mirror of https://github.com/axmolengine/axmol.git
850 lines
31 KiB
C++
850 lines
31 KiB
C++
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#include "config.h"
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#include "voice.h"
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#include <algorithm>
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#include <array>
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#include <atomic>
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#include <cassert>
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#include <cstdint>
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#include <iterator>
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#include <memory>
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#include <new>
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#include <stdlib.h>
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#include <utility>
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#include <vector>
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#include "albyte.h"
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#include "alnumeric.h"
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#include "aloptional.h"
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#include "alspan.h"
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#include "alstring.h"
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#include "ambidefs.h"
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#include "async_event.h"
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#include "buffer_storage.h"
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#include "context.h"
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#include "cpu_caps.h"
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#include "devformat.h"
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#include "device.h"
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#include "filters/biquad.h"
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#include "filters/nfc.h"
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#include "filters/splitter.h"
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#include "fmt_traits.h"
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#include "logging.h"
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#include "mixer.h"
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#include "mixer/defs.h"
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#include "mixer/hrtfdefs.h"
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#include "opthelpers.h"
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#include "resampler_limits.h"
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#include "ringbuffer.h"
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#include "vector.h"
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#include "voice_change.h"
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struct CTag;
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#ifdef HAVE_SSE
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struct SSETag;
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#endif
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#ifdef HAVE_NEON
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struct NEONTag;
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#endif
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struct CopyTag;
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static_assert(!(sizeof(Voice::BufferLine)&15), "Voice::BufferLine must be a multiple of 16 bytes");
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Resampler ResamplerDefault{Resampler::Linear};
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namespace {
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using uint = unsigned int;
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using HrtfMixerFunc = void(*)(const float *InSamples, float2 *AccumSamples, const uint IrSize,
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const MixHrtfFilter *hrtfparams, const size_t BufferSize);
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using HrtfMixerBlendFunc = void(*)(const float *InSamples, float2 *AccumSamples,
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const uint IrSize, const HrtfFilter *oldparams, const MixHrtfFilter *newparams,
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const size_t BufferSize);
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HrtfMixerFunc MixHrtfSamples{MixHrtf_<CTag>};
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HrtfMixerBlendFunc MixHrtfBlendSamples{MixHrtfBlend_<CTag>};
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inline MixerFunc SelectMixer()
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return Mix_<NEONTag>;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return Mix_<SSETag>;
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#endif
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return Mix_<CTag>;
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}
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inline HrtfMixerFunc SelectHrtfMixer()
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixHrtf_<NEONTag>;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixHrtf_<SSETag>;
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#endif
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return MixHrtf_<CTag>;
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}
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inline HrtfMixerBlendFunc SelectHrtfBlendMixer()
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{
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#ifdef HAVE_NEON
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if((CPUCapFlags&CPU_CAP_NEON))
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return MixHrtfBlend_<NEONTag>;
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#endif
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#ifdef HAVE_SSE
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if((CPUCapFlags&CPU_CAP_SSE))
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return MixHrtfBlend_<SSETag>;
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#endif
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return MixHrtfBlend_<CTag>;
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}
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} // namespace
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void Voice::InitMixer(al::optional<std::string> resampler)
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{
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if(resampler)
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{
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struct ResamplerEntry {
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const char name[16];
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const Resampler resampler;
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};
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constexpr ResamplerEntry ResamplerList[]{
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{ "none", Resampler::Point },
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{ "point", Resampler::Point },
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{ "linear", Resampler::Linear },
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{ "cubic", Resampler::Cubic },
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{ "bsinc12", Resampler::BSinc12 },
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{ "fast_bsinc12", Resampler::FastBSinc12 },
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{ "bsinc24", Resampler::BSinc24 },
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{ "fast_bsinc24", Resampler::FastBSinc24 },
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};
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const char *str{resampler->c_str()};
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if(al::strcasecmp(str, "bsinc") == 0)
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{
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WARN("Resampler option \"%s\" is deprecated, using bsinc12\n", str);
