Use std::string for the disconnect event message
[openal-soft.git] / alc / alu.cpp
blob7f8503dce8ad3dfcfe08a3859be214037f4c6132
1 /**
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
21 #include "config.h"
23 #include "alu.h"
25 #include <algorithm>
26 #include <array>
27 #include <atomic>
28 #include <cassert>
29 #include <chrono>
30 #include <climits>
31 #include <cstdarg>
32 #include <cstdint>
33 #include <cstdio>
34 #include <cstdlib>
35 #include <functional>
36 #include <iterator>
37 #include <limits>
38 #include <memory>
39 #include <new>
40 #include <optional>
41 #include <utility>
43 #include "almalloc.h"
44 #include "alnumbers.h"
45 #include "alnumeric.h"
46 #include "alspan.h"
47 #include "alstring.h"
48 #include "atomic.h"
49 #include "core/ambidefs.h"
50 #include "core/async_event.h"
51 #include "core/bformatdec.h"
52 #include "core/bs2b.h"
53 #include "core/bsinc_defs.h"
54 #include "core/bsinc_tables.h"
55 #include "core/bufferline.h"
56 #include "core/buffer_storage.h"
57 #include "core/context.h"
58 #include "core/cpu_caps.h"
59 #include "core/cubic_tables.h"
60 #include "core/devformat.h"
61 #include "core/device.h"
62 #include "core/effects/base.h"
63 #include "core/effectslot.h"
64 #include "core/filters/biquad.h"
65 #include "core/filters/nfc.h"
66 #include "core/fpu_ctrl.h"
67 #include "core/hrtf.h"
68 #include "core/mastering.h"
69 #include "core/mixer.h"
70 #include "core/mixer/defs.h"
71 #include "core/mixer/hrtfdefs.h"
72 #include "core/resampler_limits.h"
73 #include "core/uhjfilter.h"
74 #include "core/voice.h"
75 #include "core/voice_change.h"
76 #include "intrusive_ptr.h"
77 #include "opthelpers.h"
78 #include "ringbuffer.h"
79 #include "strutils.h"
80 #include "vecmat.h"
81 #include "vector.h"
83 struct CTag;
84 #ifdef HAVE_SSE
85 struct SSETag;
86 #endif
87 #ifdef HAVE_SSE2
88 struct SSE2Tag;
89 #endif
90 #ifdef HAVE_SSE4_1
91 struct SSE4Tag;
92 #endif
93 #ifdef HAVE_NEON
94 struct NEONTag;
95 #endif
96 struct PointTag;
97 struct LerpTag;
98 struct CubicTag;
99 struct BSincTag;
100 struct FastBSincTag;
103 static_assert(!(MaxResamplerPadding&1), "MaxResamplerPadding is not a multiple of two");
106 namespace {
108 using uint = unsigned int;
109 using namespace std::chrono;
110 using namespace std::string_view_literals;
112 float InitConeScale()
114 float ret{1.0f};
115 if(auto optval = al::getenv("__ALSOFT_HALF_ANGLE_CONES"))
117 if(al::case_compare(*optval, "true"sv) == 0
118 || strtol(optval->c_str(), nullptr, 0) == 1)
119 ret *= 0.5f;
121 return ret;
123 /* Cone scalar */
124 const float ConeScale{InitConeScale()};
126 /* Localized scalars for mono sources (initialized in aluInit, after
127 * configuration is loaded).
129 float XScale{1.0f};
130 float YScale{1.0f};
131 float ZScale{1.0f};
133 /* Source distance scale for NFC filters. */
134 float NfcScale{1.0f};
137 using HrtfDirectMixerFunc = void(*)(const FloatBufferSpan LeftOut, const FloatBufferSpan RightOut,
138 const al::span<const FloatBufferLine> InSamples, const al::span<float2> AccumSamples,
139 const al::span<float,BufferLineSize> TempBuf, const al::span<HrtfChannelState> ChanState,
140 const size_t IrSize, const size_t SamplesToDo);
142 HrtfDirectMixerFunc MixDirectHrtf{MixDirectHrtf_<CTag>};
144 inline HrtfDirectMixerFunc SelectHrtfMixer()
146 #ifdef HAVE_NEON
147 if((CPUCapFlags&CPU_CAP_NEON))
148 return MixDirectHrtf_<NEONTag>;
149 #endif
150 #ifdef HAVE_SSE
151 if((CPUCapFlags&CPU_CAP_SSE))
152 return MixDirectHrtf_<SSETag>;
153 #endif
155 return MixDirectHrtf_<CTag>;
159 inline void BsincPrepare(const uint increment, BsincState *state, const BSincTable *table)
161 size_t si{BSincScaleCount - 1};
162 float sf{0.0f};
164 if(increment > MixerFracOne)
166 sf = MixerFracOne/static_cast<float>(increment) - table->scaleBase;
167 sf = std::max(0.0f, BSincScaleCount*sf*table->scaleRange - 1.0f);
168 si = float2uint(sf);
169 /* The interpolation factor is fit to this diagonally-symmetric curve
170 * to reduce the transition ripple caused by interpolating different
171 * scales of the sinc function.
173 sf = 1.0f - std::cos(std::asin(sf - static_cast<float>(si)));
176 state->sf = sf;
177 state->m = table->m[si];
178 state->l = (state->m/2) - 1;
179 state->filter = table->Tab.subspan(table->filterOffset[si]);
182 inline ResamplerFunc SelectResampler(Resampler resampler, uint increment)
184 switch(resampler)
186 case Resampler::Point:
187 return Resample_<PointTag,CTag>;
188 case Resampler::Linear:
189 #ifdef HAVE_NEON
190 if((CPUCapFlags&CPU_CAP_NEON))
191 return Resample_<LerpTag,NEONTag>;
192 #endif
193 #ifdef HAVE_SSE4_1
194 if((CPUCapFlags&CPU_CAP_SSE4_1))
195 return Resample_<LerpTag,SSE4Tag>;
196 #endif
197 #ifdef HAVE_SSE2
198 if((CPUCapFlags&CPU_CAP_SSE2))
199 return Resample_<LerpTag,SSE2Tag>;
200 #endif
201 return Resample_<LerpTag,CTag>;
202 case Resampler::Spline:
203 case Resampler::Gaussian:
204 #ifdef HAVE_NEON
205 if((CPUCapFlags&CPU_CAP_NEON))
206 return Resample_<CubicTag,NEONTag>;
207 #endif
208 #ifdef HAVE_SSE4_1
209 if((CPUCapFlags&CPU_CAP_SSE4_1))
210 return Resample_<CubicTag,SSE4Tag>;
211 #endif
212 #ifdef HAVE_SSE2
213 if((CPUCapFlags&CPU_CAP_SSE2))
214 return Resample_<CubicTag,SSE2Tag>;
215 #endif
216 #ifdef HAVE_SSE
217 if((CPUCapFlags&CPU_CAP_SSE))
218 return Resample_<CubicTag,SSETag>;
219 #endif
220 return Resample_<CubicTag,CTag>;
221 case Resampler::BSinc12:
222 case Resampler::BSinc24:
223 if(increment > MixerFracOne)
225 #ifdef HAVE_NEON
226 if((CPUCapFlags&CPU_CAP_NEON))
227 return Resample_<BSincTag,NEONTag>;
228 #endif
229 #ifdef HAVE_SSE
230 if((CPUCapFlags&CPU_CAP_SSE))
231 return Resample_<BSincTag,SSETag>;
232 #endif
233 return Resample_<BSincTag,CTag>;
235 /* fall-through */
236 case Resampler::FastBSinc12:
237 case Resampler::FastBSinc24:
238 #ifdef HAVE_NEON
239 if((CPUCapFlags&CPU_CAP_NEON))
240 return Resample_<FastBSincTag,NEONTag>;
241 #endif
242 #ifdef HAVE_SSE
243 if((CPUCapFlags&CPU_CAP_SSE))
244 return Resample_<FastBSincTag,SSETag>;
245 #endif
246 return Resample_<FastBSincTag,CTag>;
249 return Resample_<PointTag,CTag>;
252 } // namespace
254 void aluInit(CompatFlagBitset flags, const float nfcscale)
256 MixDirectHrtf = SelectHrtfMixer();
257 XScale = flags.test(CompatFlags::ReverseX) ? -1.0f : 1.0f;
258 YScale = flags.test(CompatFlags::ReverseY) ? -1.0f : 1.0f;
259 ZScale = flags.test(CompatFlags::ReverseZ) ? -1.0f : 1.0f;
261 NfcScale = std::clamp(nfcscale, 0.0001f, 10000.0f);
265 ResamplerFunc PrepareResampler(Resampler resampler, uint increment, InterpState *state)
267 switch(resampler)
269 case Resampler::Point:
270 case Resampler::Linear:
271 break;
272 case Resampler::Spline:
273 state->emplace<CubicState>(al::span{gSplineFilter.mTable});
274 break;
275 case Resampler::Gaussian:
276 state->emplace<CubicState>(al::span{gGaussianFilter.mTable});
277 break;
278 case Resampler::FastBSinc12:
279 case Resampler::BSinc12:
280 BsincPrepare(increment, &state->emplace<BsincState>(), &gBSinc12);
281 break;
282 case Resampler::FastBSinc24:
283 case Resampler::BSinc24:
284 BsincPrepare(increment, &state->emplace<BsincState>(), &gBSinc24);
285 break;
287 return SelectResampler(resampler, increment);
291 void DeviceBase::ProcessHrtf(const size_t SamplesToDo)
293 /* HRTF is stereo output only. */
294 const size_t lidx{RealOut.ChannelIndex[FrontLeft]};
295 const size_t ridx{RealOut.ChannelIndex[FrontRight]};
297 MixDirectHrtf(RealOut.Buffer[lidx], RealOut.Buffer[ridx], Dry.Buffer, HrtfAccumData,
298 mHrtfState->mTemp, mHrtfState->mChannels, mHrtfState->mIrSize, SamplesToDo);
301 void DeviceBase::ProcessAmbiDec(const size_t SamplesToDo)
303 AmbiDecoder->process(RealOut.Buffer, Dry.Buffer, SamplesToDo);
306 void DeviceBase::ProcessAmbiDecStablized(const size_t SamplesToDo)
308 /* Decode with front image stablization. */
309 const size_t lidx{RealOut.ChannelIndex[FrontLeft]};
310 const size_t ridx{RealOut.ChannelIndex[FrontRight]};
311 const size_t cidx{RealOut.ChannelIndex[FrontCenter]};
313 AmbiDecoder->processStablize(RealOut.Buffer, Dry.Buffer, lidx, ridx, cidx, SamplesToDo);
316 void DeviceBase::ProcessUhj(const size_t SamplesToDo)
318 /* UHJ is stereo output only. */
319 const size_t lidx{RealOut.ChannelIndex[FrontLeft]};
320 const size_t ridx{RealOut.ChannelIndex[FrontRight]};
322 /* Encode to stereo-compatible 2-channel UHJ output. */
323 mUhjEncoder->encode(RealOut.Buffer[lidx].data(), RealOut.Buffer[ridx].data(),
324 {{Dry.Buffer[0].data(), Dry.Buffer[1].data(), Dry.Buffer[2].data()}}, SamplesToDo);
327 void DeviceBase::ProcessBs2b(const size_t SamplesToDo)
329 /* First, decode the ambisonic mix to the "real" output. */
330 AmbiDecoder->process(RealOut.Buffer, Dry.Buffer, SamplesToDo);
332 /* BS2B is stereo output only. */
333 const size_t lidx{RealOut.ChannelIndex[FrontLeft]};
334 const size_t ridx{RealOut.ChannelIndex[FrontRight]};
336 /* Now apply the BS2B binaural/crossfeed filter. */
337 Bs2b->cross_feed(RealOut.Buffer[lidx].data(), RealOut.Buffer[ridx].data(), SamplesToDo);
341 namespace {
343 /* This RNG method was created based on the math found in opusdec. It's quick,
344 * and starting with a seed value of 22222, is suitable for generating
345 * whitenoise.
