0];
296 N
[1] = Listener
->Forward
[1];
297 N
[2] = Listener
->Forward
[2];
299 V
[0] = Listener
->Up
[0];
300 V
[1] = Listener
->Up
[1];
301 V
[2] = Listener
->Up
[2];
303 /* Build and normalize right-vector */
304 aluCrossproductd(N
, V
, U
);
307 aluMatrixdSet(&Listener
->Params
.Matrix
,
308 U
[0], V
[0], -N
[0], 0.0,
309 U
[1], V
[1], -N
[1], 0.0,
310 U
[2], V
[2], -N
[2], 0.0,
314 P
[0] = Listener
->Position
.v
[0];
315 P
[1] = Listener
->Position
.v
[1];
316 P
[2] = Listener
->Position
.v
[2];
317 aluMatrixdDouble3(P
, 1.0, &Listener
->Params
.Matrix
);
318 aluMatrixdSetRow(&Listener
->Params
.Matrix
, 3, -P
[0], -P
[1], -P
[2], 1.0f
);
320 Listener
->Params
.Velocity
= aluMatrixdVector(&Listener
->Params
.Matrix
, &Listener
->Velocity
);
323 ALvoid
CalcNonAttnSourceParams(ALvoice
*voice
, const ALsource
*ALSource
, const ALCcontext
*ALContext
)
325 static const struct ChanMap MonoMap
[1] = {
326 { FrontCenter
, 0.0f
, 0.0f
}
328 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
329 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) }
331 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
332 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) }
334 { FrontLeft
, DEG2RAD( -45.0f
), DEG2RAD(0.0f
) },
335 { FrontRight
, DEG2RAD( 45.0f
), DEG2RAD(0.0f
) },
336 { BackLeft
, DEG2RAD(-135.0f
), DEG2RAD(0.0f
) },
337 { BackRight
, DEG2RAD( 135.0f
), DEG2RAD(0.0f
) }
339 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
340 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
341 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
343 { SideLeft
, DEG2RAD(-110.0f
), DEG2RAD(0.0f
) },
344 { SideRight
, DEG2RAD( 110.0f
), DEG2RAD(0.0f
) }
346 { FrontLeft
, DEG2RAD(-30.0f
), DEG2RAD(0.0f
) },
347 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
348 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
350 { BackCenter
, DEG2RAD(180.0f
), DEG2RAD(0.0f
) },
351 { SideLeft
, DEG2RAD(-90.0f
), DEG2RAD(0.0f
) },
352 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
354 { FrontLeft
, DEG2RAD( -30.0f
), DEG2RAD(0.0f
) },
355 { FrontRight
, DEG2RAD( 30.0f
), DEG2RAD(0.0f
) },
356 { FrontCenter
, DEG2RAD( 0.0f
), DEG2RAD(0.0f
) },
358 { BackLeft
, DEG2RAD(-150.0f
), DEG2RAD(0.0f
) },
359 { BackRight
, DEG2RAD( 150.0f
), DEG2RAD(0.0f
) },
360 { SideLeft
, DEG2RAD( -90.0f
), DEG2RAD(0.0f
) },
361 { SideRight
, DEG2RAD( 90.0f
), DEG2RAD(0.0f
) }
364 const ALCdevice
*Device
= ALContext
->Device
;
365 ALfloat SourceVolume
,ListenerGain
,MinVolume
,MaxVolume
;
366 ALbufferlistitem
*BufferListItem
;
367 enum FmtChannels Channels
;
368 ALfloat DryGain
, DryGainHF
, DryGainLF
;
369 ALfloat WetGain
[MAX_SENDS
];
370 ALfloat WetGainHF
[MAX_SENDS
];
371 ALfloat WetGainLF
[MAX_SENDS
];
372 ALeffectslot
*SendSlots
[MAX_SENDS
];
373 ALuint NumSends
, Frequency
;
375 const struct ChanMap
*chans
= NULL
;
376 ALuint num_channels
= 0;
377 ALboolean DirectChannels
;
378 ALboolean isbformat
= AL_FALSE
;
382 /* Get device properties */
383 NumSends
= Device
->NumAuxSends
;
384 Frequency
= Device
->Frequency
;
386 /* Get listener properties */
387 ListenerGain
= ALContext
->Listener
->Gain
;
389 /* Get source properties */
390 SourceVolume
= ALSource
->Gain
;
391 MinVolume
= ALSource
->MinGain
;
392 MaxVolume
= ALSource
->MaxGain
;
393 Pitch
= ALSource
->Pitch
;
394 Relative
= ALSource
->HeadRelative
;
395 DirectChannels
= ALSource
->DirectChannels
;
397 voice
->Direct
.OutBuffer
= Device
->DryBuffer
;
398 voice
->Direct
.OutChannels
= Device
->NumChannels
;
399 for(i
= 0;i
< NumSends
;i
++)
401 SendSlots
[i
] = ALSource
->Send
[i
].Slot
;
402 if(!SendSlots
[i
] && i
== 0)
403 SendSlots
[i
] = Device
->DefaultSlot
;
404 if(!SendSlots
[i
] || SendSlots
[i
]->EffectType
== AL_EFFECT_NULL
)
407 voice
->Send
[i
].OutBuffer
= NULL
;
408 voice
->Send
[i
].OutChannels
= 0;
412 voice
->Send
[i
].OutBuffer
= SendSlots
[i
]->WetBuffer
;
413 voice
->Send
[i
].OutChannels
= SendSlots
[i
]->NumChannels
;
417 /* Calculate the stepping value */
419 BufferListItem
= ATOMIC_LOAD(&ALSource
->queue
);
420 while(BufferListItem
!= NULL
)
423 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
425 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
426 if(Pitch
> (ALfloat
)MAX_PITCH
)
427 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
429 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
430 BsincPrepare(voice
->Step
, &voice
->SincState
);
432 Channels
= ALBuffer
->FmtChannels
;
435 BufferListItem
= BufferListItem
->next
;
438 /* Calculate gains */
439 DryGain
= clampf(SourceVolume
, MinVolume
, MaxVolume
);
440 DryGain
*= ALSource
->Direct
.Gain
* ListenerGain
;
441 DryGainHF
= ALSource
->Direct
.GainHF
;
442 DryGainLF
= ALSource
->Direct
.GainLF
;
443 for(i
= 0;i
< NumSends
;i
++)
445 WetGain
[i
] = clampf(SourceVolume
, MinVolume
, MaxVolume
);
446 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
447 WetGainHF
[i
] = ALSource
->Send
[i
].GainHF
;
448 WetGainLF
[i
] = ALSource
->Send
[i
].GainLF
;
491 DirectChannels
= AL_FALSE
;
497 DirectChannels
= AL_FALSE
;
503 ALfloat N
[3], V
[3], U
[3];
508 N
[0] = ALSource
->Orientation
[0][0];
509 N
[1] = ALSource
->Orientation
[0][1];
510 N
[2] = ALSource
->Orientation
[0][2];
512 V
[0] = ALSource
->Orientation
[1][0];
513 V
[1] = ALSource
->Orientation
[1][1];
514 V
[2] = ALSource
->Orientation
[1][2];
518 const aluMatrixd
*lmatrix
= &ALContext
->Listener
->Params
.Matrix
;
519 aluMatrixdFloat3(N
, 0.0f
, lmatrix
);
520 aluMatrixdFloat3(V
, 0.0f
, lmatrix
);
522 /* Build and normalize right-vector */
523 aluCrossproduct(N
, V
, U
);
526 /* Build a rotate + conversion matrix (B-Format -> N3D), and include
527 * scaling for first-order content on second- or third-order output.