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str = "bsinc12";
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}
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else if(al::strcasecmp(str, "sinc4") == 0 || al::strcasecmp(str, "sinc8") == 0)
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{
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WARN("Resampler option \"%s\" is deprecated, using cubic\n", str);
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str = "cubic";
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}
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auto iter = std::find_if(std::begin(ResamplerList), std::end(ResamplerList),
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[str](const ResamplerEntry &entry) -> bool
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{ return al::strcasecmp(str, entry.name) == 0; });
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if(iter == std::end(ResamplerList))
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ERR("Invalid resampler: %s\n", str);
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else
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ResamplerDefault = iter->resampler;
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}
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MixSamples = SelectMixer();
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MixHrtfBlendSamples = SelectHrtfBlendMixer();
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MixHrtfSamples = SelectHrtfMixer();
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}
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namespace {
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void SendSourceStoppedEvent(ContextBase *context, uint id)
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{
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RingBuffer *ring{context->mAsyncEvents.get()};
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auto evt_vec = ring->getWriteVector();
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if(evt_vec.first.len < 1) return;
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AsyncEvent *evt{::new(evt_vec.first.buf) AsyncEvent{EventType_SourceStateChange}};
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evt->u.srcstate.id = id;
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evt->u.srcstate.state = AsyncEvent::SrcState::Stop;
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ring->writeAdvance(1);
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}
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const float *DoFilters(BiquadFilter &lpfilter, BiquadFilter &hpfilter, float *dst,
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const al::span<const float> src, int type)
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{
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switch(type)
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{
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case AF_None:
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lpfilter.clear();
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hpfilter.clear();
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break;
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case AF_LowPass:
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lpfilter.process(src, dst);
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hpfilter.clear();
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return dst;
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case AF_HighPass:
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lpfilter.clear();
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hpfilter.process(src, dst);
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return dst;
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case AF_BandPass:
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DualBiquad{lpfilter, hpfilter}.process(src, dst);
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return dst;
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}
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return src.data();
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}
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void LoadSamples(const al::span<Voice::BufferLine> dstSamples, const size_t dstOffset,
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const al::byte *src, const size_t srcOffset, const FmtType srctype, const FmtChannels srcchans,
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const size_t samples) noexcept
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{
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#define HANDLE_FMT(T) case T: \
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{ \
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constexpr size_t sampleSize{sizeof(al::FmtTypeTraits<T>::Type)}; \
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if(srcchans == FmtUHJ2) \
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{ \
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constexpr size_t srcstep{2u}; \
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src += srcOffset*srcstep*sampleSize; \
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al::LoadSampleArray<T>(dstSamples[0].data() + dstOffset, src, \
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srcstep, samples); \
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al::LoadSampleArray<T>(dstSamples[1].data() + dstOffset, \
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src + sampleSize, srcstep, samples); \
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std::fill_n(dstSamples[2].data() + dstOffset, samples, 0.0f); \
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} \
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else \
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{ \
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const size_t srcstep{dstSamples.size()}; \
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src += srcOffset*srcstep*sampleSize; \
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for(auto &dst : dstSamples) \
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{ \
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al::LoadSampleArray<T>(dst.data() + dstOffset, src, srcstep, \
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samples); \
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src += sampleSize; \
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} \
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} \
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} \
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break
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switch(srctype)
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{
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HANDLE_FMT(FmtUByte);
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HANDLE_FMT(FmtShort);
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HANDLE_FMT(FmtFloat);
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HANDLE_FMT(FmtDouble);
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HANDLE_FMT(FmtMulaw);
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HANDLE_FMT(FmtAlaw);
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}
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#undef HANDLE_FMT
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}
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void LoadBufferStatic(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem,
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const size_t dataPosInt, const FmtType sampleType, const FmtChannels sampleChannels,
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const size_t samplesToLoad, const al::span<Voice::BufferLine> voiceSamples)
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{
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const uint loopStart{buffer->mLoopStart};
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const uint loopEnd{buffer->mLoopEnd};
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ASSUME(loopEnd > loopStart);
