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
43 #include "alnumbers.h"
44 #include "alnumeric.h"
48 #include "core/ambidefs.h"
49 #include "core/async_event.h"
50 #include "core/bformatdec.h"
51 #include "core/bs2b.h"
52 #include "core/bsinc_defs.h"
53 #include "core/bsinc_tables.h"
54 #include "core/bufferline.h"
55 #include "core/buffer_storage.h"
56 #include "core/context.h"
57 #include "core/cpu_caps.h"
58 #include "core/devformat.h"
59 #include "core/device.h"
60 #include "core/effects/base.h"
61 #include "core/effectslot.h"
62 #include "core/filters/biquad.h"
63 #include "core/filters/nfc.h"
64 #include "core/fpu_ctrl.h"
65 #include "core/hrtf.h"
66 #include "core/mastering.h"
67 #include "core/mixer.h"
68 #include "core/mixer/defs.h"
69 #include "core/mixer/hrtfdefs.h"
70 #include "core/resampler_limits.h"
71 #include "core/uhjfilter.h"
72 #include "core/voice.h"
73 #include "core/voice_change.h"
74 #include "intrusive_ptr.h"
75 #include "opthelpers.h"
76 #include "ringbuffer.h"
102 static_assert(!(MaxResamplerPadding
&1), "MaxResamplerPadding is not a multiple of two");
107 using uint
= unsigned int;
109 constexpr uint MaxPitch
{10};
111 static_assert((BufferLineSize
-1)/MaxPitch
> 0, "MaxPitch is too large for BufferLineSize!");
112 static_assert((INT_MAX
>>MixerFracBits
)/MaxPitch
> BufferLineSize
,
113 "MaxPitch and/or BufferLineSize are too large for MixerFracBits!");
115 using namespace std::placeholders
;
117 float InitConeScale()
120 if(auto optval
= al::getenv("__ALSOFT_HALF_ANGLE_CONES"))
122 if(al::strcasecmp(optval
->c_str(), "true") == 0
123 || strtol(optval
->c_str(), nullptr, 0) == 1)
129 const float ConeScale
{InitConeScale()};
131 /* Localized scalars for mono sources (initialized in aluInit, after
132 * configuration is loaded).
145 using HrtfDirectMixerFunc
= void(*)(const FloatBufferSpan LeftOut
, const FloatBufferSpan RightOut
,
146 const al::span
<const FloatBufferLine
> InSamples
, float2
*AccumSamples
, float *TempBuf
,
147 HrtfChannelState
*ChanState
, const size_t IrSize
, const size_t BufferSize
);
149 HrtfDirectMixerFunc MixDirectHrtf
{MixDirectHrtf_
<CTag
>};
151 inline HrtfDirectMixerFunc
SelectHrtfMixer(void)
154 if((CPUCapFlags
&CPU_CAP_NEON
))
155 return MixDirectHrtf_
<NEONTag
>;
158 if((CPUCapFlags
&CPU_CAP_SSE
))
159 return MixDirectHrtf_
<SSETag
>;
162 return MixDirectHrtf_
<CTag
>;
166 inline void BsincPrepare(const uint increment
, BsincState
*state
, const BSincTable
*table
)
168 size_t si
{BSincScaleCount
- 1};
171 if(increment
> MixerFracOne
)
173 sf
= MixerFracOne
/static_cast<float>(increment
) - table
->scaleBase
;
174 sf
= maxf(0.0f
, BSincScaleCount
*sf
*table
->scaleRange
- 1.0f
);
176 /* The interpolation factor is fit to this diagonally-symmetric curve
177 * to reduce the transition ripple caused by interpolating different
178 * scales of the sinc function.
180 sf
= 1.0f
- std::cos(std::asin(sf
- static_cast<float>(si
)));
184 state
->m
= table
->m
[si
];
185 state
->l
= (state
->m
/2) - 1;
186 state
->filter
= table
->Tab
+ table
->filterOffset
[si
];
189 inline ResamplerFunc
SelectResampler(Resampler resampler
, uint increment
)
193 case Resampler::Point
:
194 return Resample_
<PointTag
,CTag
>;
195 case Resampler::Linear
:
197 if((CPUCapFlags
&CPU_CAP_NEON
))
198 return Resample_
<LerpTag
,NEONTag
>;
201 if((CPUCapFlags
&CPU_CAP_SSE4_1
))
202 return Resample_
<LerpTag
,SSE4Tag
>;
205 if((CPUCapFlags
&CPU_CAP_SSE2
))
206 return Resample_
<LerpTag
,SSE2Tag
>;
208 return Resample_
<LerpTag
,CTag
>;
209 case Resampler::Cubic
:
210 return Resample_
<CubicTag
,CTag
>;
211 case Resampler::BSinc12
:
212 case Resampler::BSinc24
:
213 if(increment
> MixerFracOne
)
216 if((CPUCapFlags
&CPU_CAP_NEON
))
217 return Resample_
<BSincTag
,NEONTag
>;
220 if((CPUCapFlags
&CPU_CAP_SSE
))
221 return Resample_
<BSincTag
,SSETag
>;
223 return Resample_
<BSincTag
,CTag
>;
226 case Resampler::FastBSinc12
:
227 case Resampler::FastBSinc24
:
229 if((CPUCapFlags
&CPU_CAP_NEON
))
230 return Resample_
<FastBSincTag
,NEONTag
>;
233 if((CPUCapFlags
&CPU_CAP_SSE
))
234 return Resample_
<FastBSincTag
,SSETag
>;
236 return Resample_
<FastBSincTag
,CTag
>;
239 return Resample_
<PointTag
,CTag
>;
244 void aluInit(CompatFlagBitset flags
)
246 MixDirectHrtf
= SelectHrtfMixer();
247 XScale
= flags
.test(CompatFlags::ReverseX
) ? -1.0f
: 1.0f
;
248 YScale
= flags
.test(CompatFlags::ReverseY
) ? -1.0f
: 1.0f
;
249 ZScale
= flags
.test(CompatFlags::ReverseZ
) ? -1.0f
: 1.0f
;
253 ResamplerFunc
PrepareResampler(Resampler resampler
, uint increment
, InterpState
*state
)
257 case Resampler::Point
:
258 case Resampler::Linear
:
259 case Resampler::Cubic
:
261 case Resampler::FastBSinc12
:
262 case Resampler::BSinc12
:
263 BsincPrepare(increment
, &state
->bsinc
, &bsinc12
);
265 case Resampler::FastBSinc24
:
266 case Resampler::BSinc24
:
267 BsincPrepare(increment
, &state
->bsinc
, &bsinc24
);
270 return SelectResampler(resampler
, increment
);
274 void DeviceBase::ProcessHrtf(const size_t SamplesToDo
)
276 /* HRTF is stereo output only. */
277 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
278 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
280 MixDirectHrtf(RealOut
.Buffer
[lidx
], RealOut
.Buffer
[ridx
], Dry
.Buffer
, HrtfAccumData
,
281 mHrtfState
->mTemp
.data(), mHrtfState
->mChannels
.data(), mHrtfState
->mIrSize
, SamplesToDo
);
284 void DeviceBase::ProcessAmbiDec(const size_t SamplesToDo
)
286 AmbiDecoder
->process(RealOut
.Buffer
, Dry
.Buffer
.data(), SamplesToDo
);
289 void DeviceBase::ProcessAmbiDecStablized(const size_t SamplesToDo
)
291 /* Decode with front image stablization. */
292 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
293 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
294 const uint cidx
{RealOut
.ChannelIndex
[FrontCenter
]};
296 AmbiDecoder
->processStablize(RealOut
.Buffer
, Dry
.Buffer
.data(), lidx
, ridx
, cidx
,
300 void DeviceBase::ProcessUhj(const size_t SamplesToDo
)
302 /* UHJ is stereo output only. */
303 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
304 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
306 /* Encode to stereo-compatible 2-channel UHJ output. */
307 mUhjEncoder
->encode(RealOut
.Buffer
[lidx
].data(), RealOut
.Buffer
[ridx
].data(),
308 {{Dry
.Buffer
[0].data(), Dry
.Buffer
[1].data(), Dry
.Buffer
[2].data()}}, SamplesToDo
);
311 void DeviceBase::ProcessBs2b(const size_t SamplesToDo
)
313 /* First, decode the ambisonic mix to the "real" output. */
314 AmbiDecoder
->process(RealOut
.Buffer
, Dry
.Buffer
.data(), SamplesToDo
);
316 /* BS2B is stereo output only. */
317 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
318 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
320 /* Now apply the BS2B binaural/crossfeed filter. */
321 bs2b_cross_feed(Bs2b
.get(), RealOut
.Buffer
[lidx
].data(), RealOut
.Buffer
[ridx
].data(),
328 using AmbiRotateMatrix
= std::array
<std::array
<float,MaxAmbiChannels
>,MaxAmbiChannels
>;
330 /* This RNG method was created based on the math found in opusdec. It's quick,
331 * and starting with a seed value of 22222, is suitable for generating
334 inline uint
dither_rng(uint
*seed
) noexcept
336 *seed
= (*seed
* 96314165) + 907633515;
341 inline auto& GetAmbiScales(AmbiScaling scaletype
) noexcept
345 case AmbiScaling::FuMa
: return AmbiScale::FromFuMa();
346 case AmbiScaling::SN3D
: return AmbiScale::FromSN3D();
347 case AmbiScaling::UHJ