529 scale
= Device
->AmbiScale
* 1.732050808f
;
530 aluMatrixfSet(&matrix
,
531 1.414213562f
, 0.0f
, 0.0f
, 0.0f
,
532 0.0f
, -N
[0]*scale
, N
[1]*scale
, -N
[2]*scale
,
533 0.0f
, U
[0]*scale
, -U
[1]*scale
, U
[2]*scale
,
534 0.0f
, -V
[0]*scale
, V
[1]*scale
, -V
[2]*scale
537 for(c
= 0;c
< num_channels
;c
++)
538 ComputeFirstOrderGains(Device
->AmbiCoeffs
, Device
->NumChannels
, matrix
.m
[c
], DryGain
,
539 voice
->Direct
.Gains
[c
].Target
);
541 /* Rebuild the matrix, without the second- or third-order output
542 * scaling (effects take first-order content, and will do the scaling
543 * themselves when mixing to the output).
545 scale
= 1.732050808f
;
546 aluMatrixfSetRow(&matrix
, 1, 0.0f
, -N
[0]*scale
, N
[1]*scale
, -N
[2]*scale
);
547 aluMatrixfSetRow(&matrix
, 2, 0.0f
, U
[0]*scale
, -U
[1]*scale
, U
[2]*scale
);
548 aluMatrixfSetRow(&matrix
, 3, 0.0f
, -V
[0]*scale
, V
[1]*scale
, -V
[2]*scale
);
549 for(i
= 0;i
< NumSends
;i
++)
553 for(c
= 0;c
< num_channels
;c
++)
555 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
556 voice
->Send
[i
].Gains
[c
].Target
[j
] = 0.0f
;
561 for(c
= 0;c
< num_channels
;c
++)
563 const ALeffectslot
*Slot
= SendSlots
[i
];
564 ComputeFirstOrderGains(Slot
->AmbiCoeffs
, Slot
->NumChannels
, matrix
.m
[c
],
565 WetGain
[i
], voice
->Send
[i
].Gains
[c
].Target
);
570 voice
->IsHrtf
= AL_FALSE
;
574 ALfloat coeffs
[MAX_AMBI_COEFFS
];
578 if(Device
->Hrtf
|| Device
->Uhj_Encoder
)
580 /* DirectChannels with HRTF enabled. Skip the virtual channels
581 * and write FrontLeft and FrontRight inputs to the first and
584 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
585 voice
->Direct
.OutChannels
= 2;
586 for(c
= 0;c
< num_channels
;c
++)
588 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
589 voice
->Direct
.Gains
[c
].Target
[j
] = 0.0f
;
591 if(chans
[c
].channel
== FrontLeft
)
592 voice
->Direct
.Gains
[c
].Target
[0] = DryGain
;
593 else if(chans
[c
].channel
== FrontRight
)
594 voice
->Direct
.Gains
[c
].Target
[1] = DryGain
;
597 else for(c
= 0;c
< num_channels
;c
++)
600 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
601 voice
->Direct
.Gains
[c
].Target
[j
] = 0.0f
;
602 if((idx
=GetChannelIdxByName(Device
, chans
[c
].channel
)) != -1)
603 voice
->Direct
.Gains
[c
].Target
[idx
] = DryGain
;
606 /* Auxiliary sends still use normal panning since they mix to B-Format, which can't
608 for(c
= 0;c
< num_channels
;c
++)
610 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, coeffs
);
612 for(i
= 0;i
< NumSends
;i
++)
616 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
617 voice
->Send
[i
].Gains
[c
].Target
[j
] = 0.0f
;
621 const ALeffectslot
*Slot
= SendSlots
[i
];
622 ComputePanningGains(Slot
->AmbiCoeffs
, Slot
->NumChannels
, coeffs
,
623 WetGain
[i
], voice
->Send
[i
].Gains
[c
].Target
);
628 voice
->IsHrtf
= AL_FALSE
;
630 else if(Device
->Hrtf_Mode
== FullHrtf
)
632 /* Full HRTF rendering. Skip the virtual channels and render each
633 * input channel to the real outputs.