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/* If current pos is beyond the loop range, do not loop */
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if(!bufferLoopItem || dataPosInt >= loopEnd)
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{
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/* Load what's left to play from the buffer */
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const size_t remaining{minz(samplesToLoad, buffer->mSampleLen-dataPosInt)};
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LoadSamples(voiceSamples, MaxResamplerEdge, buffer->mSamples, dataPosInt, sampleType,
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sampleChannels, remaining);
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if(const size_t toFill{samplesToLoad - remaining})
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{
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for(auto &chanbuffer : voiceSamples)
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{
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auto srcsamples = chanbuffer.data() + MaxResamplerEdge - 1 + remaining;
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std::fill_n(srcsamples + 1, toFill, *srcsamples);
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}
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}
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}
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else
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{
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/* Load what's left of this loop iteration */
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const size_t remaining{minz(samplesToLoad, loopEnd-dataPosInt)};
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LoadSamples(voiceSamples, MaxResamplerEdge, buffer->mSamples, dataPosInt, sampleType,
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sampleChannels, remaining);
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/* Load repeats of the loop to fill the buffer. */
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const auto loopSize = static_cast<size_t>(loopEnd - loopStart);
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size_t samplesLoaded{remaining};
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while(const size_t toFill{minz(samplesToLoad - samplesLoaded, loopSize)})
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{
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LoadSamples(voiceSamples, MaxResamplerEdge + samplesLoaded, buffer->mSamples,
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loopStart, sampleType, sampleChannels, toFill);
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samplesLoaded += toFill;
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}
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}
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}
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void LoadBufferCallback(VoiceBufferItem *buffer, const size_t numCallbackSamples,
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const FmtType sampleType, const FmtChannels sampleChannels, const size_t samplesToLoad,
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const al::span<Voice::BufferLine> voiceSamples)
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{
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/* Load what's left to play from the buffer */
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const size_t remaining{minz(samplesToLoad, numCallbackSamples)};
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LoadSamples(voiceSamples, MaxResamplerEdge, buffer->mSamples, 0, sampleType, sampleChannels,
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remaining);
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if(const size_t toFill{samplesToLoad - remaining})
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{
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for(auto &chanbuffer : voiceSamples)
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{
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auto srcsamples = chanbuffer.data() + MaxResamplerEdge - 1 + remaining;
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std::fill_n(srcsamples + 1, toFill, *srcsamples);
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}
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}
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}
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void LoadBufferQueue(VoiceBufferItem *buffer, VoiceBufferItem *bufferLoopItem,
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size_t dataPosInt, const FmtType sampleType, const FmtChannels sampleChannels,
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const size_t samplesToLoad, const al::span<Voice::BufferLine> voiceSamples)
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{
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/* Crawl the buffer queue to fill in the temp buffer */
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size_t samplesLoaded{0};
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while(buffer && samplesLoaded != samplesToLoad)
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{
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if(dataPosInt >= buffer->mSampleLen)
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{
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dataPosInt -= buffer->mSampleLen;
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buffer = buffer->mNext.load(std::memory_order_acquire);
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if(!buffer) buffer = bufferLoopItem;
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continue;
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}
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const size_t remaining{minz(samplesToLoad-samplesLoaded, buffer->mSampleLen-dataPosInt)};
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LoadSamples(voiceSamples, MaxResamplerEdge+samplesLoaded, buffer->mSamples, dataPosInt,
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sampleType, sampleChannels, remaining);
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samplesLoaded += remaining;
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if(samplesLoaded == samplesToLoad)
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break;
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dataPosInt = 0;
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buffer = buffer->mNext.load(std::memory_order_acquire);
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if(!buffer) buffer = bufferLoopItem;
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}
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if(const size_t toFill{samplesToLoad - samplesLoaded})
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{
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size_t chanidx{0};
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for(auto &chanbuffer : voiceSamples)
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{
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auto srcsamples = chanbuffer.data() + MaxResamplerEdge - 1 + samplesLoaded;
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std::fill_n(srcsamples + 1, toFill, *srcsamples);
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++chanidx;
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}
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}
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}
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void DoHrtfMix(const float *samples, const uint DstBufferSize, DirectParams &parms,
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const float TargetGain, const uint Counter, uint OutPos, DeviceBase *Device)
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{
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const uint IrSize{Device->mIrSize};
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auto &HrtfSamples = Device->HrtfSourceData;
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/* Source HRTF mixing needs to include the direct delay so it remains
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* aligned with the direct mix's HRTF filtering.
