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;
108 using namespace std::chrono
;
110 constexpr uint MaxPitch
{10};
112 static_assert((BufferLineSize
-1)/MaxPitch
> 0, "MaxPitch is too large for BufferLineSize!");
113 static_assert((INT_MAX
>>MixerFracBits
)/MaxPitch
> BufferLineSize
,
114 "MaxPitch and/or BufferLineSize are too large for MixerFracBits!");
116 using namespace std::placeholders
;
118 float InitConeScale()
121 if(auto optval
= al::getenv("__ALSOFT_HALF_ANGLE_CONES"))
123 if(al::strcasecmp(optval
->c_str(), "true") == 0
124 || strtol(optval
->c_str(), nullptr, 0) == 1)
130 const float ConeScale
{InitConeScale()};
132 /* Localized scalars for mono sources (initialized in aluInit, after
133 * configuration is loaded).
139 /* Source distance scale for NFC filters. */
140 float NfcScale
{1.0f
};
149 using HrtfDirectMixerFunc
= void(*)(const FloatBufferSpan LeftOut
, const FloatBufferSpan RightOut
,
150 const al::span
<const FloatBufferLine
> InSamples
, float2
*AccumSamples
, float *TempBuf
,
151 HrtfChannelState
*ChanState
, const size_t IrSize
, const size_t BufferSize
);
153 HrtfDirectMixerFunc MixDirectHrtf
{MixDirectHrtf_
<CTag
>};
155 inline HrtfDirectMixerFunc
SelectHrtfMixer(void)
158 if((CPUCapFlags
&CPU_CAP_NEON
))
159 return MixDirectHrtf_
<NEONTag
>;
162 if((CPUCapFlags
&CPU_CAP_SSE
))
163 return MixDirectHrtf_
<SSETag
>;
166 return MixDirectHrtf_
<CTag
>;
170 inline void BsincPrepare(const uint increment
, BsincState
*state
, const BSincTable
*table
)
172 size_t si
{BSincScaleCount
- 1};
175 if(increment
> MixerFracOne
)
177 sf
= MixerFracOne
/static_cast<float>(increment
) - table
->scaleBase
;
178 sf
= maxf(0.0f
, BSincScaleCount
*sf
*table
->scaleRange
- 1.0f
);
180 /* The interpolation factor is fit to this diagonally-symmetric curve
181 * to reduce the transition ripple caused by interpolating different
182 * scales of the sinc function.
184 sf
= 1.0f
- std::cos(std::asin(sf
- static_cast<float>(si
)));
188 state
->m
= table
->m
[si
];
189 state
->l
= (state
->m
/2) - 1;
190 state
->filter
= table
->Tab
+ table
->filterOffset
[si
];
193 inline ResamplerFunc
SelectResampler(Resampler resampler
, uint increment
)
197 case Resampler::Point
:
198 return Resample_
<PointTag
,CTag
>;
199 case Resampler::Linear
:
201 if((CPUCapFlags
&CPU_CAP_NEON
))
202 return Resample_
<LerpTag
,NEONTag
>;
205 if((CPUCapFlags
&CPU_CAP_SSE4_1
))
206 return Resample_
<LerpTag
,SSE4Tag
>;
209 if((CPUCapFlags
&CPU_CAP_SSE2
))
210 return Resample_
<LerpTag
,SSE2Tag
>;
212 return Resample_
<LerpTag
,CTag
>;
213 case Resampler::Cubic
:
214 return Resample_
<CubicTag
,CTag
>;
215 case Resampler::BSinc12
:
216 case Resampler::BSinc24
:
217 if(increment
> MixerFracOne
)
220 if((CPUCapFlags
&CPU_CAP_NEON
))
221 return Resample_
<BSincTag
,NEONTag
>;
224 if((CPUCapFlags
&CPU_CAP_SSE
))
225 return Resample_
<BSincTag
,SSETag
>;
227 return Resample_
<BSincTag
,CTag
>;
230 case Resampler::FastBSinc12
:
231 case Resampler::FastBSinc24
:
233 if((CPUCapFlags
&CPU_CAP_NEON
))
234 return Resample_
<FastBSincTag
,NEONTag
>;
237 if((CPUCapFlags
&CPU_CAP_SSE
))
238 return Resample_
<FastBSincTag
,SSETag
>;
240 return Resample_
<FastBSincTag
,CTag
>;
243 return Resample_
<PointTag
,CTag
>;
248 void aluInit(CompatFlagBitset flags
, const float nfcscale
)
250 MixDirectHrtf
= SelectHrtfMixer();
251 XScale
= flags
.test(CompatFlags::ReverseX
) ? -1.0f
: 1.0f
;
252 YScale
= flags
.test(CompatFlags::ReverseY
) ? -1.0f
: 1.0f
;
253 ZScale
= flags
.test(CompatFlags::ReverseZ
) ? -1.0f
: 1.0f
;
255 NfcScale
= clampf(nfcscale
, 0.0001f
, 10000.0f
);
259 ResamplerFunc
PrepareResampler(Resampler resampler
, uint increment
, InterpState
*state
)
263 case Resampler::Point
:
264 case Resampler::Linear
:
265 case Resampler::Cubic
:
267 case Resampler::FastBSinc12
:
268 case Resampler::BSinc12
:
269 BsincPrepare(increment
, &state
->bsinc
, &bsinc12
);
271 case Resampler::FastBSinc24
:
272 case Resampler::BSinc24
:
273 BsincPrepare(increment
, &state
->bsinc
, &bsinc24
);
276 return SelectResampler(resampler
, increment
);
280 void DeviceBase::ProcessHrtf(const size_t SamplesToDo
)
282 /* HRTF is stereo output only. */
283 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
284 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
286 MixDirectHrtf(RealOut
.Buffer
[lidx
], RealOut
.Buffer
[ridx
], Dry
.Buffer
, HrtfAccumData
,
287 mHrtfState
->mTemp
.data(), mHrtfState
->mChannels
.data(), mHrtfState
->mIrSize
, SamplesToDo
);
290 void DeviceBase::ProcessAmbiDec(const size_t SamplesToDo
)
292 AmbiDecoder
->process(RealOut
.Buffer
, Dry
.Buffer
.data(), SamplesToDo
);
295 void DeviceBase::ProcessAmbiDecStablized(const size_t SamplesToDo
)
297 /* Decode with front image stablization. */
298 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
299 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
300 const uint cidx
{RealOut
.ChannelIndex
[FrontCenter
]};
302 AmbiDecoder
->processStablize(RealOut
.Buffer
, Dry
.Buffer
.data(), lidx
, ridx
, cidx
,
306 void DeviceBase::ProcessUhj(const size_t SamplesToDo
)
308 /* UHJ is stereo output only. */
309 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
310 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
312 /* Encode to stereo-compatible 2-channel UHJ output. */
313 mUhjEncoder
->encode(RealOut
.Buffer
[lidx
].data(), RealOut
.Buffer
[ridx
].data(),
314 {{Dry
.Buffer
[0].data(), Dry
.Buffer
[1].data(), Dry
.Buffer
[2].data()}}, SamplesToDo
);
317 void DeviceBase::ProcessBs2b(const size_t SamplesToDo
)
319 /* First, decode the ambisonic mix to the "real" output. */
320 AmbiDecoder
->process(RealOut
.Buffer
, Dry
.Buffer
.data(), SamplesToDo
);
322 /* BS2B is stereo output only. */
323 const uint lidx
{RealOut
.ChannelIndex
[FrontLeft
]};
324 const uint ridx
{RealOut
.ChannelIndex
[FrontRight
]};
326 /* Now apply the BS2B binaural/crossfeed filter. */
327 bs2b_cross_feed(Bs2b
.get(), RealOut
.Buffer
[lidx
].data(), RealOut
.Buffer
[ridx
].data(),
334 /* This RNG method was created based on the math found in opusdec. It's quick,
335 * and starting with a seed value of 22222, is suitable for generating
338 inline uint
dither_rng(uint
*seed
) noexcept
340 *seed
= (*seed
* 96314165) + 907633515;
345 /* Ambisonic upsampler function. It's effectively a matrix multiply. It takes
346 * an 'upsampler' and 'rotator' as the input matrices, resulting in a matrix
347 * that behaves as if the B-Format input was first decoded to a speaker array
348 * at its input order, encoded back into the higher order mix, then finally
351 void UpsampleBFormatTransform(size_t coeffs_order
,
352 const al::span
<const std::array
<float,MaxAmbiChannels
>> matrix1
,
353 const al::span
<std::array
<float,MaxAmbiChannels
>,MaxAmbiChannels
> coeffs
)
355 auto copy_coeffs
= [coeffs
]() noexcept
357 std::array
<std::array
<float,MaxAmbiChannels
>,MaxAmbiChannels
> res
{};
358 for(size_t i
{0};i
< MaxAmbiChannels
;++i
)
362 const auto matrix2
= copy_coeffs();
364 const size_t num_chans
{AmbiChannelsFromOrder(coeffs_order
)};
365 for(size_t i
{0};i
< matrix1
.size();++i
)
367 for(size_t j
{0};j
< num_chans
;++j
)
370 for(size_t k
{0};k
< num_chans
;++k
)
371 sum
+= matrix1
[i
][k
] * matrix2
[j
][k
];
378 inline auto& GetAmbiScales(AmbiScaling scaletype
) noexcept
382 case AmbiScaling::FuMa
: return AmbiScale::FromFuMa();
383 case AmbiScaling::SN3D
: return AmbiScale::FromSN3D();
384 case AmbiScaling::UHJ
: return AmbiScale::FromUHJ();
385 case AmbiScaling::N3D
: break;
387 return AmbiScale::FromN3D();
390 inline auto& GetAmbiLayout(AmbiLayout layouttype
) noexcept
392 if(layouttype
== AmbiLayout::FuMa
) return AmbiIndex::FromFuMa();
393 return AmbiIndex::FromACN();
396 inline auto& GetAmbi2DLayout(AmbiLayout layouttype
) noexcept
398 if(layouttype
== AmbiLayout::FuMa
) return AmbiIndex::FromFuMa2D();
399 return AmbiIndex::FromACN2D();
403 bool CalcContextParams(ContextBase
*ctx
)
405 ContextProps
*props
{ctx
->mParams
.ContextUpdate
.exchange(nullptr, std::memory_order_acq_rel
)};
406 if(!props
) return false;
408 const alu::Vector pos
{props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
};
409 ctx
->mParams
.Position
= pos
;
412 alu::Vector N
{props
->OrientAt
[0], props
->OrientAt
[1], props
->OrientAt
[2], 0.0f
};
414 alu::Vector V
{props
->OrientUp
[0], props
->OrientUp
[1], props
->OrientUp
[2], 0.0f
};
416 /* Build and normalize right-vector */
417 alu::Vector U
{N
.cross_product(V
)};
420 const alu::Matrix rot
{
421 U
[0], V
[0], -N
[0], 0.0,
422 U
[1], V
[1], -N
[1], 0.0,
423 U
[2], V
[2], -N
[2], 0.0,
425 const alu::Vector vel
{props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0};
427 ctx
->mParams
.Matrix
= rot
;
428 ctx
->mParams
.Velocity
= rot
* vel
;
430 ctx
->mParams
.Gain
= props
->Gain
* ctx
->mGainBoost
;
431 ctx
->mParams
.MetersPerUnit
= props
->MetersPerUnit
;
432 ctx
->mParams
.AirAbsorptionGainHF
= props
->AirAbsorptionGainHF
;
434 ctx
->mParams
.DopplerFactor
= props
->DopplerFactor
;
435 ctx
->mParams
.SpeedOfSound
= props
->SpeedOfSound
* props
->DopplerVelocity
;
437 ctx
->mParams
.SourceDistanceModel
= props
->SourceDistanceModel
;
438 ctx
->mParams
.mDistanceModel
= props
->mDistanceModel
;
440 AtomicReplaceHead(ctx
->mFreeContextProps
, props
);
444 bool CalcEffectSlotParams(EffectSlot
*slot
, EffectSlot
**sorted_slots
, ContextBase
*context
)
446 EffectSlotProps
*props
{slot
->Update
.exchange(nullptr, std::memory_order_acq_rel
)};
447 if(!props
) return false;
449 /* If the effect slot target changed, clear the first sorted entry to force
452 if(slot
->Target
!= props
->Target
)
453 *sorted_slots
= nullptr;
454 slot
->Gain
= props
->Gain
;
455 slot
->AuxSendAuto
= props
->AuxSendAuto
;
456 slot
->Target
= props
->Target
;
457 slot
->EffectType
= props
->Type
;
458 slot
->mEffectProps
= props
->Props
;
459 if(props
->Type
== EffectSlotType::Reverb
|| props
->Type
== EffectSlotType::EAXReverb
)
461 slot
->RoomRolloff
= props
->Props
.Reverb
.RoomRolloffFactor
;
462 slot
->DecayTime
= props
->Props
.Reverb
.DecayTime
;
463 slot
->DecayLFRatio
= props
->Props
.Reverb
.DecayLFRatio
;
464 slot
->DecayHFRatio
= props
->Props
.Reverb
.DecayHFRatio
;
465 slot
->DecayHFLimit
= props
->Props
.Reverb
.DecayHFLimit
;
466 slot
->AirAbsorptionGainHF
= props
->Props
.Reverb
.AirAbsorptionGainHF
;
470 slot
->RoomRolloff
= 0.0f
;
471 slot
->DecayTime
= 0.0f
;
472 slot
->DecayLFRatio
= 0.0f
;
473 slot
->DecayHFRatio
= 0.0f
;
474 slot
->DecayHFLimit
= false;
475 slot
->AirAbsorptionGainHF
= 1.0f
;
478 EffectState
*state
{props
->State
.release()};
479 EffectState
*oldstate
{slot
->mEffectState
.release()};
480 slot
->mEffectState
.reset(state
);
482 /* Only release the old state if it won't get deleted, since we can't be
483 * deleting/freeing anything in the mixer.