347 inline uint dither_rng(uint *seed) noexcept
349 *seed = (*seed * 96314165) + 907633515;
350 return *seed;
354 /* Ambisonic upsampler function. It's effectively a matrix multiply. It takes
355 * an 'upsampler' and 'rotator' as the input matrices, and creates a matrix
356 * that behaves as if the B-Format input was first decoded to a speaker array
357 * at its input order, encoded back into the higher order mix, then finally
358 * rotated.
360 void UpsampleBFormatTransform(
361 const al::span<std::array<float,MaxAmbiChannels>,MaxAmbiChannels> output,
362 const al::span<const std::array<float,MaxAmbiChannels>> upsampler,
363 const al::span<const std::array<float,MaxAmbiChannels>,MaxAmbiChannels> rotator,
364 size_t ambi_order)
366 const size_t num_chans{AmbiChannelsFromOrder(ambi_order)};
367 for(size_t i{0};i < upsampler.size();++i)
368 output[i].fill(0.0f);
369 for(size_t i{0};i < upsampler.size();++i)
371 for(size_t k{0};k < num_chans;++k)
373 const float a{upsampler[i][k]};
374 /* Write the full number of channels. The compiler will have an
375 * easier time optimizing if it has a fixed length.
377 std::transform(rotator[k].cbegin(), rotator[k].cend(), output[i].cbegin(),
378 output[i].begin(), [a](float rot, float dst) noexcept { return rot*a + dst; });
384 constexpr auto GetAmbiScales(AmbiScaling scaletype) noexcept
386 switch(scaletype)
388 case AmbiScaling::FuMa: return al::span{AmbiScale::FromFuMa};
389 case AmbiScaling::SN3D: return al::span{AmbiScale::FromSN3D};
390 case AmbiScaling::UHJ: return al::span{AmbiScale::FromUHJ};
391 case AmbiScaling::N3D: break;
393 return al::span{AmbiScale::FromN3D};
396 constexpr auto GetAmbiLayout(AmbiLayout layouttype) noexcept
398 if(layouttype == AmbiLayout::FuMa) return al::span{AmbiIndex::FromFuMa};
399 return al::span{AmbiIndex::FromACN};
402 constexpr auto GetAmbi2DLayout(AmbiLayout layouttype) noexcept
404 if(layouttype == AmbiLayout::FuMa) return al::span{AmbiIndex::FromFuMa2D};
405 return al::span{AmbiIndex::FromACN2D};
409 bool CalcContextParams(ContextBase *ctx)
411 ContextProps *props{ctx->mParams.ContextUpdate.exchange(nullptr, std::memory_order_acq_rel)};
412 if(!props) return false;
414 const alu::Vector pos{props->Position[0], props->Position[1], props->Position[2], 1.0f};
415 ctx->mParams.Position = pos;
417 /* AT then UP */
418 alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
419 N.normalize();
420 alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
421 V.normalize();
422 /* Build and normalize right-vector */
423 alu::Vector U{N.cross_product(V)};
424 U.normalize();
426 const alu::Matrix rot{
427 U[0], V[0], -N[0], 0.0,
428 U[1], V[1], -N[1], 0.0,
429 U[2], V[2], -N[2], 0.0,
430 0.0, 0.0, 0.0, 1.0};
431 const alu::Vector vel{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0};
433 ctx->mParams.Matrix = rot;
434 ctx->mParams.Velocity = rot * vel;
436 ctx->mParams.Gain = props->Gain * ctx->mGainBoost;
437 ctx->mParams.MetersPerUnit = props->MetersPerUnit;
438 ctx->mParams.AirAbsorptionGainHF = props->AirAbsorptionGainHF;
440 ctx->mParams.DopplerFactor = props->DopplerFactor;
441 ctx->mParams.SpeedOfSound = props->SpeedOfSound * props->DopplerVelocity;
443 ctx->mParams.SourceDistanceModel = props->SourceDistanceModel;
444 ctx->mParams.mDistanceModel = props->mDistanceModel;
446 AtomicReplaceHead(ctx->mFreeContextProps, props);
447 return true;
450 bool CalcEffectSlotParams(EffectSlot *slot, EffectSlot **sorted_slots, ContextBase *context)
452 EffectSlotProps *props{slot->Update.exchange(nullptr, std::memory_order_acq_rel)};
453 if(!props) return false;
455 /* If the effect slot target changed, clear the first sorted entry to force
456 * a re-sort.
458 if(slot->Target != props->Target)
459 *sorted_slots = nullptr;
460 slot->Gain = props->Gain;
461 slot->AuxSendAuto = props->AuxSendAuto;
462 slot->Target = props->Target;
463 slot->EffectType = props->Type;
464 slot->mEffectProps = props->Props;
465 if(auto *reverbprops = std::get_if<ReverbProps>(&props->Props))
467 slot->RoomRolloff = reverbprops->RoomRolloffFactor;
468 slot->DecayTime = reverbprops->DecayTime;
469 slot->DecayLFRatio = reverbprops->DecayLFRatio;
470 slot->DecayHFRatio = reverbprops->DecayHFRatio;
471 slot->DecayHFLimit = reverbprops->DecayHFLimit;
472 slot->AirAbsorptionGainHF = reverbprops->AirAbsorptionGainHF;
474 else
476 slot->RoomRolloff = 0.0f;
477 slot->DecayTime = 0.0f;
478 slot->DecayLFRatio = 0.0f;
479 slot->DecayHFRatio = 0.0f;
480 slot->DecayHFLimit = false;
481 slot->AirAbsorptionGainHF = 1.0f;
484 EffectState *state{props->State.release()};
485 EffectState *oldstate{slot->mEffectState.release()};
486 slot->mEffectState.reset(state);
488 /* Only release the old state if it won't get deleted, since we can't be
489 * deleting/freeing anything in the mixer.
491 if(!oldstate->releaseIfNoDelete())
493 /* Otherwise, if it would be deleted send it off with a release event. */
494 RingBuffer *ring{context->mAsyncEvents.get()};
495 auto evt_vec = ring->getWriteVector();
496 if(evt_vec.first.len > 0) LIKELY
498 auto &evt = InitAsyncEvent<AsyncEffectReleaseEvent>(evt_vec.first.buf);
499 evt.mEffectState = oldstate;
500 ring->writeAdvance(1);
502 else
504 /* If writing the event failed, the queue was probably full. Store
505 * the old state in the property object where it can eventually be
506 * cleaned up sometime later (not ideal, but better than blocking
507 * or leaking).
509 props->State.reset(oldstate);
513 AtomicReplaceHead(context->mFreeEffectSlotProps, props);
515 const auto output = [slot,context]() -> EffectTarget
517 if(EffectSlot *target{slot->Target})
518 return EffectTarget{&target->Wet, nullptr};
519 DeviceBase *device{context->mDevice};
520 return EffectTarget{&device->Dry, &device->RealOut};
521 }();
522 state->update(context, slot, &slot->mEffectProps, output);
523 return true;
527 /* Scales the azimuth of the given vector by 3 if it's in front. Effectively
528 * scales +/-30 degrees to +/-90 degrees, leaving > +90 and < -90 alone.
530 inline std::array<float,3> ScaleAzimuthFront3(std::array<float,3> pos)
532 if(pos[2] < 0.0f)
534 /* Normalize the length of the x,z components for a 2D vector of the
535 * azimuth angle. Negate Z since {0,0,-1} is angle 0.
537 const float len2d{std::sqrt(pos[0]*pos[0] + pos[2]*pos[2])};
538 float x{pos[0] / len2d};
539 float z{-pos[2] / len2d};
541 /* Z > cos(pi/6) = -30 < azimuth < 30 degrees. */
542 if(z > 0.866025403785f)
544 /* Triple the angle represented by x,z. */
545 x = x*3.0f - x*x*x*4.0f;
546 z = z*z*z*4.0f - z*3.0f;
548 /* Scale the vector back to fit in 3D. */
549 pos[0] = x * len2d;
550 pos[2] = -z * len2d;
552 else
554 /* If azimuth >= 30 degrees, clamp to 90 degrees. */
555 pos[0] = std::copysign(len2d, pos[0]);
556 pos[2] = 0.0f;
559 return pos;
562 /* Scales the azimuth of the given vector by 1.5 (3/2) if it's in front. */
563 inline std::array<float,3> ScaleAzimuthFront3_2(std::array<float,3> pos)
565 if(pos[2] < 0.0f)
567 const float len2d{std::sqrt(pos[0]*pos[0] + pos[2]*pos[2])};
568 float x{pos[0] / len2d};
569 float z{-pos[2] / len2d};
571 /* Z > cos(pi/3) = -60 < azimuth < 60 degrees. */
572 if(z > 0.5f)
574 /* Halve the angle represented by x,z. */
575 x = std::copysign(std::sqrt((1.0f - z) * 0.5f), x);
576 z = std::sqrt((1.0f + z) * 0.5f);
578 /* Triple the angle represented by x,z. */
579 x = x*3.0f - x*x*x*4.0f;
580 z = z*z*z*4.0f - z*3.0f;
582 /* Scale the vector back to fit in 3D. */
583 pos[0] = x * len2d;
584 pos[2] = -z * len2d;
586 else
588 /* If azimuth >= 60 degrees, clamp to 90 degrees. */
589 pos[0] = std::copysign(len2d, pos[0]);
590 pos[2] = 0.0f;
593 return pos;
597 /* Begin ambisonic rotation helpers.
599 * Rotating first-order B-Format just needs a straight-forward X/Y/Z rotation
600 * matrix. Higher orders, however, are more complicated. The method implemented
601 * here is a recursive algorithm (the rotation for first-order is used to help
602 * generate the second-order rotation, which helps generate the third-order
603 * rotation, etc).
605 * Adapted from
606 * <https://github.com/polarch/Spherical-Harmonic-Transform/blob/master/getSHrotMtx.m>,
607 * provided under the BSD 3-Clause license.
609 * Copyright (c) 2015, Archontis Politis
610 * Copyright (c) 2019, Christopher Robinson
612 * The u, v, and w coefficients used for generating higher-order rotations are
613 * precomputed since they're constant. The second-order coefficients are
614 * followed by the third-order coefficients, etc.
616 constexpr size_t CalcRotatorSize(size_t l) noexcept
618 if(l >= 2)
619 return (l*2 + 1)*(l*2 + 1) + CalcRotatorSize(l-1);
620 return 0;
623 struct RotatorCoeffs {
624 struct CoeffValues {
625 float u, v, w;
627 std::array<CoeffValues,CalcRotatorSize(MaxAmbiOrder)> mCoeffs{};
629 RotatorCoeffs()
631 auto coeffs = mCoeffs.begin();
633 for(int l=2;l <= MaxAmbiOrder;++l)
635 for(int n{-l};n <= l;++n)
637 for(int m{-l};m <= l;++m)
639 /* compute u,v,w terms of Eq.8.1 (Table I)
641 * const bool d{m == 0}; // the delta function d_m0
642 * const double denom{(std::abs(n) == l) ?