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*/
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float2 *AccumSamples{Device->HrtfAccumData + HrtfDirectDelay};
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/* Copy the HRTF history and new input samples into a temp buffer. */
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auto src_iter = std::copy(parms.Hrtf.History.begin(), parms.Hrtf.History.end(),
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std::begin(HrtfSamples));
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std::copy_n(samples, DstBufferSize, src_iter);
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/* Copy the last used samples back into the history buffer for later. */
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std::copy_n(std::begin(HrtfSamples) + DstBufferSize, parms.Hrtf.History.size(),
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parms.Hrtf.History.begin());
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/* If fading and this is the first mixing pass, fade between the IRs. */
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uint fademix{0u};
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if(Counter && OutPos == 0)
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{
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fademix = minu(DstBufferSize, Counter);
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float gain{TargetGain};
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/* The new coefficients need to fade in completely since they're
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* replacing the old ones. To keep the gain fading consistent,
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* interpolate between the old and new target gains given how much of
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* the fade time this mix handles.
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*/
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if(Counter > fademix)
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{
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const float a{static_cast<float>(fademix) / static_cast<float>(Counter)};
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gain = lerp(parms.Hrtf.Old.Gain, TargetGain, a);
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}
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MixHrtfFilter hrtfparams{
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parms.Hrtf.Target.Coeffs,
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parms.Hrtf.Target.Delay,
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0.0f, gain / static_cast<float>(fademix)};
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MixHrtfBlendSamples(HrtfSamples, AccumSamples+OutPos, IrSize, &parms.Hrtf.Old, &hrtfparams,
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fademix);
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/* Update the old parameters with the result. */
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parms.Hrtf.Old = parms.Hrtf.Target;
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parms.Hrtf.Old.Gain = gain;
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OutPos += fademix;
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}
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if(fademix < DstBufferSize)
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{
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const uint todo{DstBufferSize - fademix};
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float gain{TargetGain};
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/* Interpolate the target gain if the gain fading lasts longer than
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* this mix.
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*/
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if(Counter > DstBufferSize)
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{
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const float a{static_cast<float>(todo) / static_cast<float>(Counter-fademix)};
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gain = lerp(parms.Hrtf.Old.Gain, TargetGain, a);
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}
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MixHrtfFilter hrtfparams{
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parms.Hrtf.Target.Coeffs,
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parms.Hrtf.Target.Delay,
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parms.Hrtf.Old.Gain,
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(gain - parms.Hrtf.Old.Gain) / static_cast<float>(todo)};
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MixHrtfSamples(HrtfSamples+fademix, AccumSamples+OutPos, IrSize, &hrtfparams, todo);
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/* Store the now-current gain for next time. */
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parms.Hrtf.Old.Gain = gain;
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}
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}
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void DoNfcMix(const al::span<const float> samples, FloatBufferLine *OutBuffer, DirectParams &parms,
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const float *TargetGains, const uint Counter, const uint OutPos, DeviceBase *Device)
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{
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using FilterProc = void (NfcFilter::*)(const al::span<const float>, float*);
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static constexpr FilterProc NfcProcess[MaxAmbiOrder+1]{
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nullptr, &NfcFilter::process1, &NfcFilter::process2, &NfcFilter::process3};
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float *CurrentGains{parms.Gains.Current.data()};
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MixSamples(samples, {OutBuffer, 1u}, CurrentGains, TargetGains, Counter, OutPos);
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++OutBuffer;
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++CurrentGains;
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++TargetGains;
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const al::span<float> nfcsamples{Device->NfcSampleData, samples.size()};
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size_t order{1};
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while(const size_t chancount{Device->NumChannelsPerOrder[order]})
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{
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(parms.NFCtrlFilter.*NfcProcess[order])(samples, nfcsamples.data());
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MixSamples(nfcsamples, {OutBuffer, chancount}, CurrentGains, TargetGains, Counter, OutPos);
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OutBuffer += chancount;
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CurrentGains += chancount;
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TargetGains += chancount;
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if(++order == MaxAmbiOrder+1)
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break;
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}
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}
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} // namespace
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void Voice::mix(const State vstate, ContextBase *Context, const uint SamplesToDo)
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{
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static constexpr std::array<float,MAX_OUTPUT_CHANNELS> SilentTarget{};
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ASSUME(SamplesToDo > 0);
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/* Get voice info */
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uint DataPosInt{mPosition.load(std::memory_order_relaxed)};
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uint DataPosFrac{mPositionFrac.load(std::memory_order_relaxed)};
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VoiceBufferItem *BufferListItem{mCurrentBuffer.load(std::memory_order_relaxed)};
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VoiceBufferItem *BufferLoopItem{mLoopBuffer.load(std::memory_order_relaxed)};
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const uint increment{mStep};
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if UNLIKELY(increment < 1)
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{
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/* If the voice is supposed to be stopping but can't be mixed, just
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* stop it before bailing.