485 if(!oldstate
->releaseIfNoDelete())
487 /* Otherwise, if it would be deleted send it off with a release event. */
488 RingBuffer
*ring
{context
->mAsyncEvents
.get()};
489 auto evt_vec
= ring
->getWriteVector();
490 if(evt_vec
.first
.len
> 0) [[likely
]]
492 AsyncEvent
*evt
{al::construct_at(reinterpret_cast<AsyncEvent
*>(evt_vec
.first
.buf
),
493 AsyncEvent::ReleaseEffectState
)};
494 evt
->u
.mEffectState
= oldstate
;
495 ring
->writeAdvance(1);
499 /* If writing the event failed, the queue was probably full. Store
500 * the old state in the property object where it can eventually be
501 * cleaned up sometime later (not ideal, but better than blocking
504 props
->State
.reset(oldstate
);
508 AtomicReplaceHead(context
->mFreeEffectslotProps
, props
);
511 if(EffectSlot
*target
{slot
->Target
})
512 output
= EffectTarget
{&target
->Wet
, nullptr};
515 DeviceBase
*device
{context
->mDevice
};
516 output
= EffectTarget
{&device
->Dry
, &device
->RealOut
};
518 state
->update(context
, slot
, &slot
->mEffectProps
, output
);
523 /* Scales the given azimuth toward the side (+/- pi/2 radians) for positions in
526 inline float ScaleAzimuthFront(float azimuth
, float scale
)
528 const float abs_azi
{std::fabs(azimuth
)};
529 if(!(abs_azi
>= al::numbers::pi_v
<float>*0.5f
))
530 return std::copysign(minf(abs_azi
*scale
, al::numbers::pi_v
<float>*0.5f
), azimuth
);
534 /* Wraps the given value in radians to stay between [-pi,+pi] */
535 inline float WrapRadians(float r
)
537 static constexpr float Pi
{al::numbers::pi_v
<float>};
538 static constexpr float Pi2
{Pi
*2.0f
};
539 if(r
> Pi
) return std::fmod(Pi
+r
, Pi2
) - Pi
;
540 if(r
< -Pi
) return Pi
- std::fmod(Pi
-r
, Pi2
);
544 /* Begin ambisonic rotation helpers.
546 * Rotating first-order B-Format just needs a straight-forward X/Y/Z rotation
547 * matrix. Higher orders, however, are more complicated. The method implemented
548 * here is a recursive algorithm (the rotation for first-order is used to help
549 * generate the second-order rotation, which helps generate the third-order
553 * <https://github.com/polarch/Spherical-Harmonic-Transform/blob/master/getSHrotMtx.m>,
554 * provided under the BSD 3-Clause license.
556 * Copyright (c) 2015, Archontis Politis
557 * Copyright (c) 2019, Christopher Robinson
559 * The u, v, and w coefficients used for generating higher-order rotations are
560 * precomputed since they're constant. The second-order coefficients are
561 * followed by the third-order coefficients, etc.
564 constexpr size_t CalcRotatorSize()
565 { return (L
*2 + 1)*(L
*2 + 1) + CalcRotatorSize
<L
-1>(); }
567 template<> constexpr size_t CalcRotatorSize
<0>() = delete;
568 template<> constexpr size_t CalcRotatorSize
<1>() = delete;
569 template<> constexpr size_t CalcRotatorSize
<2>() { return 5*5; }
571 struct RotatorCoeffs
{
575 std::array
<CoeffValues
,CalcRotatorSize
<MaxAmbiOrder
>()> mCoeffs
{};
579 auto coeffs
= mCoeffs
.begin();
581 for(int l
=2;l
<= MaxAmbiOrder
;++l
)
583 for(int m
{-l
};m
<= l
;++m
)
585 for(int n
{-l
};n
<= l
;++n
)
587 // compute u,v,w terms of Eq.8.1 (Table I)
588 const bool d
{m
== 0}; // the delta function d_m0
589 const float denom
{static_cast<float>((std::abs(n
) == l
) ?
590 (2*l
) * (2*l
- 1) : (l
*l
- n
*n
))};
592 const int abs_m
{std::abs(m
)};
593 coeffs
->u
= std::sqrt(static_cast<float>(l
*l
- m
*m
)/denom
);
594 coeffs
->v
= std::sqrt(static_cast<float>(l
+abs_m
-1) *
595 static_cast<float>(l
+abs_m
) / denom
) * (1.0f
+d
) * (1.0f
- 2.0f
*d
) * 0.5f
;
596 coeffs
->w
= std::sqrt(static_cast<float>(l
-abs_m
-1) *
597 static_cast<float>(l
-abs_m
) / denom
) * (1.0f
-d
) * -0.5f
;
604 const RotatorCoeffs RotatorCoeffArray
{};
607 * Given the matrix, pre-filled with the (zeroth- and) first-order rotation
608 * coefficients, this fills in the coefficients for the higher orders up to and
609 * including the given order. The matrix is in ACN layout.