: return AmbiScale::FromUHJ();
348 case AmbiScaling::N3D
: break;
350 return AmbiScale::FromN3D();
353 inline auto& GetAmbiLayout(AmbiLayout layouttype
) noexcept
355 if(layouttype
== AmbiLayout::FuMa
) return AmbiIndex::FromFuMa();
356 return AmbiIndex::FromACN();
359 inline auto& GetAmbi2DLayout(AmbiLayout layouttype
) noexcept
361 if(layouttype
== AmbiLayout::FuMa
) return AmbiIndex::FromFuMa2D();
362 return AmbiIndex::FromACN2D();
366 bool CalcContextParams(ContextBase
*ctx
)
368 ContextProps
*props
{ctx
->mParams
.ContextUpdate
.exchange(nullptr, std::memory_order_acq_rel
)};
369 if(!props
) return false;
371 const alu::Vector pos
{props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
};
372 ctx
->mParams
.Position
= pos
;
375 alu::Vector N
{props
->OrientAt
[0], props
->OrientAt
[1], props
->OrientAt
[2], 0.0f
};
377 alu::Vector V
{props
->OrientUp
[0], props
->OrientUp
[1], props
->OrientUp
[2], 0.0f
};
379 /* Build and normalize right-vector */
380 alu::Vector U
{N
.cross_product(V
)};
383 const alu::Matrix rot
{
384 U
[0], V
[0], -N
[0], 0.0,
385 U
[1], V
[1], -N
[1], 0.0,
386 U
[2], V
[2], -N
[2], 0.0,
388 const alu::Vector vel
{props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0};
390 ctx
->mParams
.Matrix
= rot
;
391 ctx
->mParams
.Velocity
= rot
* vel
;
393 ctx
->mParams
.Gain
= props
->Gain
* ctx
->mGainBoost
;
394 ctx
->mParams
.MetersPerUnit
= props
->MetersPerUnit
;
395 ctx
->mParams
.AirAbsorptionGainHF
= props
->AirAbsorptionGainHF
;
397 ctx
->mParams
.DopplerFactor
= props
->DopplerFactor
;
398 ctx
->mParams
.SpeedOfSound
= props
->SpeedOfSound
* props
->DopplerVelocity
;
400 ctx
->mParams
.SourceDistanceModel
= props
->SourceDistanceModel
;
401 ctx
->mParams
.mDistanceModel
= props
->mDistanceModel
;
403 AtomicReplaceHead(ctx
->mFreeContextProps
, props
);
407 bool CalcEffectSlotParams(EffectSlot
*slot
, EffectSlot
**sorted_slots
, ContextBase
*context
)
409 EffectSlotProps
*props
{slot
->Update
.exchange(nullptr, std::memory_order_acq_rel
)};
410 if(!props
) return false;
412 /* If the effect slot target changed, clear the first sorted entry to force
415 if(slot
->Target
!= props
->Target
)
416 *sorted_slots
= nullptr;
417 slot
->Gain
= props
->Gain
;
418 slot
->AuxSendAuto
= props
->AuxSendAuto
;
419 slot
->Target
= props
->Target
;
420 slot
->EffectType
= props
->Type
;
421 slot
->mEffectProps
= props
->Props
;
422 if(props
->Type
== EffectSlotType::Reverb
|| props
->Type
== EffectSlotType::EAXReverb
)
424 slot
->RoomRolloff
= props
->Props
.Reverb
.RoomRolloffFactor
;
425 slot
->DecayTime
= props
->Props
.Reverb
.DecayTime
;
426 slot
->DecayLFRatio
= props
->Props
.Reverb
.DecayLFRatio
;
427 slot
->DecayHFRatio
= props
->Props
.Reverb
.DecayHFRatio
;
428 slot
->DecayHFLimit
= props
->Props
.Reverb
.DecayHFLimit
;
429 slot
->AirAbsorptionGainHF
= props
->Props
.Reverb
.AirAbsorptionGainHF
;
433 slot
->RoomRolloff
= 0.0f
;
434 slot
->DecayTime
= 0.0f
;
435 slot
->DecayLFRatio
= 0.0f
;
436 slot
->DecayHFRatio
= 0.0f
;
437 slot
->DecayHFLimit
= false;
438 slot
->AirAbsorptionGainHF
= 1.0f
;
441 EffectState
*state
{props
->State
.release()};
442 EffectState
*oldstate
{slot
->mEffectState
};
443 slot
->mEffectState
= state
;
445 /* Only release the old state if it won't get deleted, since we can't be
446 * deleting/freeing anything in the mixer.
448 if(!oldstate
->releaseIfNoDelete())
450 /* Otherwise, if it would be deleted send it off with a release event. */
451 RingBuffer
*ring
{context
->mAsyncEvents
.get()};
452 auto evt_vec
= ring
->getWriteVector();
453 if LIKELY(evt_vec
.first
.len
> 0)
455 AsyncEvent
*evt
{al::construct_at(reinterpret_cast<AsyncEvent
*>(evt_vec
.first
.buf
),
456 AsyncEvent::ReleaseEffectState
)};
457 evt
->u
.mEffectState
= oldstate
;
458 ring
->writeAdvance(1);
462 /* If writing the event failed, the queue was probably full. Store
463 * the old state in the property object where it can eventually be
464 * cleaned up sometime later (not ideal, but better than blocking
467 props
->State
.reset(oldstate
);
471 AtomicReplaceHead(context
->mFreeEffectslotProps
, props
);
474 if(EffectSlot
*target
{slot
->Target
})
475 output
= EffectTarget
{&target
->Wet
, nullptr};
478 DeviceBase
*device
{context
->mDevice
};
479 output
= EffectTarget
{&device
->Dry
, &device
->RealOut
};
481 state
->update(context
, slot
, &slot
->mEffectProps
, output
);
486 /* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
489 inline float ScaleAzimuthFront(float azimuth
, float scale
)
491 const float abs_azi
{std::fabs(azimuth
)};
492 if(!(abs_azi
>= al::numbers::pi_v
<float>*0.5f
))
493 return std::copysign(minf(abs_azi
*scale
, al::numbers::pi_v
<float>*0.5f
), azimuth
);
497 /* Wraps the given value in radians to stay between [-pi,+pi] */
498 inline float WrapRadians(float r
)
500 static constexpr float Pi
{al::numbers::pi_v
<float>};
501 static constexpr float Pi2
{Pi
*2.0f
};
502 if(r
> Pi
) return std::fmod(Pi
+r
, Pi2
) - Pi
;
503 if(r
< -Pi
) return Pi
- std::fmod(Pi
-r
, Pi2
);
507 /* Begin ambisonic rotation helpers.
509 * Rotating first-order B-Format just needs a straight-forward X/Y/Z rotation
510 * matrix. Higher orders, however, are more complicated. The method implemented
511 * here is a recursive algorithm (the rotation for first-order is used to help
512 * generate the second-order rotation, which helps generate the third-order
516 * <https://github.com/polarch/Spherical-Harmonic-Transform/blob/master/getSHrotMtx.m>,
517 * provided under the BSD 3-Clause license.
519 * Copyright (c) 2015, Archontis Politis
520 * Copyright (c) 2019, Christopher Robinson
522 * The u, v, and w coefficients used for generating higher-order rotations are
523 * precomputed since they're constant. The second-order coefficients are
524 * followed by the third-order coefficients, etc.
526 struct RotatorCoeffs
{
529 template<size_t N0
, size_t N1
>
530 static std::array
<RotatorCoeffs
,N0
+N1
> ConcatArrays(const std::array
<RotatorCoeffs
,N0
> &lhs
,
531 const std::array
<RotatorCoeffs
,N1
> &rhs
)
533 std::array
<RotatorCoeffs
,N0
+N1
> ret
;
534 auto iter
= std::copy(lhs
.cbegin(), lhs
.cend(), ret
.begin());
535 std::copy(rhs
.cbegin(), rhs
.cend(), iter
);
539 template<int l
, int num_elems
=l
*2+1>
540 static std::array
<RotatorCoeffs
,num_elems
*num_elems
> GenCoeffs()
542 std::array
<RotatorCoeffs
,num_elems
*num_elems
> ret
{};
543 auto coeffs
= ret
.begin();
545 for(int m
{-l
};m
<= l
;++m
)
547 for(int n
{-l
};n
<= l
;++n
)
549 // compute u,v,w terms of Eq.8.1 (Table I)
550 const bool d
{m
== 0}; // the delta function d_m0
551 const float denom
{static_cast<float>((std::abs(n
) == l
) ?
552 (2*l
) * (2*l
- 1) : (l
*l
- n
*n
))};
554 const int abs_m
{std::abs(m
)};
555 coeffs
->u
= std::sqrt(static_cast<float>(l
*l
- m
*m
)/denom
);
556 coeffs
->v
= std::sqrt(static_cast<float>(l
+abs_m
-1) * static_cast<float>(l
+abs_m
) /
557 denom
) * (1.0f
+d
) * (1.0f
- 2.0f
*d
) * 0.5f
;
558 coeffs
->w
= std::sqrt(static_cast<float>(l
-abs_m
-1) * static_cast<float>(l
-abs_m
) /
559 denom
) * (1.0f
-d
) * -0.5f
;
567 const auto RotatorCoeffArray
= RotatorCoeffs::ConcatArrays(RotatorCoeffs::GenCoeffs
<2>(),
568 RotatorCoeffs::GenCoeffs
<3>());
571 * Given the matrix, pre-filled with the (zeroth- and) first-order rotation
572 * coefficients, this fills in the coefficients for the higher orders up to and
573 * including the given order. The matrix is in ACN layout.