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*/
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if(vstate == Stopping)
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mPlayState.store(Stopped, std::memory_order_release);
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return;
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}
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DeviceBase *Device{Context->mDevice};
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const uint NumSends{Device->NumAuxSends};
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ResamplerFunc Resample{(increment == MixerFracOne && DataPosFrac == 0) ?
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Resample_<CopyTag,CTag> : mResampler};
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uint Counter{(mFlags&VoiceIsFading) ? SamplesToDo : 0};
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if(!Counter)
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{
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/* No fading, just overwrite the old/current params. */
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for(auto &chandata : mChans)
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{
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{
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DirectParams &parms = chandata.mDryParams;
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if(!(mFlags&VoiceHasHrtf))
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parms.Gains.Current = parms.Gains.Target;
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else
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parms.Hrtf.Old = parms.Hrtf.Target;
|
|
}
|
|
for(uint send{0};send < NumSends;++send)
|
|
{
|
|
if(mSend[send].Buffer.empty())
|
|
continue;
|
|
|
|
SendParams &parms = chandata.mWetParams[send];
|
|
parms.Gains.Current = parms.Gains.Target;
|
|
}
|
|
}
|
|
}
|
|
else if UNLIKELY(!BufferListItem)
|
|
Counter = std::min(Counter, 64u);
|
|
|
|
const uint PostPadding{MaxResamplerEdge +
|
|
((mFmtChannels==FmtUHJ2 || mFmtChannels==FmtUHJ3 || mFmtChannels==FmtUHJ4)
|
|
? uint{UhjDecoder::sFilterDelay} : 0u)};
|
|
uint buffers_done{0u};
|
|
uint OutPos{0u};
|
|
do {
|
|
/* Figure out how many buffer samples will be needed */
|
|
uint DstBufferSize{SamplesToDo - OutPos};
|
|
uint SrcBufferSize;
|
|
|
|
if(increment <= MixerFracOne)
|
|
{
|
|
/* Calculate the last written dst sample pos. */
|
|
uint64_t DataSize64{DstBufferSize - 1};
|
|
/* Calculate the last read src sample pos. */
|
|
DataSize64 = (DataSize64*increment + DataPosFrac) >> MixerFracBits;
|
|
/* +1 to get the src sample count, include padding. */
|
|
DataSize64 += 1 + PostPadding;
|
|
|
|
/* Result is guaranteed to be <= BufferLineSize+ResamplerPrePadding
|
|
* since we won't use more src samples than dst samples+padding.
|
|
*/
|
|
SrcBufferSize = static_cast<uint>(DataSize64);
|
|
}
|
|
else
|
|
{
|
|
uint64_t DataSize64{DstBufferSize};
|
|
/* Calculate the end src sample pos, include padding. */
|
|
DataSize64 = (DataSize64*increment + DataPosFrac) >> MixerFracBits;
|
|
DataSize64 += PostPadding;
|
|
|
|
if(DataSize64 <= LineSize - MaxResamplerEdge)
|
|
SrcBufferSize = static_cast<uint>(DataSize64);
|
|
else
|
|
{
|
|
/* If the source size got saturated, we can't fill the desired
|
|
* dst size. Figure out how many samples we can actually mix.
|
|
*/
|
|
SrcBufferSize = LineSize - MaxResamplerEdge;
|
|
|
|
DataSize64 = SrcBufferSize - PostPadding;
|
|
DataSize64 = ((DataSize64<<MixerFracBits) - DataPosFrac) / increment;
|
|
if(DataSize64 < DstBufferSize)
|
|
{
|
|
/* Some mixers require being 16-byte aligned, so also limit
|
|
* to a multiple of 4 samples to maintain alignment.