611 void AmbiRotator(AmbiRotateMatrix
&matrix
, const int order
)
613 /* Don't do anything for < 2nd order. */
614 if(order
< 2) return;
616 auto P
= [](const int i
, const int l
, const int a
, const int n
, const size_t last_band
,
617 const AmbiRotateMatrix
&R
)
619 const float ri1
{ R
[static_cast<uint
>(i
+2)][ 1+2]};
620 const float rim1
{R
[static_cast<uint
>(i
+2)][-1+2]};
621 const float ri0
{ R
[static_cast<uint
>(i
+2)][ 0+2]};
623 auto vec
= R
[static_cast<uint
>(a
+l
-1) + last_band
].cbegin() + last_band
;
625 return ri1
*vec
[0] + rim1
*vec
[static_cast<uint
>(l
-1)*size_t{2}];
627 return ri1
*vec
[static_cast<uint
>(l
-1)*size_t{2}] - rim1
*vec
[0];
628 return ri0
*vec
[static_cast<uint
>(n
+l
-1)];
631 auto U
= [P
](const int l
, const int m
, const int n
, const size_t last_band
,
632 const AmbiRotateMatrix
&R
)
634 return P(0, l
, m
, n
, last_band
, R
);
636 auto V
= [P
](const int l
, const int m
, const int n
, const size_t last_band
,
637 const AmbiRotateMatrix
&R
)
639 using namespace al::numbers
;
642 const bool d
{m
== 1};
643 const float p0
{P( 1, l
, m
-1, n
, last_band
, R
)};
644 const float p1
{P(-1, l
, -m
+1, n
, last_band
, R
)};
645 return d
? p0
*sqrt2_v
<float> : (p0
- p1
);
647 const bool d
{m
== -1};
648 const float p0
{P( 1, l
, m
+1, n
, last_band
, R
)};
649 const float p1
{P(-1, l
, -m
-1, n
, last_band
, R
)};
650 return d
? p1
*sqrt2_v
<float> : (p0
+ p1
);
652 auto W
= [P
](const int l
, const int m
, const int n
, const size_t last_band
,
653 const AmbiRotateMatrix
&R
)
658 const float p0
{P( 1, l
, m
+1, n
, last_band
, R
)};
659 const float p1
{P(-1, l
, -m
-1, n
, last_band
, R
)};
662 const float p0
{P( 1, l
, m
-1, n
, last_band
, R
)};
663 const float p1
{P(-1, l
, -m
+1, n
, last_band
, R
)};
667 // compute rotation matrix of each subsequent band recursively
668 auto coeffs
= RotatorCoeffArray
.mCoeffs
.cbegin();
669 size_t band_idx
{4}, last_band
{1};
670 for(int l
{2};l
<= order
;++l
)
673 for(int m
{-l
};m
<= l
;++m
,++y
)
676 for(int n
{-l
};n
<= l
;++n
,++x
)
681 const float u
{coeffs
->u
};
682 if(u
!= 0.0f
) r
+= u
* U(l
, m
, n
, last_band
, matrix
);
683 const float v
{coeffs
->v
};
684 if(v
!= 0.0f
) r
+= v
* V(l
, m
, n
, last_band
, matrix
);
685 const float w
{coeffs
->w
};
686 if(w
!= 0.0f
) r
+= w
* W(l
, m
, n
, last_band
, matrix
);
692 last_band
= band_idx
;
693 band_idx
+= static_cast<uint
>(l
)*size_t{2} + 1;
696 /* End ambisonic rotation helpers. */
699 constexpr float Deg2Rad(float x
) noexcept
700 { return static_cast<float>(al::numbers::pi
/ 180.0 * x
); }
702 struct GainTriplet
{ float Base
, HF
, LF
; };
704 void CalcPanningAndFilters(Voice
*voice
, const float xpos
, const float ypos
, const float zpos
,
705 const float Distance
, const float Spread
, const GainTriplet
&DryGain
,
706 const al::span
<const GainTriplet
,MAX_SENDS
> WetGain
, EffectSlot
*(&SendSlots
)[MAX_SENDS
],
707 const VoiceProps
*props
, const ContextParams
&Context
, DeviceBase
*Device
)
709 static constexpr ChanMap MonoMap
[1]{
710 { FrontCenter
, 0.0f
, 0.0f
}
712 { BackLeft
, Deg2Rad(-150.0f
), Deg2Rad(0.0f
) },
713 { BackRight
, Deg2Rad( 150.0f
), Deg2Rad(0.0f
) }
715 { FrontLeft
, Deg2Rad( -45.0f
), Deg2Rad(0.0f
) },
716 { FrontRight
, Deg2Rad( 45.0f
), Deg2Rad(0.0f
) },
717 { BackLeft
, Deg2Rad(-135.0f
), Deg2Rad(0.0f
) },
718 { BackRight
, Deg2Rad( 135.0f
), Deg2Rad(0.0f
) }
720 { FrontLeft
, Deg2Rad( -30.0f
), Deg2Rad(0.0f
) },
721 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) },
722 { FrontCenter
, Deg2Rad( 0.0f
), Deg2Rad(0.0f
) },
724 { SideLeft
, Deg2Rad(-110.0f
), Deg2Rad(0.0f
) },
725 { SideRight
, Deg2Rad( 110.0f
), Deg2Rad(0.0f
) }
727 { FrontLeft
, Deg2Rad(-30.0f
), Deg2Rad(0.0f
) },
728 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) },
729 { FrontCenter
, Deg2Rad( 0.0f
), Deg2Rad(0.0f
) },
731 { BackCenter
, Deg2Rad(180.0f
), Deg2Rad(0.0f
) },
732 { SideLeft
, Deg2Rad(-90.0f
), Deg2Rad(0.0f
) },
733 { SideRight
, Deg2Rad( 90.0f
), Deg2Rad(0.0f
) }
735 { FrontLeft
, Deg2Rad( -30.0f
), Deg2Rad(0.0f
) },
736 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) },
737 { FrontCenter
, Deg2Rad( 0.0f
), Deg2Rad(0.0f
) },
739 { BackLeft
, Deg2Rad(-150.0f
), Deg2Rad(0.0f
) },
740 { BackRight
, Deg2Rad( 150.0f
), Deg2Rad(0.0f
) },
741 { SideLeft
, Deg2Rad( -90.0f
), Deg2Rad(0.0f
) },
742 { SideRight
, Deg2Rad( 90.0f
), Deg2Rad(0.0f
) }
745 ChanMap StereoMap
[2]{
746 { FrontLeft
, Deg2Rad(-30.0f
), Deg2Rad(0.0f
) },
747 { FrontRight
, Deg2Rad( 30.0f
), Deg2Rad(0.0f
) }
750 const auto Frequency
= static_cast<float>(Device
->Frequency
);
751 const uint NumSends
{Device
->NumAuxSends
};
753 const size_t num_channels
{voice
->mChans
.size()};
754 ASSUME(num_channels
> 0);
756 for(auto &chandata
: voice
->mChans
)
758 chandata
.mDryParams
.Hrtf
.Target
= HrtfFilter
{};
759 chandata
.mDryParams
.Gains
.Target
.fill(0.0f
);
760 std::for_each(chandata
.mWetParams
.begin(), chandata
.mWetParams
.begin()+NumSends
,
761 [](SendParams
¶ms
) -> void { params
.Gains
.Target
.fill(0.0f
); });
764 DirectMode DirectChannels
{props
->DirectChannels
};
765 const ChanMap
*chans
{nullptr};
766 switch(voice
->mFmtChannels
)
770 /* Mono buffers are never played direct. */
771 DirectChannels
= DirectMode::Off
;
775 if(DirectChannels
== DirectMode::Off
)
777 /* Convert counter-clockwise to clock-wise, and wrap between
780 StereoMap
[0].angle
= WrapRadians(-props
->StereoPan
[0]);
781 StereoMap
[1].angle
= WrapRadians(-props
->StereoPan
[1]);
786 case FmtRear
: chans
= RearMap
; break;
787 case FmtQuad
: chans
= QuadMap
; break;
788 case FmtX51
: chans
= X51Map
; break;
789 case FmtX61
: chans
= X61Map
; break;
790 case FmtX71
: chans
= X71Map
; break;
798 DirectChannels
= DirectMode::Off
;
802 voice
->mFlags
.reset(VoiceHasHrtf
).reset(VoiceHasNfc
);
803 if(auto *decoder
{voice
->mDecoder
.get()})
804 decoder
->mWidthControl
= minf(props
->EnhWidth
, 0.7f
);
806 if(IsAmbisonic(voice
->mFmtChannels
))
808 /* Special handling for B-Format and UHJ sources. */
810 if(Device
->AvgSpeakerDist
> 0.0f
&& voice
->mFmtChannels
!= FmtUHJ2
811 && voice
->mFmtChannels
!= FmtSuperStereo
)
813 if(!(Distance
> std::numeric_limits
<float>::epsilon()))
815 /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
816 * is what we want for FOA input. The first channel may have
817 * been previously re-adjusted if panned, so reset it.
819 voice
->mChans
[0].mDryParams
.NFCtrlFilter
.adjust(0.0f
);
823 /* Clamp the distance for really close sources, to prevent
826 const float mdist
{maxf(Distance
*NfcScale
, Device
->AvgSpeakerDist
/4.0f
)};
827 const float w0
{SpeedOfSoundMetersPerSec
/ (mdist
* Frequency
)};
829 /* Only need to adjust the first channel of a B-Format source. */
830 voice
->mChans
[0].mDryParams
.NFCtrlFilter
.adjust(w0
);
833 voice
->mFlags
.set(VoiceHasNfc
);
836 /* Panning a B-Format sound toward some direction is easy. Just pan the
837 * first (W) channel as a normal mono sound. The angular spread is used
838 * as a directional scalar to blend between full coverage and full
841 const float coverage
{!(Distance
> std::numeric_limits
<float>::epsilon()) ? 1.0f
:
842 (al::numbers::inv_pi_v
<float>/2.0f
* Spread
)};
844 auto calc_coeffs
= [xpos
,ypos
,zpos
](RenderMode mode
)
846 if(mode
!= RenderMode::Pairwise
)
847 return CalcDirectionCoeffs({xpos
, ypos
, zpos
});
849 /* Clamp Y, in case rounding errors caused it to end up outside
852 const float ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
853 /* Negate Z for right-handed coords with -Z in front. */
854 const float az
{std::atan2(xpos
, -zpos
)};
856 /* A scalar of 1.5 for plain stereo results in +/-60 degrees
857 * being moved to +/-90 degrees for direct right and left
860 return CalcAngleCoeffs(ScaleAzimuthFront(az
, 1.5f
), ev
, 0.0f
);
862 auto&& scales
= GetAmbiScales(voice
->mAmbiScaling
);
863 auto coeffs
= calc_coeffs(Device
->mRenderMode
);
864 /* Scale the panned W signal based on the coverage (full coverage means
865 * no panned signal). Scale the panned W signal according to channel
868 std::transform(coeffs
.begin(), coeffs
.end(), coeffs
.begin(),
869 [scale
=(1.0f
-coverage
)*scales
[0]](const float c
){ return c
* scale
; });
871 if(!(coverage
> 0.0f
))
873 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
874 voice
->mChans
[0].mDryParams
.Gains
.Target
);
875 for(uint i
{0};i
< NumSends
;i
++)
877 if(const EffectSlot
*Slot
{SendSlots
[i
]})
878 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
*scales
[0],
879 voice
->mChans
[0].mWetParams
[i
].Gains
.Target
);
884 /* Local B-Format sources have their XYZ channels rotated according
885 * to the orientation.
888 alu::Vector N
{props
->OrientAt
[0], props
->OrientAt
[1], props
->OrientAt
[2], 0.0f
};
890 alu::Vector V
{props
->OrientUp
[0], props
->OrientUp
[1], props
->OrientUp
[2], 0.0f
};
892 if(!props
->HeadRelative
)
894 N
= Context
.Matrix
* N
;
895 V
= Context
.Matrix
* V
;
897 /* Build and normalize right-vector */
898 alu::Vector U
{N
.cross_product(V
)};
901 /* Build a rotation matrix. Manually fill the zeroth- and first-
902 * order elements, then construct the rotation for the higher
905 AmbiRotateMatrix
&shrot
= Device
->mAmbiRotateMatrix
;
909 shrot
[1][1] = U
[0]; shrot
[1][2] = -V
[0]; shrot
[1][3] = -N
[0];
910 shrot
[2][1] = -U
[1]; shrot
[2][2] = V
[1]; shrot
[2][3] = N
[1];
911 shrot
[3][1] = U
[2]; shrot
[3][2] = -V
[2]; shrot
[3][3] = -N
[2];
912 AmbiRotator(shrot
, static_cast<int>(Device
->mAmbiOrder
));
913 /* If the device is higher order than the voice, "upsample" the
916 * NOTE: Starting with second-order, a 2D upsample needs to be
917 * applied with a 2D source and 3D output, even when they're the
918 * same order. This is because higher orders have a height offset
919 * on various channels (i.e. when elevation=0, those height-related
920 * channels should be non-0).
922 if(Device
->mAmbiOrder
> voice
->mAmbiOrder
923 || (Device
->mAmbiOrder
>= 2 && !Device
->m2DMixing
924 && Is2DAmbisonic(voice
->mFmtChannels
)))
926 if(voice
->mAmbiOrder
== 1)
928 auto&& upsampler
= Is2DAmbisonic(voice
->mFmtChannels
) ?
