/** * OpenAL cross platform audio library * Copyright (C) 1999-2007 by authors. * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. * Or go to http://www.gnu.org/copyleft/lgpl.html */ #include "config.h" #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "AL/al.h" #include "AL/alc.h" #include "al/buffer.h" #include "al/event.h" #include "al/source.h" #include "alcmain.h" #include "albyte.h" #include "alconfig.h" #include "alcontext.h" #include "alnumeric.h" #include "aloptional.h" #include "alspan.h" #include "alu.h" #include "cpu_caps.h" #include "devformat.h" #include "filters/biquad.h" #include "filters/nfc.h" #include "filters/splitter.h" #include "hrtf.h" #include "inprogext.h" #include "logging.h" #include "mixer/defs.h" #include "opthelpers.h" #include "ringbuffer.h" #include "threads.h" #include "vector.h" static_assert((INT_MAX>>FRACTIONBITS)/MAX_PITCH > BUFFERSIZE, "MAX_PITCH and/or BUFFERSIZE are too large for FRACTIONBITS!"); /* BSinc24 requires up to 23 extra samples before the current position, and 24 after. */ static_assert(MAX_RESAMPLE_PADDING >= 24, "MAX_RESAMPLE_PADDING must be at least 24!"); Resampler ResamplerDefault = LinearResampler; MixerFunc MixSamples = Mix_; RowMixerFunc MixRowSamples = MixRow_; static HrtfMixerFunc MixHrtfSamples = MixHrtf_; static HrtfMixerBlendFunc MixHrtfBlendSamples = MixHrtfBlend_; static MixerFunc SelectMixer() { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Mix_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Mix_; #endif return Mix_; } static RowMixerFunc SelectRowMixer() { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixRow_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixRow_; #endif return MixRow_; } static inline HrtfMixerFunc SelectHrtfMixer() { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixHrtf_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixHrtf_; #endif return MixHrtf_; } static inline HrtfMixerBlendFunc SelectHrtfBlendMixer() { #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return MixHrtfBlend_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return MixHrtfBlend_; #endif return MixHrtfBlend_; } ResamplerFunc SelectResampler(Resampler resampler) { switch(resampler) { case PointResampler: return Resample_; case LinearResampler: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE4_1 if((CPUCapFlags&CPU_CAP_SSE4_1)) return Resample_; #endif #ifdef HAVE_SSE2 if((CPUCapFlags&CPU_CAP_SSE2)) return Resample_; #endif return Resample_; case FIR4Resampler: return Resample_; case BSinc12Resampler: case BSinc24Resampler: #ifdef HAVE_NEON if((CPUCapFlags&CPU_CAP_NEON)) return Resample_; #endif #ifdef HAVE_SSE if((CPUCapFlags&CPU_CAP_SSE)) return Resample_; #endif return Resample_; } return Resample_; } void aluInitMixer() { if(auto resopt = ConfigValueStr(nullptr, nullptr, "resampler")) { const char *str{resopt->c_str()}; if(strcasecmp(str, "point") == 0 || strcasecmp(str, "none") == 0) ResamplerDefault = PointResampler; else if(strcasecmp(str, "linear") == 0) ResamplerDefault = LinearResampler; else if(strcasecmp(str, "cubic") == 0) ResamplerDefault = FIR4Resampler; else if(strcasecmp(str, "bsinc12") == 0) ResamplerDefault = BSinc12Resampler; else if(strcasecmp(str, "bsinc24") == 0) ResamplerDefault = BSinc24Resampler; else if(strcasecmp(str, "bsinc") == 0) { WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str); ResamplerDefault = BSinc12Resampler; } else if(strcasecmp(str, "sinc4") == 0 || strcasecmp(str, "sinc8") == 0) { WARN("Resampler option \"%s\" is deprecated, using cubic\n", str); ResamplerDefault = FIR4Resampler; } else ERR("Invalid resampler: %s\n", str); } MixHrtfBlendSamples = SelectHrtfBlendMixer(); MixHrtfSamples = SelectHrtfMixer(); MixSamples = SelectMixer(); MixRowSamples = SelectRowMixer(); } namespace { /* A quick'n'dirty lookup table to decode a muLaw-encoded byte sample into a * signed 16-bit sample */ constexpr ALshort muLawDecompressionTable[256] = { -32124,-31100,-30076,-29052,-28028,-27004,-25980,-24956, -23932,-22908,-21884,-20860,-19836,-18812,-17788,-16764, -15996,-15484,-14972,-14460,-13948,-13436,-12924,-12412, -11900,-11388,-10876,-10364, -9852, -9340, -8828, -8316, -7932, -7676, -7420, -7164, -6908, -6652, -6396, -6140, -5884, -5628, -5372, -5116, -4860, -4604, -4348, -4092, -3900, -3772, -3644, -3516, -3388, -3260, -3132, -3004, -2876, -2748, -2620, -2492, -2364, -2236, -2108, -1980, -1884, -1820, -1756, -1692, -1628, -1564, -1500, -1436, -1372, -1308, -1244, -1180, -1116, -1052, -988, -924, -876, -844, -812, -780, -748, -716, -684, -652, -620, -588, -556, -524, -492, -460, -428, -396, -372, -356, -340, -324, -308, -292, -276, -260, -244, -228, -212, -196, -180, -164, -148, -132, -120, -112, -104, -96, -88, -80, -72, -64, -56, -48, -40, -32, -24, -16, -8, 0, 32124, 31100, 30076, 29052, 28028, 27004, 25980, 24956, 23932, 22908, 21884, 20860, 19836, 18812, 17788, 16764, 15996, 15484, 14972, 14460, 13948, 13436, 12924, 12412, 11900, 11388, 10876, 10364, 9852, 9340, 8828, 8316, 7932, 7676, 7420, 7164, 6908, 6652, 6396, 6140, 5884, 5628, 5372, 5116, 4860, 4604, 4348, 4092, 3900, 3772, 3644, 3516, 3388, 3260, 3132, 3004, 2876, 2748, 2620, 2492, 2364, 2236, 2108, 1980, 1884, 1820, 1756, 1692, 1628, 1564, 1500, 1436, 1372, 1308, 1244, 1180, 1116, 1052, 988, 924, 876, 844, 812, 780, 748, 716, 684, 652, 620, 588, 556, 524, 492, 460, 428, 396, 372, 356, 340, 324, 308, 292, 276, 260, 244, 228, 212, 196, 180, 164, 148, 132, 120, 112, 104, 96, 88, 80, 72, 64, 56, 48, 40, 32, 24, 16, 8, 0 }; /* A quick'n'dirty lookup table to decode an aLaw-encoded byte sample into a * signed 16-bit sample */ constexpr ALshort aLawDecompressionTable[256] = { -5504, -5248, -6016, -5760, -4480, -4224, -4992, -4736, -7552, -7296, -8064, -7808, -6528, -6272, -7040, -6784, -2752, -2624, -3008, -2880, -2240, -2112, -2496, -2368, -3776, -3648, -4032, -3904, -3264, -3136, -3520, -3392, -22016,-20992,-24064,-23040,-17920,-16896,-19968,-18944, -30208,-29184,-32256,-31232,-26112,-25088,-28160,-27136, -11008,-10496,-12032,-11520, -8960, -8448, -9984, -9472, -15104,-14592,-16128,-15616,-13056,-12544,-14080,-13568, -344, -328, -376, -360, -280, -264, -312, -296, -472, -456, -504, -488, -408, -392, -440, -424, -88, -72, -120, -104, -24, -8, -56, -40, -216, -200, -248, -232, -152, -136, -184, -168, -1376, -1312, -1504, -1440, -1120, -1056, -1248, -1184, -1888, -1824, -2016, -1952, -1632, -1568, -1760, -1696, -688, -656, -752, -720, -560, -528, -624, -592, -944, -912, -1008, -976, -816, -784, -880, -848, 5504, 5248, 