|
|
*/
|
|
DstBufferSize = static_cast<uint>(DataSize64) & ~3u;
|
|
}
|
|
ASSUME(DstBufferSize > 0);
|
|
}
|
|
}
|
|
|
|
if((mFlags&(VoiceIsCallback|VoiceCallbackStopped)) == VoiceIsCallback && BufferListItem)
|
|
{
|
|
if(SrcBufferSize > mNumCallbackSamples)
|
|
{
|
|
const size_t byteOffset{mNumCallbackSamples*mFrameSize};
|
|
const size_t needBytes{SrcBufferSize*mFrameSize - byteOffset};
|
|
|
|
const int gotBytes{BufferListItem->mCallback(BufferListItem->mUserData,
|
|
&BufferListItem->mSamples[byteOffset], static_cast<int>(needBytes))};
|
|
if(gotBytes < 0)
|
|
mFlags |= VoiceCallbackStopped;
|
|
else if(static_cast<uint>(gotBytes) < needBytes)
|
|
{
|
|
mFlags |= VoiceCallbackStopped;
|
|
mNumCallbackSamples += static_cast<uint>(static_cast<uint>(gotBytes) /
|
|
mFrameSize);
|
|
}
|
|
else
|
|
mNumCallbackSamples = SrcBufferSize;
|
|
}
|
|
}
|
|
|
|
if UNLIKELY(!BufferListItem)
|
|
{
|
|
for(auto &chanbuffer : mVoiceSamples)
|
|
{
|
|
auto srciter = chanbuffer.data() + MaxResamplerEdge;
|
|
auto srcend = chanbuffer.data() + MaxResamplerPadding;
|
|
|
|
/* When loading from a voice that ended prematurely, only take
|
|
* the samples that get closest to 0 amplitude. This helps
|
|
* certain sounds fade out better.
|
|
*/
|
|
auto abs_lt = [](const float lhs, const float rhs) noexcept -> bool
|
|
{ return std::abs(lhs) < std::abs(rhs); };
|
|
srciter = std::min_element(srciter, srcend, abs_lt);
|
|
|
|
SrcBufferSize = SrcBufferSize - PostPadding + MaxResamplerPadding;
|
|
std::fill(srciter+1, chanbuffer.data() + SrcBufferSize, *srciter);
|
|
}
|
|
}
|
|
else
|
|
{
|
|
if((mFlags&VoiceIsStatic))
|
|
LoadBufferStatic(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, mFmtChannels,
|
|
SrcBufferSize, mVoiceSamples);
|
|
else if((mFlags&VoiceIsCallback))
|
|
LoadBufferCallback(BufferListItem, mNumCallbackSamples, mFmtType, mFmtChannels,
|
|
SrcBufferSize, mVoiceSamples);
|
|
else
|
|
LoadBufferQueue(BufferListItem, BufferLoopItem, DataPosInt, mFmtType, mFmtChannels,
|
|
SrcBufferSize, mVoiceSamples);
|
|
|
|
if(mDecoder)
|
|
{
|
|
const size_t srcOffset{(increment*DstBufferSize + DataPosFrac)>>MixerFracBits};
|
|
SrcBufferSize = SrcBufferSize - PostPadding + MaxResamplerEdge;
|
|
mDecoder->decode(mVoiceSamples, MaxResamplerEdge, SrcBufferSize, srcOffset);
|
|
}
|
|
}
|
|
|
|
auto voiceSamples = mVoiceSamples.begin();
|
|
for(auto &chandata : mChans)
|
|
{
|
|
/* Resample, then apply ambisonic upsampling as needed. */
|
|
float *ResampledData{Resample(&mResampleState,
|
|
voiceSamples->data() + MaxResamplerEdge, DataPosFrac, increment,
|
|
{Device->ResampledData, DstBufferSize})};
|
|
if((mFlags&VoiceIsAmbisonic))
|
|
chandata.mAmbiSplitter.processHfScale({ResampledData, DstBufferSize},
|
|
chandata.mAmbiScale);
|
|
|
|
/* Now filter and mix to the appropriate outputs. */
|
|
const al::span<float,BufferLineSize> FilterBuf{Device->FilteredData};
|
|
{
|
|
DirectParams &parms = chandata.mDryParams;
|
|
const float *samples{DoFilters(parms.LowPass, parms.HighPass, FilterBuf.data(),
|
|
{ResampledData, DstBufferSize}, mDirect.FilterType)};
|
|
|
|
if((mFlags&VoiceHasHrtf))
|
|
{
|
|
const float TargetGain{UNLIKELY(vstate == Stopping) ? 0.0f :
|
|
parms.Hrtf.Target.Gain};
|
|
DoHrtfMix(samples, DstBufferSize, parms, TargetGain, Counter, OutPos, Device);
|
|
}
|
|
else if((mFlags&VoiceHasNfc))
|
|
{
|
|
const float *TargetGains{UNLIKELY(vstate == Stopping) ? SilentTarget.data()
|
|
: parms.Gains.Target.data()};
|
|
DoNfcMix({samples, DstBufferSize}, mDirect.Buffer.data(), parms, TargetGains,
|
|
Counter, OutPos, Device);
|