635 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
636 voice
->Direct
.OutChannels
= 2;
637 for(c
= 0;c
< num_channels
;c
++)
639 if(chans
[c
].channel
== LFE
)
642 voice
->Direct
.Hrtf
[c
].Target
.Delay
[0] = 0;
643 voice
->Direct
.Hrtf
[c
].Target
.Delay
[1] = 0;
644 for(i
= 0;i
< HRIR_LENGTH
;i
++)
646 voice
->Direct
.Hrtf
[c
].Target
.Coeffs
[i
][0] = 0.0f
;
647 voice
->Direct
.Hrtf
[c
].Target
.Coeffs
[i
][1] = 0.0f
;
650 for(i
= 0;i
< NumSends
;i
++)
652 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
653 voice
->Send
[i
].Gains
[c
].Target
[j
] = 0.0f
;
659 /* Get the static HRIR coefficients and delays for this channel. */
660 GetLerpedHrtfCoeffs(Device
->Hrtf
,
661 chans
[c
].elevation
, chans
[c
].angle
, 1.0f
, DryGain
,
662 voice
->Direct
.Hrtf
[c
].Target
.Coeffs
,
663 voice
->Direct
.Hrtf
[c
].Target
.Delay
666 /* Normal panning for auxiliary sends. */
667 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, coeffs
);
669 for(i
= 0;i
< NumSends
;i
++)
673 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
674 voice
->Send
[i
].Gains
[c
].Target
[j
] = 0.0f
;
678 const ALeffectslot
*Slot
= SendSlots
[i
];
679 ComputePanningGains(Slot
->AmbiCoeffs
, Slot
->NumChannels
, coeffs
,
680 WetGain
[i
], voice
->Send
[i
].Gains
[c
].Target
);
685 voice
->IsHrtf
= AL_TRUE
;
689 /* Basic or no HRTF rendering. Use normal panning to the output. */
690 for(c
= 0;c
< num_channels
;c
++)
692 /* Special-case LFE */
693 if(chans
[c
].channel
== LFE
)
696 for(j
= 0;j
< MAX_OUTPUT_CHANNELS
;j
++)
697 voice
->Direct
.Gains
[c
].Target
[j
] = 0.0f
;
698 if((idx
=GetChannelIdxByName(Device
, chans
[c
].channel
)) != -1)
699 voice
->Direct
.Gains
[c
].Target
[idx
] = DryGain
;
701 for(i
= 0;i
< NumSends
;i
++)
704 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
705 voice
->Send
[i
].Gains
[c
].Target
[j
] = 0.0f
;
710 CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, coeffs
);
712 ComputePanningGains(Device
->AmbiCoeffs
, Device
->NumChannels
, coeffs
, DryGain
, voice
->Direct
.Gains
[c
].Target
);
714 for(i
= 0;i
< NumSends
;i
++)
719 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
720 voice
->Send
[i
].Gains
[c
].Target
[j
] = 0.0f
;
724 const ALeffectslot
*Slot
= SendSlots
[i
];
725 ComputePanningGains(Slot
->AmbiCoeffs
, Slot
->NumChannels
, coeffs
,
726 WetGain
[i
], voice
->Send
[i
].Gains
[c
].Target
);
731 voice
->IsHrtf
= AL_FALSE
;
736 ALfloat hfscale
= ALSource
->Direct
.HFReference
/ Frequency
;
737 ALfloat lfscale
= ALSource
->Direct
.LFReference
/ Frequency
;
738 DryGainHF
= maxf(DryGainHF
, 0.0001f
);
739 DryGainLF
= maxf(DryGainLF
, 0.0001f
);
740 for(c
= 0;c
< num_channels
;c
++)
742 voice
->Direct
.Filters
[c
].ActiveType
= AF_None
;
743 if(DryGainHF
!= 1.0f
) voice
->Direct
.Filters
[c
].ActiveType
|= AF_LowPass
;
744 if(DryGainLF
!= 1.0f
) voice
->Direct
.Filters
[c
].ActiveType
|= AF_HighPass
;
745 ALfilterState_setParams(
746 &voice
->Direct
.Filters
[c
].LowPass
, ALfilterType_HighShelf
,
747 DryGainHF
, hfscale
, calc_rcpQ_from_slope(DryGainHF
, 0.75f
)
749 ALfilterState_setParams(
750 &voice
->Direct
.Filters
[c
].HighPass
, ALfilterType_LowShelf
,
751 DryGainLF
, lfscale
, calc_rcpQ_from_slope(DryGainLF
, 0.75f
)
755 for(i
= 0;i
< NumSends
;i
++)
757 ALfloat hfscale
= ALSource
->Send
[i
].HFReference
/ Frequency
;
758 ALfloat lfscale
= ALSource
->Send
[i
].LFReference
/ Frequency
;
759 WetGainHF
[i
] = maxf(WetGainHF
[i
], 0.0001f
);
760 WetGainLF
[i
] = maxf(WetGainLF
[i
], 0.0001f
);
761 for(c
= 0;c
< num_channels
;c
++)
763 voice
->Send
[i
].Filters
[c
].ActiveType
= AF_None
;
764 if(WetGainHF
[i
] != 1.0f
) voice
->Send
[i
].Filters
[c
].ActiveType
|= AF_LowPass
;
765 if(WetGainLF
[i
] != 1.0f
) voice
->Send
[i
].Filters
[c
].ActiveType
|= AF_HighPass
;
766 ALfilterState_setParams(
767 &voice
->Send
[i
].Filters
[c
].LowPass
, ALfilterType_HighShelf
,
768 WetGainHF
[i
], hfscale
, calc_rcpQ_from_slope(WetGainHF
[i
], 0.75f
)
770 ALfilterState_setParams(
771 &voice
->Send
[i
].Filters
[c
].HighPass
, ALfilterType_LowShelf
,
772 WetGainLF
[i
], lfscale
, calc_rcpQ_from_slope(WetGainLF
[i
], 0.75f
)
778 ALvoid
CalcSourceParams(ALvoice
*voice
, const ALsource
*ALSource
, const ALCcontext
*ALContext
)
780 const ALCdevice
*Device
= ALContext
->Device
;
781 aluVector Position
, Velocity
, Direction
, SourceToListener
;
782 ALfloat InnerAngle