643 * (2*l) * (2*l - 1) : (l*l - n*n)};
645 * const int abs_m{std::abs(m)};
646 * coeffs->u = std::sqrt((l*l - m*m) / denom);
647 * coeffs->v = std::sqrt((l+abs_m-1) * (l+abs_m) / denom) *
648 * (1.0+d) * (1.0 - 2.0*d) * 0.5;
649 * coeffs->w = std::sqrt((l-abs_m-1) * (l-abs_m) / denom) *
650 * (1.0-d) * -0.5;
653 const double denom{static_cast<double>((std::abs(n) == l) ?
654 (2*l) * (2*l - 1) : (l*l - n*n))};
656 if(m == 0)
658 coeffs->u = static_cast<float>(std::sqrt(l * l / denom));
659 coeffs->v = static_cast<float>(std::sqrt((l-1) * l / denom) * -1.0);
660 coeffs->w = 0.0f;
662 else
664 const int abs_m{std::abs(m)};
665 coeffs->u = static_cast<float>(std::sqrt((l*l - m*m) / denom));
666 coeffs->v = static_cast<float>(std::sqrt((l+abs_m-1) * (l+abs_m) / denom) *
667 0.5);
668 coeffs->w = static_cast<float>(std::sqrt((l-abs_m-1) * (l-abs_m) / denom) *
669 -0.5);
671 ++coeffs;
677 const RotatorCoeffs RotatorCoeffArray{};
680 * Given the matrix, pre-filled with the (zeroth- and) first-order rotation
681 * coefficients, this fills in the coefficients for the higher orders up to and
682 * including the given order. The matrix is in ACN layout.
684 void AmbiRotator(AmbiRotateMatrix &matrix, const int order)
686 /* Don't do anything for < 2nd order. */
687 if(order < 2) return;
689 auto P = [](const int i, const int l, const int a, const int n, const size_t last_band,
690 const AmbiRotateMatrix &R)
692 const float ri1{ R[ 1+2][static_cast<size_t>(i+2_z)]};
693 const float rim1{R[-1+2][static_cast<size_t>(i+2_z)]};
694 const float ri0{ R[ 0+2][static_cast<size_t>(i+2_z)]};
696 const size_t y{last_band + static_cast<size_t>(a+l-1)};
697 if(n == -l)
698 return ri1*R[last_band][y] + rim1*R[last_band + static_cast<size_t>(l-1_z)*2][y];
699 if(n == l)
700 return ri1*R[last_band + static_cast<size_t>(l-1_z)*2][y] - rim1*R[last_band][y];
701 return ri0*R[last_band + static_cast<size_t>(l-1_z+n)][y];
704 auto U = [P](const int l, const int m, const int n, const size_t last_band,
705 const AmbiRotateMatrix &R)
707 return P(0, l, m, n, last_band, R);
709 auto V = [P](const int l, const int m, const int n, const size_t last_band,
710 const AmbiRotateMatrix &R)
712 using namespace al::numbers;
713 if(m > 0)
715 const bool d{m == 1};
716 const float p0{P( 1, l, m-1, n, last_band, R)};
717 const float p1{P(-1, l, -m+1, n, last_band, R)};
718 return d ? p0*sqrt2_v<float> : (p0 - p1);
720 const bool d{m == -1};
721 const float p0{P( 1, l, m+1, n, last_band, R)};
722 const float p1{P(-1, l, -m-1, n, last_band, R)};
723 return d ? p1*sqrt2_v<float> : (p0 + p1);
725 auto W = [P](const int l, const int m, const int n, const size_t last_band,
726 const AmbiRotateMatrix &R)
728 assert(m != 0);
729 if(m > 0)
731 const float p0{P( 1, l, m+1, n, last_band, R)};
732 const float p1{P(-1, l, -m-1, n, last_band, R)};
733 return p0 + p1;
735 const float p0{P( 1, l, m-1, n, last_band, R)};
736 const float p1{P(-1, l, -m+1, n, last_band, R)};
737 return p0 - p1;
740 // compute rotation matrix of each subsequent band recursively
741 auto coeffs = RotatorCoeffArray.mCoeffs.cbegin();
742 size_t band_idx{4}, last_band{1};
743 for(int l{2};l <= order;++l)
745 size_t y{band_idx};
746 for(int n{-l};n <= l;++n,++y)
748 size_t x{band_idx};
749 for(int m{-l};m <= l;++m,++x)
751 float r{0.0f};
753 // computes Eq.8.1
754 if(const float u{coeffs->u}; u != 0.0f)
755 r += u * U(l, m, n, last_band, matrix);
756 if(const float v{coeffs->v}; v != 0.0f)
757 r += v * V(l, m, n, last_band, matrix);
758 if(const float w{coeffs->w}; w != 0.0f)
759 r += w * W(l, m, n, last_band, matrix);
761 matrix[y][x] = r;
762 ++coeffs;
765 last_band = band_idx;
766 band_idx += static_cast<uint>(l)*2_uz + 1;
769 /* End ambisonic rotation helpers. */
772 constexpr float sin30{0.5f};
773 constexpr float cos30{0.866025403785f};
774 constexpr float sin45{al::numbers::sqrt2_v<float>*0.5f};
775 constexpr float cos45{al::numbers::sqrt2_v<float>*0.5f};
776 constexpr float sin110{ 0.939692620786f};
777 constexpr float cos110{-0.342020143326f};
779 struct ChanPosMap {
780 Channel channel;
781 std::array<float,3> pos;
785 struct GainTriplet { float Base, HF, LF; };
787 void CalcPanningAndFilters(Voice *voice, const float xpos, const float ypos, const float zpos,
788 const float Distance, const float Spread, const GainTriplet &DryGain,
789 const al::span<const GainTriplet,MaxSendCount> WetGain,
790 const al::span<EffectSlot*,MaxSendCount> SendSlots, const VoiceProps *props,
791 const ContextParams &Context, DeviceBase *Device)
793 static constexpr std::array MonoMap{
794 ChanPosMap{FrontCenter, std::array{0.0f, 0.0f, -1.0f}}
796 static constexpr std::array RearMap{
797 ChanPosMap{BackLeft, std::array{-sin30, 0.0f, cos30}},
798 ChanPosMap{BackRight, std::array{ sin30, 0.0f, cos30}},
800 static constexpr std::array QuadMap{
801 ChanPosMap{FrontLeft, std::array{-sin45, 0.0f, -cos45}},
802 ChanPosMap{FrontRight, std::array{ sin45, 0.0f, -cos45}},
803 ChanPosMap{BackLeft, std::array{-sin45, 0.0f, cos45}},
804 ChanPosMap{BackRight, std::array{ sin45, 0.0f, cos45}},
806 static constexpr std::array X51Map{
807 ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}},
808 ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}},
809 ChanPosMap{FrontCenter, std::array{ 0.0f, 0.0f, -1.0f}},
810 ChanPosMap{LFE, {}},
811 ChanPosMap{SideLeft, std::array{-sin110, 0.0f, -cos110}},
812 ChanPosMap{SideRight, std::array{ sin110, 0.0f, -cos110}},
814 static constexpr std::array X61Map{
815 ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}},
816 ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}},
817 ChanPosMap{FrontCenter, std::array{ 0.0f, 0.0f, -1.0f}},
818 ChanPosMap{LFE, {}},
819 ChanPosMap{BackCenter, std::array{ 0.0f, 0.0f, 1.0f}},
820 ChanPosMap{SideLeft, std::array{-1.0f, 0.0f, 0.0f}},
821 ChanPosMap{SideRight, std::array{ 1.0f, 0.0f, 0.0f}},
823 static constexpr std::array X71Map{
824 ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}},
825 ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}},
826 ChanPosMap{FrontCenter, std::array{ 0.0f, 0.0f, -1.0f}},
827 ChanPosMap{LFE, {}},
828 ChanPosMap{BackLeft, std::array{-sin30, 0.0f, cos30}},
829 ChanPosMap{BackRight, std::array{ sin30, 0.0f, cos30}},
830 ChanPosMap{SideLeft, std::array{ -1.0f, 0.0f, 0.0f}},
831 ChanPosMap{SideRight, std::array{ 1.0f, 0.0f, 0.0f}},
834 std::array StereoMap{
835 ChanPosMap{FrontLeft, std::array{-sin30, 0.0f, -cos30}},
836 ChanPosMap{FrontRight, std::array{ sin30, 0.0f, -cos30}},
839 const auto Frequency = static_cast<float>(Device->Frequency);
840 const uint NumSends{Device->NumAuxSends};
842 const size_t num_channels{voice->mChans.size()};
843 ASSUME(num_channels > 0);
845 for(auto &chandata : voice->mChans)
847 chandata.mDryParams.Hrtf.Target = HrtfFilter{};
848 chandata.mDryParams.Gains.Target.fill(0.0f);
849 std::for_each(chandata.mWetParams.begin(), chandata.mWetParams.begin()+NumSends,
850 [](SendParams &params) -> void { params.Gains.Target.fill(0.0f); });
853 const auto getChans = [props,&StereoMap](FmtChannels chanfmt) noexcept
854 -> std::pair<DirectMode,al::span<const ChanPosMap>>
856 switch(chanfmt)
858 case FmtMono:
859 /* Mono buffers are never played direct. */
860 return {DirectMode::Off, al::span{MonoMap}};
862 case FmtStereo:
863 case FmtMonoDup:
864 if(props->DirectChannels == DirectMode::Off)
866 for(size_t i{0};i < 2;++i)
868 /* StereoPan is counter-clockwise in radians. */
869 const float a{props->StereoPan[i]};
870 StereoMap[i].pos[0] = -std::sin(a);
871 StereoMap[i].pos[2] = -std::cos(a);
874 return {props->DirectChannels, al::span{StereoMap}};
876 case FmtRear: return {props->DirectChannels, al::span{RearMap}};
877 case FmtQuad: return {props->DirectChannels, al::span{QuadMap}};
878 case FmtX51: return {props->DirectChannels, al::span{X51Map}};
879 case FmtX61: return {props->DirectChannels, al::span{X61Map}};
880 case FmtX71: return {props->DirectChannels, al::span{X71Map}};
882 case FmtBFormat2D:
883 case FmtBFormat3D:
884 case FmtUHJ2:
885 case FmtUHJ3:
886 case FmtUHJ4:
887 case FmtSuperStereo:
888 return {DirectMode::Off, {}};
890 return {props->DirectChannels, {}};
892 const auto [DirectChannels,chans] = getChans(voice->mFmtChannels);
894 voice->mFlags.reset(VoiceHasHrtf).reset(VoiceHasNfc);
895 if(auto *decoder{voice->mDecoder.get()})
896 decoder->mWidthControl = std::min(props->EnhWidth, 0.7f);
898 const float lgain{std::min(1.0f-props->Panning, 1.0f)};
899 const float rgain{std::min(1.0f+props->Panning, 1.0f)};
900 const float mingain{std::min(lgain, rgain)};
901 auto SelectChannelGain = [lgain,rgain,mingain](const Channel chan) noexcept
903 switch(chan)
905 case FrontLeft: return lgain;
906 case FrontRight: return rgain;
907 case FrontCenter: break;
908 case LFE: break;
909 case BackLeft: return lgain;
910 case BackRight: return rgain;
911 case BackCenter: break;
912 case SideLeft: return lgain;
913 case SideRight: return rgain;
914 case TopCenter: break;
915 case TopFrontLeft: return lgain;
916 case TopFrontCenter: break;
917 case TopFrontRight: return rgain;
918 case TopBackLeft: return lgain;
919 case TopBackCenter: break;
920 case TopBackRight: return rgain;
921 case BottomFrontLeft: return lgain;
922 case BottomFrontRight: return rgain;
923 case BottomBackLeft: return lgain;
924 case BottomBackRight: return rgain;
925 case Aux0: case Aux1: case Aux2: case Aux3: case Aux4: case Aux5: case Aux6: case Aux7:
926 case Aux8: case Aux9: case Aux10: case Aux11: case Aux12: case Aux13: case Aux14:
927 case Aux15: case MaxChannels: break;
929 return mingain;
932 if(IsAmbisonic(voice->mFmtChannels))
934 /* Special handling for B-Format and UHJ sources. */
936 if(Device->AvgSpeakerDist > 0.0f && voice->mFmtChannels != FmtUHJ2
937 && voice->mFmtChannels != FmtSuperStereo)
939 if(!(Distance > std::numeric_limits<float>::epsilon()))
941 /* NOTE: The NFCtrlFilters were created with a w0 of 0, which
942 * is what we want for FOA input. The first channel may have
943 * been previously re-adjusted if panned, so reset it.