575 void AmbiRotator(AmbiRotateMatrix
&matrix
, const int order
)
577 /* Don't do anything for < 2nd order. */
578 if(order
< 2) return;
580 auto P
= [](const int i
, const int l
, const int a
, const int n
, const size_t last_band
,
581 const AmbiRotateMatrix
&R
)
583 const float ri1
{ R
[static_cast<uint
>(i
+2)][ 1+2]};
584 const float rim1
{R
[static_cast<uint
>(i
+2)][-1+2]};
585 const float ri0
{ R
[static_cast<uint
>(i
+2)][ 0+2]};
587 auto vec
= R
[static_cast<uint
>(a
+l
-1) + last_band
].cbegin() + last_band
;
589 return ri1
*vec
[0] + rim1
*vec
[static_cast<uint
>(l
-1)*size_t{2}];
591 return ri1
*vec
[static_cast<uint
>(l
-1)*size_t{2}] - rim1
*vec
[0];
592 return ri0
*vec
[static_cast<uint
>(n
+l
-1)];
595 auto U
= [P
](const int l
, const int m
, const int n
, const size_t last_band
,
596 const AmbiRotateMatrix
&R
)
598 return P(0, l
, m
, n
, last_band
, R
);
600 auto V
= [P
](const int l
, const int m
, const int n
, const size_t last_band
,
601 const AmbiRotateMatrix
&R
)
603 using namespace al::numbers
;
606 const bool d
{m
== 1};
607 const float p0
{P( 1, l
, m
-1, n
, last_band
, R
)};
608 const float p1
{P(-1, l
, -m
+1, n
, last_band
, R
)};
609 return d
? p0
*sqrt2_v
<float> : (p0
- p1
);
611 const bool d
{m
== -1};
612 const float p0
{P( 1, l
, m
+1, n
, last_band
, R
)};
613 const float p1
{P(-1, l
, -m
-1, n
, last_band
, R
)};
614 return d
? p1
*sqrt2_v
<float> : (p0
+ p1
);
616 auto W
= [P
](const int l
, const int m
, const int n
, const size_t last_band
,
617 const AmbiRotateMatrix
&R
)
622 const float p0
{P( 1, l
, m
+1, n
, last_band
, R
)};
623 const float p1
{P(-1, l
, -m
-1, n
, last_band
, R
)};
626 const float p0
{P( 1, l
, m
-1, n
, last_band
, R
)};
627 const float p1
{P(-1, l
, -m
+1, n
, last_band
, R
)};
631 // compute rotation matrix of each subsequent band recursively
632 auto coeffs
= RotatorCoeffArray
.cbegin();
633 size_t band_idx
{4}, last_band
{1};
634 for(int l
{2};l
<= order
;++l
)
637 for(int m
{-l
};m
<= l
;++m
,++y
)
640 for(int n
{-l
};n
<= l
;++n
,++x
)
645 const float u
{coeffs
->u
};
646 if(u
!= 0.0f
) r
+= u
* U(l
, m
, n
, last_band
, matrix
);
647 const float v
{coeffs
->v
};
648 if(v
!= 0.0f
) r
+= v
* V(l
, m
, n
, last_band
, matrix
);
649 const float w
{coeffs
->w
};
650 if(w
!= 0.0f
) r
+= w
* W(l
, m
, n
, last_band
, matrix
);
656 last_band
= band_idx
;
657 band_idx
+= static_cast<uint
>(l
)*size_t{2} + 1;
660 /* End ambisonic rotation helpers. */
663 constexpr float Deg2Rad(float x
) noexcept
664 { return static_cast<float>(al::numbers::pi
/ 180.0 * x
); }
666 struct GainTriplet
{ float Base
, HF
, LF
; };
668 void CalcPanningAndFilters(Voice
*voice
, const float xpos
, const float ypos
, const float zpos
,
669 const float Distance
, const float Spread
, const GainTriplet
&DryGain
,
670 const al::span
<const GainTriplet
,MAX_SENDS
> WetGain
, EffectSlot
*(&SendSlots
)[MAX_SENDS
],
671 const VoiceProps
*props
, const ContextParams
&Context
, const DeviceBase
*Device
)
673 static constexpr ChanMap MonoMap
[1]{
674 { FrontCenter
, 0.0f
, 0.0f
}
676 { BackLeft
, Deg2Rad(-150.0f
), Deg2Rad(0.0f
) },
677 { BackRight
, Deg2Rad( 150.0f
), Deg2Rad(0.0f
) }
679 { FrontLeft
, Deg2Rad( -45.0f
), Deg2Rad(0.0f
) },
680 { FrontRight
, Deg2Rad( 45.0f
), Deg2Rad(0.0f
) },
681 { BackLeft
, Deg2Rad(-135.0f
), Deg2Rad(0.0f
) },
682 { BackRight
, Deg2Rad( 135.0f
), Deg2Rad(0.0f
) }
684 { FrontLeft
, Deg2Rad( -30.0f
), Deg2Rad(0.0f
) },
685 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) },
686 { FrontCenter
, Deg2Rad( 0.0f
), Deg2Rad(0.0f
) },
688 { SideLeft
, Deg2Rad(-110.0f
), Deg2Rad(0.0f
) },
689 { SideRight
, Deg2Rad( 110.0f
), Deg2Rad(0.0f
) }
691 { FrontLeft
, Deg2Rad(-30.0f
), Deg2Rad(0.0f
) },
692 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) },
693 { FrontCenter
, Deg2Rad( 0.0f
), Deg2Rad(0.0f
) },
695 { BackCenter
, Deg2Rad(180.0f
), Deg2Rad(0.0f
) },
696 { SideLeft
, Deg2Rad(-90.0f
), Deg2Rad(0.0f
) },
697 { SideRight
, Deg2Rad( 90.0f
), Deg2Rad(0.0f
) }
699 { FrontLeft
, Deg2Rad( -30.0f
), Deg2Rad(0.0f
) },
700 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) },
701 { FrontCenter
, Deg2Rad( 0.0f
), Deg2Rad(0.0f
) },
703 { BackLeft
, Deg2Rad(-150.0f
), Deg2Rad(0.0f
) },
704 { BackRight
, Deg2Rad( 150.0f
), Deg2Rad(0.0f
) },
705 { SideLeft
, Deg2Rad( -90.0f
), Deg2Rad(0.0f
) },
706 { SideRight
, Deg2Rad( 90.0f
), Deg2Rad(0.0f
) }
709 ChanMap StereoMap
[2]{
710 { FrontLeft
, Deg2Rad(-30.0f
), Deg2Rad(0.0f
) },
711 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) }
714 const auto Frequency
= static_cast<float>(Device
->Frequency
);
715 const uint NumSends
{Device
->NumAuxSends
};
717 const size_t num_channels
{voice
->mChans
.size()};
718 ASSUME(num_channels
> 0);
720 for(auto &chandata
: voice
->mChans
)
722 chandata
.mDryParams
.Hrtf
.Target
= HrtfFilter
{};
723 chandata
.mDryParams
.Gains
.Target
.fill(0.0f
);
724 std::for_each(chandata
.mWetParams
.begin(), chandata
.mWetParams
.begin()+NumSends
,
725 [](SendParams
¶ms
) -> void { params
.Gains
.Target
.fill(0.0f
); });
728 DirectMode DirectChannels
{props
->DirectChannels
};
729 const ChanMap
*chans
{nullptr};
730 switch(voice
->mFmtChannels
)
734 /* Mono buffers are never played direct. */
735 DirectChannels
= DirectMode::Off
;
739 if(DirectChannels
== DirectMode::Off
)
741 /* Convert counter-clockwise to clock-wise, and wrap between
744 StereoMap
[0].angle
= WrapRadians(-props
->StereoPan
[0]);
745 StereoMap
[1].angle
= WrapRadians(-props
->StereoPan
[1]);
750 case FmtRear
: chans
= RearMap
; break;
751 case FmtQuad
: chans
= QuadMap
; break;
752 case FmtX51
: chans
= X51Map
; break;
753 case FmtX61
: chans
= X61Map
; break;
754 case FmtX71
: chans
= X71Map
; break;
762 DirectChannels
= DirectMode::Off
;
766 voice
->mFlags
.reset(VoiceHasHrtf
).reset(VoiceHasNfc
);
767 if(auto *decoder
{voice
->mDecoder
.get()})
768 decoder
->setWidth(minf(props
->EnhWidth
, 0.7f
));
770 if(IsAmbisonic(voice
->mFmtChannels
))
772 /* Special handling for B-Format and UHJ sources. */
774 if(Device
->AvgSpeakerDist
> 0.0f
&& voice
->mFmtChannels
!= FmtUHJ2
775 && voice
->mFmtChannels
!= FmtSuperStereo
)
777 if(!(Distance
> std::numeric_limits
<float>::epsilon()))
779 /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
780 * is what we want for FOA input. The first channel may have
781 * been previously re-adjusted if panned, so reset it.
783 voice
->mChans
[0].mDryParams
.NFCtrlFilter
.adjust(0.0f
);
787 /* Clamp the distance for really close sources, to prevent
790 const float mdist
{maxf(Distance
, Device
->AvgSpeakerDist
/4.0f
)};
791 const float w0
{SpeedOfSoundMetersPerSec
/ (mdist
* Frequency
)};
793 /* Only need to adjust the first channel of a B-Format source. */
794 voice
->mChans
[0].mDryParams
.NFCtrlFilter
.adjust(w0
);
797 voice
->mFlags
.set(VoiceHasNfc
);
800 /* Panning a B-Format sound toward some direction is easy. Just pan the
801 * first (W) channel as a normal mono sound. The angular spread is used
802 * as a directional scalar to blend between full coverage and full
805 const float coverage
{!(Distance
> std::numeric_limits
<float>::epsilon()) ? 1.0f
:
806 (al::numbers::inv_pi_v
<float>/2.0f
* Spread
)};
808 auto calc_coeffs
= [xpos
,ypos
,zpos
](RenderMode mode
)
810 if(mode
!= RenderMode::Pairwise
)
811 return CalcDirectionCoeffs({xpos
, ypos
, zpos
}, 0.0f
);
813 /* Clamp Y, in case rounding errors caused it to end up outside
816 const float ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
817 /* Negate Z for right-handed coords with -Z in front. */
818 const float az
{std::atan2(xpos
, -zpos
)};
820 /* A scalar of 1.5 for plain stereo results in +/-60 degrees
821 * being moved to +/-90 degrees for direct right and left
824 return CalcAngleCoeffs(ScaleAzimuthFront(az
, 1.5f
), ev
, 0.0f
);
826 auto coeffs
= calc_coeffs(Device
->mRenderMode
);
827 std::transform(coeffs
.begin()+1, coeffs
.end(), coeffs
.begin()+1,
828 std::bind(std::multiplies
<float>{}, _1
, 1.0f
-coverage
));
830 /* NOTE: W needs to be scaled according to channel scaling. */
831 auto&& scales
= GetAmbiScales(voice
->mAmbiScaling
);
832 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
*scales
[0],
833 voice
->mChans
[0].mDryParams
.Gains
.Target
);
834 for(uint i
{0};i
< NumSends
;i
++)
836 if(const EffectSlot
*Slot
{SendSlots
[i
]})
837 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
*scales
[0],
838 voice
->mChans
[0].mWetParams
[i
].Gains
.Target
);
843 /* Local B-Format sources have their XYZ channels rotated according
844 * to the orientation.