,OuterAngle
,Angle
,Distance
,ClampedDist
;
783 ALfloat MinVolume
,MaxVolume
,MinDist
,MaxDist
,Rolloff
;
784 ALfloat ConeVolume
,ConeHF
,SourceVolume
,ListenerGain
;
785 ALfloat DopplerFactor
, SpeedOfSound
;
786 ALfloat AirAbsorptionFactor
;
787 ALfloat RoomAirAbsorption
[MAX_SENDS
];
788 ALbufferlistitem
*BufferListItem
;
789 ALeffectslot
*SendSlots
[MAX_SENDS
];
791 ALfloat RoomAttenuation
[MAX_SENDS
];
792 ALfloat MetersPerUnit
;
793 ALfloat RoomRolloffBase
;
794 ALfloat RoomRolloff
[MAX_SENDS
];
795 ALfloat DecayDistance
[MAX_SENDS
];
799 ALboolean DryGainHFAuto
;
800 ALfloat WetGain
[MAX_SENDS
];
801 ALfloat WetGainHF
[MAX_SENDS
];
802 ALfloat WetGainLF
[MAX_SENDS
];
803 ALboolean WetGainAuto
;
804 ALboolean WetGainHFAuto
;
812 for(i
= 0;i
< MAX_SENDS
;i
++)
818 /* Get context/device properties */
819 DopplerFactor
= ALContext
->DopplerFactor
* ALSource
->DopplerFactor
;
820 SpeedOfSound
= ALContext
->SpeedOfSound
* ALContext
->DopplerVelocity
;
821 NumSends
= Device
->NumAuxSends
;
822 Frequency
= Device
->Frequency
;
824 /* Get listener properties */
825 ListenerGain
= ALContext
->Listener
->Gain
;
826 MetersPerUnit
= ALContext
->Listener
->MetersPerUnit
;
828 /* Get source properties */
829 SourceVolume
= ALSource
->Gain
;
830 MinVolume
= ALSource
->MinGain
;
831 MaxVolume
= ALSource
->MaxGain
;
832 Pitch
= ALSource
->Pitch
;
833 Position
= ALSource
->Position
;
834 Direction
= ALSource
->Direction
;
835 Velocity
= ALSource
->Velocity
;
836 MinDist
= ALSource
->RefDistance
;
837 MaxDist
= ALSource
->MaxDistance
;
838 Rolloff
= ALSource
->RollOffFactor
;
839 InnerAngle
= ALSource
->InnerAngle
;
840 OuterAngle
= ALSource
->OuterAngle
;
841 AirAbsorptionFactor
= ALSource
->AirAbsorptionFactor
;
842 DryGainHFAuto
= ALSource
->DryGainHFAuto
;
843 WetGainAuto
= ALSource
->WetGainAuto
;
844 WetGainHFAuto
= ALSource
->WetGainHFAuto
;
845 RoomRolloffBase
= ALSource
->RoomRolloffFactor
;
847 voice
->Direct
.OutBuffer
= Device
->DryBuffer
;
848 voice
->Direct
.OutChannels
= Device
->NumChannels
;
849 for(i
= 0;i
< NumSends
;i
++)
851 SendSlots
[i
] = ALSource
->Send
[i
].Slot
;
853 if(!SendSlots
[i
] && i
== 0)
854 SendSlots
[i
] = Device
->DefaultSlot
;
855 if(!SendSlots
[i
] || SendSlots
[i
]->EffectType
== AL_EFFECT_NULL
)
858 RoomRolloff
[i
] = 0.0f
;
859 DecayDistance
[i
] = 0.0f
;
860 RoomAirAbsorption
[i
] = 1.0f
;
862 else if(SendSlots
[i
]->AuxSendAuto
)
864 RoomRolloff
[i
] = RoomRolloffBase
;
865 if(IsReverbEffect(SendSlots
[i
]->EffectType
))
867 RoomRolloff
[i
] += SendSlots
[i
]->EffectProps
.Reverb
.RoomRolloffFactor
;
868 DecayDistance
[i
] = SendSlots
[i
]->EffectProps
.Reverb
.DecayTime
*
869 SPEEDOFSOUNDMETRESPERSEC
;
870 RoomAirAbsorption
[i
] = SendSlots
[i
]->EffectProps
.Reverb
.AirAbsorptionGainHF
;
874 DecayDistance
[i
] = 0.0f
;
875 RoomAirAbsorption
[i
] = 1.0f
;
880 /* If the slot's auxiliary send auto is off, the data sent to the
881 * effect slot is the same as the dry path, sans filter effects */
882 RoomRolloff
[i
] = Rolloff
;
883 DecayDistance
[i
] = 0.0f
;
884 RoomAirAbsorption
[i
] = AIRABSORBGAINHF
;
889 voice
->Send
[i
].OutBuffer
= NULL
;
890 voice
->Send
[i
].OutChannels
= 0;
894 voice
->Send
[i
].OutBuffer
= SendSlots
[i
]->WetBuffer
;
895 voice
->Send
[i
].OutChannels
= SendSlots
[i
]->NumChannels
;
899 /* Transform source to listener space (convert to head relative) */
900 if(ALSource
->HeadRelative
== AL_FALSE
)
902 const aluMatrixd
*Matrix
= &ALContext
->Listener
->Params
.Matrix
;
903 /* Transform source vectors */
904 Position
= aluMatrixdVector(Matrix
, &Position
);
905 Velocity
= aluMatrixdVector(Matrix
, &Velocity
);
906 Direction
= aluMatrixdVector(Matrix
, &Direction
);
910 const aluVector
*lvelocity
= &ALContext
->Listener
->Params
.Velocity
;
911 /* Offset the source velocity to be relative of the listener velocity */
912 Velocity
.v
[0] += lvelocity
->v
[0];
913 Velocity
.v
[1] += lvelocity
->v
[1];
914 Velocity
.v
[2] += lvelocity
->v
[2];
917 aluNormalize(Direction
.v
);
918 SourceToListener
.v
[0] = -Position
.v
[0];
919 SourceToListener
.v
[1] = -Position
.v
[1];
920 SourceToListener
.v
[2] = -Position
.v
[2];
921 SourceToListener
.v
[3] = 0.0f
;
922 Distance
= aluNormalize(SourceToListener
.v
);
924 /* Calculate distance attenuation */
925 ClampedDist
= Distance
;
928 for(i
= 0;i
< NumSends
;i
++)
929 RoomAttenuation
