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update openal-soft
sync point: master-ac5d40e40a0155351fe1be4aab30017b6a13a859
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365 changed files with 76053 additions and 53126 deletions
567
Engine/lib/openal-soft/Alc/effects/convolution.cpp
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567
Engine/lib/openal-soft/Alc/effects/convolution.cpp
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#include "config.h"
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#include <stdint.h>
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#ifdef HAVE_SSE_INTRINSICS
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#include <xmmintrin.h>
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#elif defined(HAVE_NEON)
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#include <arm_neon.h>
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#endif
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#include "alcmain.h"
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#include "alcomplex.h"
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#include "alcontext.h"
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#include "almalloc.h"
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#include "alspan.h"
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#include "bformatdec.h"
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#include "buffer_storage.h"
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#include "core/ambidefs.h"
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#include "core/filters/splitter.h"
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#include "core/fmt_traits.h"
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#include "core/logging.h"
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#include "effects/base.h"
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#include "effectslot.h"
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#include "math_defs.h"
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#include "polyphase_resampler.h"
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namespace {
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/* Convolution reverb is implemented using a segmented overlap-add method. The
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* impulse response is broken up into multiple segments of 128 samples, and
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* each segment has an FFT applied with a 256-sample buffer (the latter half
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* left silent) to get its frequency-domain response. The resulting response
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* has its positive/non-mirrored frequencies saved (129 bins) in each segment.
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*
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* Input samples are similarly broken up into 128-sample segments, with an FFT
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* applied to each new incoming segment to get its 129 bins. A history of FFT'd
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* input segments is maintained, equal to the length of the impulse response.
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*
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* To apply the reverberation, each impulse response segment is convolved with
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* its paired input segment (using complex multiplies, far cheaper than FIRs),
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* accumulating into a 256-bin FFT buffer. The input history is then shifted to
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* align with later impulse response segments for next time.
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*
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* An inverse FFT is then applied to the accumulated FFT buffer to get a 256-
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* sample time-domain response for output, which is split in two halves. The
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* first half is the 128-sample output, and the second half is a 128-sample
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* (really, 127) delayed extension, which gets added to the output next time.
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* Convolving two time-domain responses of lengths N and M results in a time-
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* domain signal of length N+M-1, and this holds true regardless of the
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* convolution being applied in the frequency domain, so these "overflow"
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* samples need to be accounted for.
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*
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* To avoid a delay with gathering enough input samples to apply an FFT with,
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* the first segment is applied directly in the time-domain as the samples come
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* in. Once enough have been retrieved, the FFT is applied on the input and
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* it's paired with the remaining (FFT'd) filter segments for processing.
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*/
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void LoadSamples(double *RESTRICT dst, const al::byte *src, const size_t srcstep, FmtType srctype,
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const size_t samples) noexcept
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{
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#define HANDLE_FMT(T) case T: al::LoadSampleArray<T>(dst, src, srcstep, samples); 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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inline auto& GetAmbiScales(AmbiScaling scaletype) noexcept
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{
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if(scaletype == AmbiScaling::FuMa) return AmbiScale::FromFuMa();
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if(scaletype == AmbiScaling::SN3D) return AmbiScale::FromSN3D();
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return AmbiScale::FromN3D();
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}
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inline auto& GetAmbiLayout(AmbiLayout layouttype) noexcept
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{
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if(layouttype == AmbiLayout::FuMa) return AmbiIndex::FromFuMa();
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return AmbiIndex::FromACN();
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}
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inline auto& GetAmbi2DLayout(AmbiLayout layouttype) noexcept
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{
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if(layouttype == AmbiLayout::FuMa) return AmbiIndex::FromFuMa2D();
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return AmbiIndex::FromACN2D();
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}
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struct ChanMap {
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Channel channel;
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float angle;
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float elevation;
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};
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using complex_d = std::complex<double>;
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constexpr size_t ConvolveUpdateSize{256};
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constexpr size_t ConvolveUpdateSamples{ConvolveUpdateSize / 2};
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void apply_fir(al::span<float> dst, const float *RESTRICT src, const float *RESTRICT filter)
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{
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#ifdef HAVE_SSE_INTRINSICS
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for(float &output : dst)
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{
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__m128 r4{_mm_setzero_ps()};
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for(size_t j{0};j < ConvolveUpdateSamples;j+=4)
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{
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const __m128 coeffs{_mm_load_ps(&filter[j])};
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const __m128 s{_mm_loadu_ps(&src[j])};
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r4 = _mm_add_ps(r4, _mm_mul_ps(s, coeffs));
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}
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r4 = _mm_add_ps(r4, _mm_shuffle_ps(r4, r4, _MM_SHUFFLE(0, 1, 2, 3)));
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r4 = _mm_add_ps(r4, _mm_movehl_ps(r4, r4));