847 alu::Vector N
{props
->OrientAt
[0], props
->OrientAt
[1], props
->OrientAt
[2], 0.0f
};
849 alu::Vector V
{props
->OrientUp
[0], props
->OrientUp
[1], props
->OrientUp
[2], 0.0f
};
851 if(!props
->HeadRelative
)
853 N
= Context
.Matrix
* N
;
854 V
= Context
.Matrix
* V
;
856 /* Build and normalize right-vector */
857 alu::Vector U
{N
.cross_product(V
)};
860 /* Build a rotation matrix. Manually fill the zeroth- and first-
861 * order elements, then construct the rotation for the higher
864 AmbiRotateMatrix shrot
{};
866 shrot
[1][1] = U
[0]; shrot
[1][2] = -V
[0]; shrot
[1][3] = -N
[0];
867 shrot
[2][1] = -U
[1]; shrot
[2][2] = V
[1]; shrot
[2][3] = N
[1];
868 shrot
[3][1] = U
[2]; shrot
[3][2] = -V
[2]; shrot
[3][3] = -N
[2];
869 AmbiRotator(shrot
, static_cast<int>(minu(voice
->mAmbiOrder
, Device
->mAmbiOrder
)));
871 /* Convert the rotation matrix for input ordering and scaling, and
872 * whether input is 2D or 3D.
874 const uint8_t *index_map
{Is2DAmbisonic(voice
->mFmtChannels
) ?
875 GetAmbi2DLayout(voice
->mAmbiLayout
).data() :
876 GetAmbiLayout(voice
->mAmbiLayout
).data()};
878 static const uint8_t ChansPerOrder
[MaxAmbiOrder
+1]{1, 3, 5, 7,};
879 static const uint8_t OrderOffset
[MaxAmbiOrder
+1]{0, 1, 4, 9,};
880 for(size_t c
{1};c
< num_channels
;c
++)
882 const size_t acn
{index_map
[c
]};
883 const size_t order
{AmbiIndex::OrderFromChannel()[acn
]};
884 const size_t tocopy
{ChansPerOrder
[order
]};
885 const size_t offset
{OrderOffset
[order
]};
886 const float scale
{scales
[acn
] * coverage
};
887 auto in
= shrot
.cbegin() + offset
;
889 coeffs
= std::array
<float,MaxAmbiChannels
>{};
890 for(size_t x
{0};x
< tocopy
;++x
)
891 coeffs
[offset
+x
] = in
[x
][acn
] * scale
;
893 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
894 voice
->mChans
[c
].mDryParams
.Gains
.Target
);
896 for(uint i
{0};i
< NumSends
;i
++)
898 if(const EffectSlot
*Slot
{SendSlots
[i
]})
899 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
900 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
905 else if(DirectChannels
!= DirectMode::Off
&& !Device
->RealOut
.RemixMap
.empty())
907 /* Direct source channels always play local. Skip the virtual channels
908 * and write inputs to the matching real outputs.
910 voice
->mDirect
.Buffer
= Device
->RealOut
.Buffer
;
912 for(size_t c
{0};c
< num_channels
;c
++)
914 uint idx
{GetChannelIdxByName(Device
->RealOut
, chans
[c
].channel
)};
915 if(idx
!= INVALID_CHANNEL_INDEX
)
916 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
;
917 else if(DirectChannels
== DirectMode::RemixMismatch
)
919 auto match_channel
= [chans
,c
](const InputRemixMap
&map
) noexcept
-> bool
920 { return chans
[c
].channel
== map
.channel
; };
921 auto remap
= std::find_if(Device
->RealOut
.RemixMap
.cbegin(),
922 Device
->RealOut
.RemixMap
.cend(), match_channel
);
923 if(remap
!= Device
->RealOut
.RemixMap
.cend())
925 for(const auto &target
: remap
->targets
)
927 idx
= GetChannelIdxByName(Device
->RealOut
, target
.channel
);
928 if(idx
!= INVALID_CHANNEL_INDEX
)
929 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
*
936 /* Auxiliary sends still use normal channel panning since they mix to
937 * B-Format, which can't channel-match.
939 for(size_t c
{0};c
< num_channels
;c
++)
941 const auto coeffs
= CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, 0.0f
);
943 for(uint i
{0};i
< NumSends
;i
++)
945 if(const EffectSlot
*Slot
{SendSlots
[i
]})
946 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
947 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
951 else if(Device
->mRenderMode
== RenderMode::Hrtf
)
953 /* Full HRTF rendering. Skip the virtual channels and render to the
956 voice
->mDirect
.Buffer
= Device
->RealOut
.Buffer
;
958 if(Distance
> std::numeric_limits
<float>::epsilon())
960 const float ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
961 const float az
{std::atan2(xpos
, -zpos
)};
963 /* Get the HRIR coefficients and delays just once, for the given
966 GetHrtfCoeffs(Device
->mHrtf
.get(), ev
, az
, Distance
, Spread
,
967 voice
->mChans
[0].mDryParams
.Hrtf
.Target
.Coeffs
,
968 voice
->mChans
[0].mDryParams
.Hrtf
.Target
.Delay
);
969 voice
->mChans
[0].mDryParams
.Hrtf
.Target
.Gain
= DryGain
.Base
;
971 /* Remaining channels use the same results as the first. */
972 for(size_t c
{1};c
< num_channels
;c
++)
975 if(chans
[c
].channel
== LFE
) continue;
976 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
= voice
->mChans
[0].mDryParams
.Hrtf
.Target
;
979 /* Calculate the directional coefficients once, which apply to all
980 * input channels of the source sends.
982 const auto coeffs
= CalcDirectionCoeffs({xpos
, ypos
, zpos
}, Spread
);
984 for(size_t c
{0};c
< num_channels
;c
++)
987 if(chans
[c
].channel
== LFE
)
989 for(uint i
{0};i
< NumSends
;i
++)
991 if(const EffectSlot
*Slot
{SendSlots
[i
]})
992 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
993 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
999 /* Local sources on HRTF play with each channel panned to its
1000 * relative location around the listener, providing "virtual
1001 * speaker" responses.
1003 for(size_t c
{0};c
< num_channels
;c
++)
1006 if(chans
[c
].channel
== LFE
)
1009 /* Get the HRIR coefficients and delays for this channel
1012 GetHrtfCoeffs(Device
->mHrtf
.get(), chans
[c
].elevation
, chans
[c
].angle
,
1013 std::numeric_limits
<float>::infinity(), Spread
,
1014 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Coeffs
,
1015 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Delay
);
1016 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Gain
= DryGain
.Base
;
1018 /* Normal panning for auxiliary sends. */
1019 const auto coeffs
= CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, Spread
);
1021 for(uint i
{0};i
< NumSends
;i
++)
1023 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1024 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1025 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1030 voice
->mFlags
.set(VoiceHasHrtf
);
1034 /* Non-HRTF rendering. Use normal panning to the output. */
1036 if(Distance
> std::numeric_limits
<float>::epsilon())
1038 /* Calculate NFC filter coefficient if needed. */
1039 if(Device
->AvgSpeakerDist
> 0.0f
)
1041 /* Clamp the distance for really close sources, to prevent
1044 const float mdist
{maxf(Distance
, Device
->AvgSpeakerDist
/4.0f
)};
1045 const float w0
{SpeedOfSoundMetersPerSec
/ (mdist
* Frequency
)};
1047 /* Adjust NFC filters. */
1048 for(size_t c
{0};c
< num_channels
;c
++)
1049 voice
->mChans
[c
].mDryParams
.NFCtrlFilter
.adjust(w0
);
1051 voice
->mFlags
.set(VoiceHasNfc
);
1054 /* Calculate the directional coefficients once, which apply to all
1057 auto calc_coeffs
= [xpos
,ypos
,zpos
,Spread
](RenderMode mode
)
1059 if(mode
!= RenderMode::Pairwise
)
1060 return CalcDirectionCoeffs({xpos
, ypos
, zpos
}, Spread
);
1061 const float ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
1062 const float az
{std::atan2(xpos
, -zpos
)};
1063 return CalcAngleCoeffs(ScaleAzimuthFront(az
, 1.5f
), ev
, Spread
);
1065 const auto coeffs
= calc_coeffs(Device
->mRenderMode
);
1067 for(size_t c
{0};c
< num_channels
;c
++)
1069 /* Special-case LFE */
1070 if(chans
[c
].channel
== LFE
)
1072 if(Device
->Dry
.Buffer
.data() == Device
->RealOut
.Buffer
.data())
1074 const uint idx
{GetChannelIdxByName(Device
->RealOut
, chans
[c
].channel
)};
1075 if(idx
!= INVALID_CHANNEL_INDEX
)
1076 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
;
1081 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
1082 voice
->mChans
[c
].mDryParams
.Gains
.Target
);
1083 for(uint i
{0};i
< NumSends
;i
++)
1085 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1086 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1087 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1093 if(Device
->AvgSpeakerDist
> 0.0f
)
1095 /* If the source distance is 0, simulate a plane-wave by using
1096 * infinite distance, which results in a w0 of 0.