[i
] = 1.0f
;
930 switch(ALContext
->SourceDistanceModel
? ALSource
->DistanceModel
:
931 ALContext
->DistanceModel
)
933 case InverseDistanceClamped
:
934 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
935 if(MaxDist
< MinDist
)
938 case InverseDistance
:
941 ALfloat dist
= lerp(MinDist
, ClampedDist
, Rolloff
);
942 if(dist
> 0.0f
) Attenuation
= MinDist
/ dist
;
943 for(i
= 0;i
< NumSends
;i
++)
945 dist
= lerp(MinDist
, ClampedDist
, RoomRolloff
[i
]);
946 if(dist
> 0.0f
) RoomAttenuation
[i
] = MinDist
/ dist
;
951 case LinearDistanceClamped
:
952 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
953 if(MaxDist
< MinDist
)
957 if(MaxDist
!= MinDist
)
959 Attenuation
= 1.0f
- (Rolloff
*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
960 Attenuation
= maxf(Attenuation
, 0.0f
);
961 for(i
= 0;i
< NumSends
;i
++)
963 RoomAttenuation
[i
] = 1.0f
- (RoomRolloff
[i
]*(ClampedDist
-MinDist
)/(MaxDist
- MinDist
));
964 RoomAttenuation
[i
] = maxf(RoomAttenuation
[i
], 0.0f
);
969 case ExponentDistanceClamped
:
970 ClampedDist
= clampf(ClampedDist
, MinDist
, MaxDist
);
971 if(MaxDist
< MinDist
)
974 case ExponentDistance
:
975 if(ClampedDist
> 0.0f
&& MinDist
> 0.0f
)
977 Attenuation
= powf(ClampedDist
/MinDist
, -Rolloff
);
978 for(i
= 0;i
< NumSends
;i
++)
979 RoomAttenuation
[i
] = powf(ClampedDist
/MinDist
, -RoomRolloff
[i
]);
983 case DisableDistance
:
984 ClampedDist
= MinDist
;
988 /* Source Gain + Attenuation */
989 DryGain
= SourceVolume
* Attenuation
;
990 for(i
= 0;i
< NumSends
;i
++)
991 WetGain
[i
] = SourceVolume
* RoomAttenuation
[i
];
993 /* Distance-based air absorption */
994 if(AirAbsorptionFactor
> 0.0f
&& ClampedDist
> MinDist
)
996 ALfloat meters
= (ClampedDist
-MinDist
) * MetersPerUnit
;
997 DryGainHF
*= powf(AIRABSORBGAINHF
, AirAbsorptionFactor
*meters
);
998 for(i
= 0;i
< NumSends
;i
++)
999 WetGainHF
[i
] *= powf(RoomAirAbsorption
[i
], AirAbsorptionFactor
*meters
);
1004 ALfloat ApparentDist
= 1.0f
/maxf(Attenuation
, 0.00001f
) - 1.0f
;
1006 /* Apply a decay-time transformation to the wet path, based on the
1007 * attenuation of the dry path.
1009 * Using the apparent distance, based on the distance attenuation, the
1010 * initial decay of the reverb effect is calculated and applied to the
1013 for(i
= 0;i
< NumSends
;i
++)
1015 if(DecayDistance
[i
] > 0.0f
)
1016 WetGain
[i
] *= powf(0.001f
/*-60dB*/, ApparentDist
/DecayDistance
[i
]);
1020 /* Calculate directional soundcones */
1021 Angle
= RAD2DEG(acosf(aluDotproduct(&Direction
, &SourceToListener
)) * ConeScale
) * 2.0f
;
1022 if(Angle
> InnerAngle
&& Angle
<= OuterAngle
)
1024 ALfloat scale
= (Angle
-InnerAngle
) / (OuterAngle
-InnerAngle
);
1025 ConeVolume
= lerp(1.0f
, ALSource
->OuterGain
, scale
);
1026 ConeHF
= lerp(1.0f
, ALSource
->OuterGainHF
, scale
);
1028 else if(Angle
> OuterAngle
)
1030 ConeVolume
= ALSource
->OuterGain
;
1031 ConeHF
= ALSource
->OuterGainHF
;
1039 DryGain
*= ConeVolume
;
1042 for(i
= 0;i
< NumSends
;i
++)
1043 WetGain
[i
] *= ConeVolume
;
1046 DryGainHF
*= ConeHF
;
1049 for(i
= 0;i
< NumSends
;i
++)
1050 WetGainHF
[i
] *= ConeHF
;
1053 /* Clamp to Min/Max Gain */
1054 DryGain
= clampf(DryGain
, MinVolume
, MaxVolume
);
1055 for(i
= 0;i
< NumSends
;i
++)
1056 WetGain
[i
] = clampf(WetGain
[i
], MinVolume
, MaxVolume
);
1058 /* Apply gain and frequency filters */
1059 DryGain
*= ALSource
->Direct
.Gain
* ListenerGain
;
1060 DryGainHF
*= ALSource
->Direct
.GainHF
;
1061 DryGainLF
*= ALSource
->Direct
.GainLF
;
1062 for(i
= 0;i
< NumSends
;i
++)
1064 WetGain
[i
] *= ALSource
->Send
[i
].Gain
* ListenerGain
;
1065 WetGainHF
[i
] *= ALSource
->Send
[i
].GainHF
;
1066 WetGainLF
[i
] *= ALSource
->Send
[i
].GainLF
;
1069 /* Calculate velocity-based doppler effect */
1070 if(DopplerFactor
> 0.0f
)
1072 const aluVector
*lvelocity
= &ALContext
->Listener
->Params
.Velocity
;
1075 if(SpeedOfSound
< 1.0f
)
1077 DopplerFactor
*= 1.0f
/SpeedOfSound
;
1078 SpeedOfSound
= 1.0f
;
1081 VSS
= aluDotproduct(&Velocity
, &SourceToListener
) * DopplerFactor
;
1082 VLS
= aluDotproduct(lvelocity
, &SourceToListener
) * DopplerFactor
;
1084 Pitch
*= clampf(SpeedOfSound
-VLS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
) /
1085 clampf(SpeedOfSound
-VSS
, 1.0f
, SpeedOfSound
*2.0f
- 1.0f
);
1088 BufferListItem
= ATOMIC_LOAD(&ALSource
->queue
);
1089 while(BufferListItem