|
}
|
|
else
|
|
{
|
|
const float *TargetGains{UNLIKELY(vstate == Stopping) ? SilentTarget.data()
|
|
: parms.Gains.Target.data()};
|
|
MixSamples({samples, DstBufferSize}, mDirect.Buffer,
|
|
parms.Gains.Current.data(), TargetGains, Counter, OutPos);
|
|
}
|
|
}
|
|
|
|
for(uint send{0};send < NumSends;++send)
|
|
{
|
|
if(mSend[send].Buffer.empty())
|
|
continue;
|
|
|
|
SendParams &parms = chandata.mWetParams[send];
|
|
const float *samples{DoFilters(parms.LowPass, parms.HighPass, FilterBuf.data(),
|
|
{ResampledData, DstBufferSize}, mSend[send].FilterType)};
|
|
|
|
const float *TargetGains{UNLIKELY(vstate == Stopping) ? SilentTarget.data()
|
|
: parms.Gains.Target.data()};
|
|
MixSamples({samples, DstBufferSize}, mSend[send].Buffer,
|
|
parms.Gains.Current.data(), TargetGains, Counter, OutPos);
|
|
}
|
|
|
|
/* Store the last source samples used for next time. */
|
|
const size_t srcOffset{(increment*DstBufferSize + DataPosFrac)>>MixerFracBits};
|
|
std::copy_n(voiceSamples->data()+srcOffset, MaxResamplerPadding, voiceSamples->data());
|
|
++voiceSamples;
|
|
}
|
|
/* Update positions */
|
|
DataPosFrac += increment*DstBufferSize;
|
|
const uint SrcSamplesDone{DataPosFrac>>MixerFracBits};
|
|
DataPosInt += SrcSamplesDone;
|
|
DataPosFrac &= MixerFracMask;
|
|
|
|
OutPos += DstBufferSize;
|
|
Counter = maxu(DstBufferSize, Counter) - DstBufferSize;
|
|
|
|
if UNLIKELY(!BufferListItem)
|
|
{
|
|
/* Do nothing extra when there's no buffers. */
|
|
}
|
|
else if((mFlags&VoiceIsStatic))
|
|
{
|
|
if(BufferLoopItem)
|
|
{
|
|
/* Handle looping static source */
|
|
const uint LoopStart{BufferListItem->mLoopStart};
|
|
const uint LoopEnd{BufferListItem->mLoopEnd};
|
|
if(DataPosInt >= LoopEnd)
|
|
{
|
|
assert(LoopEnd > LoopStart);
|
|
DataPosInt = ((DataPosInt-LoopStart)%(LoopEnd-LoopStart)) + LoopStart;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Handle non-looping static source */
|
|
if(DataPosInt >= BufferListItem->mSampleLen)
|
|
{
|
|
BufferListItem = nullptr;
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
else if((mFlags&VoiceIsCallback))
|
|
{
|
|
if(SrcSamplesDone < mNumCallbackSamples)
|
|
{
|
|
const size_t byteOffset{SrcSamplesDone*mFrameSize};
|
|
const size_t byteEnd{mNumCallbackSamples*mFrameSize};
|
|
al::byte *data{BufferListItem->mSamples};
|
|
std::copy(data+byteOffset, data+byteEnd, data);
|
|
mNumCallbackSamples -= SrcSamplesDone;
|
|
}
|
|
else
|
|
{
|
|
BufferListItem = nullptr;
|
|
mNumCallbackSamples = 0;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
/* Handle streaming source */
|
|
do {
|
|
if(BufferListItem->mSampleLen > DataPosInt)
|
|
break;
|
|
|
|
DataPosInt -= BufferListItem->mSampleLen;
|
|
|
|
++buffers_done;
|
|
BufferListItem = BufferListItem->mNext.load(std::memory_order_relaxed);
|
|
if(!BufferListItem) BufferListItem = BufferLoopItem;
|
|
} while(BufferListItem);
|
|
}
|
|
} while(OutPos < SamplesToDo);
|
|
|
|
mFlags |= VoiceIsFading;
|
|
|
|
/* Don't update positions and buffers if we were stopping. */
|
|
if UNLIKELY(vstate == Stopping)
|
|
{
|
|
mPlayState.store(Stopped, std::memory_order_release);
|
|
return;
|
|
}
|
|
|
|
/* Capture the source ID in case it's reset for stopping. */
|
|
const uint 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(!BufferListItem)
|
|
{
|
|
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 uint 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);
|
|
}
|
|
}
|
|
|
|
if(!BufferListItem)
|
|
{
|
|
/* If the voice just ended, set it to Stopping so the next render
|
|
* ensures any residual noise fades to 0 amplitude.