1098 static constexpr float w0
{0.0f
};
1099 for(size_t c
{0};c
< num_channels
;c
++)
1100 voice
->mChans
[c
].mDryParams
.NFCtrlFilter
.adjust(w0
);
1102 voice
->mFlags
.set(VoiceHasNfc
);
1105 for(size_t c
{0};c
< num_channels
;c
++)
1107 /* Special-case LFE */
1108 if(chans
[c
].channel
== LFE
)
1110 if(Device
->Dry
.Buffer
.data() == Device
->RealOut
.Buffer
.data())
1112 const uint idx
{GetChannelIdxByName(Device
->RealOut
, chans
[c
].channel
)};
1113 if(idx
!= INVALID_CHANNEL_INDEX
)
1114 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
;
1119 const auto coeffs
= CalcAngleCoeffs((Device
->mRenderMode
== RenderMode::Pairwise
)
1120 ? ScaleAzimuthFront(chans
[c
].angle
, 3.0f
) : chans
[c
].angle
,
1121 chans
[c
].elevation
, Spread
);
1123 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
1124 voice
->mChans
[c
].mDryParams
.Gains
.Target
);
1125 for(uint i
{0};i
< NumSends
;i
++)
1127 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1128 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1129 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1136 const float hfNorm
{props
->Direct
.HFReference
/ Frequency
};
1137 const float lfNorm
{props
->Direct
.LFReference
/ Frequency
};
1139 voice
->mDirect
.FilterType
= AF_None
;
1140 if(DryGain
.HF
!= 1.0f
) voice
->mDirect
.FilterType
|= AF_LowPass
;
1141 if(DryGain
.LF
!= 1.0f
) voice
->mDirect
.FilterType
|= AF_HighPass
;
1143 auto &lowpass
= voice
->mChans
[0].mDryParams
.LowPass
;
1144 auto &highpass
= voice
->mChans
[0].mDryParams
.HighPass
;
1145 lowpass
.setParamsFromSlope(BiquadType::HighShelf
, hfNorm
, DryGain
.HF
, 1.0f
);
1146 highpass
.setParamsFromSlope(BiquadType::LowShelf
, lfNorm
, DryGain
.LF
, 1.0f
);
1147 for(size_t c
{1};c
< num_channels
;c
++)
1149 voice
->mChans
[c
].mDryParams
.LowPass
.copyParamsFrom(lowpass
);
1150 voice
->mChans
[c
].mDryParams
.HighPass
.copyParamsFrom(highpass
);
1153 for(uint i
{0};i
< NumSends
;i
++)
1155 const float hfNorm
{props
->Send
[i
].HFReference
/ Frequency
};
1156 const float lfNorm
{props
->Send
[i
].LFReference
/ Frequency
};
1158 voice
->mSend
[i
].FilterType
= AF_None
;
1159 if(WetGain
[i
].HF
!= 1.0f
) voice
->mSend
[i
].FilterType
|= AF_LowPass
;
1160 if(WetGain
[i
].LF
!= 1.0f
) voice
->mSend
[i
].FilterType
|= AF_HighPass
;
1162 auto &lowpass
= voice
->mChans
[0].mWetParams
[i
].LowPass
;
1163 auto &highpass
= voice
->mChans
[0].mWetParams
[i
].HighPass
;
1164 lowpass
.setParamsFromSlope(BiquadType::HighShelf
, hfNorm
, WetGain
[i
].HF
, 1.0f
);
1165 highpass
.setParamsFromSlope(BiquadType::LowShelf
, lfNorm
, WetGain
[i
].LF
, 1.0f
);
1166 for(size_t c
{1};c
< num_channels
;c
++)
1168 voice
->mChans
[c
].mWetParams
[i
].LowPass
.copyParamsFrom(lowpass
);
1169 voice
->mChans
[c
].mWetParams
[i
].HighPass
.copyParamsFrom(highpass
);
1174 void CalcNonAttnSourceParams(Voice
*voice
, const VoiceProps
*props
, const ContextBase
*context
)
1176 const DeviceBase
*Device
{context
->mDevice
};
1177 EffectSlot
*SendSlots
[MAX_SENDS
];
1179 voice
->mDirect
.Buffer
= Device
->Dry
.Buffer
;
1180 for(uint i
{0};i
< Device
->NumAuxSends
;i
++)
1182 SendSlots
[i
] = props
->Send
[i
].Slot
;
1183 if(!SendSlots
[i
] || SendSlots
[i
]->EffectType
== EffectSlotType::None
)
1185 SendSlots
[i
] = nullptr;
1186 voice
->mSend
[i
].Buffer
= {};
1189 voice
->mSend
[i
].Buffer
= SendSlots
[i
]->Wet
.Buffer
;
1192 /* Calculate the stepping value */
1193 const auto Pitch
= static_cast<float>(voice
->mFrequency
) /
1194 static_cast<float>(Device
->Frequency
) * props
->Pitch
;
1195 if(Pitch
> float{MaxPitch
})
1196 voice
->mStep
= MaxPitch
<<MixerFracBits
;
1198 voice
->mStep
= maxu(fastf2u(Pitch
* MixerFracOne
), 1);
1199 voice
->mResampler
= PrepareResampler(props
->mResampler
, voice
->mStep
, &voice
->mResampleState
);
1201 /* Calculate gains */
1202 GainTriplet DryGain
;
1203 DryGain
.Base
= minf(clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
) * props
->Direct
.Gain
*
1204 context
->mParams
.Gain
, GainMixMax
);
1205 DryGain
.HF
= props
->Direct
.GainHF
;
1206 DryGain
.LF
= props
->Direct
.GainLF
;
1207 GainTriplet WetGain
[MAX_SENDS
];
1208 for(uint i
{0};i
< Device
->NumAuxSends
;i
++)
1210 WetGain
[i
].Base
= minf(clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
) *
1211 props
->Send
[i
].Gain
* context
->mParams
.Gain
, GainMixMax
);
1212 WetGain
[i
].HF
= props
->Send
[i
].GainHF
;
1213 WetGain
[i
].LF
= props
->Send
[i
].GainLF
;
1216 CalcPanningAndFilters(voice
, 0.0f
, 0.0f
, -1.0f
, 0.0f
, 0.0f
, DryGain
, WetGain
, SendSlots
, props
,
1217 context
->mParams
, Device
);
1220 void CalcAttnSourceParams(Voice
*voice
, const VoiceProps
*props
, const ContextBase
*context
)
1222 const DeviceBase
*Device
{context
->mDevice
};
1223 const uint NumSends
{Device
->NumAuxSends
};
1225 /* Set mixing buffers and get send parameters. */
1226 voice
->mDirect
.Buffer
= Device
->Dry
.Buffer
;
1227 EffectSlot
*SendSlots
[MAX_SENDS
];
1228 uint UseDryAttnForRoom
{0};
1229 for(uint i
{0};i
< NumSends
;i
++)
1231 SendSlots
[i
] = props
->Send
[i
].Slot
;
1232 if(!SendSlots
[i
] || SendSlots
[i
]->EffectType
== EffectSlotType::None
)
1233 SendSlots
[i
] = nullptr;
1234 else if(!SendSlots
[i
]->AuxSendAuto
)
1236 /* If the slot's auxiliary send auto is off, the data sent to the
1237 * effect slot is the same as the dry path, sans filter effects.
1239 UseDryAttnForRoom
|= 1u<<i
;
1243 voice
->mSend
[i
].Buffer
= {};
1245 voice
->mSend
[i
].Buffer
= SendSlots
[i
]->Wet
.Buffer
;
1248 /* Transform source to listener space (convert to head relative) */
1249 alu::Vector Position
{props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
};
1250 alu::Vector Velocity
{props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
};
1251 alu::Vector Direction
{props
->Direction
[0], props
->Direction
[1], props
->Direction
[2], 0.0f
};
1252 if(!props
->HeadRelative
)
1254 /* Transform source vectors */
1255 Position
= context
->mParams
.Matrix
* (Position
- context
->mParams
.Position
);
1256 Velocity
= context
->mParams
.Matrix
* Velocity
;
1257 Direction
= context
->mParams
.Matrix
* Direction
;
1261 /* Offset the source velocity to be relative of the listener velocity */
1262 Velocity
+= context
->mParams
.Velocity
;
1265 const bool directional
{Direction
.normalize() > 0.0f
};
1266 alu::Vector ToSource
{Position
[0], Position
[1], Position
[2], 0.0f
};
1267 const float Distance
{ToSource
.normalize()};
1269 /* Calculate distance attenuation */
1270 float ClampedDist
{Distance
};
1271 float DryGainBase
{props
->Gain
};
1272 float WetGainBase
{props
->Gain
};
1274 switch(context
->mParams
.SourceDistanceModel
? props
->mDistanceModel
1275 : context
->mParams
.mDistanceModel
)
1277 case DistanceModel::InverseClamped
:
1278 if(props
->MaxDistance
< props
->RefDistance
) break;
1279 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1281 case DistanceModel::Inverse
:
1282 if(props
->RefDistance
> 0.0f
)
1284 float dist
{lerpf(props
->RefDistance
, ClampedDist
, props
->RolloffFactor
)};
1285 if(dist
> 0.0f
) DryGainBase
*= props
->RefDistance
/ dist
;
1287 dist
= lerpf(props
->RefDistance
, ClampedDist
, props
->RoomRolloffFactor
);
1288 if(dist
> 0.0f
) WetGainBase
*= props
->RefDistance
/ dist
;
1292 case DistanceModel::LinearClamped
:
1293 if(props
->MaxDistance
< props
->RefDistance
) break;
1294 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1296 case DistanceModel::Linear
:
1297 if(props
->MaxDistance
!= props
->RefDistance
)
1299 float attn
{(ClampedDist
-props
->RefDistance
) /
1300 (props
->MaxDistance
-props
->RefDistance
) * props
->RolloffFactor
};
1301 DryGainBase
*= maxf(1.0f
- attn
, 0.0f
);
1303 attn
= (ClampedDist
-props
->RefDistance
) /
1304 (props
->MaxDistance
-props
->RefDistance
) * props
->RoomRolloffFactor
;
1305 WetGainBase
*= maxf(1.0f
- attn
, 0.0f
);
1309 case DistanceModel::ExponentClamped
:
1310 if(props
->MaxDistance
< props
->RefDistance
) break;
1311 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1313 case DistanceModel::Exponent
:
1314 if(ClampedDist
> 0.0f
&& props
->RefDistance
> 0.0f
)
1316 const float dist_ratio
{ClampedDist
/props
->RefDistance
};
1317 DryGainBase
*= std::pow(dist_ratio
, -props
->RolloffFactor
);
1318 WetGainBase
*= std::pow(dist_ratio
, -props
->RoomRolloffFactor
);
1322 case DistanceModel::Disable
:
1326 /* Calculate directional soundcones */
1327 float ConeHF
{1.0f
}, WetConeHF
{1.0f
};
1328 if(directional
&& props
->InnerAngle
< 360.0f
)
1330 static constexpr float Rad2Deg
{static_cast<float>(180.0 / al::numbers::pi
)};
1331 const float Angle
{Rad2Deg
*2.0f
* std::acos(-Direction
.dot_product(ToSource
)) * ConeScale
};
1333 float ConeGain
{1.0f
};
1334 if(Angle
>= props
->OuterAngle
)
1336 ConeGain
= props
->OuterGain
;
1337 ConeHF
= lerpf(1.0f
, props
->OuterGainHF
, props
->DryGainHFAuto
);
1339 else if(Angle
>= props
->InnerAngle
)
1341 const float scale
{(Angle
-props
->InnerAngle
) / (props
->OuterAngle
-props
->InnerAngle
)};
1342 ConeGain
= lerpf(1.0f
, props
->OuterGain
, scale
);
1343 ConeHF
= lerpf(1.0f
, props
->OuterGainHF
, scale
* props
->DryGainHFAuto
);
1346 DryGainBase
*= ConeGain
;
1347 WetGainBase
*= lerpf(1.0f
, ConeGain
, props
->WetGainAuto
);
1349 WetConeHF
= lerpf(1.0f
, ConeHF
, props
->WetGainHFAuto
);
1352 /* Apply gain and frequency filters */
1353 DryGainBase
= clampf(DryGainBase
, props
->MinGain
, props
->MaxGain
) * context
->mParams
.Gain
;
1354 WetGainBase
= clampf(WetGainBase
, props
->MinGain
, props
->MaxGain
) * context
->mParams
.Gain
;
1356 GainTriplet DryGain
{};
1357 DryGain
.Base
= minf(DryGainBase
* props
->Direct
.Gain
, GainMixMax
);
1358 DryGain
.HF
= ConeHF
* props
->Direct
.GainHF
;
1359 DryGain
.LF
= props
->Direct
.GainLF
;
1360 GainTriplet WetGain
[MAX_SENDS
]{};
1361 for(uint i
{0};i
< NumSends
;i
++)
1363 /* If this effect slot's Auxiliary Send Auto is off, then use the dry
1364 * path distance and cone attenuation, otherwise use the wet (room)
1365 * path distance and cone attenuation. The send filter is used instead
1366 * of the direct filter, regardless.