!= NULL
)
1092 if((ALBuffer
=BufferListItem
->buffer
) != NULL
)
1094 /* Calculate fixed-point stepping value, based on the pitch, buffer
1095 * frequency, and output frequency. */
1096 Pitch
= Pitch
* ALBuffer
->Frequency
/ Frequency
;
1097 if(Pitch
> (ALfloat
)MAX_PITCH
)
1098 voice
->Step
= MAX_PITCH
<<FRACTIONBITS
;
1100 voice
->Step
= maxi(fastf2i(Pitch
*FRACTIONONE
+ 0.5f
), 1);
1101 BsincPrepare(voice
->Step
, &voice
->SincState
);
1105 BufferListItem
= BufferListItem
->next
;
1108 if(Device
->Hrtf_Mode
== FullHrtf
)
1110 /* Full HRTF rendering. Skip the virtual channels and render to the
1113 aluVector dir
= {{ 0.0f
, 0.0f
, -1.0f
, 0.0f
}};
1114 ALfloat ev
= 0.0f
, az
= 0.0f
;
1115 ALfloat radius
= ALSource
->Radius
;
1116 ALfloat dirfact
= 1.0f
;
1117 ALfloat coeffs
[MAX_AMBI_COEFFS
];
1119 voice
->Direct
.OutBuffer
+= voice
->Direct
.OutChannels
;
1120 voice
->Direct
.OutChannels
= 2;
1122 if(Distance
> FLT_EPSILON
)
1124 dir
.v
[0] = -SourceToListener
.v
[0];
1125 dir
.v
[1] = -SourceToListener
.v
[1];
1126 dir
.v
[2] = -SourceToListener
.v
[2] * ZScale
;
1128 /* Calculate elevation and azimuth only when the source is not at
1129 * the listener. This prevents +0 and -0 Z from producing
1130 * inconsistent panning. Also, clamp Y in case FP precision errors
1131 * cause it to land outside of -1..+1. */
1132 ev
= asinf(clampf(dir
.v
[1], -1.0f
, 1.0f
));
1133 az
= atan2f(dir
.v
[0], -dir
.v
[2]);
1137 if(radius
>= Distance
)
1138 dirfact
*= Distance
/ radius
* 0.5f
;
1140 dirfact
*= 1.0f
- (asinf(radius
/ Distance
) / F_PI
);
1143 /* Get the HRIR coefficients and delays. */
1144 GetLerpedHrtfCoeffs(Device
->Hrtf
, ev
, az
, dirfact
, DryGain
,
1145 voice
->Direct
.Hrtf
[0].Target
.Coeffs
,
1146 voice
->Direct
.Hrtf
[0].Target
.Delay
);
1148 dir
.v
[0] *= dirfact
;
1149 dir
.v
[1] *= dirfact
;
1150 dir
.v
[2] *= dirfact
;
1151 CalcDirectionCoeffs(dir
.v
, coeffs
);
1153 for(i
= 0;i
< NumSends
;i
++)
1158 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
1159 voice
->Send
[i
].Gains
[0].Target
[j
] = 0.0f
;
1163 const ALeffectslot
*Slot
= SendSlots
[i
];
1164 ComputePanningGains(Slot
->AmbiCoeffs
, Slot
->NumChannels
, coeffs
,
1165 WetGain
[i
], voice
->Send
[i
].Gains
[0].Target
);
1169 voice
->IsHrtf
= AL_TRUE
;
1173 /* Basic or no HRTF rendering. Use normal panning to the output. */
1174 ALfloat dir
[3] = { 0.0f
, 0.0f
, -1.0f
};
1175 ALfloat radius
= ALSource
->Radius
;
1176 ALfloat coeffs
[MAX_AMBI_COEFFS
];
1178 /* Get the localized direction, and compute panned gains. */
1179 if(Distance
> FLT_EPSILON
)
1181 dir
[0] = -SourceToListener
.v
[0];
1182 dir
[1] = -SourceToListener
.v
[1];
1183 dir
[2] = -SourceToListener
.v
[2] * ZScale
;
1188 if(radius
>= Distance
)
1189 dirfact
= Distance
/ radius
* 0.5f
;
1191 dirfact
= 1.0f
- (asinf(radius
/ Distance
) / F_PI
);
1196 CalcDirectionCoeffs(dir
, coeffs
);
1198 ComputePanningGains(Device
->AmbiCoeffs
, Device
->NumChannels
, coeffs
, DryGain
,
1199 voice
->Direct
.Gains
[0].Target
);
1201 for(i
= 0;i
< NumSends
;i
++)
1206 for(j
= 0;j
< MAX_EFFECT_CHANNELS
;j
++)
1207 voice
->Send
[i
].Gains
[0].Target
[j
] = 0.0f
;
1211 const ALeffectslot
*Slot
= SendSlots
[i
];
1212 ComputePanningGains(Slot
->AmbiCoeffs
, Slot
->NumChannels
, coeffs
,
1213 WetGain
[i
], voice
->Send
[i
].Gains
[0].Target
);
1217 voice
->IsHrtf
= AL_FALSE
;
1221 ALfloat hfscale
= ALSource
->Direct
.HFReference
/ Frequency
;
1222 ALfloat lfscale
= ALSource
->Direct
.LFReference
/ Frequency
;
1223 DryGainHF
= maxf(DryGainHF
, 0.0001f
);
1224 DryGainLF
= maxf(DryGainLF
, 0.0001f
);
1225 voice
->Direct
.Filters
[0].ActiveType
= AF_None
;
1226 if(DryGainHF
!= 1.0f
) voice
->Direct
.Filters
[0].ActiveType
|= AF_LowPass
;
1227 if(DryGainLF
!= 1.0f
) voice
->Direct
.Filters
[0].ActiveType
|= AF_HighPass
;
1228 ALfilterState_setParams(
1229 &voice
->Direct
.Filters
[0].LowPass
, ALfilterType_HighShelf
,
1230 DryGainHF
, hfscale
, calc_rcpQ_from_slope(DryGainHF
, 0.75f
)
1232 ALfilterState_setParams(
1233 &voice
->Direct
.Filters
[0].HighPass
, ALfilterType_LowShelf
,
1234 DryGainLF
, lfscale
, calc_rcpQ_from_slope(DryGainLF
, 0.75f
)
1237 for(i
= 0;i
< NumSends
;i
++)
1239 ALfloat hfscale
= ALSource
->Send
[i
].HFReference
/ Frequency
;
1240 ALfloat lfscale
= ALSource
->Send
[i
].LFReference
/ Frequency
;
1241 WetGainHF
[i
] = maxf(WetGainHF
[i
], 0.0001f
);