945 voice->mChans[0].mDryParams.NFCtrlFilter.adjust(0.0f);
947 else
949 /* Clamp the distance for really close sources, to prevent
950 * excessive bass.
952 const float mdist{std::max(Distance*NfcScale, Device->AvgSpeakerDist/4.0f)};
953 const float w0{SpeedOfSoundMetersPerSec / (mdist * Frequency)};
955 /* Only need to adjust the first channel of a B-Format source. */
956 voice->mChans[0].mDryParams.NFCtrlFilter.adjust(w0);
959 voice->mFlags.set(VoiceHasNfc);
962 /* Panning a B-Format sound toward some direction is easy. Just pan the
963 * first (W) channel as a normal mono sound. The angular spread is used
964 * as a directional scalar to blend between full coverage and full
965 * panning.
967 const float coverage{!(Distance > std::numeric_limits<float>::epsilon()) ? 1.0f :
968 (al::numbers::inv_pi_v<float>/2.0f * Spread)};
970 auto calc_coeffs = [xpos,ypos,zpos](RenderMode mode)
972 if(mode != RenderMode::Pairwise)
973 return CalcDirectionCoeffs(std::array{xpos, ypos, zpos}, 0.0f);
974 const auto pos = ScaleAzimuthFront3_2(std::array{xpos, ypos, zpos});
975 return CalcDirectionCoeffs(pos, 0.0f);
977 const auto scales = GetAmbiScales(voice->mAmbiScaling);
978 auto coeffs = calc_coeffs(Device->mRenderMode);
980 if(!(coverage > 0.0f))
982 ComputePanGains(&Device->Dry, coeffs, DryGain.Base*scales[0],
983 voice->mChans[0].mDryParams.Gains.Target);
984 for(uint i{0};i < NumSends;i++)
986 if(const EffectSlot *Slot{SendSlots[i]})
987 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base*scales[0],
988 voice->mChans[0].mWetParams[i].Gains.Target);
991 else
993 /* Local B-Format sources have their XYZ channels rotated according
994 * to the orientation.
996 /* AT then UP */
997 alu::Vector N{props->OrientAt[0], props->OrientAt[1], props->OrientAt[2], 0.0f};
998 N.normalize();
999 alu::Vector V{props->OrientUp[0], props->OrientUp[1], props->OrientUp[2], 0.0f};
1000 V.normalize();
1001 if(!props->HeadRelative)
1003 N = Context.Matrix * N;
1004 V = Context.Matrix * V;
1006 /* Build and normalize right-vector */
1007 alu::Vector U{N.cross_product(V)};
1008 U.normalize();
1010 /* Build a rotation matrix. Manually fill the zeroth- and first-
1011 * order elements, then construct the rotation for the higher
1012 * orders.
1014 AmbiRotateMatrix &shrot = Device->mAmbiRotateMatrix;
1015 shrot.fill(AmbiRotateMatrix::value_type{});
1017 shrot[0][0] = 1.0f;
1018 shrot[1][1] = U[0]; shrot[1][2] = -U[1]; shrot[1][3] = U[2];
1019 shrot[2][1] = -V[0]; shrot[2][2] = V[1]; shrot[2][3] = -V[2];
1020 shrot[3][1] = -N[0]; shrot[3][2] = N[1]; shrot[3][3] = -N[2];
1021 AmbiRotator(shrot, static_cast<int>(Device->mAmbiOrder));
1023 /* If the device is higher order than the voice, "upsample" the
1024 * matrix.
1026 * NOTE: Starting with second-order, a 2D upsample needs to be
1027 * applied with a 2D source and 3D output, even when they're the
1028 * same order. This is because higher orders have a height offset
1029 * on various channels (i.e. when elevation=0, those height-related
1030 * channels should be non-0).
1032 AmbiRotateMatrix &mixmatrix = Device->mAmbiRotateMatrix2;
1033 if(Device->mAmbiOrder > voice->mAmbiOrder
1034 || (Device->mAmbiOrder >= 2 && !Device->m2DMixing
1035 && Is2DAmbisonic(voice->mFmtChannels)))
1037 if(voice->mAmbiOrder == 1)
1039 const auto upsampler = Is2DAmbisonic(voice->mFmtChannels) ?
1040 al::span{AmbiScale::FirstOrder2DUp} : al::span{AmbiScale::FirstOrderUp};
1041 UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder);
1043 else if(voice->mAmbiOrder == 2)
1045 const auto upsampler = Is2DAmbisonic(voice->mFmtChannels) ?
1046 al::span{AmbiScale::SecondOrder2DUp} : al::span{AmbiScale::SecondOrderUp};
1047 UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder);
1049 else if(voice->mAmbiOrder == 3)
1051 const auto upsampler = Is2DAmbisonic(voice->mFmtChannels) ?
1052 al::span{AmbiScale::ThirdOrder2DUp} : al::span{AmbiScale::ThirdOrderUp};
1053 UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder);
1055 else if(voice->mAmbiOrder == 4)
1057 const auto upsampler = al::span{AmbiScale::FourthOrder2DUp};
1058 UpsampleBFormatTransform(mixmatrix, upsampler, shrot, Device->mAmbiOrder);
1060 else
1061 al::unreachable();
1063 else
1064 mixmatrix = shrot;
1066 /* Convert the rotation matrix for input ordering and scaling, and
1067 * whether input is 2D or 3D.
1069 const auto index_map = Is2DAmbisonic(voice->mFmtChannels) ?
1070 GetAmbi2DLayout(voice->mAmbiLayout).subspan(0) :
1071 GetAmbiLayout(voice->mAmbiLayout).subspan(0);
1073 /* Scale the panned W signal inversely to coverage (full coverage
1074 * means no panned signal), and according to the channel scaling.
1076 std::for_each(coeffs.begin(), coeffs.end(),
1077 [scale=(1.0f-coverage)*scales[0]](float &coeff) noexcept { coeff *= scale; });
1079 for(size_t c{0};c < num_channels;c++)
1081 const size_t acn{index_map[c]};
1082 const float scale{scales[acn] * coverage};
1084 /* For channel 0, combine the B-Format signal (scaled according
1085 * to the coverage amount) with the directional pan. For all
1086 * other channels, use just the (scaled) B-Format signal.
1088 std::transform(mixmatrix[acn].cbegin(), mixmatrix[acn].cend(), coeffs.begin(),
1089 coeffs.begin(), [scale](const float in, const float coeff) noexcept
1090 { return in*scale + coeff; });
1092 ComputePanGains(&Device->Dry, coeffs, DryGain.Base,
1093 voice->mChans[c].mDryParams.Gains.Target);
1095 for(uint i{0};i < NumSends;i++)
1097 if(const EffectSlot *Slot{SendSlots[i]})
1098 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base,
1099 voice->mChans[c].mWetParams[i].Gains.Target);
1102 coeffs = std::array<float,MaxAmbiChannels>{};
1106 else if(DirectChannels != DirectMode::Off && !Device->RealOut.RemixMap.empty())
1108 /* Direct source channels always play local. Skip the virtual channels
1109 * and write inputs to the matching real outputs.
1111 voice->mDirect.Buffer = Device->RealOut.Buffer;
1113 for(size_t c{0};c < num_channels;c++)
1115 const float pangain{SelectChannelGain(chans[c].channel)};
1116 if(uint idx{Device->channelIdxByName(chans[c].channel)}; idx != InvalidChannelIndex)
1117 voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base * pangain;
1118 else if(DirectChannels == DirectMode::RemixMismatch)
1120 auto match_channel = [channel=chans[c].channel](const InputRemixMap &map) noexcept
1121 { return channel == map.channel; };
1122 auto remap = std::find_if(Device->RealOut.RemixMap.cbegin(),
1123 Device->RealOut.RemixMap.cend(), match_channel);
1124 if(remap != Device->RealOut.RemixMap.cend())
1126 for(const auto &target : remap->targets)
1128 idx = Device->channelIdxByName(target.channel);
1129 if(idx != InvalidChannelIndex)
1130 voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base * pangain
1131 * target.mix;
1137 /* Auxiliary sends still use normal channel panning since they mix to
1138 * B-Format, which can't channel-match.
1140 for(size_t c{0};c < num_channels;c++)
1142 /* Skip LFE */
1143 if(chans[c].channel == LFE)
1144 continue;
1146 const float pangain{SelectChannelGain(chans[c].channel)};
1147 const auto coeffs = CalcDirectionCoeffs(chans[c].pos, 0.0f);
1149 for(uint i{0};i < NumSends;i++)
1151 if(const EffectSlot *Slot{SendSlots[i]})
1152 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * pangain,
1153 voice->mChans[c].mWetParams[i].Gains.Target);
1157 else if(Device->mRenderMode == RenderMode::Hrtf)
1159 /* Full HRTF rendering. Skip the virtual channels and render to the
1160 * real outputs.
1162 voice->mDirect.Buffer = Device->RealOut.Buffer;
1164 if(Distance > std::numeric_limits<float>::epsilon())
1166 if(voice->mFmtChannels == FmtMono)
1168 const float src_ev{std::asin(std::clamp(ypos, -1.0f, 1.0f))};
1169 const float src_az{std::atan2(xpos, -zpos)};
1171 Device->mHrtf->getCoeffs(src_ev, src_az, Distance*NfcScale, Spread,
1172 voice->mChans[0].mDryParams.Hrtf.Target.Coeffs,
1173 voice->mChans[0].mDryParams.Hrtf.Target.Delay);
1174 voice->mChans[0].mDryParams.Hrtf.Target.Gain = DryGain.Base;
1176 const auto coeffs = CalcDirectionCoeffs(std::array{xpos, ypos, zpos}, Spread);
1177 for(uint i{0};i < NumSends;i++)
1179 if(const EffectSlot *Slot{SendSlots[i]})
1180 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base,
1181 voice->mChans[0].mWetParams[i].Gains.Target);
1184 else for(size_t c{0};c < num_channels;c++)
1186 using namespace al::numbers;
1188 /* Skip LFE */
1189 if(chans[c].channel == LFE) continue;
1190 const float pangain{SelectChannelGain(chans[c].channel)};
1192 /* Warp the channel position toward the source position as the
1193 * source spread decreases. With no spread, all channels are at
1194 * the source position, at full spread (pi*2), each channel is
1195 * left unchanged.