929 AmbiScale::FirstOrder2DUp
: AmbiScale::FirstOrderUp
;
930 UpsampleBFormatTransform(Device
->mAmbiOrder
, upsampler
, shrot
);
932 else if(voice
->mAmbiOrder
== 2)
934 auto&& upsampler
= Is2DAmbisonic(voice
->mFmtChannels
) ?
935 AmbiScale::SecondOrder2DUp
: AmbiScale::SecondOrderUp
;
936 UpsampleBFormatTransform(Device
->mAmbiOrder
, upsampler
, shrot
);
938 else if(voice
->mAmbiOrder
== 3)
940 auto&& upsampler
= Is2DAmbisonic(voice
->mFmtChannels
) ?
941 AmbiScale::ThirdOrder2DUp
: AmbiScale::ThirdOrderUp
;
942 UpsampleBFormatTransform(Device
->mAmbiOrder
, upsampler
, shrot
);
944 else if(voice
->mAmbiOrder
== 4)
946 auto&& upsampler
= AmbiScale::FourthOrder2DUp
;
947 UpsampleBFormatTransform(Device
->mAmbiOrder
, upsampler
, shrot
);
951 /* Convert the rotation matrix for input ordering and scaling, and
952 * whether input is 2D or 3D.
954 const uint8_t *index_map
{Is2DAmbisonic(voice
->mFmtChannels
) ?
955 GetAmbi2DLayout(voice
->mAmbiLayout
).data() :
956 GetAmbiLayout(voice
->mAmbiLayout
).data()};
958 static const uint8_t OrderOffset
[MaxAmbiOrder
+1]{0, 1, 4, 9,};
959 for(size_t c
{0};c
< num_channels
;c
++)
961 const size_t acn
{index_map
[c
]};
962 const size_t order
{AmbiIndex::OrderFromChannel()[acn
]};
963 const float scale
{scales
[acn
] * coverage
};
965 /* For channel 0, combine the B-Format signal (scaled according
966 * to the coverage amount) with the directional pan. For all
967 * other channels, use just the (scaled) B-Format signal.
969 for(size_t x
{OrderOffset
[order
]};x
< MaxAmbiChannels
;++x
)
970 coeffs
[x
] += shrot
[x
][acn
] * scale
;
972 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
973 voice
->mChans
[c
].mDryParams
.Gains
.Target
);
975 for(uint i
{0};i
< NumSends
;i
++)
977 if(const EffectSlot
*Slot
{SendSlots
[i
]})
978 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
979 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
982 coeffs
= std::array
<float,MaxAmbiChannels
>{};
986 else if(DirectChannels
!= DirectMode::Off
&& !Device
->RealOut
.RemixMap
.empty())
988 /* Direct source channels always play local. Skip the virtual channels
989 * and write inputs to the matching real outputs.
991 voice
->mDirect
.Buffer
= Device
->RealOut
.Buffer
;
993 for(size_t c
{0};c
< num_channels
;c
++)
995 uint idx
{Device
->channelIdxByName(chans
[c
].channel
)};
996 if(idx
!= InvalidChannelIndex
)
997 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
;
998 else if(DirectChannels
== DirectMode::RemixMismatch
)
1000 auto match_channel
= [chans
,c
](const InputRemixMap
&map
) noexcept
-> bool
1001 { return chans
[c
].channel
== map
.channel
; };
1002 auto remap
= std::find_if(Device
->RealOut
.RemixMap
.cbegin(),
1003 Device
->RealOut
.RemixMap
.cend(), match_channel
);
1004 if(remap
!= Device
->RealOut
.RemixMap
.cend())
1006 for(const auto &target
: remap
->targets
)
1008 idx
= Device
->channelIdxByName(target
.channel
);
1009 if(idx
!= InvalidChannelIndex
)
1010 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
*
1017 /* Auxiliary sends still use normal channel panning since they mix to
1018 * B-Format, which can't channel-match.
1020 for(size_t c
{0};c
< num_channels
;c
++)
1023 if(chans
[c
].channel
== LFE
)
1026 const auto coeffs
= CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, 0.0f
);
1028 for(uint i
{0};i
< NumSends
;i
++)
1030 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1031 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1032 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1036 else if(Device
->mRenderMode
== RenderMode::Hrtf
)
1038 /* Full HRTF rendering. Skip the virtual channels and render to the
1041 voice
->mDirect
.Buffer
= Device
->RealOut
.Buffer
;
1043 if(Distance
> std::numeric_limits
<float>::epsilon())
1045 const float src_ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
1046 const float src_az
{std::atan2(xpos
, -zpos
)};
1048 if(voice
->mFmtChannels
== FmtMono
)
1050 Device
->mHrtf
->getCoeffs(src_ev
, src_az
, Distance
*NfcScale
, Spread
,
1051 voice
->mChans
[0].mDryParams
.Hrtf
.Target
.Coeffs
,
1052 voice
->mChans
[0].mDryParams
.Hrtf
.Target
.Delay
);
1053 voice
->mChans
[0].mDryParams
.Hrtf
.Target
.Gain
= DryGain
.Base
;
1055 const auto coeffs
= CalcAngleCoeffs(src_az
, src_ev
, Spread
);
1056 for(uint i
{0};i
< NumSends
;i
++)
1058 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1059 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1060 voice
->mChans
[0].mWetParams
[i
].Gains
.Target
);
1063 else for(size_t c
{0};c
< num_channels
;c
++)
1065 using namespace al::numbers
;
1068 if(chans
[c
].channel
== LFE
) continue;
1070 /* Warp the channel position toward the source position as the
1071 * source spread decreases. With no spread, all channels are at
1072 * the source position, at full spread (pi*2), each channel is
1075 const float ev
{lerpf(src_ev
, chans
[c
].elevation
, inv_pi_v
<float>/2.0f
* Spread
)};
1077 float az
{chans
[c
].angle
- src_az
};
1078 if(az
< -pi_v
<float>) az
+= pi_v
<float>*2.0f
;
1079 else if(az
> pi_v
<float>) az
-= pi_v
<float>*2.0f
;
1081 az
*= inv_pi_v
<float>/2.0f
* Spread
;
1084 if(az
< -pi_v
<float>) az
+= pi_v
<float>*2.0f
;
1085 else if(az
> pi_v
<float>) az
-= pi_v
<float>*2.0f
;
1087 Device
->mHrtf
->getCoeffs(ev
, az
, Distance
*NfcScale
, 0.0f
,
1088 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Coeffs
,
1089 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Delay
);
1090 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Gain
= DryGain
.Base
;
1092 const auto coeffs
= CalcAngleCoeffs(az
, ev
, 0.0f
);
1093 for(uint i
{0};i
< NumSends
;i
++)
1095 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1096 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1097 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1103 /* With no distance, spread is only meaningful for mono sources
1104 * where it can be 0 or full (non-mono sources are always full
1107 const float spread
{Spread
* (voice
->mFmtChannels
== FmtMono
)};
1109 /* Local sources on HRTF play with each channel panned to its
1110 * relative location around the listener, providing "virtual
1111 * speaker" responses.
1113 for(size_t c
{0};c
< num_channels
;c
++)
1116 if(chans
[c
].channel
== LFE
)
1119 /* Get the HRIR coefficients and delays for this channel
1122 Device
->mHrtf
->getCoeffs(chans
[c
].elevation
, chans
[c
].angle
,
1123 std::numeric_limits
<float>::infinity(), spread
,
1124 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Coeffs
,
1125 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Delay
);
1126 voice
->mChans
[c
].mDryParams
.Hrtf
.Target
.Gain
= DryGain
.Base
;
1128 /* Normal panning for auxiliary sends. */
1129 const auto coeffs
= CalcAngleCoeffs(chans
[c
].angle
, chans
[c
].elevation
, spread
);
1131 for(uint i
{0};i
< NumSends
;i
++)
1133 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1134 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1135 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1140 voice
->mFlags
.set(VoiceHasHrtf
);
1144 /* Non-HRTF rendering. Use normal panning to the output. */
1146 if(Distance
> std::numeric_limits
<float>::epsilon())
1148 /* Calculate NFC filter coefficient if needed. */
1149 if(Device
->AvgSpeakerDist
> 0.0f
)
1151 /* Clamp the distance for really close sources, to prevent
1154 const float mdist
{maxf(Distance
*NfcScale
, Device
->AvgSpeakerDist
/4.0f
)};
1155 const float w0
{SpeedOfSoundMetersPerSec
/ (mdist
* Frequency
)};
1157 /* Adjust NFC filters. */
1158 for(size_t c
{0};c
< num_channels
;c
++)
1159 voice
->mChans
[c
].mDryParams
.NFCtrlFilter
.adjust(w0
);
1161 voice
->mFlags
.set(VoiceHasNfc
);
1164 if(voice
->mFmtChannels
== FmtMono
)
1166 auto calc_coeffs
= [xpos
,ypos
,zpos
,Spread
](RenderMode mode
)
1168 if(mode
!= RenderMode::Pairwise
)
1169 return CalcDirectionCoeffs({xpos
, ypos
, zpos
}, Spread
);
1170 const float ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
1171 const float az
{std::atan2(xpos
, -zpos
)};
1172 return CalcAngleCoeffs(ScaleAzimuthFront(az
, 1.5f
), ev
, Spread
);
1174 const auto coeffs
= calc_coeffs(Device
->mRenderMode
);
1176 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
1177 voice
->mChans
[0].mDryParams
.Gains
.Target
);
1178 for(uint i
{0};i
< NumSends
;i
++)
1180 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1181 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1182 voice
->mChans
[0].mWetParams
[i
].Gains
.Target
);
1187 using namespace al::numbers
;
1189 const float src_ev
{std::asin(clampf(ypos
, -1.0f
, 1.0f
))};
1190 const float src_az
{std::atan2(xpos
, -zpos
)};
1192 for(size_t c
{0};c
< num_channels
;c
++)
1194 /* Special-case LFE */
1195 if(chans
[c
].channel
== LFE
)
1197 if(Device
->Dry
.Buffer
.data() == Device
->RealOut
.Buffer
.data())
1199 const uint idx
{Device
->channelIdxByName(chans
[c
].channel
)};
1200 if(idx
!= InvalidChannelIndex
)
1201 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
;
1206 /* Warp the channel position toward the source position as
1207 * the spread decreases. With no spread, all channels are
1208 * at the source position, at full spread (pi*2), each
1209 * channel position is left unchanged.