1368 const bool use_room
{!(UseDryAttnForRoom
&(1u<<i
))};
1369 const float gain
{use_room
? WetGainBase
: DryGainBase
};
1370 WetGain
[i
].Base
= minf(gain
* props
->Send
[i
].Gain
, GainMixMax
);
1371 WetGain
[i
].HF
= (use_room
? WetConeHF
: ConeHF
) * props
->Send
[i
].GainHF
;
1372 WetGain
[i
].LF
= props
->Send
[i
].GainLF
;
1375 /* Distance-based air absorption and initial send decay. */
1376 if(likely(Distance
> props
->RefDistance
))
1378 const float distance_base
{(Distance
-props
->RefDistance
) * props
->RolloffFactor
};
1379 const float absorption
{distance_base
* context
->mParams
.MetersPerUnit
*
1380 props
->AirAbsorptionFactor
};
1381 if(absorption
> std::numeric_limits
<float>::epsilon())
1383 const float hfattn
{std::pow(context
->mParams
.AirAbsorptionGainHF
, absorption
)};
1384 DryGain
.HF
*= hfattn
;
1385 for(uint i
{0u};i
< NumSends
;++i
)
1386 WetGain
[i
].HF
*= hfattn
;
1389 /* If the source's Auxiliary Send Filter Gain Auto is off, no extra
1390 * adjustment is applied to the send gains.
1392 for(uint i
{props
->WetGainAuto
? 0u : NumSends
};i
< NumSends
;++i
)
1397 auto calc_attenuation
= [](float distance
, float refdist
, float rolloff
) noexcept
1399 const float dist
{lerpf(refdist
, distance
, rolloff
)};
1400 if(dist
> refdist
) return refdist
/ dist
;
1404 /* The reverb effect's room rolloff factor always applies to an
1405 * inverse distance rolloff model.
1407 WetGain
[i
].Base
*= calc_attenuation(Distance
, props
->RefDistance
,
1408 SendSlots
[i
]->RoomRolloff
);
1410 /* If this effect slot's Auxiliary Send Auto is off, don't apply
1411 * the automatic initial reverb decay (should the reverb's room
1412 * rolloff still apply?).
1414 if(!SendSlots
[i
]->AuxSendAuto
)
1417 GainTriplet DecayDistance
;
1418 /* Calculate the distances to where this effect's decay reaches
1421 DecayDistance
.Base
= SendSlots
[i
]->DecayTime
* SpeedOfSoundMetersPerSec
;
1422 DecayDistance
.LF
= DecayDistance
.Base
* SendSlots
[i
]->DecayLFRatio
;
1423 DecayDistance
.HF
= DecayDistance
.Base
* SendSlots
[i
]->DecayHFRatio
;
1424 if(SendSlots
[i
]->DecayHFLimit
)
1426 const float airAbsorption
{SendSlots
[i
]->AirAbsorptionGainHF
};
1427 if(airAbsorption
< 1.0f
)
1429 /* Calculate the distance to where this effect's air
1430 * absorption reaches -60dB, and limit the effect's HF
1431 * decay distance (so it doesn't take any longer to decay
1432 * than the air would allow).
1434 static constexpr float log10_decaygain
{-3.0f
/*std::log10(ReverbDecayGain)*/};
1435 const float absorb_dist
{log10_decaygain
/ std::log10(airAbsorption
)};
1436 DecayDistance
.HF
= minf(absorb_dist
, DecayDistance
.HF
);
1440 const float baseAttn
= calc_attenuation(Distance
, props
->RefDistance
,
1441 props
->RolloffFactor
);
1443 /* Apply a decay-time transformation to the wet path, based on the
1444 * source distance. The initial decay of the reverb effect is
1445 * calculated and applied to the wet path.
1447 const float fact
{distance_base
/ DecayDistance
.Base
};
1448 const float gain
{std::pow(ReverbDecayGain
, fact
)*(1.0f
-baseAttn
) + baseAttn
};
1449 WetGain
[i
].Base
*= gain
;
1453 const float hffact
{distance_base
/ DecayDistance
.HF
};
1454 const float gainhf
{std::pow(ReverbDecayGain
, hffact
)*(1.0f
-baseAttn
) + baseAttn
};
1455 WetGain
[i
].HF
*= minf(gainhf
/gain
, 1.0f
);
1456 const float lffact
{distance_base
/ DecayDistance
.LF
};
1457 const float gainlf
{std::pow(ReverbDecayGain
, lffact
)*(1.0f
-baseAttn
) + baseAttn
};
1458 WetGain
[i
].LF
*= minf(gainlf
/gain
, 1.0f
);
1464 /* Initial source pitch */
1465 float Pitch
{props
->Pitch
};
1467 /* Calculate velocity-based doppler effect */
1468 float DopplerFactor
{props
->DopplerFactor
* context
->mParams
.DopplerFactor
};
1469 if(DopplerFactor
> 0.0f
)
1471 const alu::Vector
&lvelocity
= context
->mParams
.Velocity
;
1472 float vss
{Velocity
.dot_product(ToSource
) * -DopplerFactor
};
1473 float vls
{lvelocity
.dot_product(ToSource
) * -DopplerFactor
};
1475 const float SpeedOfSound
{context
->mParams
.SpeedOfSound
};
1476 if(!(vls
< SpeedOfSound
))
1478 /* Listener moving away from the source at the speed of sound.
1479 * Sound waves can't catch it.
1483 else if(!(vss
< SpeedOfSound
))
1485 /* Source moving toward the listener at the speed of sound. Sound
1486 * waves bunch up to extreme frequencies.
1488 Pitch
= std::numeric_limits
<float>::infinity();
1492 /* Source and listener movement is nominal. Calculate the proper
1495 Pitch
*= (SpeedOfSound
-vls
) / (SpeedOfSound
-vss
);
1499 /* Adjust pitch based on the buffer and output frequencies, and calculate
1500 * fixed-point stepping value.
1502 Pitch
*= static_cast<float>(voice
->mFrequency
) / static_cast<float>(Device
->Frequency
);
1503 if(Pitch
> float{MaxPitch
})
1504 voice
->mStep
= MaxPitch
<<MixerFracBits
;
1506 voice
->mStep
= maxu(fastf2u(Pitch
* MixerFracOne
), 1);
1507 voice
->mResampler
= PrepareResampler(props
->mResampler
, voice
->mStep
, &voice
->mResampleState
);
1510 if(props
->Radius
> Distance
)
1511 spread
= al::numbers::pi_v
<float>*2.0f
- Distance
/props
->Radius
*al::numbers::pi_v
<float>;
1512 else if(Distance
> 0.0f
)
1513 spread
= std::asin(props
->Radius
/Distance
) * 2.0f
;
1515 CalcPanningAndFilters(voice
, ToSource
[0]*XScale
, ToSource
[1]*YScale
, ToSource
[2]*ZScale
,
1516 Distance
*context
->mParams
.MetersPerUnit
, spread
, DryGain
, WetGain
, SendSlots
, props
,
1517 context
->mParams
, Device
);
1520 void CalcSourceParams(Voice
*voice
, ContextBase
*context
, bool force
)
1522 VoicePropsItem
*props
{voice
->mUpdate
.exchange(nullptr, std::memory_order_acq_rel
)};
1523 if(!props
&& !force
) return;
1527 voice
->mProps
= *props
;
1529 AtomicReplaceHead(context
->mFreeVoiceProps
, props
);
1532 if((voice
->mProps
.DirectChannels
!= DirectMode::Off
&& voice
->mFmtChannels
!= FmtMono
1533 && !IsAmbisonic(voice
->mFmtChannels
))
1534 || voice
->mProps
.mSpatializeMode
== SpatializeMode::Off
1535 || (voice
->mProps
.mSpatializeMode
==SpatializeMode::Auto
&& voice
->mFmtChannels
!= FmtMono
))
1536 CalcNonAttnSourceParams(voice
, &voice
->mProps
, context
);
1538 CalcAttnSourceParams(voice
, &voice
->mProps
, context
);
1542 void SendSourceStateEvent(ContextBase
*context
, uint id
, VChangeState state
)
1544 RingBuffer
*ring
{context
->mAsyncEvents
.get()};
1545 auto evt_vec
= ring
->getWriteVector();
1546 if(evt_vec
.first
.len
< 1) return;
1548 AsyncEvent
*evt
{al::construct_at(reinterpret_cast<AsyncEvent
*>(evt_vec
.first
.buf
),
1549 AsyncEvent::SourceStateChange
)};
1550 evt
->u
.srcstate
.id
= id
;
1553 case VChangeState::Reset
:
1554 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Reset
;
1556 case VChangeState::Stop
:
1557 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Stop
;
1559 case VChangeState::Play
:
1560 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Play
;
1562 case VChangeState::Pause
:
1563 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Pause
;
1565 /* Shouldn't happen. */
1566 case VChangeState::Restart
:
1570 ring
->writeAdvance(1);
1573 void ProcessVoiceChanges(ContextBase
*ctx
)
1575 VoiceChange
*cur
{ctx
->mCurrentVoiceChange
.load(std::memory_order_acquire
)};
1576 VoiceChange
*next
{cur
->mNext
.load(std::memory_order_acquire
)};
1579 const uint enabledevt
{ctx
->mEnabledEvts
.load(std::memory_order_acquire
)};
1583 bool sendevt
{false};
1584 if(cur
->mState
== VChangeState::Reset
|| cur
->mState
== VChangeState::Stop
)
1586 if(Voice
*voice
{cur
->mVoice
})
1588 voice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
1589 voice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
1590 /* A source ID indicates the voice was playing or paused, which
1591 * gets a reset/stop event.