|
|
*/
|
|
mPlayState.store(Stopping, std::memory_order_release);
|
|
if((enabledevt&EventType_SourceStateChange))
|
|
SendSourceStoppedEvent(Context, SourceID);
|
|
}
|
|
}
|
|
|
|
void Voice::prepare(DeviceBase *device)
|
|
{
|
|
if((mFmtChannels == FmtUHJ2 || mFmtChannels == FmtUHJ3 || mFmtChannels==FmtUHJ4) && !mDecoder)
|
|
mDecoder = std::make_unique<UhjDecoder>();
|
|
else if(mFmtChannels != FmtUHJ2 && mFmtChannels != FmtUHJ3 && mFmtChannels != FmtUHJ4)
|
|
mDecoder = nullptr;
|
|
|
|
/* Clear the stepping value explicitly so the mixer knows not to mix this
|
|
* until the update gets applied.
|
|
*/
|
|
mStep = 0;
|
|
|
|
/* Make sure the sample history is cleared. */
|
|
std::fill(mVoiceSamples.begin(), mVoiceSamples.end(), BufferLine{});
|
|
|
|
/* Don't need to set the VoiceIsAmbisonic flag if the device is not higher
|
|
* order than the voice. No HF scaling is necessary to mix it.
|
|
*/
|
|
if(mAmbiOrder && device->mAmbiOrder > mAmbiOrder)
|
|
{
|
|
const uint8_t *OrderFromChan{(mFmtChannels == FmtBFormat2D) ?
|
|
AmbiIndex::OrderFrom2DChannel().data() : AmbiIndex::OrderFromChannel().data()};
|
|
const auto scales = AmbiScale::GetHFOrderScales(mAmbiOrder, device->mAmbiOrder);
|
|
|
|
const BandSplitter splitter{device->mXOverFreq / static_cast<float>(device->Frequency)};
|
|
for(auto &chandata : mChans)
|
|
{
|
|
chandata.mAmbiScale = scales[*(OrderFromChan++)];
|
|
chandata.mAmbiSplitter = splitter;
|
|
chandata.mDryParams = DirectParams{};
|
|
std::fill_n(chandata.mWetParams.begin(), device->NumAuxSends, SendParams{});
|
|
}
|
|
mFlags |= VoiceIsAmbisonic;
|
|
}
|
|
else
|
|
{
|
|
for(auto &chandata : mChans)
|
|
{
|
|
chandata.mDryParams = DirectParams{};
|
|
std::fill_n(chandata.mWetParams.begin(), device->NumAuxSends, SendParams{});
|
|
}
|
|
mFlags &= ~VoiceIsAmbisonic;
|
|
}
|
|
|
|
if(device->AvgSpeakerDist > 0.0f)
|
|
{
|
|
const float w1{SpeedOfSoundMetersPerSec /
|
|
(device->AvgSpeakerDist * static_cast<float>(device->Frequency))};
|
|
for(auto &chandata : mChans)
|
|
chandata.mDryParams.NFCtrlFilter.init(w1);
|
|
}
|
|
}
|