1211 const float ev
{lerpf(src_ev
, chans
[c
].elevation
,
1212 inv_pi_v
<float>/2.0f
* Spread
)};
1214 float az
{chans
[c
].angle
- src_az
};
1215 if(az
< -pi_v
<float>) az
+= pi_v
<float>*2.0f
;
1216 else if(az
> pi_v
<float>) az
-= pi_v
<float>*2.0f
;
1218 az
*= inv_pi_v
<float>/2.0f
* Spread
;
1221 if(az
< -pi_v
<float>) az
+= pi_v
<float>*2.0f
;
1222 else if(az
> pi_v
<float>) az
-= pi_v
<float>*2.0f
;
1224 if(Device
->mRenderMode
== RenderMode::Pairwise
)
1225 az
= ScaleAzimuthFront(az
, 3.0f
);
1226 const auto coeffs
= CalcAngleCoeffs(az
, ev
, 0.0f
);
1228 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
1229 voice
->mChans
[c
].mDryParams
.Gains
.Target
);
1230 for(uint i
{0};i
< NumSends
;i
++)
1232 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1233 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1234 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1241 if(Device
->AvgSpeakerDist
> 0.0f
)
1243 /* If the source distance is 0, simulate a plane-wave by using
1244 * infinite distance, which results in a w0 of 0.
1246 static constexpr float w0
{0.0f
};
1247 for(size_t c
{0};c
< num_channels
;c
++)
1248 voice
->mChans
[c
].mDryParams
.NFCtrlFilter
.adjust(w0
);
1250 voice
->mFlags
.set(VoiceHasNfc
);
1253 /* With no distance, spread is only meaningful for mono sources
1254 * where it can be 0 or full (non-mono sources are always full
1257 const float spread
{Spread
* (voice
->mFmtChannels
== FmtMono
)};
1258 for(size_t c
{0};c
< num_channels
;c
++)
1260 /* Special-case LFE */
1261 if(chans
[c
].channel
== LFE
)
1263 if(Device
->Dry
.Buffer
.data() == Device
->RealOut
.Buffer
.data())
1265 const uint idx
{Device
->channelIdxByName(chans
[c
].channel
)};
1266 if(idx
!= InvalidChannelIndex
)
1267 voice
->mChans
[c
].mDryParams
.Gains
.Target
[idx
] = DryGain
.Base
;
1272 const auto coeffs
= CalcAngleCoeffs((Device
->mRenderMode
== RenderMode::Pairwise
)
1273 ? ScaleAzimuthFront(chans
[c
].angle
, 3.0f
) : chans
[c
].angle
,
1274 chans
[c
].elevation
, spread
);
1276 ComputePanGains(&Device
->Dry
, coeffs
.data(), DryGain
.Base
,
1277 voice
->mChans
[c
].mDryParams
.Gains
.Target
);
1278 for(uint i
{0};i
< NumSends
;i
++)
1280 if(const EffectSlot
*Slot
{SendSlots
[i
]})
1281 ComputePanGains(&Slot
->Wet
, coeffs
.data(), WetGain
[i
].Base
,
1282 voice
->mChans
[c
].mWetParams
[i
].Gains
.Target
);
1289 const float hfNorm
{props
->Direct
.HFReference
/ Frequency
};
1290 const float lfNorm
{props
->Direct
.LFReference
/ Frequency
};
1292 voice
->mDirect
.FilterType
= AF_None
;
1293 if(DryGain
.HF
!= 1.0f
) voice
->mDirect
.FilterType
|= AF_LowPass
;
1294 if(DryGain
.LF
!= 1.0f
) voice
->mDirect
.FilterType
|= AF_HighPass
;
1296 auto &lowpass
= voice
->mChans
[0].mDryParams
.LowPass
;
1297 auto &highpass
= voice
->mChans
[0].mDryParams
.HighPass
;
1298 lowpass
.setParamsFromSlope(BiquadType::HighShelf
, hfNorm
, DryGain
.HF
, 1.0f
);
1299 highpass
.setParamsFromSlope(BiquadType::LowShelf
, lfNorm
, DryGain
.LF
, 1.0f
);
1300 for(size_t c
{1};c
< num_channels
;c
++)
1302 voice
->mChans
[c
].mDryParams
.LowPass
.copyParamsFrom(lowpass
);
1303 voice
->mChans
[c
].mDryParams
.HighPass
.copyParamsFrom(highpass
);
1306 for(uint i
{0};i
< NumSends
;i
++)
1308 const float hfNorm
{props
->Send
[i
].HFReference
/ Frequency
};
1309 const float lfNorm
{props
->Send
[i
].LFReference
/ Frequency
};
1311 voice
->mSend
[i
].FilterType
= AF_None
;
1312 if(WetGain
[i
].HF
!= 1.0f
) voice
->mSend
[i
].FilterType
|= AF_LowPass
;
1313 if(WetGain
[i
].LF
!= 1.0f
) voice
->mSend
[i
].FilterType
|= AF_HighPass
;
1315 auto &lowpass
= voice
->mChans
[0].mWetParams
[i
].LowPass
;
1316 auto &highpass
= voice
->mChans
[0].mWetParams
[i
].HighPass
;
1317 lowpass
.setParamsFromSlope(BiquadType::HighShelf
, hfNorm
, WetGain
[i
].HF
, 1.0f
);
1318 highpass
.setParamsFromSlope(BiquadType::LowShelf
, lfNorm
, WetGain
[i
].LF
, 1.0f
);
1319 for(size_t c
{1};c
< num_channels
;c
++)
1321 voice
->mChans
[c
].mWetParams
[i
].LowPass
.copyParamsFrom(lowpass
);
1322 voice
->mChans
[c
].mWetParams
[i
].HighPass
.copyParamsFrom(highpass
);
1327 void CalcNonAttnSourceParams(Voice
*voice
, const VoiceProps
*props
, const ContextBase
*context
)
1329 DeviceBase
*Device
{context
->mDevice
};
1330 EffectSlot
*SendSlots
[MAX_SENDS
];
1332 voice
->mDirect
.Buffer
= Device
->Dry
.Buffer
;
1333 for(uint i
{0};i
< Device
->NumAuxSends
;i
++)
1335 SendSlots
[i
] = props
->Send
[i
].Slot
;
1336 if(!SendSlots
[i
] || SendSlots
[i
]->EffectType
== EffectSlotType::None
)
1338 SendSlots
[i
] = nullptr;
1339 voice
->mSend
[i
].Buffer
= {};
1342 voice
->mSend
[i
].Buffer
= SendSlots
[i
]->Wet
.Buffer
;
1345 /* Calculate the stepping value */
1346 const auto Pitch
= static_cast<float>(voice
->mFrequency
) /
1347 static_cast<float>(Device
->Frequency
) * props
->Pitch
;
1348 if(Pitch
> float{MaxPitch
})
1349 voice
->mStep
= MaxPitch
<<MixerFracBits
;
1351 voice
->mStep
= maxu(fastf2u(Pitch
* MixerFracOne
), 1);
1352 voice
->mResampler
= PrepareResampler(props
->mResampler
, voice
->mStep
, &voice
->mResampleState
);
1354 /* Calculate gains */
1355 GainTriplet DryGain
;
1356 DryGain
.Base
= minf(clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
) * props
->Direct
.Gain
*
1357 context
->mParams
.Gain
, GainMixMax
);
1358 DryGain
.HF
= props
->Direct
.GainHF
;
1359 DryGain
.LF
= props
->Direct
.GainLF
;
1360 GainTriplet WetGain
[MAX_SENDS
];
1361 for(uint i
{0};i
< Device
->NumAuxSends
;i
++)
1363 WetGain
[i
].Base
= minf(clampf(props
->Gain
, props
->MinGain
, props
->MaxGain
) *
1364 props
->Send
[i
].Gain
* context
->mParams
.Gain
, GainMixMax
);
1365 WetGain
[i
].HF
= props
->Send
[i
].GainHF
;
1366 WetGain
[i
].LF
= props
->Send
[i
].GainLF
;
1369 CalcPanningAndFilters(voice
, 0.0f
, 0.0f
, -1.0f
, 0.0f
, 0.0f
, DryGain
, WetGain
, SendSlots
, props
,
1370 context
->mParams
, Device
);
1373 void CalcAttnSourceParams(Voice
*voice
, const VoiceProps
*props
, const ContextBase
*context
)
1375 DeviceBase
*Device
{context
->mDevice
};
1376 const uint NumSends
{Device
->NumAuxSends
};
1378 /* Set mixing buffers and get send parameters. */
1379 voice
->mDirect
.Buffer
= Device
->Dry
.Buffer
;
1380 EffectSlot
*SendSlots
[MAX_SENDS
];
1381 uint UseDryAttnForRoom
{0};
1382 for(uint i
{0};i
< NumSends
;i
++)
1384 SendSlots
[i
] = props
->Send
[i
].Slot
;
1385 if(!SendSlots
[i
] || SendSlots
[i
]->EffectType
== EffectSlotType::None
)
1386 SendSlots
[i
] = nullptr;
1387 else if(!SendSlots
[i
]->AuxSendAuto
)
1389 /* If the slot's auxiliary send auto is off, the data sent to the
1390 * effect slot is the same as the dry path, sans filter effects.