6016, 5760, 4480, 4224, 4992, 4736, 7552, 7296, 8064, 7808, 6528, 6272, 7040, 6784, 2752, 2624, 3008, 2880, 2240, 2112, 2496, 2368, 3776, 3648, 4032, 3904, 3264, 3136, 3520, 3392, 22016, 20992, 24064, 23040, 17920, 16896, 19968, 18944, 30208, 29184, 32256, 31232, 26112, 25088, 28160, 27136, 11008, 10496, 12032, 11520, 8960, 8448, 9984, 9472, 15104, 14592, 16128, 15616, 13056, 12544, 14080, 13568, 344, 328, 376, 360, 280, 264, 312, 296, 472, 456, 504, 488, 408, 392, 440, 424, 88, 72, 120, 104, 24, 8, 56, 40, 216, 200, 248, 232, 152, 136, 184, 168, 1376, 1312, 1504, 1440, 1120, 1056, 1248, 1184, 1888, 1824, 2016, 1952, 1632, 1568, 1760, 1696, 688, 656, 752, 720, 560, 528, 624, 592, 944, 912, 1008, 976, 816, 784, 880, 848 }; template struct FmtTypeTraits { }; template<> struct FmtTypeTraits { using Type = ALubyte; static constexpr ALfloat to_float(const Type val) { return (val-128) * (1.0f/128.0f); } }; template<> struct FmtTypeTraits { using Type = ALshort; static constexpr ALfloat to_float(const Type val) { return val * (1.0f/32768.0f); } }; template<> struct FmtTypeTraits { using Type = ALfloat; static constexpr ALfloat to_float(const Type val) { return val; } }; template<> struct FmtTypeTraits { using Type = ALdouble; static constexpr ALfloat to_float(const Type val) { return static_cast(val); } }; template<> struct FmtTypeTraits { using Type = ALubyte; static constexpr ALfloat to_float(const Type val) { return muLawDecompressionTable[val] * (1.0f/32768.0f); } }; template<> struct FmtTypeTraits { using Type = ALubyte; static constexpr ALfloat to_float(const Type val) { return aLawDecompressionTable[val] * (1.0f/32768.0f); } }; void SendSourceStoppedEvent(ALCcontext *context, ALuint id) { RingBuffer *ring{context->mAsyncEvents.get()}; auto evt_vec = ring->getWriteVector(); if(evt_vec.first.len < 1) return; AsyncEvent *evt{new (evt_vec.first.buf) AsyncEvent{EventType_SourceStateChange}}; evt->u.srcstate.id = id; evt->u.srcstate.state = AL_STOPPED; ring->writeAdvance(1); context->mEventSem.post(); } const ALfloat *DoFilters(BiquadFilter *lpfilter, BiquadFilter *hpfilter, ALfloat *dst, const ALfloat *src, const size_t numsamples, int type) { switch(type) { case AF_None: lpfilter->clear(); hpfilter->clear(); break; case AF_LowPass: lpfilter->process(dst, src, numsamples); hpfilter->clear(); return dst; case AF_HighPass: lpfilter->clear(); hpfilter->process(dst, src, numsamples); return dst; case AF_BandPass: lpfilter->process(dst, src, numsamples); hpfilter->process(dst, dst, numsamples); return dst; } return src; } template inline void LoadSampleArray(ALfloat *RESTRICT dst, const al::byte *src, const size_t srcstep, const size_t samples) { using SampleType = typename FmtTypeTraits::Type; const SampleType *RESTRICT ssrc{reinterpret_cast(src)}; for(size_t i{0u};i < samples;i++) dst[i] = FmtTypeTraits::to_float(ssrc[i*srcstep]); } void LoadSamples(ALfloat *RESTRICT dst, const al::byte *src, const size_t srcstep, FmtType srctype, const size_t samples) { #define HANDLE_FMT(T) case T: LoadSampleArray(dst, src, srcstep, samples); break switch(srctype) { HANDLE_FMT(FmtUByte); HANDLE_FMT(FmtShort); HANDLE_FMT(FmtFloat); HANDLE_FMT(FmtDouble); HANDLE_FMT(FmtMulaw); HANDLE_FMT(FmtAlaw); } #undef HANDLE_FMT } ALfloat *LoadBufferStatic(ALbufferlistitem *BufferListItem, ALbufferlistitem *&BufferLoopItem, const size_t NumChannels, const size_t SampleSize, const size_t chan, size_t DataPosInt, al::span SrcBuffer) { const ALbuffer *Buffer{BufferListItem->mBuffer}; const ALuint LoopStart{Buffer->LoopStart}; const ALuint LoopEnd{Buffer->LoopEnd}; ASSUME(LoopEnd > LoopStart); /* If current pos is beyond the loop range, do not loop */ if(!BufferLoopItem || DataPosInt >= LoopEnd) { BufferLoopItem = nullptr; /* Load what's left to play from the buffer */ const size_t DataSize{minz(SrcBuffer.size(), Buffer->SampleLen-DataPosInt)}; const al::byte *Data{Buffer->mData.data()}; Data += (DataPosInt*NumChannels + chan)*SampleSize; LoadSamples(SrcBuffer.data(), Data, NumChannels, Buffer->mFmtType, DataSize); SrcBuffer = SrcBuffer.subspan(DataSize); } else { /* Load what's left of this loop iteration */ const size_t DataSize{minz(SrcBuffer.size(), LoopEnd-DataPosInt)}; const al::byte *Data{Buffer->mData.data()}; Data += (DataPosInt*NumChannels + chan)*SampleSize; LoadSamples(SrcBuffer.data(), Data, NumChannels, Buffer->mFmtType, DataSize); SrcBuffer = SrcBuffer.subspan(DataSize); /* Load any repeats of the loop we can to fill the buffer. */ const auto LoopSize = static_cast(LoopEnd - LoopStart); while(!SrcBuffer.empty()) { const size_t DataSize{minz(SrcBuffer.size(), LoopSize)}; const al::byte *Data{Buffer->mData.data()}; Data += (LoopStart*NumChannels + chan)*SampleSize; LoadSamples(SrcBuffer.data(), Data, NumChannels, Buffer->mFmtType, DataSize); SrcBuffer = SrcBuffer.subspan(DataSize); } } return SrcBuffer.begin(); } ALfloat *LoadBufferQueue(ALbufferlistitem *BufferListItem, ALbufferlistitem *BufferLoopItem, const size_t NumChannels, const size_t SampleSize, const size_t chan, size_t DataPosInt, al::span SrcBuffer) { /* Crawl the buffer queue to fill in the temp buffer */ while(BufferListItem && !SrcBuffer.empty()) { ALbuffer *Buffer{BufferListItem->mBuffer}; if(!(Buffer && DataPosInt < Buffer->SampleLen)) { if(Buffer) DataPosInt -= Buffer->SampleLen; BufferListItem = BufferListItem->mNext.load(std::memory_order_acquire); if(!BufferListItem) BufferListItem = BufferLoopItem; continue; } const size_t DataSize{minz(SrcBuffer.size(), Buffer->SampleLen-DataPosInt)}; const al::byte *Data{Buffer->mData.data()}; Data += (DataPosInt*NumChannels + chan)*SampleSize; LoadSamples(SrcBuffer.data(), Data, NumChannels, Buffer->mFmtType, DataSize); SrcBuffer = SrcBuffer.subspan(DataSize); if(SrcBuffer.empty()) break; DataPosInt = 0; BufferListItem = BufferListItem->mNext.load(std::memory_order_acquire); if(!BufferListItem) BufferListItem = BufferLoopItem; } return SrcBuffer.begin(); } } // namespace void ALvoice::mix(State vstate, ALCcontext *Context, const ALuint SamplesToDo) { static constexpr ALfloat SilentTarget[MAX_OUTPUT_CHANNELS]{}; ASSUME(SamplesToDo > 0); /* Get voice info */ const bool isstatic{(mFlags&VOICE_IS_STATIC) != 0}; ALuint DataPosInt{mPosition.load(std::memory_order_relaxed)}; ALuint DataPosFrac{mPositionFrac.load(std::memory_order_relaxed)}; ALbufferlistitem *BufferListItem{mCurrentBuffer.load(std::memory_order_relaxed)}; ALbufferlistitem *BufferLoopItem{mLoopBuffer.load(std::memory_order_relaxed)}; const auto NumChannels = static_cast(mNumChannels); const auto SampleSize = static_cast(mSampleSize); const auto increment = static_cast(mStep); if(increment < 1) return; ASSUME(NumChannels > 0); ASSUME(SampleSize > 0); ASSUME(increment > 0); ALCdevice *Device{Context->mDevice.get()}; const ALsizei NumSends{Device->NumAuxSends}; const ALsizei IrSize{Device->mHrtf ? Device->mHrtf->irSize : 0}; ASSUME(NumSends >= 0); ASSUME(IrSize >= 0); ResamplerFunc Resample{(increment == FRACTIONONE && DataPosFrac == 0) ? Resample_ : mResampler}; ALuint Counter{(mFlags&VOICE_IS_FADING) ? SamplesToDo : 0}; if(!Counter) { /* No fading, just overwrite the old/current params. */ for(ALuint chan{0};chan < NumChannels;chan++) { ChannelData &chandata = mChans[chan]; DirectParams &parms = chandata.mDryParams; if(!(mFlags&VOICE_HAS_HRTF)) std::copy(std::begin(parms.Gains.Target), std::end(parms.Gains.Target), std::begin(parms.Gains.Current)); else parms.Hrtf.Old = parms.Hrtf.Target; for(ALsizei send{0};send < NumSends;++send) { if(mSend[send].Buffer.empty()) continue; SendParams &parms = chandata.mWetParams[send]; std::copy(std::begin(parms.Gains.Target), std::end(parms.Gains.Target), std::begin(parms.Gains.Current)); } } } else if((mFlags&VOICE_HAS_HRTF)) { for(ALuint chan{0};chan < NumChannels;chan++) { DirectParams &parms = mChans[chan].mDryParams; if(!(parms.Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD)) { /* The old HRTF params are silent, so overwrite the old * coefficients with the new, and reset the old gain to 0. The * future mix will then fade from silence. */ parms.Hrtf.Old = parms.Hrtf.Target; parms.Hrtf.Old.Gain = 0.0f; } } } ALuint buffers_done{0u}; ALuint OutPos{0u}; do { /* Figure out how many buffer samples will be needed */ ALuint DstBufferSize{SamplesToDo - OutPos}; /* Calculate the last written dst sample pos. */ uint64_t DataSize64{DstBufferSize - 1}; /* Calculate the last read src sample pos. */ DataSize64 = (DataSize64*increment + DataPosFrac) >> FRACTIONBITS; /* +1 to get the src sample count, include padding. */ DataSize64 += 1 + MAX_RESAMPLE_PADDING*2; auto SrcBufferSize = static_cast( minu64(DataSize64, BUFFERSIZE + MAX_RESAMPLE_PADDING*2 + 1)); if(SrcBufferSize > BUFFERSIZE + MAX_RESAMPLE_PADDING*2) { SrcBufferSize = BUFFERSIZE + MAX_RESAMPLE_PADDING*2; /* If the source buffer got saturated, we can't fill the desired * dst size. Figure out how many samples we can actually mix from * this. */ DataSize64 = SrcBufferSize - MAX_RESAMPLE_PADDING*2; DataSize64 = ((DataSize64<(minu64(DataSize64, DstBufferSize)); /* Some mixers like having a multiple of 4, so try to give that * unless this is the last update. */ if(DstBufferSize < SamplesToDo-OutPos) DstBufferSize &= ~3u; } ASSUME(DstBufferSize > 0); for(ALuint chan{0};chan < NumChannels;chan++) { ChannelData &chandata = mChans[chan]; const al::span SrcData{Device->SourceData, SrcBufferSize}; /* Load the previous samples into the source data first, then load * what we can from the buffer queue. */ auto srciter = std::copy_n(chandata.mPrevSamples.begin(), MAX_RESAMPLE_PADDING, SrcData.begin()); if UNLIKELY(!BufferListItem) srciter = std::copy(chandata.mPrevSamples.begin()+MAX_RESAMPLE_PADDING, chandata.mPrevSamples.end(), srciter); else if(isstatic) srciter = LoadBufferStatic(BufferListItem, BufferLoopItem, NumChannels, SampleSize, chan, DataPosInt, {srciter, SrcData.end()}); else srciter = LoadBufferQueue(BufferListItem, BufferLoopItem, NumChannels, SampleSize, chan, DataPosInt, {srciter, SrcData.end()}); if UNLIKELY(srciter != SrcData.end()) { /* If the source buffer wasn't filled, copy the last sample for * the remaining buffer. Ideally it should have ended with * silence, but if not the gain fading should help avoid clicks * from sudden amplitude changes. */ const ALfloat sample{*(srciter-1)}; std::fill(srciter, SrcData.end(), sample); } /* Store the last source samples used for next time. */ std::copy_n(&SrcData[(increment*DstBufferSize + DataPosFrac)>>FRACTIONBITS], chandata.mPrevSamples.size(), chandata.mPrevSamples.begin()); /* Resample, then apply ambisonic upsampling as needed. */ const ALfloat *ResampledData{Resample(&mResampleState, &SrcData[MAX_RESAMPLE_PADDING], DataPosFrac, static_cast(increment), {Device->ResampledData, DstBufferSize})}; if((mFlags&VOICE_IS_AMBISONIC)) { const ALfloat hfscale{chandata.mAmbiScale}; /* Beware the evil const_cast. It's safe since it's pointing to * either SourceData or ResampledData (both non-const), but the * resample method takes the source as const float* and may * return it without copying to output, making it currently * unavoidable. */ chandata.mAmbiSplitter.applyHfScale(const_cast(ResampledData), hfscale, DstBufferSize); } /* Now filter and mix to the appropriate outputs. */ ALfloat (&FilterBuf)[BUFFERSIZE] = Device->FilteredData; { DirectParams &parms = chandata.mDryParams; const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass, FilterBuf, ResampledData, DstBufferSize, mDirect.FilterType)}; if((mFlags&VOICE_HAS_HRTF)) { const ALuint OutLIdx{GetChannelIdxByName(Device->RealOut, FrontLeft)}; const ALuint OutRIdx{GetChannelIdxByName(Device->RealOut, FrontRight)}; auto &HrtfSamples = Device->HrtfSourceData; auto &AccumSamples = Device->HrtfAccumData; const ALfloat TargetGain{UNLIKELY(vstate == ALvoice::Stopping) ? 0.0f : parms.Hrtf.Target.Gain}; ALuint fademix{0u}; /* Copy the HRTF history and new input samples into a temp * buffer. */ auto src_iter = std::copy(parms.Hrtf.State.History.begin(), parms.Hrtf.State.History.end(), std::begin(HrtfSamples)); std::copy_n(samples, DstBufferSize, src_iter); /* Copy the last used samples back into the history buffer * for later. */ std::copy_n(std::begin(HrtfSamples) + DstBufferSize, parms.Hrtf.State.History.size(), parms.Hrtf.State.History.begin()); /* Copy the current filtered values being accumulated into * the temp buffer. */ auto accum_iter = std::copy_n(parms.Hrtf.State.Values.begin(), parms.Hrtf.State.Values.size(), std::begin(AccumSamples)); /* Clear the accumulation buffer that will start getting * filled in. */ std::fill_n(accum_iter, DstBufferSize, float2{}); /* If fading, the old gain is not silence, and this is the * first mixing pass, fade between the IRs. */ if(Counter && (parms.Hrtf.Old.Gain > GAIN_SILENCE_THRESHOLD) && OutPos == 0) { fademix = minu(DstBufferSize, 128); ALfloat gain{TargetGain}; /* The new coefficients need to fade in completely * since they're replacing the old ones. To keep the * gain fading consistent, interpolate between the old * and new target gains