1197 const float a{1.0f - (inv_pi_v<float>/2.0f)*Spread};
1198 std::array pos{
1199 lerpf(chans[c].pos[0], xpos, a),
1200 lerpf(chans[c].pos[1], ypos, a),
1201 lerpf(chans[c].pos[2], zpos, a)};
1202 const float len{std::sqrt(pos[0]*pos[0] + pos[1]*pos[1] + pos[2]*pos[2])};
1203 if(len < 1.0f)
1205 pos[0] /= len;
1206 pos[1] /= len;
1207 pos[2] /= len;
1210 const float ev{std::asin(std::clamp(pos[1], -1.0f, 1.0f))};
1211 const float az{std::atan2(pos[0], -pos[2])};
1213 Device->mHrtf->getCoeffs(ev, az, Distance*NfcScale, 0.0f,
1214 voice->mChans[c].mDryParams.Hrtf.Target.Coeffs,
1215 voice->mChans[c].mDryParams.Hrtf.Target.Delay);
1216 voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain.Base * pangain;
1218 const auto coeffs = CalcDirectionCoeffs(pos, 0.0f);
1219 for(uint i{0};i < NumSends;i++)
1221 if(const EffectSlot *Slot{SendSlots[i]})
1222 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * pangain,
1223 voice->mChans[c].mWetParams[i].Gains.Target);
1227 else
1229 /* With no distance, spread is only meaningful for mono sources
1230 * where it can be 0 or full (non-mono sources are always full
1231 * spread here).
1233 const float spread{Spread * float(voice->mFmtChannels == FmtMono)};
1235 /* Local sources on HRTF play with each channel panned to its
1236 * relative location around the listener, providing "virtual
1237 * speaker" responses.
1239 for(size_t c{0};c < num_channels;c++)
1241 /* Skip LFE */
1242 if(chans[c].channel == LFE)
1243 continue;
1244 const float pangain{SelectChannelGain(chans[c].channel)};
1246 /* Get the HRIR coefficients and delays for this channel
1247 * position.
1249 const float ev{std::asin(chans[c].pos[1])};
1250 const float az{std::atan2(chans[c].pos[0], -chans[c].pos[2])};
1252 Device->mHrtf->getCoeffs(ev, az, std::numeric_limits<float>::infinity(), spread,
1253 voice->mChans[c].mDryParams.Hrtf.Target.Coeffs,
1254 voice->mChans[c].mDryParams.Hrtf.Target.Delay);
1255 voice->mChans[c].mDryParams.Hrtf.Target.Gain = DryGain.Base * pangain;
1257 /* Normal panning for auxiliary sends. */
1258 const auto coeffs = CalcDirectionCoeffs(chans[c].pos, spread);
1260 for(uint i{0};i < NumSends;i++)
1262 if(const EffectSlot *Slot{SendSlots[i]})
1263 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * pangain,
1264 voice->mChans[c].mWetParams[i].Gains.Target);
1269 voice->mFlags.set(VoiceHasHrtf);
1271 else
1273 /* Non-HRTF rendering. Use normal panning to the output. */
1275 if(Distance > std::numeric_limits<float>::epsilon())
1277 /* Calculate NFC filter coefficient if needed. */
1278 if(Device->AvgSpeakerDist > 0.0f)
1280 /* Clamp the distance for really close sources, to prevent
1281 * excessive bass.
1283 const float mdist{std::max(Distance*NfcScale, Device->AvgSpeakerDist/4.0f)};
1284 const float w0{SpeedOfSoundMetersPerSec / (mdist * Frequency)};
1286 /* Adjust NFC filters. */
1287 for(size_t c{0};c < num_channels;c++)
1288 voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
1290 voice->mFlags.set(VoiceHasNfc);
1293 if(voice->mFmtChannels == FmtMono)
1295 auto calc_coeffs = [xpos,ypos,zpos,Spread](RenderMode mode)
1297 if(mode != RenderMode::Pairwise)
1298 return CalcDirectionCoeffs(std::array{xpos, ypos, zpos}, Spread);
1299 const auto pos = ScaleAzimuthFront3_2(std::array{xpos, ypos, zpos});
1300 return CalcDirectionCoeffs(pos, Spread);
1302 const auto coeffs = calc_coeffs(Device->mRenderMode);
1304 ComputePanGains(&Device->Dry, coeffs, DryGain.Base,
1305 voice->mChans[0].mDryParams.Gains.Target);
1306 for(uint i{0};i < NumSends;i++)
1308 if(const EffectSlot *Slot{SendSlots[i]})
1309 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base,
1310 voice->mChans[0].mWetParams[i].Gains.Target);
1313 else
1315 using namespace al::numbers;
1317 for(size_t c{0};c < num_channels;c++)
1319 const float pangain{SelectChannelGain(chans[c].channel)};
1321 /* Special-case LFE */
1322 if(chans[c].channel == LFE)
1324 if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
1326 const uint idx{Device->channelIdxByName(chans[c].channel)};
1327 if(idx != InvalidChannelIndex)
1328 voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base
1329 * pangain;
1331 continue;
1334 /* Warp the channel position toward the source position as
1335 * the spread decreases. With no spread, all channels are
1336 * at the source position, at full spread (pi*2), each
1337 * channel position is left unchanged.
1339 const float a{1.0f - (inv_pi_v<float>/2.0f)*Spread};
1340 std::array pos{
1341 lerpf(chans[c].pos[0], xpos, a),
1342 lerpf(chans[c].pos[1], ypos, a),
1343 lerpf(chans[c].pos[2], zpos, a)};
1344 const float len{std::sqrt(pos[0]*pos[0] + pos[1]*pos[1] + pos[2]*pos[2])};
1345 if(len < 1.0f)
1347 pos[0] /= len;
1348 pos[1] /= len;
1349 pos[2] /= len;
1352 if(Device->mRenderMode == RenderMode::Pairwise)
1353 pos = ScaleAzimuthFront3(pos);
1354 const auto coeffs = CalcDirectionCoeffs(pos, 0.0f);
1356 ComputePanGains(&Device->Dry, coeffs, DryGain.Base * pangain,
1357 voice->mChans[c].mDryParams.Gains.Target);
1358 for(uint i{0};i < NumSends;i++)
1360 if(const EffectSlot *Slot{SendSlots[i]})
1361 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * pangain,
1362 voice->mChans[c].mWetParams[i].Gains.Target);
1367 else
1369 if(Device->AvgSpeakerDist > 0.0f)
1371 /* If the source distance is 0, simulate a plane-wave by using
1372 * infinite distance, which results in a w0 of 0.
1374 static constexpr float w0{0.0f};
1375 for(size_t c{0};c < num_channels;c++)
1376 voice->mChans[c].mDryParams.NFCtrlFilter.adjust(w0);
1378 voice->mFlags.set(VoiceHasNfc);
1381 /* With no distance, spread is only meaningful for mono sources
1382 * where it can be 0 or full (non-mono sources are always full
1383 * spread here).
1385 const float spread{Spread * float(voice->mFmtChannels == FmtMono)};
1386 for(size_t c{0};c < num_channels;c++)
1388 const float pangain{SelectChannelGain(chans[c].channel)};
1390 /* Special-case LFE */
1391 if(chans[c].channel == LFE)
1393 if(Device->Dry.Buffer.data() == Device->RealOut.Buffer.data())
1395 const uint idx{Device->channelIdxByName(chans[c].channel)};
1396 if(idx != InvalidChannelIndex)
1397 voice->mChans[c].mDryParams.Gains.Target[idx] = DryGain.Base * pangain;
1399 continue;
1402 const auto coeffs = CalcDirectionCoeffs((Device->mRenderMode==RenderMode::Pairwise)
1403 ? ScaleAzimuthFront3(chans[c].pos) : chans[c].pos, spread);
1405 ComputePanGains(&Device->Dry, coeffs, DryGain.Base * pangain,
1406 voice->mChans[c].mDryParams.Gains.Target);
1407 for(uint i{0};i < NumSends;i++)
1409 if(const EffectSlot *Slot{SendSlots[i]})
1410 ComputePanGains(&Slot->Wet, coeffs, WetGain[i].Base * pangain,
1411 voice->mChans[c].mWetParams[i].Gains.Target);
1418 const float hfNorm{props->Direct.HFReference / Frequency};
1419 const float lfNorm{props->Direct.LFReference / Frequency};
1421 voice->mDirect.FilterType = AF_None;
1422 if(DryGain.HF != 1.0f) voice->mDirect.FilterType |= AF_LowPass;
1423 if(DryGain.LF != 1.0f) voice->mDirect.FilterType |= AF_HighPass;
1425 auto &lowpass = voice->mChans[0].mDryParams.LowPass;
1426 auto &highpass = voice->mChans[0].mDryParams.HighPass;
1427 lowpass.setParamsFromSlope(BiquadType::HighShelf, hfNorm, DryGain.HF, 1.0f);
1428 highpass.setParamsFromSlope(BiquadType::LowShelf, lfNorm, DryGain.LF, 1.0f);
1429 for(size_t c{1};c < num_channels;c++)
1431 voice->mChans[c].mDryParams.LowPass.copyParamsFrom(lowpass);
1432 voice->mChans[c].mDryParams.HighPass.copyParamsFrom(highpass);
1435 for(uint i{0};i < NumSends;i++)
1437 const float hfNorm{props->Send[i].HFReference / Frequency};
1438 const float lfNorm{props->Send[i].LFReference / Frequency};
1440 voice->mSend[i].FilterType = AF_None;
1441 if(WetGain[i].HF != 1.0f) voice->mSend[i].FilterType |= AF_LowPass;
1442 if(WetGain[i].LF != 1.0f) voice->mSend[i].FilterType |= AF_HighPass;
1444 auto &lowpass = voice->mChans[0].mWetParams[i].LowPass;
1445 auto &highpass = voice->mChans[0].mWetParams[i].HighPass;
1446 lowpass.setParamsFromSlope(BiquadType::HighShelf, hfNorm, WetGain[i].HF, 1.0f);
1447 highpass.setParamsFromSlope(BiquadType::LowShelf, lfNorm, WetGain[i].LF, 1.0f);
1448 for(size_t c{1};c < num_channels;c++)
1450 voice->mChans[c].mWetParams[i].LowPass.copyParamsFrom(lowpass);
1451 voice->mChans[c].mWetParams[i].HighPass.copyParamsFrom(highpass);
1456 void CalcNonAttnSourceParams(Voice *voice, const VoiceProps *props, const ContextBase *context)
1458 DeviceBase *Device{context->mDevice};
1459 std::array<EffectSlot*,MaxSendCount> SendSlots{};
1461 voice->mDirect.Buffer = Device->Dry.Buffer;
1462 for(uint i{0};i < Device->NumAuxSends;i++)
1464 SendSlots[i] = props->Send[i].Slot;
1465 if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None)
1467 SendSlots[i] = nullptr;
1468 voice->mSend[i].Buffer = {};
1470 else
1471 voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
1474 /* Calculate the stepping value */
1475 const auto Pitch = static_cast<float>(voice->mFrequency) /
1476 static_cast<float>(Device->Frequency) * props->Pitch;
1477 if(Pitch > float{MaxPitch})
1478 voice->mStep = MaxPitch<<MixerFracBits;
1479 else
1480 voice->mStep = std::max(fastf2u(Pitch * MixerFracOne), 1u);
1481 voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState);
1483 /* Calculate gains */
1484 GainTriplet DryGain{};
1485 DryGain.Base = std::min(std::clamp(props->Gain, props->MinGain, props->MaxGain) *
1486 props->Direct.Gain * context->mParams.Gain, GainMixMax);
1487 DryGain.HF = props->Direct.GainHF;
1488 DryGain.LF = props->Direct.GainLF;
1490 std::array<GainTriplet,MaxSendCount> WetGain{};
1491 for(uint i{0};i < Device->NumAuxSends;i++)
1493 WetGain[i].Base = std::min(std::clamp(props->Gain, props->MinGain, props->MaxGain) *
1494 props->Send[i].Gain * context->mParams.Gain, GainMixMax);
1495 WetGain[i].HF = props->Send[i].GainHF;
1496 WetGain[i].LF = props->Send[i].GainLF;
1499 CalcPanningAndFilters(voice, 0.0f, 0.0f, -1.0f, 0.0f, 0.0f, DryGain, WetGain, SendSlots, props,
1500 context->mParams, Device);
1503 void CalcAttnSourceParams(Voice *voice, const VoiceProps *props, const ContextBase *context)
1505 DeviceBase *Device{context->mDevice};
1506 const uint NumSends{Device->NumAuxSends};
1508 /* Set mixing buffers and get send parameters. */
1509 voice->mDirect.Buffer = Device->Dry.Buffer;
1510 std::array<EffectSlot*,MaxSendCount> SendSlots{};
1511 std::array<float,MaxSendCount> RoomRolloff{};
1512 std::bitset<MaxSendCount> UseDryAttnForRoom{0};
1513 for(uint i{0};i < NumSends;i++)
1515 SendSlots[i] = props->Send[i].Slot;
1516 if(!SendSlots[i] || SendSlots[i]->EffectType == EffectSlotType::None)
1517 SendSlots[i] = nullptr;
1518 else if(SendSlots[i]->AuxSendAuto)
1520 /* NOTE: Contrary to the EFX docs, the effect's room rolloff factor
1521 * applies to the selected distance model along with the source's
1522 * room rolloff factor, not necessarily the inverse distance model.