1593 sendevt
= voice
->mSourceID
.exchange(0u, std::memory_order_relaxed
) != 0u;
1594 Voice::State oldvstate
{Voice::Playing
};
1595 voice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1596 std::memory_order_relaxed
, std::memory_order_acquire
);
1597 voice
->mPendingChange
.store(false, std::memory_order_release
);
1599 /* Reset state change events are always sent, even if the voice is
1600 * already stopped or even if there is no voice.
1602 sendevt
|= (cur
->mState
== VChangeState::Reset
);
1604 else if(cur
->mState
== VChangeState::Pause
)
1606 Voice
*voice
{cur
->mVoice
};
1607 Voice::State oldvstate
{Voice::Playing
};
1608 sendevt
= voice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1609 std::memory_order_release
, std::memory_order_acquire
);
1611 else if(cur
->mState
== VChangeState::Play
)
1613 /* NOTE: When playing a voice, sending a source state change event
1614 * depends if there's an old voice to stop and if that stop is
1615 * successful. If there is no old voice, a playing event is always
1616 * sent. If there is an old voice, an event is sent only if the
1617 * voice is already stopped.
1619 if(Voice
*oldvoice
{cur
->mOldVoice
})
1621 oldvoice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
1622 oldvoice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
1623 oldvoice
->mSourceID
.store(0u, std::memory_order_relaxed
);
1624 Voice::State oldvstate
{Voice::Playing
};
1625 sendevt
= !oldvoice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1626 std::memory_order_relaxed
, std::memory_order_acquire
);
1627 oldvoice
->mPendingChange
.store(false, std::memory_order_release
);
1632 Voice
*voice
{cur
->mVoice
};
1633 voice
->mPlayState
.store(Voice::Playing
, std::memory_order_release
);
1635 else if(cur
->mState
== VChangeState::Restart
)
1637 /* Restarting a voice never sends a source change event. */
1638 Voice
*oldvoice
{cur
->mOldVoice
};
1639 oldvoice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
1640 oldvoice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
1641 /* If there's no sourceID, the old voice finished so don't start
1642 * the new one at its new offset.
1644 if(oldvoice
->mSourceID
.exchange(0u, std::memory_order_relaxed
) != 0u)
1646 /* Otherwise, set the voice to stopping if it's not already (it
1647 * might already be, if paused), and play the new voice as
1650 Voice::State oldvstate
{Voice::Playing
};
1651 oldvoice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1652 std::memory_order_relaxed
, std::memory_order_acquire
);
1654 Voice
*voice
{cur
->mVoice
};
1655 voice
->mPlayState
.store((oldvstate
== Voice::Playing
) ? Voice::Playing
1656 : Voice::Stopped
, std::memory_order_release
);
1658 oldvoice
->mPendingChange
.store(false, std::memory_order_release
);
1660 if(sendevt
&& (enabledevt
&AsyncEvent::SourceStateChange
))
1661 SendSourceStateEvent(ctx
, cur
->mSourceID
, cur
->mState
);
1663 next
= cur
->mNext
.load(std::memory_order_acquire
);
1665 ctx
->mCurrentVoiceChange
.store(cur
, std::memory_order_release
);
1668 void ProcessParamUpdates(ContextBase
*ctx
, const EffectSlotArray
&slots
,
1669 const al::span
<Voice
*> voices
)
1671 ProcessVoiceChanges(ctx
);
1673 IncrementRef(ctx
->mUpdateCount
);
1674 if LIKELY(!ctx
->mHoldUpdates
.load(std::memory_order_acquire
))
1676 bool force
{CalcContextParams(ctx
)};
1677 auto sorted_slots
= const_cast<EffectSlot
**>(slots
.data() + slots
.size());
1678 for(EffectSlot
*slot
: slots
)
1679 force
|= CalcEffectSlotParams(slot
, sorted_slots
, ctx
);
1681 for(Voice
*voice
: voices
)
1683 /* Only update voices that have a source. */
1684 if(voice
->mSourceID
.load(std::memory_order_relaxed
) != 0)
1685 CalcSourceParams(voice
, ctx
, force
);
1688 IncrementRef(ctx
->mUpdateCount
);
1691 void ProcessContexts(DeviceBase
*device
, const uint SamplesToDo
)
1693 ASSUME(SamplesToDo
> 0);
1695 for(ContextBase
*ctx
: *device
->mContexts
.load(std::memory_order_acquire
))
1697 const EffectSlotArray
&auxslots
= *ctx
->mActiveAuxSlots
.load(std::memory_order_acquire
);
1698 const al::span
<Voice
*> voices
{ctx
->getVoicesSpanAcquired()};
1700 /* Process pending propery updates for objects on the context. */
1701 ProcessParamUpdates(ctx
, auxslots
, voices
);
1703 /* Clear auxiliary effect slot mixing buffers. */
1704 for(EffectSlot
*slot
: auxslots
)
1706 for(auto &buffer
: slot
->Wet
.Buffer
)
1710 /* Process voices that have a playing source. */
1711 for(Voice
*voice
: voices
)
1713 const Voice::State vstate
{voice
->mPlayState
.load(std::memory_order_acquire
)};
1714 if(vstate
!= Voice::Stopped
&& vstate
!= Voice::Pending
)
1715 voice
->mix(vstate
, ctx
, SamplesToDo
);
1718 /* Process effects. */
1719 if(const size_t num_slots
{auxslots
.size()})
1721 auto slots
= auxslots
.data();
1722 auto slots_end
= slots
+ num_slots
;
1724 /* Sort the slots into extra storage, so that effect slots come
1725 * before their effect slot target (or their targets' target).
1727 const al::span
<EffectSlot
*> sorted_slots
{const_cast<EffectSlot
**>(slots_end
),
1729 /* Skip sorting if it has already been done. */
1730 if(!sorted_slots
[0])
1732 /* First, copy the slots to the sorted list, then partition the
1733 * sorted list so that all slots without a target slot go to
1736 std::copy(slots
, slots_end
, sorted_slots
.begin());
1737 auto split_point
= std::partition(sorted_slots
.begin(), sorted_slots
.end(),
1738 [](const EffectSlot
*slot
) noexcept
-> bool
1739 { return slot
->Target
!= nullptr; });
1740 /* There must be at least one slot without a slot target. */
1741 assert(split_point
!= sorted_slots
.end());
1743 /* Simple case: no more than 1 slot has a target slot. Either
1744 * all slots go right to the output, or the remaining one must
1745 * target an already-partitioned slot.
1747 if(split_point
- sorted_slots
.begin() > 1)
1749 /* At least two slots target other slots. Starting from the
1750 * back of the sorted list, continue partitioning the front
1751 * of the list given each target until all targets are
1752 * accounted for. This ensures all slots without a target
1753 * go last, all slots directly targeting those last slots
1754 * go second-to-last, all slots directly targeting those
1755 * second-last slots go third-to-last, etc.
1757 auto next_target
= sorted_slots
.end();
1759 /* This shouldn't happen, but if there's unsorted slots
1760 * left that don't target any sorted slots, they can't
1761 * contribute to the output, so leave them.
1763 if UNLIKELY(next_target
== split_point
)
1767 split_point
= std::partition(sorted_slots
.begin(), split_point
,
1768 [next_target
](const EffectSlot
*slot
) noexcept
-> bool
1769 { return slot
->Target
!= *next_target
; });
1770 } while(split_point
- sorted_slots
.begin() > 1);
1774 for(const EffectSlot
*slot
: sorted_slots
)
1776 EffectState
*state
{slot
->mEffectState
};
1777 state
->process(SamplesToDo
, slot
->Wet
.Buffer
, state
->mOutTarget
);
1781 /* Signal the event handler if there are any events to read. */
1782 RingBuffer
*ring
{ctx
->mAsyncEvents
.get()};
1783 if(ring
->readSpace() > 0)
1784 ctx
->mEventSem
.post();
1789 void ApplyDistanceComp(const al::span
<FloatBufferLine
> Samples
, const size_t SamplesToDo
,
1790 const DistanceComp::ChanData
*distcomp
)
1792 ASSUME(SamplesToDo
> 0);
1794 for(auto &chanbuffer
: Samples
)
1796 const float gain
{distcomp
->Gain
};
1797 const size_t base
{distcomp
->Length
};
1798 float *distbuf
{al::assume_aligned
<16>(distcomp
->Buffer
)};
1804 float *inout
{al::assume_aligned
<16>(chanbuffer
.data())};
1805 auto inout_end
= inout
+ SamplesToDo
;
1806 if LIKELY(SamplesToDo
>= base
)
1808 auto delay_end
= std::rotate(inout
, inout_end
- base
, inout_end
);
1809 std::swap_ranges(inout
, delay_end
, distbuf
);
1813 auto delay_start
= std::swap_ranges(inout
, inout_end
, distbuf
);
1814 std::rotate(distbuf
, delay_start
, distbuf
+ base
);
1816 std::transform(inout
, inout_end
, inout
, std::bind(std::multiplies
<float>{}, _1
, gain
));
1820 void ApplyDither(const al::span
<FloatBufferLine
> Samples
, uint
*dither_seed
,
1821 const float quant_scale
, const size_t SamplesToDo
)
1823 ASSUME(SamplesToDo
> 0);
1825 /* Dithering. Generate whitenoise (uniform distribution of random values
1826 * between -1 and +1) and add it to the sample values, after scaling up to
1827 * the desired quantization depth amd before rounding.