1242 WetGainLF
[i
] = maxf(WetGainLF
[i
], 0.0001f
);
1243 voice
->Send
[i
].Filters
[0].ActiveType
= AF_None
;
1244 if(WetGainHF
[i
] != 1.0f
) voice
->Send
[i
].Filters
[0].ActiveType
|= AF_LowPass
;
1245 if(WetGainLF
[i
] != 1.0f
) voice
->Send
[i
].Filters
[0].ActiveType
|= AF_HighPass
;
1246 ALfilterState_setParams(
1247 &voice
->Send
[i
].Filters
[0].LowPass
, ALfilterType_HighShelf
,
1248 WetGainHF
[i
], hfscale
, calc_rcpQ_from_slope(WetGainHF
[i
], 0.75f
)
1250 ALfilterState_setParams(
1251 &voice
->Send
[i
].Filters
[0].HighPass
, ALfilterType_LowShelf
,
1252 WetGainLF
[i
], lfscale
, calc_rcpQ_from_slope(WetGainLF
[i
], 0.75f
)
1258 void UpdateContextSources(ALCcontext
*ctx
)
1260 ALvoice
*voice
, *voice_end
;
1263 if(ATOMIC_EXCHANGE(ALenum
, &ctx
->UpdateSources
, AL_FALSE
))
1265 CalcListenerParams(ctx
->Listener
);
1267 voice
= ctx
->Voices
;
1268 voice_end
= voice
+ ctx
->VoiceCount
;
1269 for(;voice
!= voice_end
;++voice
)
1271 if(!(source
=voice
->Source
)) continue;
1272 if(source
->state
!= AL_PLAYING
&& source
->state
!= AL_PAUSED
)
1273 voice
->Source
= NULL
;
1276 ATOMIC_STORE(&source
->NeedsUpdate
, AL_FALSE
);
1277 voice
->Update(voice
, source
, ctx
);
1283 voice
= ctx
->Voices
;
1284 voice_end
= voice
+ ctx
->VoiceCount
;
1285 for(;voice
!= voice_end
;++voice
)
1287 if(!(source
=voice
->Source
)) continue;
1288 if(source
->state
!= AL_PLAYING
&& source
->state
!= AL_PAUSED
)
1289 voice
->Source
= NULL
;
1290 else if(ATOMIC_EXCHANGE(ALenum
, &source
->NeedsUpdate
, AL_FALSE
))
1291 voice
->Update(voice
, source
, ctx
);
1297 /* Specialized function to clamp to [-1, +1] with only one branch. This also
1298 * converts NaN to 0. */
1299 static inline ALfloat
aluClampf(ALfloat val
)
1301 if(fabsf(val
) <= 1.0f
) return val
;
1302 return (ALfloat
)((0.0f
< val
) - (val
< 0.0f
));
1305 static inline ALfloat
aluF2F(ALfloat val
)
1308 static inline ALint
aluF2I(ALfloat val
)
1310 /* Floats only have a 24-bit mantissa, so [-16777215, +16777215] is the max
1311 * integer range normalized floats can be safely converted to.
1313 return fastf2i(aluClampf(val
)*16777215.0f
)<<7;
1315 static inline ALuint
aluF2UI(ALfloat val
)
1316 { return aluF2I(val
)+2147483648u; }
1318 static inline ALshort
aluF2S(ALfloat val
)
1319 { return fastf2i(aluClampf(val
)*32767.0f
); }
1320 static inline ALushort
aluF2US(ALfloat val
)
1321 { return aluF2S(val
)+32768; }
1323 static inline ALbyte
aluF2B(ALfloat val
)
1324 { return fastf2i(aluClampf(val
)*127.0f
); }
1325 static inline ALubyte
aluF2UB(ALfloat val
)
1326 { return aluF2B(val
)+128; }
1328 #define DECL_TEMPLATE(T, func) \
1329 static void Write_##T(ALfloatBUFFERSIZE *InBuffer, ALvoid *OutBuffer, \
1330 ALuint SamplesToDo, ALuint numchans) \
1333 for(j = 0;j < numchans;j++) \
1335 const ALfloat *in = InBuffer[j]; \
1336 T *restrict out = (T*)OutBuffer + j; \
1337 for(i = 0;i < SamplesToDo;i++) \
1338 out[i*numchans] = func(in[i]); \
1342 DECL_TEMPLATE(ALfloat
, aluF2F
)
1343 DECL_TEMPLATE(ALuint
, aluF2UI
)
1344 DECL_TEMPLATE(ALint
, aluF2I
)
1345 DECL_TEMPLATE(ALushort
, aluF2US
)
1346 DECL_TEMPLATE(ALshort
, aluF2S
)
1347 DECL_TEMPLATE(ALubyte
, aluF2UB
)
1348 DECL_TEMPLATE(ALbyte
, aluF2B
)
1350 #undef DECL_TEMPLATE
1353 ALvoid
aluMixData(ALCdevice
*device
, ALvoid
*buffer
, ALsizei size
)
1356 ALvoice
*voice
, *voice_end
;
1363 SetMixerFPUMode(&oldMode
);
1367 ALfloat (*OutBuffer
)[BUFFERSIZE
];
1370 IncrementRef(&device
->MixCount
);
1372 OutBuffer
= device
->DryBuffer
;
1373 OutChannels
= device
->NumChannels
;
1375 SamplesToDo
= minu(size
, BUFFERSIZE
);
1376 for(c
= 0;c
< OutChannels
;c
++)
1377 memset(OutBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1378 if(device
->Hrtf
|| device
->Uhj_Encoder
)
1380 /* Set OutBuffer/OutChannels to correspond to the actual output
1381 * with HRTF. Make sure to clear them too. */
1382 OutBuffer
+= OutChannels
;
1384 for(c
= 0;c
< OutChannels
;c
++)
1385 memset(OutBuffer
[c
], 0, SamplesToDo
*sizeof(ALfloat
));
1388 V0(device
->Backend
,lock
)();
1390 if((slot
=device
->DefaultSlot
) != NULL
)
1392 if(ATOMIC_EXCHANGE(ALenum
, &slot
->NeedsUpdate
, AL_FALSE
))
1393 V(slot
->EffectState
,update
)(device
, slot
);
1394 for(i
= 0;i
< slot
->NumChannels
;i
++)
1395 memset(slot
->WetBuffer
[i
], 0, SamplesToDo
*sizeof(ALfloat
));
1398 ctx
= ATOMIC_LOAD(&device
->ContextList
);
1401 if(!ctx