1524 * Generic Software also applies these rolloff factors regardless
1525 * of any setting. It doesn't seem to use the effect slot's send
1526 * auto for anything, though as far as I understand, it's supposed
1527 * to control whether the send gets the same gain/gainhf as the
1528 * direct path (excluding the filter).
1530 RoomRolloff[i] = props->RoomRolloffFactor + SendSlots[i]->RoomRolloff;
1532 else
1533 UseDryAttnForRoom.set(i);
1535 if(!SendSlots[i])
1536 voice->mSend[i].Buffer = {};
1537 else
1538 voice->mSend[i].Buffer = SendSlots[i]->Wet.Buffer;
1541 /* Transform source to listener space (convert to head relative) */
1542 alu::Vector Position{props->Position[0], props->Position[1], props->Position[2], 1.0f};
1543 alu::Vector Velocity{props->Velocity[0], props->Velocity[1], props->Velocity[2], 0.0f};
1544 alu::Vector Direction{props->Direction[0], props->Direction[1], props->Direction[2], 0.0f};
1545 if(!props->HeadRelative)
1547 /* Transform source vectors */
1548 Position = context->mParams.Matrix * (Position - context->mParams.Position);
1549 Velocity = context->mParams.Matrix * Velocity;
1550 Direction = context->mParams.Matrix * Direction;
1552 else
1554 /* Offset the source velocity to be relative of the listener velocity */
1555 Velocity += context->mParams.Velocity;
1558 const bool directional{Direction.normalize() > 0.0f};
1559 alu::Vector ToSource{Position[0], Position[1], Position[2], 0.0f};
1560 const float Distance{ToSource.normalize()};
1562 /* Calculate distance attenuation */
1563 float ClampedDist{Distance};
1564 float DryGainBase{props->Gain};
1565 std::array<float,MaxSendCount> WetGainBase{};
1566 WetGainBase.fill(props->Gain);
1568 float DryAttnBase{1.0f};
1569 switch(context->mParams.SourceDistanceModel ? props->mDistanceModel
1570 : context->mParams.mDistanceModel)
1572 case DistanceModel::InverseClamped:
1573 if(props->MaxDistance < props->RefDistance) break;
1574 ClampedDist = std::clamp(ClampedDist, props->RefDistance, props->MaxDistance);
1575 /*fall-through*/
1576 case DistanceModel::Inverse:
1577 if(props->RefDistance > 0.0f)
1579 float dist{lerpf(props->RefDistance, ClampedDist, props->RolloffFactor)};
1580 if(dist > 0.0f)
1582 DryAttnBase = props->RefDistance / dist;
1583 DryGainBase *= DryAttnBase;
1586 for(size_t i{0};i < NumSends;++i)
1588 dist = lerpf(props->RefDistance, ClampedDist, RoomRolloff[i]);
1589 if(dist > 0.0f) WetGainBase[i] *= props->RefDistance / dist;
1592 break;
1594 case DistanceModel::LinearClamped:
1595 if(props->MaxDistance < props->RefDistance) break;
1596 ClampedDist = std::clamp(ClampedDist, props->RefDistance, props->MaxDistance);
1597 /*fall-through*/
1598 case DistanceModel::Linear:
1599 if(props->MaxDistance != props->RefDistance)
1601 float attn{(ClampedDist-props->RefDistance) /
1602 (props->MaxDistance-props->RefDistance) * props->RolloffFactor};
1603 DryAttnBase = std::max(1.0f - attn, 0.0f);
1604 DryGainBase *= DryAttnBase;
1606 for(size_t i{0};i < NumSends;++i)
1608 attn = (ClampedDist-props->RefDistance) /
1609 (props->MaxDistance-props->RefDistance) * RoomRolloff[i];
1610 WetGainBase[i] *= std::max(1.0f - attn, 0.0f);
1613 break;
1615 case DistanceModel::ExponentClamped:
1616 if(props->MaxDistance < props->RefDistance) break;
1617 ClampedDist = std::clamp(ClampedDist, props->RefDistance, props->MaxDistance);
1618 /*fall-through*/
1619 case DistanceModel::Exponent:
1620 if(ClampedDist > 0.0f && props->RefDistance > 0.0f)
1622 const float dist_ratio{ClampedDist/props->RefDistance};
1623 DryAttnBase = std::pow(dist_ratio, -props->RolloffFactor);
1624 DryGainBase *= DryAttnBase;
1625 for(size_t i{0};i < NumSends;++i)
1626 WetGainBase[i] *= std::pow(dist_ratio, -RoomRolloff[i]);
1628 break;
1630 case DistanceModel::Disable:
1631 break;
1634 /* Calculate directional soundcones */
1635 float ConeHF{1.0f}, WetCone{1.0f}, WetConeHF{1.0f};
1636 if(directional && props->InnerAngle < 360.0f)
1638 static constexpr float Rad2Deg{static_cast<float>(180.0 / al::numbers::pi)};
1639 const float Angle{Rad2Deg*2.0f * std::acos(-Direction.dot_product(ToSource)) * ConeScale};
1641 float ConeGain{1.0f};
1642 if(Angle >= props->OuterAngle)
1644 ConeGain = props->OuterGain;
1645 if(props->DryGainHFAuto)
1646 ConeHF = props->OuterGainHF;
1648 else if(Angle >= props->InnerAngle)
1650 const float scale{(Angle-props->InnerAngle) / (props->OuterAngle-props->InnerAngle)};
1651 ConeGain = lerpf(1.0f, props->OuterGain, scale);
1652 if(props->DryGainHFAuto)
1653 ConeHF = lerpf(1.0f, props->OuterGainHF, scale);
1656 DryGainBase *= ConeGain;
1657 if(props->WetGainAuto)
1658 WetCone = ConeGain;
1659 if(props->WetGainHFAuto)
1660 WetConeHF = ConeHF;
1663 /* Apply gain and frequency filters */
1664 GainTriplet DryGain{};
1665 DryGainBase = std::clamp(DryGainBase, props->MinGain, props->MaxGain) * context->mParams.Gain;
1666 DryGain.Base = std::min(DryGainBase * props->Direct.Gain, GainMixMax);
1667 DryGain.HF = ConeHF * props->Direct.GainHF;
1668 DryGain.LF = props->Direct.GainLF;
1670 std::array<GainTriplet,MaxSendCount> WetGain{};
1671 for(uint i{0};i < NumSends;i++)
1673 WetGainBase[i] = std::clamp(WetGainBase[i]*WetCone, props->MinGain, props->MaxGain) *
1674 context->mParams.Gain;
1675 /* If this effect slot's Auxiliary Send Auto is off, then use the dry
1676 * path distance and cone attenuation, otherwise use the wet (room)
1677 * path distance and cone attenuation. The send filter is used instead
1678 * of the direct filter, regardless.
1680 const bool use_room{!UseDryAttnForRoom.test(i)};
1681 const float gain{use_room ? WetGainBase[i] : DryGainBase};
1682 WetGain[i].Base = std::min(gain * props->Send[i].Gain, GainMixMax);
1683 WetGain[i].HF = (use_room ? WetConeHF : ConeHF) * props->Send[i].GainHF;
1684 WetGain[i].LF = props->Send[i].GainLF;
1687 /* Distance-based air absorption and initial send decay. */
1688 if(Distance > props->RefDistance) LIKELY
1690 const float distance_base{(Distance-props->RefDistance) * props->RolloffFactor};
1691 const float distance_meters{distance_base * context->mParams.MetersPerUnit};
1692 const float dryabsorb{distance_meters * props->AirAbsorptionFactor};
1693 if(dryabsorb > std::numeric_limits<float>::epsilon())
1694 DryGain.HF *= std::pow(context->mParams.AirAbsorptionGainHF, dryabsorb);
1696 /* If the source's Auxiliary Send Filter Gain Auto is off, no extra
1697 * adjustment is applied to the send gains.
1699 for(uint i{props->WetGainAuto ? 0u : NumSends};i < NumSends;++i)
1701 if(!SendSlots[i] || !(SendSlots[i]->DecayTime > 0.0f))
1702 continue;
1704 if(distance_meters > std::numeric_limits<float>::epsilon())
1705 WetGain[i].HF *= std::pow(SendSlots[i]->AirAbsorptionGainHF, distance_meters);
1707 /* If this effect slot's Auxiliary Send Auto is off, don't apply
1708 * the automatic initial reverb decay.
1710 * NOTE: Generic Software applies the initial decay regardless of
1711 * this setting. It doesn't seem to use it for anything, only the
1712 * source's send filter gain auto flag affects this.
1714 if(!SendSlots[i]->AuxSendAuto)
1715 continue;
1717 const float DecayDistance{SendSlots[i]->DecayTime * SpeedOfSoundMetersPerSec};
1719 /* Apply a decay-time transformation to the wet path, based on the
1720 * source distance. The initial decay of the reverb effect is
1721 * calculated and applied to the wet path.
1723 * FIXME: This is very likely not correct. It more likely should
1724 * work by calculating a rolloff dynamically based on the reverb
1725 * parameters (and source distance?) and add it to the room rolloff
1726 * with the reverb and source rolloff parameters.
1728 const float baseAttn{DryAttnBase};
1729 const float fact{distance_base / DecayDistance};
1730 const float gain{std::pow(ReverbDecayGain, fact)*(1.0f-baseAttn) + baseAttn};
1731 WetGain[i].Base *= gain;
1736 /* Initial source pitch */
1737 float Pitch{props->Pitch};
1739 /* Calculate velocity-based doppler effect */
1740 float DopplerFactor{props->DopplerFactor * context->mParams.DopplerFactor};
1741 if(DopplerFactor > 0.0f)
1743 const alu::Vector &lvelocity = context->mParams.Velocity;
1744 float vss{Velocity.dot_product(ToSource) * -DopplerFactor};
1745 float vls{lvelocity.dot_product(ToSource) * -DopplerFactor};
1747 const float SpeedOfSound{context->mParams.SpeedOfSound};
1748 if(!(vls < SpeedOfSound))
1750 /* Listener moving away from the source at the speed of sound.