1829 const float invscale
{1.0f
/ quant_scale
};
1830 uint seed
{*dither_seed
};
1831 auto dither_sample
= [&seed
,invscale
,quant_scale
](const float sample
) noexcept
-> float
1833 float val
{sample
* quant_scale
};
1834 uint rng0
{dither_rng(&seed
)};
1835 uint rng1
{dither_rng(&seed
)};
1836 val
+= static_cast<float>(rng0
*(1.0/UINT_MAX
) - rng1
*(1.0/UINT_MAX
));
1837 return fast_roundf(val
) * invscale
;
1839 for(FloatBufferLine
&inout
: Samples
)
1840 std::transform(inout
.begin(), inout
.begin()+SamplesToDo
, inout
.begin(), dither_sample
);
1841 *dither_seed
= seed
;
1845 /* Base template left undefined. Should be marked =delete, but Clang 3.8.1
1846 * chokes on that given the inline specializations.
1848 template<typename T
>
1849 inline T
SampleConv(float) noexcept
;
1851 template<> inline float SampleConv(float val
) noexcept
1853 template<> inline int32_t SampleConv(float val
) noexcept
1855 /* Floats have a 23-bit mantissa, plus an implied 1 bit and a sign bit.
1856 * This means a normalized float has at most 25 bits of signed precision.
1857 * When scaling and clamping for a signed 32-bit integer, these following
1858 * values are the best a float can give.
1860 return fastf2i(clampf(val
*2147483648.0f
, -2147483648.0f
, 2147483520.0f
));
1862 template<> inline int16_t SampleConv(float val
) noexcept
1863 { return static_cast<int16_t>(fastf2i(clampf(val
*32768.0f
, -32768.0f
, 32767.0f
))); }
1864 template<> inline int8_t SampleConv(float val
) noexcept
1865 { return static_cast<int8_t>(fastf2i(clampf(val
*128.0f
, -128.0f
, 127.0f
))); }
1867 /* Define unsigned output variations. */
1868 template<> inline uint32_t SampleConv(float val
) noexcept
1869 { return static_cast<uint32_t>(SampleConv
<int32_t>(val
)) + 2147483648u; }
1870 template<> inline uint16_t SampleConv(float val
) noexcept
1871 { return static_cast<uint16_t>(SampleConv
<int16_t>(val
) + 32768); }
1872 template<> inline uint8_t SampleConv(float val
) noexcept
1873 { return static_cast<uint8_t>(SampleConv
<int8_t>(val
) + 128); }
1875 template<DevFmtType T
>
1876 void Write(const al::span
<const FloatBufferLine
> InBuffer
, void *OutBuffer
, const size_t Offset
,
1877 const size_t SamplesToDo
, const size_t FrameStep
)
1879 ASSUME(FrameStep
> 0);
1880 ASSUME(SamplesToDo
> 0);
1882 DevFmtType_t
<T
> *outbase
{static_cast<DevFmtType_t
<T
>*>(OutBuffer
) + Offset
*FrameStep
};
1884 for(const FloatBufferLine
&inbuf
: InBuffer
)
1886 DevFmtType_t
<T
> *out
{outbase
++};
1887 auto conv_sample
= [FrameStep
,&out
](const float s
) noexcept
-> void
1889 *out
= SampleConv
<DevFmtType_t
<T
>>(s
);
1892 std::for_each(inbuf
.begin(), inbuf
.begin()+SamplesToDo
, conv_sample
);
1895 if(const size_t extra
{FrameStep
- c
})
1897 const auto silence
= SampleConv
<DevFmtType_t
<T
>>(0.0f
);
1898 for(size_t i
{0};i
< SamplesToDo
;++i
)
1900 std::fill_n(outbase
, extra
, silence
);
1901 outbase
+= FrameStep
;
1908 uint
DeviceBase::renderSamples(const uint numSamples
)
1910 const uint samplesToDo
{minu(numSamples
, BufferLineSize
)};
1912 /* Clear main mixing buffers. */
1913 for(FloatBufferLine
&buffer
: MixBuffer
)
1916 /* Increment the mix count at the start (lsb should now be 1). */
1917 IncrementRef(MixCount
);
1919 /* Process and mix each context's sources and effects. */
1920 ProcessContexts(this, samplesToDo
);
1922 /* Increment the clock time. Every second's worth of samples is converted
1923 * and added to clock base so that large sample counts don't overflow
1924 * during conversion. This also guarantees a stable conversion.
1926 SamplesDone
+= samplesToDo
;
1927 ClockBase
+= std::chrono::seconds
{SamplesDone
/ Frequency
};
1928 SamplesDone
%= Frequency
;
1930 /* Increment the mix count at the end (lsb should now be 0). */
1931 IncrementRef(MixCount
);
1933 /* Apply any needed post-process for finalizing the Dry mix to the RealOut
1934 * (Ambisonic decode, UHJ encode, etc).
1936 postProcess(samplesToDo
);
1938 /* Apply compression, limiting sample amplitude if needed or desired. */
1939 if(Limiter
) Limiter
->process(samplesToDo
, RealOut
.Buffer
.data());
1941 /* Apply delays and attenuation for mismatched speaker distances. */
1943 ApplyDistanceComp(RealOut
.Buffer
, samplesToDo
, ChannelDelays
->mChannels
.data());
1945 /* Apply dithering. The compressor should have left enough headroom for the
1946 * dither noise to not saturate.
1948 if(DitherDepth
> 0.0f
)
1949 ApplyDither(RealOut
.Buffer
, &DitherSeed
, DitherDepth
, samplesToDo
);
1954 void DeviceBase::renderSamples(const al::span
<float*> outBuffers
, const uint numSamples
)
1956 FPUCtl mixer_mode
{};
1958 while(const uint todo
{numSamples
- total
})
1960 const uint samplesToDo
{renderSamples(todo
)};
1962 auto *srcbuf
= RealOut
.Buffer
.data();
1963 for(auto *dstbuf
: outBuffers
)
1965 std::copy_n(srcbuf
->data(), samplesToDo
, dstbuf
+ total
);
1969 total
+= samplesToDo
;
1973 void DeviceBase::renderSamples(void *outBuffer
, const uint numSamples
, const size_t frameStep
)
1975 FPUCtl mixer_mode
{};
1977 while(const uint todo
{numSamples
- total
})
1979 const uint samplesToDo
{renderSamples(todo
)};
1981 if LIKELY(outBuffer
)
1983 /* Finally, interleave and convert samples, writing to the device's
1988 #define HANDLE_WRITE(T) case T: \
1989 Write<T>(RealOut.Buffer, outBuffer, total, samplesToDo, frameStep); break;
1990 HANDLE_WRITE(DevFmtByte
)
1991 HANDLE_WRITE(DevFmtUByte
)
1992 HANDLE_WRITE(DevFmtShort
)
1993 HANDLE_WRITE(DevFmtUShort
)
1994 HANDLE_WRITE(DevFmtInt
)
1995 HANDLE_WRITE(DevFmtUInt
)
1996 HANDLE_WRITE(DevFmtFloat
)
2001 total
+= samplesToDo
;
2005 void DeviceBase::handleDisconnect(const char *msg
, ...)
2007 IncrementRef(MixCount
);
2008 if(Connected
.exchange(false, std::memory_order_acq_rel
))
2010 AsyncEvent evt
{AsyncEvent::Disconnected
};
2013 va_start(args
, msg
);
2014 int msglen
{vsnprintf(evt
.u
.disconnect
.msg
, sizeof(evt
.u
.disconnect
.msg
), msg
, args
)};
2017 if(msglen
< 0 || static_cast<size_t>(msglen
) >= sizeof(evt
.u
.disconnect
.msg
))
2018 evt
.u
.disconnect
.msg
[sizeof(evt
.u
.disconnect
.msg
)-1] = 0;
2020 for(ContextBase
*ctx
: *mContexts
.load())
2022 const uint enabledevt
{ctx
->mEnabledEvts
.load(std::memory_order_acquire
)};
2023 if((enabledevt
&AsyncEvent::Disconnected
))
2025 RingBuffer
*ring
{ctx
->mAsyncEvents
.get()};
2026 auto evt_data
= ring
->getWriteVector().first
;
2027 if(evt_data
.len
> 0)
2029 al::construct_at(reinterpret_cast<AsyncEvent
*>(evt_data
.buf
), evt
);
2030 ring
->writeAdvance(1);
2031 ctx
->mEventSem
.post();
2035 if(!ctx
->mStopVoicesOnDisconnect
)
2037 ProcessVoiceChanges(ctx
);
2041 auto voicelist
= ctx
->getVoicesSpanAcquired();
2042 auto stop_voice
= [](Voice
*voice
) -> void
2044 voice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
2045 voice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
2046 voice
->mSourceID
.store(0u, std::memory_order_relaxed
);
2047 voice
->mPlayState
.store(Voice::Stopped
, std::memory_order_release
);
2049 std::for_each(voicelist
.begin(), voicelist
.end(), stop_voice
);
2052 IncrementRef(MixCount
);