given how much of the fade time * this mix handles. */ if LIKELY(Counter > fademix) { const ALfloat a{static_cast(fademix) / static_cast(Counter)}; gain = lerp(parms.Hrtf.Old.Gain, TargetGain, a); } MixHrtfFilter hrtfparams; hrtfparams.Coeffs = &parms.Hrtf.Target.Coeffs; hrtfparams.Delay[0] = parms.Hrtf.Target.Delay[0]; hrtfparams.Delay[1] = parms.Hrtf.Target.Delay[1]; hrtfparams.Gain = 0.0f; hrtfparams.GainStep = gain / static_cast(fademix); MixHrtfBlendSamples(mDirect.Buffer[OutLIdx], mDirect.Buffer[OutRIdx], HrtfSamples, AccumSamples, OutPos, IrSize, &parms.Hrtf.Old, &hrtfparams, fademix); /* Update the old parameters with the result. */ parms.Hrtf.Old = parms.Hrtf.Target; if(fademix < Counter) parms.Hrtf.Old.Gain = hrtfparams.Gain; else parms.Hrtf.Old.Gain = TargetGain; } if LIKELY(fademix < DstBufferSize) { const ALuint todo{DstBufferSize - fademix}; ALfloat gain{TargetGain}; /* Interpolate the target gain if the gain fading lasts * longer than this mix. */ if(Counter > DstBufferSize) { const ALfloat a{static_cast(todo) / static_cast(Counter-fademix)}; gain = lerp(parms.Hrtf.Old.Gain, TargetGain, a); } MixHrtfFilter hrtfparams; hrtfparams.Coeffs = &parms.Hrtf.Target.Coeffs; hrtfparams.Delay[0] = parms.Hrtf.Target.Delay[0]; hrtfparams.Delay[1] = parms.Hrtf.Target.Delay[1]; hrtfparams.Gain = parms.Hrtf.Old.Gain; hrtfparams.GainStep = (gain - parms.Hrtf.Old.Gain) / static_cast(todo); MixHrtfSamples(mDirect.Buffer[OutLIdx], mDirect.Buffer[OutRIdx], HrtfSamples+fademix, AccumSamples+fademix, OutPos+fademix, IrSize, &hrtfparams, todo); /* Store the interpolated gain or the final target gain * depending if the fade is done. */ if(DstBufferSize < Counter) parms.Hrtf.Old.Gain = gain; else parms.Hrtf.Old.Gain = TargetGain; } /* Copy the new in-progress accumulation values back for * the next mix. */ std::copy_n(std::begin(AccumSamples) + DstBufferSize, parms.Hrtf.State.Values.size(), parms.Hrtf.State.Values.begin()); } else if((mFlags&VOICE_HAS_NFC)) { const ALfloat *TargetGains{UNLIKELY(vstate == ALvoice::Stopping) ? SilentTarget : parms.Gains.Target}; const size_t outcount{Device->NumChannelsPerOrder[0]}; MixSamples({samples, DstBufferSize}, mDirect.Buffer.first(outcount), parms.Gains.Current, TargetGains, Counter, OutPos); const al::span nfcsamples{Device->NfcSampleData, DstBufferSize}; size_t chanoffset{outcount}; using FilterProc = void (NfcFilter::*)(float*,const float*,const size_t); auto apply_nfc = [this,&parms,samples,TargetGains,Counter,OutPos,&chanoffset,nfcsamples](const FilterProc process, const size_t outcount) -> void { if(outcount < 1) return; (parms.NFCtrlFilter.*process)(nfcsamples.data(), samples, nfcsamples.size()); MixSamples(nfcsamples, mDirect.Buffer.subspan(chanoffset, outcount), parms.Gains.Current+chanoffset, TargetGains+chanoffset, Counter, OutPos); chanoffset += outcount; }; apply_nfc(&NfcFilter::process1, Device->NumChannelsPerOrder[1]); apply_nfc(&NfcFilter::process2, Device->NumChannelsPerOrder[2]); apply_nfc(&NfcFilter::process3, Device->NumChannelsPerOrder[3]); } else { const ALfloat *TargetGains{UNLIKELY(vstate == ALvoice::Stopping) ? SilentTarget : parms.Gains.Target}; MixSamples({samples, DstBufferSize}, mDirect.Buffer, parms.Gains.Current, TargetGains, Counter, OutPos); } } for(ALsizei