1392 UseDryAttnForRoom
|= 1u<<i
;
1396 voice
->mSend
[i
].Buffer
= {};
1398 voice
->mSend
[i
].Buffer
= SendSlots
[i
]->Wet
.Buffer
;
1401 /* Transform source to listener space (convert to head relative) */
1402 alu::Vector Position
{props
->Position
[0], props
->Position
[1], props
->Position
[2], 1.0f
};
1403 alu::Vector Velocity
{props
->Velocity
[0], props
->Velocity
[1], props
->Velocity
[2], 0.0f
};
1404 alu::Vector Direction
{props
->Direction
[0], props
->Direction
[1], props
->Direction
[2], 0.0f
};
1405 if(!props
->HeadRelative
)
1407 /* Transform source vectors */
1408 Position
= context
->mParams
.Matrix
* (Position
- context
->mParams
.Position
);
1409 Velocity
= context
->mParams
.Matrix
* Velocity
;
1410 Direction
= context
->mParams
.Matrix
* Direction
;
1414 /* Offset the source velocity to be relative of the listener velocity */
1415 Velocity
+= context
->mParams
.Velocity
;
1418 const bool directional
{Direction
.normalize() > 0.0f
};
1419 alu::Vector ToSource
{Position
[0], Position
[1], Position
[2], 0.0f
};
1420 const float Distance
{ToSource
.normalize()};
1422 /* Calculate distance attenuation */
1423 float ClampedDist
{Distance
};
1424 float DryGainBase
{props
->Gain
};
1425 float WetGainBase
{props
->Gain
};
1427 switch(context
->mParams
.SourceDistanceModel
? props
->mDistanceModel
1428 : context
->mParams
.mDistanceModel
)
1430 case DistanceModel::InverseClamped
:
1431 if(props
->MaxDistance
< props
->RefDistance
) break;
1432 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1434 case DistanceModel::Inverse
:
1435 if(props
->RefDistance
> 0.0f
)
1437 float dist
{lerpf(props
->RefDistance
, ClampedDist
, props
->RolloffFactor
)};
1438 if(dist
> 0.0f
) DryGainBase
*= props
->RefDistance
/ dist
;
1440 dist
= lerpf(props
->RefDistance
, ClampedDist
, props
->RoomRolloffFactor
);
1441 if(dist
> 0.0f
) WetGainBase
*= props
->RefDistance
/ dist
;
1445 case DistanceModel::LinearClamped
:
1446 if(props
->MaxDistance
< props
->RefDistance
) break;
1447 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1449 case DistanceModel::Linear
:
1450 if(props
->MaxDistance
!= props
->RefDistance
)
1452 float attn
{(ClampedDist
-props
->RefDistance
) /
1453 (props
->MaxDistance
-props
->RefDistance
) * props
->RolloffFactor
};
1454 DryGainBase
*= maxf(1.0f
- attn
, 0.0f
);
1456 attn
= (ClampedDist
-props
->RefDistance
) /
1457 (props
->MaxDistance
-props
->RefDistance
) * props
->RoomRolloffFactor
;
1458 WetGainBase
*= maxf(1.0f
- attn
, 0.0f
);
1462 case DistanceModel::ExponentClamped
:
1463 if(props
->MaxDistance
< props
->RefDistance
) break;
1464 ClampedDist
= clampf(ClampedDist
, props
->RefDistance
, props
->MaxDistance
);
1466 case DistanceModel::Exponent
:
1467 if(ClampedDist
> 0.0f
&& props
->RefDistance
> 0.0f
)
1469 const float dist_ratio
{ClampedDist
/props
->RefDistance
};
1470 DryGainBase
*= std::pow(dist_ratio
, -props
->RolloffFactor
);
1471 WetGainBase
*= std::pow(dist_ratio
, -props
->RoomRolloffFactor
);
1475 case DistanceModel::Disable
:
1479 /* Calculate directional soundcones */
1480 float ConeHF
{1.0f
}, WetConeHF
{1.0f
};
1481 if(directional
&& props
->InnerAngle
< 360.0f
)
1483 static constexpr float Rad2Deg
{static_cast<float>(180.0 / al::numbers::pi
)};
1484 const float Angle
{Rad2Deg
*2.0f
* std::acos(-Direction
.dot_product(ToSource
)) * ConeScale
};
1486 float ConeGain
{1.0f
};
1487 if(Angle
>= props
->OuterAngle
)
1489 ConeGain
= props
->OuterGain
;
1490 ConeHF
= lerpf(1.0f
, props
->OuterGainHF
, props
->DryGainHFAuto
);
1492 else if(Angle
>= props
->InnerAngle
)
1494 const float scale
{(Angle
-props
->InnerAngle
) / (props
->OuterAngle
-props
->InnerAngle
)};
1495 ConeGain
= lerpf(1.0f
, props
->OuterGain
, scale
);
1496 ConeHF
= lerpf(1.0f
, props
->OuterGainHF
, scale
* props
->DryGainHFAuto
);
1499 DryGainBase
*= ConeGain
;
1500 WetGainBase
*= lerpf(1.0f
, ConeGain
, props
->WetGainAuto
);
1502 WetConeHF
= lerpf(1.0f
, ConeHF
, props
->WetGainHFAuto
);
1505 /* Apply gain and frequency filters */
1506 DryGainBase
= clampf(DryGainBase
, props
->MinGain
, props
->MaxGain
) * context
->mParams
.Gain
;
1507 WetGainBase
= clampf(WetGainBase
, props
->MinGain
, props
->MaxGain
) * context
->mParams
.Gain
;
1509 GainTriplet DryGain
{};
1510 DryGain
.Base
= minf(DryGainBase
* props
->Direct
.Gain
, GainMixMax
);
1511 DryGain
.HF
= ConeHF
* props
->Direct
.GainHF
;
1512 DryGain
.LF
= props
->Direct
.GainLF
;
1513 GainTriplet WetGain
[MAX_SENDS
]{};
1514 for(uint i
{0};i
< NumSends
;i
++)
1516 /* If this effect slot's Auxiliary Send Auto is off, then use the dry
1517 * path distance and cone attenuation, otherwise use the wet (room)
1518 * path distance and cone attenuation. The send filter is used instead
1519 * of the direct filter, regardless.
1521 const bool use_room
{!(UseDryAttnForRoom
&(1u<<i
))};
1522 const float gain
{use_room
? WetGainBase
: DryGainBase
};
1523 WetGain
[i
].Base
= minf(gain
* props
->Send
[i
].Gain
, GainMixMax
);
1524 WetGain
[i
].HF
= (use_room
? WetConeHF
: ConeHF
) * props
->Send
[i
].GainHF
;
1525 WetGain
[i
].LF
= props
->Send
[i
].GainLF
;
1528 /* Distance-based air absorption and initial send decay. */
1529 if(Distance
> props
->RefDistance
) [[likely
]]
1531 const float distance_base
{(Distance
-props
->RefDistance
) * props
->RolloffFactor
};
1532 const float distance_meters
{distance_base
* context
->mParams
.MetersPerUnit
};
1533 const float dryabsorb
{distance_meters
* props
->AirAbsorptionFactor
};
1534 if(dryabsorb
> std::numeric_limits
<float>::epsilon())
1535 DryGain
.HF
*= std::pow(context
->mParams
.AirAbsorptionGainHF
, dryabsorb
);
1537 /* If the source's Auxiliary Send Filter Gain Auto is off, no extra
1538 * adjustment is applied to the send gains.
1540 for(uint i
{props
->WetGainAuto
? 0u : NumSends
};i
< NumSends
;++i
)
1542 if(!SendSlots
[i
] || !(SendSlots
[i
]->DecayTime
> 0.0f
))
1545 auto calc_attenuation
= [](float distance
, float refdist
, float rolloff
) noexcept
1547 const float dist
{lerpf(refdist
, distance
, rolloff
)};
1548 if(dist
> refdist
) return refdist
/ dist
;
1552 /* The reverb effect's room rolloff factor always applies to an
1553 * inverse distance rolloff model.
1555 WetGain
[i
].Base
*= calc_attenuation(Distance
, props
->RefDistance
,
1556 SendSlots
[i
]->RoomRolloff
);
1558 if(distance_meters
> std::numeric_limits
<float>::epsilon())
1559 WetGain
[i
].HF
*= std::pow(SendSlots
[i
]->AirAbsorptionGainHF
, distance_meters
);
1561 /* If this effect slot's Auxiliary Send Auto is off, don't apply
1562 * the automatic initial reverb decay (should the reverb's room
1563 * rolloff still apply?).
1565 if(!SendSlots
[i
]->AuxSendAuto
)
1568 GainTriplet DecayDistance
;
1569 /* Calculate the distances to where this effect's decay reaches
1572 DecayDistance
.Base
= SendSlots
[i
]->DecayTime
* SpeedOfSoundMetersPerSec
;
1573 DecayDistance
.LF
= DecayDistance
.Base
* SendSlots
[i
]->DecayLFRatio
;
1574 DecayDistance
.HF
= DecayDistance
.Base
* SendSlots
[i
]->DecayHFRatio
;
1575 if(SendSlots
[i
]->DecayHFLimit
)
1577 const float airAbsorption
{SendSlots
[i
]->AirAbsorptionGainHF
};
1578 if(airAbsorption
< 1.0f
)
1580 /* Calculate the distance to where this effect's air
1581 * absorption reaches -60dB, and limit the effect's HF
1582 * decay distance (so it doesn't take any longer to decay
1583 * than the air would allow).
1585 static constexpr float log10_decaygain
{-3.0f
/*std::log10(ReverbDecayGain)*/};
1586 const float absorb_dist
{log10_decaygain
/ std::log10(airAbsorption
)};
1587 DecayDistance
.HF
= minf(absorb_dist
, DecayDistance
.HF
);
1591 const float baseAttn
= calc_attenuation(Distance
, props
->RefDistance
,
1592 props
->RolloffFactor
);
1594 /* Apply a decay-time transformation to the wet path, based on the
1595 * source distance. The initial decay of the reverb effect is
1596 * calculated and applied to the wet path.
1598 const float fact
{distance_base
/ DecayDistance
.Base
};
1599 const float gain
{std::pow(ReverbDecayGain
, fact
)*(1.0f
-baseAttn
) + baseAttn
};
1600 WetGain
[i
].Base
*= gain
;
1604 const float hffact
{distance_base
/ DecayDistance
.HF
};
1605 const float gainhf
{std::pow(ReverbDecayGain
, hffact
)*(1.0f
-baseAttn
) + baseAttn
};
1606 WetGain
[i
].HF
*= minf(gainhf
/gain
, 1.0f
);
1607 const float lffact
{distance_base
/ DecayDistance
.LF
};
1608 const float gainlf
{std::pow(ReverbDecayGain
, lffact
)*(1.0f
-baseAttn
) + baseAttn
};
1609 WetGain
[i
].LF
*= minf(gainlf
/gain
, 1.0f
);
1615 /* Initial source pitch */
1616 float Pitch
{props
->Pitch
};
1618 /* Calculate velocity-based doppler effect */
1619 float DopplerFactor
{props
->DopplerFactor
* context
->mParams
.DopplerFactor
};
1620 if(DopplerFactor
> 0.0f
)
1622 const alu::Vector
&lvelocity
= context
->mParams
.Velocity
;
1623 float vss
{Velocity
.dot_product(ToSource
) * -DopplerFactor
};
1624 float vls
{lvelocity
.dot_product(ToSource
) * -DopplerFactor
};
1626 const float SpeedOfSound
{context
->mParams
.SpeedOfSound
};
1627 if(!(vls
< SpeedOfSound
))
1629 /* Listener moving away from the source at the speed of sound.
1630 * Sound waves can't catch it.
1634 else if(!(vss
< SpeedOfSound
))
1636 /* Source moving toward the listener at the speed of sound. Sound
1637 * waves bunch up to extreme frequencies.
1639 Pitch
= std::numeric_limits
<float>::infinity();
1643 /* Source and listener movement is nominal. Calculate the proper
1646 Pitch
*= (SpeedOfSound
-vls
) / (SpeedOfSound
-vss
);
1650 /* Adjust pitch based on the buffer and output frequencies, and calculate
1651 * fixed-point stepping value.