1751 * Sound waves can't catch it.
1753 Pitch = 0.0f;
1755 else if(!(vss < SpeedOfSound))
1757 /* Source moving toward the listener at the speed of sound. Sound
1758 * waves bunch up to extreme frequencies.
1760 Pitch = std::numeric_limits<float>::infinity();
1762 else
1764 /* Source and listener movement is nominal. Calculate the proper
1765 * doppler shift.
1767 Pitch *= (SpeedOfSound-vls) / (SpeedOfSound-vss);
1771 /* Adjust pitch based on the buffer and output frequencies, and calculate
1772 * fixed-point stepping value.
1774 Pitch *= static_cast<float>(voice->mFrequency) / static_cast<float>(Device->Frequency);
1775 if(Pitch > float{MaxPitch})
1776 voice->mStep = MaxPitch<<MixerFracBits;
1777 else
1778 voice->mStep = std::max(fastf2u(Pitch * MixerFracOne), 1u);
1779 voice->mResampler = PrepareResampler(props->mResampler, voice->mStep, &voice->mResampleState);
1781 float spread{0.0f};
1782 if(props->Radius > Distance)
1783 spread = al::numbers::pi_v<float>*2.0f - Distance/props->Radius*al::numbers::pi_v<float>;
1784 else if(Distance > 0.0f)
1785 spread = std::asin(props->Radius/Distance) * 2.0f;
1787 CalcPanningAndFilters(voice, ToSource[0]*XScale, ToSource[1]*YScale, ToSource[2]*ZScale,
1788 Distance, spread, DryGain, WetGain, SendSlots, props, context->mParams, Device);
1791 void CalcSourceParams(Voice *voice, ContextBase *context, bool force)
1793 VoicePropsItem *props{voice->mUpdate.exchange(nullptr, std::memory_order_acq_rel)};
1794 if(!props && !force) return;
1796 if(props)
1798 voice->mProps = static_cast<VoiceProps&>(*props);
1800 AtomicReplaceHead(context->mFreeVoiceProps, props);
1803 if((voice->mProps.DirectChannels != DirectMode::Off && voice->mFmtChannels != FmtMono
1804 && !IsAmbisonic(voice->mFmtChannels))
1805 || voice->mProps.mSpatializeMode == SpatializeMode::Off
1806 || (voice->mProps.mSpatializeMode==SpatializeMode::Auto && voice->mFmtChannels != FmtMono))
1807 CalcNonAttnSourceParams(voice, &voice->mProps, context);
1808 else
1809 CalcAttnSourceParams(voice, &voice->mProps, context);
1813 void SendSourceStateEvent(ContextBase *context, uint id, VChangeState state)
1815 RingBuffer *ring{context->mAsyncEvents.get()};
1816 auto evt_vec = ring->getWriteVector();
1817 if(evt_vec.first.len < 1) return;
1819 auto &evt = InitAsyncEvent<AsyncSourceStateEvent>(evt_vec.first.buf);
1820 evt.mId = id;
1821 switch(state)
1823 case VChangeState::Reset:
1824 evt.mState = AsyncSrcState::Reset;
1825 break;
1826 case VChangeState::Stop:
1827 evt.mState = AsyncSrcState::Stop;
1828 break;
1829 case VChangeState::Play:
1830 evt.mState = AsyncSrcState::Play;
1831 break;
1832 case VChangeState::Pause:
1833 evt.mState = AsyncSrcState::Pause;
1834 break;
1835 /* Shouldn't happen. */
1836 case VChangeState::Restart:
1837 al::unreachable();
1840 ring->writeAdvance(1);
1843 void ProcessVoiceChanges(ContextBase *ctx)
1845 VoiceChange *cur{ctx->mCurrentVoiceChange.load(std::memory_order_acquire)};
1846 VoiceChange *next{cur->mNext.load(std::memory_order_acquire)};
1847 if(!next) return;
1849 const auto enabledevt = ctx->mEnabledEvts.load(std::memory_order_acquire);
1850 do {
1851 cur = next;
1853 bool sendevt{false};
1854 if(cur->mState == VChangeState::Reset || cur->mState == VChangeState::Stop)
1856 if(Voice *voice{cur->mVoice})
1858 voice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
1859 voice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
1860 /* A source ID indicates the voice was playing or paused, which
1861 * gets a reset/stop event.
1863 sendevt = voice->mSourceID.exchange(0u, std::memory_order_relaxed) != 0u;
1864 Voice::State oldvstate{Voice::Playing};
1865 voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
1866 std::memory_order_relaxed, std::memory_order_acquire);
1867 voice->mPendingChange.store(false, std::memory_order_release);
1869 /* Reset state change events are always sent, even if the voice is
1870 * already stopped or even if there is no voice.
1872 sendevt |= (cur->mState == VChangeState::Reset);
1874 else if(cur->mState == VChangeState::Pause)
1876 Voice *voice{cur->mVoice};
1877 Voice::State oldvstate{Voice::Playing};
1878 sendevt = voice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
1879 std::memory_order_release, std::memory_order_acquire);
1881 else if(cur->mState == VChangeState::Play)
1883 /* NOTE: When playing a voice, sending a source state change event
1884 * depends if there's an old voice to stop and if that stop is
1885 * successful. If there is no old voice, a playing event is always
1886 * sent. If there is an old voice, an event is sent only if the
1887 * voice is already stopped.
1889 if(Voice *oldvoice{cur->mOldVoice})
1891 oldvoice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
1892 oldvoice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
1893 oldvoice->mSourceID.store(0u, std::memory_order_relaxed);
1894 Voice::State oldvstate{Voice::Playing};
1895 sendevt = !oldvoice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
1896 std::memory_order_relaxed, std::memory_order_acquire);
1897 oldvoice->mPendingChange.store(false, std::memory_order_release);
1899 else
1900 sendevt = true;
1902 Voice *voice{cur->mVoice};
1903 voice->mPlayState.store(Voice::Playing, std::memory_order_release);
1905 else if(cur->mState == VChangeState::Restart)
1907 /* Restarting a voice never sends a source change event. */
1908 Voice *oldvoice{cur->mOldVoice};
1909 oldvoice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
1910 oldvoice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
1911 /* If there's no sourceID, the old voice finished so don't start
1912 * the new one at its new offset.
1914 if(oldvoice->mSourceID.exchange(0u, std::memory_order_relaxed) != 0u)
1916 /* Otherwise, set the voice to stopping if it's not already (it
1917 * might already be, if paused), and play the new voice as
1918 * appropriate.
1920 Voice::State oldvstate{Voice::Playing};
1921 oldvoice->mPlayState.compare_exchange_strong(oldvstate, Voice::Stopping,
1922 std::memory_order_relaxed, std::memory_order_acquire);
1924 Voice *voice{cur->mVoice};
1925 voice->mPlayState.store((oldvstate == Voice::Playing) ? Voice::Playing
1926 : Voice::Stopped, std::memory_order_release);
1928 oldvoice->mPendingChange.store(false, std::memory_order_release);
1930 if(sendevt && enabledevt.test(al::to_underlying(AsyncEnableBits::SourceState)))
1931 SendSourceStateEvent(ctx, cur->mSourceID, cur->mState);
1933 next = cur->mNext.load(std::memory_order_acquire);
1934 } while(next);
1935 ctx->mCurrentVoiceChange.store(cur, std::memory_order_release);
1938 void ProcessParamUpdates(ContextBase *ctx, const al::span<EffectSlot*> slots,
1939 const al::span<EffectSlot*> sorted_slots, const al::span<Voice*> voices)
1941 ProcessVoiceChanges(ctx);
1943 IncrementRef(ctx->mUpdateCount);
1944 if(!ctx->mHoldUpdates.load(std::memory_order_acquire)) LIKELY
1946 bool force{CalcContextParams(ctx)};
1947 auto sorted_slot_base = al::to_address(sorted_slots.begin());
1948 for(EffectSlot *slot : slots)
1949 force |= CalcEffectSlotParams(slot, sorted_slot_base, ctx);
1951 for(Voice *voice : voices)
1953 /* Only update voices that have a source. */
1954 if(voice->mSourceID.load(std::memory_order_relaxed) != 0)
1955 CalcSourceParams(voice, ctx, force);
1958 IncrementRef(ctx->mUpdateCount);
1961 void ProcessContexts(DeviceBase *device, const uint SamplesToDo)
1963 ASSUME(SamplesToDo > 0);
1965 const nanoseconds curtime{device->mClockBase.load(std::memory_order_relaxed) +
1966 nanoseconds{seconds{device->mSamplesDone.load(std::memory_order_relaxed)}}/
1967 device->Frequency};
1969 for(ContextBase *ctx : *device->mContexts.load(std::memory_order_acquire))
1971 const auto auxslotspan = al::span{*ctx->mActiveAuxSlots.load(std::memory_order_acquire)};
1972 const auto auxslots = auxslotspan.first(auxslotspan.size()>>1);
1973 const auto sorted_slots = auxslotspan.last(auxslotspan.size()>>1);
1974 const al::span<Voice*> voices{ctx->getVoicesSpanAcquired()};
1976 /* Process pending property updates for objects on the context. */
1977 ProcessParamUpdates(ctx, auxslots, sorted_slots, voices);
1979 /* Clear auxiliary effect slot mixing buffers. */
1980 for(EffectSlot *slot : auxslots)
1982 for(auto &buffer : slot->Wet.Buffer)
1983 buffer.fill(0.0f);
1986 /* Process voices that have a playing source. */
1987 for(Voice *voice : voices)
1989 const Voice::State vstate{voice->mPlayState.load(std::memory_order_acquire)};
1990 if(vstate != Voice::Stopped && vstate != Voice::Pending)
1991 voice->mix(vstate, ctx, curtime, SamplesToDo);
1994 /* Process effects. */
1995 if(!auxslots.empty())
1997 /* Sort the slots into extra storage, so that effect slots come
1998 * before their effect slot target (or their targets' target). Skip
1999 * sorting if it has already been done.
2001 if(!sorted_slots[0])
2003 /* First, copy the slots to the sorted list, then partition the
2004 * sorted list so that all slots without a target slot go to
2005 * the end.
2007 std::copy(auxslots.begin(), auxslots.end(), sorted_slots.begin());
2008 auto split_point = std::partition(sorted_slots.begin(), sorted_slots.end(),
2009 [](const EffectSlot *slot) noexcept -> bool
2010 { return slot->Target != nullptr; });
2011 /* There must be at least one slot without a slot target. */
2012 assert(split_point != sorted_slots.end());
2014 /* Simple case: no more than 1 slot has a target slot. Either
2015 * all slots go right to the output, or the remaining one must
2016 * target an already-partitioned slot.
2018 if(split_point - sorted_slots.begin() > 1)
2020 /* At least two slots target other slots. Starting from the
2021 * back of the sorted list, continue partitioning the front
2022 * of the list given each target until all targets are
2023 * accounted for. This ensures all slots without a target
2024 * go last, all slots directly targeting those last slots
2025 * go second-to-last, all slots directly targeting those
2026 * second-last slots go third-to-last, etc.
2028 auto next_target = sorted_slots.end();
2029 do {
2030 /* This shouldn't happen, but if there's unsorted slots
2031 * left that don't target any sorted slots, they can't
2032 * contribute to the output, so leave them.