send{0};send < NumSends;++send) { if(mSend[send].Buffer.empty()) continue; SendParams &parms = chandata.mWetParams[send]; const ALfloat *samples{DoFilters(&parms.LowPass, &parms.HighPass, FilterBuf, ResampledData, DstBufferSize, mSend[send].FilterType)}; const ALfloat *TargetGains{UNLIKELY(vstate==ALvoice::Stopping) ? SilentTarget : parms.Gains.Target}; MixSamples({samples, DstBufferSize}, mSend[send].Buffer, parms.Gains.Current, TargetGains, Counter, OutPos); } } /* Update positions */ DataPosFrac += increment*DstBufferSize; DataPosInt += DataPosFrac>>FRACTIONBITS; DataPosFrac &= FRACTIONMASK; OutPos += DstBufferSize; Counter = maxu(DstBufferSize, Counter) - DstBufferSize; if UNLIKELY(!BufferListItem) { /* Do nothing extra when there's no buffers. */ } else if(isstatic) { if(BufferLoopItem) { /* Handle looping static source */ const ALbuffer *Buffer{BufferListItem->mBuffer}; const ALuint LoopStart{Buffer->LoopStart}; const ALuint LoopEnd{Buffer->LoopEnd}; if(DataPosInt >= LoopEnd) { assert(LoopEnd > LoopStart); DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart; } } else { /* Handle non-looping static source */ if(DataPosInt >= BufferListItem->mSampleLen) { if LIKELY(vstate == ALvoice::Playing) vstate = ALvoice::Stopped; BufferListItem = nullptr; break; } } } else while(1) { /* Handle streaming source */ if(BufferListItem->mSampleLen > DataPosInt) break; DataPosInt -= BufferListItem->mSampleLen; ++buffers_done; BufferListItem = BufferListItem->mNext.load(std::memory_order_relaxed); if(!BufferListItem && !(BufferListItem=BufferLoopItem)) { if LIKELY(vstate == ALvoice::Playing) vstate = ALvoice::Stopped; break; } } } while(OutPos < SamplesToDo); mFlags |= VOICE_IS_FADING; /* Don't update positions and buffers if we were stopping. */ if UNLIKELY(vstate == ALvoice::Stopping) { mPlayState.store(ALvoice::Stopped, std::memory_order_release); return; } /* Capture the source ID in case it's reset for stopping. */ const ALuint SourceID{mSourceID.load(std::memory_order_relaxed)}; /* Update voice info */ mPosition.store(DataPosInt, std::memory_order_relaxed); mPositionFrac.store(DataPosFrac, std::memory_order_relaxed); mCurrentBuffer.store(BufferListItem, std::memory_order_relaxed); if(vstate == ALvoice::Stopped) { mLoopBuffer.store(nullptr, std::memory_order_relaxed); mSourceID.store(0u, std::memory_order_relaxed); } std::atomic_thread_fence(std::memory_order_release); /* Send any events now, after the position/buffer info was updated. */ const ALbitfieldSOFT enabledevt{Context->mEnabledEvts.load(std::memory_order_acquire)}; if(buffers_done > 0 && (enabledevt&EventType_BufferCompleted)) { RingBuffer *ring{Context->mAsyncEvents.get()}; auto evt_vec = ring->getWriteVector(); if(evt_vec.first.len > 0) { AsyncEvent *evt{new (evt_vec.first.buf) AsyncEvent{EventType_BufferCompleted}}; evt->u.bufcomp.id = SourceID; evt->u.bufcomp.count = buffers_done; ring->writeAdvance(1); Context->mEventSem.post(); } } if(vstate == ALvoice::Stopped) { /* If the voice just ended, set it to Stopping so the next render * ensures any residual noise fades to 0 amplitude. */ mPlayState.store(ALvoice::Stopping, std::memory_order_release); if((enabledevt&EventType_SourceStateChange)) SendSourceStoppedEvent(Context, SourceID); } }