1653 Pitch
*= static_cast<float>(voice
->mFrequency
) / static_cast<float>(Device
->Frequency
);
1654 if(Pitch
> float{MaxPitch
})
1655 voice
->mStep
= MaxPitch
<<MixerFracBits
;
1657 voice
->mStep
= maxu(fastf2u(Pitch
* MixerFracOne
), 1);
1658 voice
->mResampler
= PrepareResampler(props
->mResampler
, voice
->mStep
, &voice
->mResampleState
);
1661 if(props
->Radius
> Distance
)
1662 spread
= al::numbers::pi_v
<float>*2.0f
- Distance
/props
->Radius
*al::numbers::pi_v
<float>;
1663 else if(Distance
> 0.0f
)
1664 spread
= std::asin(props
->Radius
/Distance
) * 2.0f
;
1666 CalcPanningAndFilters(voice
, ToSource
[0]*XScale
, ToSource
[1]*YScale
, ToSource
[2]*ZScale
,
1667 Distance
, spread
, DryGain
, WetGain
, SendSlots
, props
, context
->mParams
, Device
);
1670 void CalcSourceParams(Voice
*voice
, ContextBase
*context
, bool force
)
1672 VoicePropsItem
*props
{voice
->mUpdate
.exchange(nullptr, std::memory_order_acq_rel
)};
1673 if(!props
&& !force
) return;
1677 voice
->mProps
= *props
;
1679 AtomicReplaceHead(context
->mFreeVoiceProps
, props
);
1682 if((voice
->mProps
.DirectChannels
!= DirectMode::Off
&& voice
->mFmtChannels
!= FmtMono
1683 && !IsAmbisonic(voice
->mFmtChannels
))
1684 || voice
->mProps
.mSpatializeMode
== SpatializeMode::Off
1685 || (voice
->mProps
.mSpatializeMode
==SpatializeMode::Auto
&& voice
->mFmtChannels
!= FmtMono
))
1686 CalcNonAttnSourceParams(voice
, &voice
->mProps
, context
);
1688 CalcAttnSourceParams(voice
, &voice
->mProps
, context
);
1692 void SendSourceStateEvent(ContextBase
*context
, uint id
, VChangeState state
)
1694 RingBuffer
*ring
{context
->mAsyncEvents
.get()};
1695 auto evt_vec
= ring
->getWriteVector();
1696 if(evt_vec
.first
.len
< 1) return;
1698 AsyncEvent
*evt
{al::construct_at(reinterpret_cast<AsyncEvent
*>(evt_vec
.first
.buf
),
1699 AsyncEvent::SourceStateChange
)};
1700 evt
->u
.srcstate
.id
= id
;
1703 case VChangeState::Reset
:
1704 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Reset
;
1706 case VChangeState::Stop
:
1707 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Stop
;
1709 case VChangeState::Play
:
1710 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Play
;
1712 case VChangeState::Pause
:
1713 evt
->u
.srcstate
.state
= AsyncEvent::SrcState::Pause
;
1715 /* Shouldn't happen. */
1716 case VChangeState::Restart
:
1720 ring
->writeAdvance(1);
1723 void ProcessVoiceChanges(ContextBase
*ctx
)
1725 VoiceChange
*cur
{ctx
->mCurrentVoiceChange
.load(std::memory_order_acquire
)};
1726 VoiceChange
*next
{cur
->mNext
.load(std::memory_order_acquire
)};
1729 const auto enabledevt
= ctx
->mEnabledEvts
.load(std::memory_order_acquire
);
1733 bool sendevt
{false};
1734 if(cur
->mState
== VChangeState::Reset
|| cur
->mState
== VChangeState::Stop
)
1736 if(Voice
*voice
{cur
->mVoice
})
1738 voice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
1739 voice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
1740 /* A source ID indicates the voice was playing or paused, which
1741 * gets a reset/stop event.
1743 sendevt
= voice
->mSourceID
.exchange(0u, std::memory_order_relaxed
) != 0u;
1744 Voice::State oldvstate
{Voice::Playing
};
1745 voice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1746 std::memory_order_relaxed
, std::memory_order_acquire
);
1747 voice
->mPendingChange
.store(false, std::memory_order_release
);
1749 /* Reset state change events are always sent, even if the voice is
1750 * already stopped or even if there is no voice.
1752 sendevt
|= (cur
->mState
== VChangeState::Reset
);
1754 else if(cur
->mState
== VChangeState::Pause
)
1756 Voice
*voice
{cur
->mVoice
};
1757 Voice::State oldvstate
{Voice::Playing
};
1758 sendevt
= voice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1759 std::memory_order_release
, std::memory_order_acquire
);
1761 else if(cur
->mState
== VChangeState::Play
)
1763 /* NOTE: When playing a voice, sending a source state change event
1764 * depends if there's an old voice to stop and if that stop is
1765 * successful. If there is no old voice, a playing event is always
1766 * sent. If there is an old voice, an event is sent only if the
1767 * voice is already stopped.
1769 if(Voice
*oldvoice
{cur
->mOldVoice
})
1771 oldvoice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
1772 oldvoice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
1773 oldvoice
->mSourceID
.store(0u, std::memory_order_relaxed
);
1774 Voice::State oldvstate
{Voice::Playing
};
1775 sendevt
= !oldvoice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1776 std::memory_order_relaxed
, std::memory_order_acquire
);
1777 oldvoice
->mPendingChange
.store(false, std::memory_order_release
);
1782 Voice
*voice
{cur
->mVoice
};
1783 voice
->mPlayState
.store(Voice::Playing
, std::memory_order_release
);
1785 else if(cur
->mState
== VChangeState::Restart
)
1787 /* Restarting a voice never sends a source change event. */
1788 Voice
*oldvoice
{cur
->mOldVoice
};
1789 oldvoice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
1790 oldvoice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
1791 /* If there's no sourceID, the old voice finished so don't start
1792 * the new one at its new offset.
1794 if(oldvoice
->mSourceID
.exchange(0u, std::memory_order_relaxed
) != 0u)
1796 /* Otherwise, set the voice to stopping if it's not already (it
1797 * might already be, if paused), and play the new voice as
1800 Voice::State oldvstate
{Voice::Playing
};
1801 oldvoice
->mPlayState
.compare_exchange_strong(oldvstate
, Voice::Stopping
,
1802 std::memory_order_relaxed
, std::memory_order_acquire
);
1804 Voice
*voice
{cur
->mVoice
};
1805 voice
->mPlayState
.store((oldvstate
== Voice::Playing
) ? Voice::Playing
1806 : Voice::Stopped
, std::memory_order_release
);
1808 oldvoice
->mPendingChange
.store(false, std::memory_order_release
);
1810 if(sendevt
&& enabledevt
.test(AsyncEvent::SourceStateChange
))
1811 SendSourceStateEvent(ctx
, cur
->mSourceID
, cur
->mState
);
1813 next
= cur
->mNext
.load(std::memory_order_acquire
);
1815 ctx
->mCurrentVoiceChange
.store(cur
, std::memory_order_release
);
1818 void ProcessParamUpdates(ContextBase
*ctx
, const EffectSlotArray
&slots
,
1819 const al::span
<Voice
*> voices
)
1821 ProcessVoiceChanges(ctx
);
1823 IncrementRef(ctx
->mUpdateCount
);
1824 if(!ctx
->mHoldUpdates
.load(std::memory_order_acquire
)) [[likely
]]
1826 bool force
{CalcContextParams(ctx
)};
1827 auto sorted_slots
= const_cast<EffectSlot
**>(slots
.data() + slots
.size());
1828 for(EffectSlot
*slot
: slots
)
1829 force
|= CalcEffectSlotParams(slot
, sorted_slots
, ctx
);
1831 for(Voice
*voice
: voices
)
1833 /* Only update voices that have a source. */
1834 if(voice
->mSourceID
.load(std::memory_order_relaxed
) != 0)
1835 CalcSourceParams(voice
, ctx
, force
);
1838 IncrementRef(ctx
->mUpdateCount
);
1841 void ProcessContexts(DeviceBase
*device
, const uint SamplesToDo
)
1843 ASSUME(SamplesToDo
> 0);
1845 const nanoseconds curtime
{device
->ClockBase
+
1846 nanoseconds
{seconds
{device
->SamplesDone
}}/device
->Frequency
};
1848 for(ContextBase
*ctx
: *device
->mContexts
.load(std::memory_order_acquire
))
1850 const EffectSlotArray
&auxslots
= *ctx
->mActiveAuxSlots
.load(std::memory_order_acquire
);
1851 const al::span
<Voice
*> voices
{ctx
->getVoicesSpanAcquired()};
1853 /* Process pending propery updates for objects on the context. */
1854 ProcessParamUpdates(ctx
, auxslots
, voices
);
1856 /* Clear auxiliary effect slot mixing buffers. */
1857 for(EffectSlot
*slot
: auxslots
)
1859 for(auto &buffer
: slot
->Wet
.Buffer
)
1863 /* Process voices that have a playing source. */
1864 for(Voice
*voice
: voices
)
1866 const Voice::State vstate
{voice
->mPlayState
.load(std::memory_order_acquire
)};
1867 if(vstate
!= Voice::Stopped
&& vstate
!= Voice::Pending
)
1868 voice
->mix(vstate
, ctx
, curtime
, SamplesToDo
);
1871 /* Process effects. */
1872 if(const size_t num_slots
{auxslots
.size()})
1874 auto slots
= auxslots
.data();
1875 auto slots_end
= slots
+ num_slots
;
1877 /* Sort the slots into extra storage, so that effect slots come
1878 * before their effect slot target (or their targets' target).
1880 const al::span
<EffectSlot
*> sorted_slots
{const_cast<EffectSlot
**>(slots_end
),
1882 /* Skip sorting if it has already been done. */
1883 if(!sorted_slots
[0])
1885 /* First, copy the slots to the sorted list, then partition the
1886 * sorted list so that all slots without a target slot go to
1889 std::copy(slots
, slots_end
, sorted_slots
.begin());
1890 auto split_point
= std::partition(sorted_slots
.begin(), sorted_slots
.end(),
1891 [](const EffectSlot
*slot
) noexcept
-> bool
1892 { return slot
->Target
!= nullptr; });
1893 /* There must be at least one slot without a slot target. */
1894 assert(split_point
!= sorted_slots
.end());
1896 /* Simple case: no more than 1 slot has a target slot. Either
1897 * all slots go right to the output, or the remaining one must
1898 * target an already-partitioned slot.
1900 if(split_point
- sorted_slots
.begin() > 1)
1902 /* At least two slots target other slots. Starting from the
1903 * back of the sorted list, continue partitioning the front
1904 * of the list given each target until all targets are
1905 * accounted for. This ensures all slots without a target
1906 * go last, all slots directly targeting those last slots
1907 * go second-to-last, all slots directly targeting those
1908 * second-last slots go third-to-last, etc.
1910 auto next_target
= sorted_slots
.end();
1912 /* This shouldn't happen, but if there's unsorted slots
1913 * left that don't target any sorted slots, they can't
1914 * contribute to the output, so leave them.