2034 if(next_target == split_point) UNLIKELY
2035 break;
2037 --next_target;
2038 split_point = std::partition(sorted_slots.begin(), split_point,
2039 [next_target](const EffectSlot *slot) noexcept -> bool
2040 { return slot->Target != *next_target; });
2041 } while(split_point - sorted_slots.begin() > 1);
2045 for(const EffectSlot *slot : sorted_slots)
2047 EffectState *state{slot->mEffectState.get()};
2048 state->process(SamplesToDo, slot->Wet.Buffer, state->mOutTarget);
2052 /* Signal the event handler if there are any events to read. */
2053 RingBuffer *ring{ctx->mAsyncEvents.get()};
2054 if(ring->readSpace() > 0)
2055 ctx->mEventSem.post();
2060 void ApplyDistanceComp(const al::span<FloatBufferLine> Samples, const size_t SamplesToDo,
2061 const al::span<const DistanceComp::ChanData,MaxOutputChannels> chandata)
2063 ASSUME(SamplesToDo > 0);
2065 auto distcomp = chandata.begin();
2066 for(auto &chanbuffer : Samples)
2068 const float gain{distcomp->Gain};
2069 auto distbuf = al::span{al::assume_aligned<16>(distcomp->Buffer.data()),
2070 distcomp->Buffer.size()};
2071 ++distcomp;
2073 const size_t base{distbuf.size()};
2074 if(base < 1) continue;
2076 const auto inout = al::span{al::assume_aligned<16>(chanbuffer.data()), SamplesToDo};
2077 if(SamplesToDo >= base) LIKELY
2079 auto delay_end = std::rotate(inout.begin(), inout.end()-ptrdiff_t(base), inout.end());
2080 std::swap_ranges(inout.begin(), delay_end, distbuf.begin());
2082 else
2084 auto delay_start = std::swap_ranges(inout.begin(), inout.end(), distbuf.begin());
2085 std::rotate(distbuf.begin(), delay_start, distbuf.begin()+ptrdiff_t(base));
2087 std::transform(inout.begin(), inout.end(), inout.begin(),
2088 [gain](float s) { return s*gain; });
2092 void ApplyDither(const al::span<FloatBufferLine> Samples, uint *dither_seed,
2093 const float quant_scale, const size_t SamplesToDo)
2095 static constexpr double invRNGRange{1.0 / std::numeric_limits<uint>::max()};
2096 ASSUME(SamplesToDo > 0);
2098 /* Dithering. Generate whitenoise (uniform distribution of random values
2099 * between -1 and +1) and add it to the sample values, after scaling up to
2100 * the desired quantization depth and before rounding.
2102 const float invscale{1.0f / quant_scale};
2103 uint seed{*dither_seed};
2104 auto dither_sample = [&seed,invscale,quant_scale](const float sample) noexcept -> float
2106 float val{sample * quant_scale};
2107 uint rng0{dither_rng(&seed)};
2108 uint rng1{dither_rng(&seed)};
2109 val += static_cast<float>(rng0*invRNGRange - rng1*invRNGRange);
2110 return fast_roundf(val) * invscale;
2112 for(FloatBufferLine &inout : Samples)
2113 std::transform(inout.begin(), inout.begin()+SamplesToDo, inout.begin(), dither_sample);
2114 *dither_seed = seed;
2118 /* Base template left undefined. Should be marked =delete, but Clang 3.8.1
2119 * chokes on that given the inline specializations.
2121 template<typename T>
2122 inline T SampleConv(float) noexcept;
2124 template<> inline float SampleConv(float val) noexcept
2125 { return val; }
2126 template<> inline int32_t SampleConv(float val) noexcept
2128 /* Floats have a 23-bit mantissa, plus an implied 1 bit and a sign bit.
2129 * This means a normalized float has at most 25 bits of signed precision.
2130 * When scaling and clamping for a signed 32-bit integer, these following
2131 * values are the best a float can give.
2133 return fastf2i(std::clamp(val*2147483648.0f, -2147483648.0f, 2147483520.0f));
2135 template<> inline int16_t SampleConv(float val) noexcept
2136 { return static_cast<int16_t>(fastf2i(std::clamp(val*32768.0f, -32768.0f, 32767.0f))); }
2137 template<> inline int8_t SampleConv(float val) noexcept
2138 { return static_cast<int8_t>(fastf2i(std::clamp(val*128.0f, -128.0f, 127.0f))); }
2140 /* Define unsigned output variations. */
2141 template<> inline uint32_t SampleConv(float val) noexcept
2142 { return static_cast<uint32_t>(SampleConv<int32_t>(val)) + 2147483648u; }
2143 template<> inline uint16_t SampleConv(float val) noexcept
2144 { return static_cast<uint16_t>(SampleConv<int16_t>(val) + 32768); }
2145 template<> inline uint8_t SampleConv(float val) noexcept
2146 { return static_cast<uint8_t>(SampleConv<int8_t>(val) + 128); }
2148 template<typename T>
2149 void Write(const al::span<const FloatBufferLine> InBuffer, void *OutBuffer, const size_t Offset,
2150 const size_t SamplesToDo, const size_t FrameStep)
2152 ASSUME(FrameStep > 0);
2153 ASSUME(SamplesToDo > 0);
2155 const auto output = al::span{static_cast<T*>(OutBuffer), (Offset+SamplesToDo)*FrameStep}
2156 .subspan(Offset*FrameStep);
2157 size_t c{0};
2158 for(const FloatBufferLine &inbuf : InBuffer)
2160 auto out = output.begin();
2161 auto conv_sample = [FrameStep,c,&out](const float s) noexcept
2163 out[c] = SampleConv<T>(s);
2164 out += ptrdiff_t(FrameStep);
2166 std::for_each_n(inbuf.cbegin(), SamplesToDo, conv_sample);
2167 ++c;
2169 if(const size_t extra{FrameStep - c})
2171 const auto silence = SampleConv<T>(0.0f);
2172 for(size_t i{0};i < SamplesToDo;++i)
2173 std::fill_n(&output[i*FrameStep + c], extra, silence);
2177 } // namespace
2179 uint DeviceBase::renderSamples(const uint numSamples)
2181 const uint samplesToDo{std::min(numSamples, uint{BufferLineSize})};
2183 /* Clear main mixing buffers. */
2184 for(FloatBufferLine &buffer : MixBuffer)
2185 buffer.fill(0.0f);
2188 const auto mixLock = getWriteMixLock();
2190 /* Process and mix each context's sources and effects. */
2191 ProcessContexts(this, samplesToDo);
2193 /* Every second's worth of samples is converted and added to clock base
2194 * so that large sample counts don't overflow during conversion. This
2195 * also guarantees a stable conversion.
2197 auto samplesDone = mSamplesDone.load(std::memory_order_relaxed) + samplesToDo;
2198 auto clockBase = mClockBase.load(std::memory_order_relaxed) +
2199 std::chrono::seconds{samplesDone/Frequency};
2200 mSamplesDone.store(samplesDone%Frequency, std::memory_order_relaxed);
2201 mClockBase.store(clockBase, std::memory_order_relaxed);
2204 /* Apply any needed post-process for finalizing the Dry mix to the RealOut
2205 * (Ambisonic decode, UHJ encode, etc).
2207 postProcess(samplesToDo);
2209 /* Apply compression, limiting sample amplitude if needed or desired. */
2210 if(Limiter) Limiter->process(samplesToDo, RealOut.Buffer.data());
2212 /* Apply delays and attenuation for mismatched speaker distances. */
2213 if(ChannelDelays)
2214 ApplyDistanceComp(RealOut.Buffer, samplesToDo, ChannelDelays->mChannels);
2216 /* Apply dithering. The compressor should have left enough headroom for the
2217 * dither noise to not saturate.
2219 if(DitherDepth > 0.0f)
2220 ApplyDither(RealOut.Buffer, &DitherSeed, DitherDepth, samplesToDo);
2222 return samplesToDo;
2225 void DeviceBase::renderSamples(const al::span<float*> outBuffers, const uint numSamples)
2227 FPUCtl mixer_mode{};
2228 uint total{0};
2229 while(const uint todo{numSamples - total})
2231 const uint samplesToDo{renderSamples(todo)};
2233 auto srcbuf = RealOut.Buffer.cbegin();
2234 for(auto *dstbuf : outBuffers)
2236 const auto dst = al::span{dstbuf, numSamples}.subspan(total);
2237 std::copy_n(srcbuf->cbegin(), samplesToDo, dst.begin());
2238 ++srcbuf;
2241 total += samplesToDo;
2245 void DeviceBase::renderSamples(void *outBuffer, const uint numSamples, const size_t frameStep)
2247 FPUCtl mixer_mode{};
2248 uint total{0};
2249 while(const uint todo{numSamples - total})
2251 const uint samplesToDo{renderSamples(todo)};
2253 if(outBuffer) LIKELY
2255 /* Finally, interleave and convert samples, writing to the device's
2256 * output buffer.
2258 switch(FmtType)
2260 #define HANDLE_WRITE(T) case T: \
2261 Write<DevFmtType_t<T>>(RealOut.Buffer, outBuffer, total, samplesToDo, frameStep); break;
2262 HANDLE_WRITE(DevFmtByte)
2263 HANDLE_WRITE(DevFmtUByte)
2264 HANDLE_WRITE(DevFmtShort)
2265 HANDLE_WRITE(DevFmtUShort)
2266 HANDLE_WRITE(DevFmtInt)
2267 HANDLE_WRITE(DevFmtUInt)
2268 HANDLE_WRITE(DevFmtFloat)
2269 #undef HANDLE_WRITE
2273 total += samplesToDo;
2277 void DeviceBase::handleDisconnect(const char *msg, ...)
2279 const auto mixLock = getWriteMixLock();
2281 if(Connected.exchange(false, std::memory_order_acq_rel))
2283 AsyncEvent evt{std::in_place_type<AsyncDisconnectEvent>};
2284 auto &disconnect = std::get<AsyncDisconnectEvent>(evt);
2286 /* NOLINTBEGIN(*-array-to-pointer-decay) */
2287 va_list args, args2;
2288 va_start(args, msg);
2289 va_copy(args2, args);
2290 if(int msglen{vsnprintf(nullptr, 0, msg, args)}; msglen > 0)
2292 disconnect.msg.resize(static_cast<uint>(msglen)+1_uz);
2293 vsnprintf(disconnect.msg.data(), disconnect.msg.size(), msg, args2);
2295 else
2296 disconnect.msg = "<failed constructing message>";
2297 va_end(args2);
2298 va_end(args);
2299 /* NOLINTEND(*-array-to-pointer-decay) */
2301 while(!disconnect.msg.empty() && disconnect.msg.back() == '\0')
2302 disconnect.msg.pop_back();
2304 for(ContextBase *ctx : *mContexts.load())
2306 RingBuffer *ring{ctx->mAsyncEvents.get()};
2307 auto evt_data = ring->getWriteVector().first;
2308 if(evt_data.len > 0)
2310 al::construct_at(reinterpret_cast<AsyncEvent*>(evt_data.buf), evt);
2311 ring->writeAdvance(1);
2312 ctx->mEventSem.post();
2315 if(!ctx->mStopVoicesOnDisconnect.load())
2317 ProcessVoiceChanges(ctx);
2318 continue;
2321 auto voicelist = ctx->getVoicesSpanAcquired();
2322 auto stop_voice = [](Voice *voice) -> void
2324 voice->mCurrentBuffer.store(nullptr, std::memory_order_relaxed);
2325 voice->mLoopBuffer.store(nullptr, std::memory_order_relaxed);
2326 voice->mSourceID.store(0u, std::memory_order_relaxed);
2327 voice->mPlayState.store(Voice::Stopped, std::memory_order_release);
2329 std::for_each(voicelist.begin(), voicelist.end(), stop_voice);