->DeferUpdates
)
1403 UpdateContextSources(ctx
);
1404 #define UPDATE_SLOT(iter) do { \
1405 if(ATOMIC_EXCHANGE(ALenum, &(*iter)->NeedsUpdate, AL_FALSE)) \
1406 V((*iter)->EffectState,update)(device, *iter); \
1407 for(i = 0;i < (*iter)->NumChannels;i++) \
1408 memset((*iter)->WetBuffer[i], 0, SamplesToDo*sizeof(ALfloat)); \
1410 VECTOR_FOR_EACH(ALeffectslot
*, ctx
->ActiveAuxSlots
, UPDATE_SLOT
);
1415 #define CLEAR_WET_BUFFER(iter) do { \
1416 for(i = 0;i < (*iter)->NumChannels;i++) \
1417 memset((*iter)->WetBuffer[i], 0, SamplesToDo*sizeof(ALfloat)); \
1419 VECTOR_FOR_EACH(ALeffectslot
*, ctx
->ActiveAuxSlots
, CLEAR_WET_BUFFER
);
1420 #undef CLEAR_WET_BUFFER
1423 /* source processing */
1424 voice
= ctx
->Voices
;
1425 voice_end
= voice
+ ctx
->VoiceCount
;
1426 for(;voice
!= voice_end
;++voice
)
1428 source
= voice
->Source
;
1429 if(source
&& source
->state
== AL_PLAYING
)
1430 MixSource(voice
, source
, device
, SamplesToDo
);
1433 /* effect slot processing */
1434 c
= VECTOR_SIZE(ctx
->ActiveAuxSlots
);
1435 for(i
= 0;i
< c
;i
++)
1437 const ALeffectslot
*slot
= VECTOR_ELEM(ctx
->ActiveAuxSlots
, i
);
1438 ALeffectState
*state
= slot
->EffectState
;
1439 V(state
,process
)(SamplesToDo
, slot
->WetBuffer
, device
->DryBuffer
,
1440 device
->NumChannels
);
1446 if(device
->DefaultSlot
!= NULL
)
1448 const ALeffectslot
*slot
= device
->DefaultSlot
;
1449 ALeffectState
*state
= slot
->EffectState
;
1450 V(state
,process
)(SamplesToDo
, slot
->WetBuffer
, device
->DryBuffer
,
1451 device
->NumChannels
);
1454 /* Increment the clock time. Every second's worth of samples is
1455 * converted and added to clock base so that large sample counts don't
1456 * overflow during conversion. This also guarantees an exact, stable
1458 device
->SamplesDone
+= SamplesToDo
;
1459 device
->ClockBase
+= (device
->SamplesDone
/device
->Frequency
) * DEVICE_CLOCK_RES
;
1460 device
->SamplesDone
%= device
->Frequency
;
1461 V0(device
->Backend
,unlock
)();
1465 HrtfMixerFunc HrtfMix
= SelectHrtfMixer();
1466 ALuint irsize
= GetHrtfIrSize(device
->Hrtf
);
1467 MixHrtfParams hrtfparams
;
1468 memset(&hrtfparams
, 0, sizeof(hrtfparams
));
1469 for(c
= 0;c
< device
->NumChannels
;c
++)
1471 hrtfparams
.Current
= &device
->Hrtf_Params
[c
];
1472 hrtfparams
.Target
= &device
->Hrtf_Params
[c
];
1473 HrtfMix(OutBuffer
, device
->DryBuffer
[c
], 0, device
->Hrtf_Offset
,
1474 0, irsize
, &hrtfparams
, &device
->Hrtf_State
[c
], SamplesToDo
1477 device
->Hrtf_Offset
+= SamplesToDo
;
1481 if(device
->Uhj_Encoder
)
1483 /* Encode to stereo-compatible 2-channel UHJ output. */
1484 EncodeUhj2(device
->Uhj_Encoder
, OutBuffer
, device
->DryBuffer
, SamplesToDo
);
1488 /* Apply binaural/crossfeed filter */
1489 for(i
= 0;i
< SamplesToDo
;i
++)
1492 samples
[0] = OutBuffer
[0][i
];
1493 samples
[1] = OutBuffer
[1][i
];
1494 bs2b_cross_feed(device
->Bs2b
, samples
);
1495 OutBuffer
[0][i
] = samples
[0];
1496 OutBuffer
[1][i
] = samples
[1];
1503 #define WRITE(T, a, b, c, d) do { \
1504 Write_##T((a), (b), (c), (d)); \
1505 buffer = (T*)buffer + (c)*(d); \
1507 switch(device
->FmtType
)
1510 WRITE(ALbyte
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1513 WRITE(ALubyte
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1516 WRITE(ALshort
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1519 WRITE(ALushort
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1522 WRITE(ALint
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1525 WRITE(ALuint
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1528 WRITE(ALfloat
, OutBuffer
, buffer
, SamplesToDo
, OutChannels
);
1534 size
-= SamplesToDo
;
1535 IncrementRef(&device
->MixCount
);
1538 RestoreFPUMode(&oldMode
);
1542 ALvoid
aluHandleDisconnect(ALCdevice
*device
)
1544 ALCcontext
*Context
;
1546 device
->Connected
= ALC_FALSE
;
1548 Context
= ATOMIC_LOAD(&device
->ContextList
);
1551 ALvoice
*voice
, *voice_end
;
1553 voice
= Context
->Voices
;
1554 voice_end
= voice
+ Context
->VoiceCount
;
1555 while(voice
!= voice_end
)
1557 ALsource
*source
= voice
->Source
;
1558 voice
->Source
= NULL
;
1560 if(source
&& source
->state
== AL_PLAYING
)
1562 source
->state
= AL_STOPPED
;
1563 ATOMIC_STORE(&source
->current_buffer
, NULL
);
1564 source
->position
= 0;
1565 source
->position_fraction
= 0;
1570 Context
->VoiceCount
= 0;
1572 Context
= Context
->next
;