1916 if(next_target
== split_point
) [[unlikely
]]
1920 split_point
= std::partition(sorted_slots
.begin(), split_point
,
1921 [next_target
](const EffectSlot
*slot
) noexcept
-> bool
1922 { return slot
->Target
!= *next_target
; });
1923 } while(split_point
- sorted_slots
.begin() > 1);
1927 for(const EffectSlot
*slot
: sorted_slots
)
1929 EffectState
*state
{slot
->mEffectState
.get()};
1930 state
->process(SamplesToDo
, slot
->Wet
.Buffer
, state
->mOutTarget
);
1934 /* Signal the event handler if there are any events to read. */
1935 RingBuffer
*ring
{ctx
->mAsyncEvents
.get()};
1936 if(ring
->readSpace() > 0)
1937 ctx
->mEventSem
.post();
1942 void ApplyDistanceComp(const al::span
<FloatBufferLine
> Samples
, const size_t SamplesToDo
,
1943 const DistanceComp::ChanData
*distcomp
)
1945 ASSUME(SamplesToDo
> 0);
1947 for(auto &chanbuffer
: Samples
)
1949 const float gain
{distcomp
->Gain
};
1950 const size_t base
{distcomp
->Length
};
1951 float *distbuf
{al::assume_aligned
<16>(distcomp
->Buffer
)};
1957 float *inout
{al::assume_aligned
<16>(chanbuffer
.data())};
1958 auto inout_end
= inout
+ SamplesToDo
;
1959 if(SamplesToDo
>= base
) [[likely
]]
1961 auto delay_end
= std::rotate(inout
, inout_end
- base
, inout_end
);
1962 std::swap_ranges(inout
, delay_end
, distbuf
);
1966 auto delay_start
= std::swap_ranges(inout
, inout_end
, distbuf
);
1967 std::rotate(distbuf
, delay_start
, distbuf
+ base
);
1969 std::transform(inout
, inout_end
, inout
, [gain
](auto a
){ return a
* gain
; });
1973 void ApplyDither(const al::span
<FloatBufferLine
> Samples
, uint
*dither_seed
,
1974 const float quant_scale
, const size_t SamplesToDo
)
1976 ASSUME(SamplesToDo
> 0);
1978 /* Dithering. Generate whitenoise (uniform distribution of random values
1979 * between -1 and +1) and add it to the sample values, after scaling up to
1980 * the desired quantization depth amd before rounding.
1982 const float invscale
{1.0f
/ quant_scale
};
1983 uint seed
{*dither_seed
};
1984 auto dither_sample
= [&seed
,invscale
,quant_scale
](const float sample
) noexcept
-> float
1986 float val
{sample
* quant_scale
};
1987 uint rng0
{dither_rng(&seed
)};
1988 uint rng1
{dither_rng(&seed
)};
1989 val
+= static_cast<float>(rng0
*(1.0/UINT_MAX
) - rng1
*(1.0/UINT_MAX
));
1990 return fast_roundf(val
) * invscale
;
1992 for(FloatBufferLine
&inout
: Samples
)
1993 std::transform(inout
.begin(), inout
.begin()+SamplesToDo
, inout
.begin(), dither_sample
);
1994 *dither_seed
= seed
;
1998 /* Base template left undefined. Should be marked =delete, but Clang 3.8.1
1999 * chokes on that given the inline specializations.
2001 template<typename T
>
2002 inline T
SampleConv(float) noexcept
;
2004 template<> inline float SampleConv(float val
) noexcept
2006 template<> inline int32_t SampleConv(float val
) noexcept
2008 /* Floats have a 23-bit mantissa, plus an implied 1 bit and a sign bit.
2009 * This means a normalized float has at most 25 bits of signed precision.
2010 * When scaling and clamping for a signed 32-bit integer, these following
2011 * values are the best a float can give.
2013 return fastf2i(clampf(val
*2147483648.0f
, -2147483648.0f
, 2147483520.0f
));
2015 template<> inline int16_t SampleConv(float val
) noexcept
2016 { return static_cast<int16_t>(fastf2i(clampf(val
*32768.0f
, -32768.0f
, 32767.0f
))); }
2017 template<> inline int8_t SampleConv(float val
) noexcept
2018 { return static_cast<int8_t>(fastf2i(clampf(val
*128.0f
, -128.0f
, 127.0f
))); }
2020 /* Define unsigned output variations. */
2021 template<> inline uint32_t SampleConv(float val
) noexcept
2022 { return static_cast<uint32_t>(SampleConv
<int32_t>(val
)) + 2147483648u; }
2023 template<> inline uint16_t SampleConv(float val
) noexcept
2024 { return static_cast<uint16_t>(SampleConv
<int16_t>(val
) + 32768); }
2025 template<> inline uint8_t SampleConv(float val
) noexcept
2026 { return static_cast<uint8_t>(SampleConv
<int8_t>(val
) + 128); }
2028 template<DevFmtType T
>
2029 void Write(const al::span
<const FloatBufferLine
> InBuffer
, void *OutBuffer
, const size_t Offset
,
2030 const size_t SamplesToDo
, const size_t FrameStep
)
2032 ASSUME(FrameStep
> 0);
2033 ASSUME(SamplesToDo
> 0);
2035 DevFmtType_t
<T
> *outbase
{static_cast<DevFmtType_t
<T
>*>(OutBuffer
) + Offset
*FrameStep
};
2037 for(const FloatBufferLine
&inbuf
: InBuffer
)
2039 DevFmtType_t
<T
> *out
{outbase
++};
2040 auto conv_sample
= [FrameStep
,&out
](const float s
) noexcept
-> void
2042 *out
= SampleConv
<DevFmtType_t
<T
>>(s
);
2045 std::for_each(inbuf
.begin(), inbuf
.begin()+SamplesToDo
, conv_sample
);
2048 if(const size_t extra
{FrameStep
- c
})
2050 const auto silence
= SampleConv
<DevFmtType_t
<T
>>(0.0f
);
2051 for(size_t i
{0};i
< SamplesToDo
;++i
)
2053 std::fill_n(outbase
, extra
, silence
);
2054 outbase
+= FrameStep
;
2061 uint
DeviceBase::renderSamples(const uint numSamples
)
2063 const uint samplesToDo
{minu(numSamples
, BufferLineSize
)};
2065 /* Clear main mixing buffers. */
2066 for(FloatBufferLine
&buffer
: MixBuffer
)
2069 /* Increment the mix count at the start (lsb should now be 1). */
2070 IncrementRef(MixCount
);
2072 /* Process and mix each context's sources and effects. */
2073 ProcessContexts(this, samplesToDo
);
2075 /* Increment the clock time. Every second's worth of samples is converted
2076 * and added to clock base so that large sample counts don't overflow
2077 * during conversion. This also guarantees a stable conversion.
2079 SamplesDone
+= samplesToDo
;
2080 ClockBase
+= std::chrono::seconds
{SamplesDone
/ Frequency
};
2081 SamplesDone
%= Frequency
;
2083 /* Increment the mix count at the end (lsb should now be 0). */
2084 IncrementRef(MixCount
);
2086 /* Apply any needed post-process for finalizing the Dry mix to the RealOut
2087 * (Ambisonic decode, UHJ encode, etc).
2089 postProcess(samplesToDo
);
2091 /* Apply compression, limiting sample amplitude if needed or desired. */
2092 if(Limiter
) Limiter
->process(samplesToDo
, RealOut
.Buffer
.data());
2094 /* Apply delays and attenuation for mismatched speaker distances. */
2096 ApplyDistanceComp(RealOut
.Buffer
, samplesToDo
, ChannelDelays
->mChannels
.data());
2098 /* Apply dithering. The compressor should have left enough headroom for the
2099 * dither noise to not saturate.
2101 if(DitherDepth
> 0.0f
)
2102 ApplyDither(RealOut
.Buffer
, &DitherSeed
, DitherDepth
, samplesToDo
);
2107 void DeviceBase::renderSamples(const al::span
<float*> outBuffers
, const uint numSamples
)
2109 FPUCtl mixer_mode
{};
2111 while(const uint todo
{numSamples
- total
})
2113 const uint samplesToDo
{renderSamples(todo
)};
2115 auto *srcbuf
= RealOut
.Buffer
.data();
2116 for(auto *dstbuf
: outBuffers
)
2118 std::copy_n(srcbuf
->data(), samplesToDo
, dstbuf
+ total
);
2122 total
+= samplesToDo
;
2126 void DeviceBase::renderSamples(void *outBuffer
, const uint numSamples
, const size_t frameStep
)
2128 FPUCtl mixer_mode
{};
2130 while(const uint todo
{numSamples
- total
})
2132 const uint samplesToDo
{renderSamples(todo
)};
2134 if(outBuffer
) [[likely
]]
2136 /* Finally, interleave and convert samples, writing to the device's
2141 #define HANDLE_WRITE(T) case T: \
2142 Write<T>(RealOut.Buffer, outBuffer, total, samplesToDo, frameStep); break;
2143 HANDLE_WRITE(DevFmtByte
)
2144 HANDLE_WRITE(DevFmtUByte
)
2145 HANDLE_WRITE(DevFmtShort
)
2146 HANDLE_WRITE(DevFmtUShort
)
2147 HANDLE_WRITE(DevFmtInt
)
2148 HANDLE_WRITE(DevFmtUInt
)
2149 HANDLE_WRITE(DevFmtFloat
)
2154 total
+= samplesToDo
;
2158 void DeviceBase::handleDisconnect(const char *msg
, ...)
2160 IncrementRef(MixCount
);
2161 if(Connected
.exchange(false, std::memory_order_acq_rel
))
2163 AsyncEvent evt
{AsyncEvent::Disconnected
};
2166 va_start(args
, msg
);
2167 int msglen
{vsnprintf(evt
.u
.disconnect
.msg
, sizeof(evt
.u
.disconnect
.msg
), msg
, args
)};
2170 if(msglen
< 0 || static_cast<size_t>(msglen
) >= sizeof(evt
.u
.disconnect
.msg
))
2171 evt
.u
.disconnect
.msg
[sizeof(evt
.u
.disconnect
.msg
)-1] = 0;
2173 for(ContextBase
*ctx
: *mContexts
.load())
2175 if(ctx
->mEnabledEvts
.load(std::memory_order_acquire
).test(AsyncEvent::Disconnected
))
2177 RingBuffer
*ring
{ctx
->mAsyncEvents
.get()};
2178 auto evt_data
= ring
->getWriteVector().first
;
2179 if(evt_data
.len
> 0)
2181 al::construct_at(reinterpret_cast<AsyncEvent
*>(evt_data
.buf
), evt
);
2182 ring
->writeAdvance(1);
2183 ctx
->mEventSem
.post();
2187 if(!ctx
->mStopVoicesOnDisconnect
)
2189 ProcessVoiceChanges(ctx
);
2193 auto voicelist
= ctx
->getVoicesSpanAcquired();
2194 auto stop_voice
= [](Voice
*voice
) -> void
2196 voice
->mCurrentBuffer
.store(nullptr, std::memory_order_relaxed
);
2197 voice
->mLoopBuffer
.store(nullptr, std::memory_order_relaxed
);
2198 voice
->mSourceID
.store(0u, std::memory_order_relaxed
);
2199 voice
->mPlayState
.store(Voice::Stopped
, std::memory_order_release
);
2201 std::for_each(voicelist
.begin(), voicelist
.end(), stop_voice
);
2204 IncrementRef(MixCount
);