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output = _mm_cvtss_f32(r4);
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++src;
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}
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#elif defined(HAVE_NEON)
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for(float &output : dst)
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{
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float32x4_t r4{vdupq_n_f32(0.0f)};
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for(size_t j{0};j < ConvolveUpdateSamples;j+=4)
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r4 = vmlaq_f32(r4, vld1q_f32(&src[j]), vld1q_f32(&filter[j]));
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r4 = vaddq_f32(r4, vrev64q_f32(r4));
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output = vget_lane_f32(vadd_f32(vget_low_f32(r4), vget_high_f32(r4)), 0);
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++src;
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}
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#else
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for(float &output : dst)
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{
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float ret{0.0f};
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for(size_t j{0};j < ConvolveUpdateSamples;++j)
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ret += src[j] * filter[j];
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output = ret;
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++src;
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}
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#endif
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}
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struct ConvolutionState final : public EffectState {
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FmtChannels mChannels{};
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AmbiLayout mAmbiLayout{};
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AmbiScaling mAmbiScaling{};
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uint mAmbiOrder{};
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size_t mFifoPos{0};
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std::array<float,ConvolveUpdateSamples*2> mInput{};
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al::vector<std::array<float,ConvolveUpdateSamples>,16> mFilter;
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al::vector<std::array<float,ConvolveUpdateSamples*2>,16> mOutput;
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alignas(16) std::array<complex_d,ConvolveUpdateSize> mFftBuffer{};
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size_t mCurrentSegment{0};
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size_t mNumConvolveSegs{0};
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struct ChannelData {
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alignas(16) FloatBufferLine mBuffer{};
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float mHfScale{};
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BandSplitter mFilter{};
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float Current[MAX_OUTPUT_CHANNELS]{};
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float Target[MAX_OUTPUT_CHANNELS]{};
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};
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using ChannelDataArray = al::FlexArray<ChannelData>;
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std::unique_ptr<ChannelDataArray> mChans;
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std::unique_ptr<complex_d[]> mComplexData;
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ConvolutionState() = default;
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~ConvolutionState() override = default;
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void NormalMix(const al::span<FloatBufferLine> samplesOut, const size_t samplesToDo);
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void UpsampleMix(const al::span<FloatBufferLine> samplesOut, const size_t samplesToDo);
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void (ConvolutionState::*mMix)(const al::span<FloatBufferLine>,const size_t)
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{&ConvolutionState::NormalMix};
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void deviceUpdate(const ALCdevice *device, const Buffer &buffer) override;
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void update(const ALCcontext *context, const EffectSlot *slot, const EffectProps *props,
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const EffectTarget target) override;
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void process(const size_t samplesToDo, const al::span<const FloatBufferLine> samplesIn,
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const al::span<FloatBufferLine> samplesOut) override;
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DEF_NEWDEL(ConvolutionState)
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};
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void ConvolutionState::NormalMix(const al::span<FloatBufferLine> samplesOut,
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const size_t samplesToDo)
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{
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for(auto &chan : *mChans)
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MixSamples({chan.mBuffer.data(), samplesToDo}, samplesOut, chan.Current, chan.Target,
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samplesToDo, 0);
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}
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void ConvolutionState::UpsampleMix(const al::span<FloatBufferLine> samplesOut,
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const size_t samplesToDo)
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{
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for(auto &chan : *mChans)
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{
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const al::span<float> src{chan.mBuffer.data(), samplesToDo};
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chan.mFilter.processHfScale(src, chan.mHfScale);
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MixSamples(src, samplesOut, chan.Current, chan.Target, samplesToDo, 0);
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}
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}
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void ConvolutionState::deviceUpdate(const ALCdevice *device, const Buffer &buffer)
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{
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constexpr uint MaxConvolveAmbiOrder{1u};
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mFifoPos = 0;
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mInput.fill(0.0f);
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decltype(mFilter){}.swap(mFilter);
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decltype(mOutput){}.swap(mOutput);
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mFftBuffer.fill(complex_d{});
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mCurrentSegment = 0;
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mNumConvolveSegs = 0;
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mChans = nullptr;
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mComplexData = nullptr;
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/* An empty buffer doesn't need a convolution filter. */
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if(!buffer.storage || buffer.storage->mSampleLen < 1) return;
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constexpr size_t m{ConvolveUpdateSize/2 + 1};
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auto bytesPerSample = BytesFromFmt(buffer.storage->mType);
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auto realChannels = ChannelsFromFmt(buffer.storage->mChannels, buffer.storage->mAmbiOrder);
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auto numChannels = ChannelsFromFmt(buffer.storage->mChannels,
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minu(buffer.storage->mAmbiOrder, MaxConvolveAmbiOrder));
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mChans = ChannelDataArray::Create(numChannels);
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/* The impulse response needs to have the same sample rate as the input and
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* output. The bsinc24 resampler is decent, but there is high-frequency
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* attenation that some people may be able to pick up on. Since this is
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* called very infrequently, go ahead and use the polyphase resampler.
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*/
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PPhaseResampler resampler;
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if(device->Frequency != buffer.storage->mSampleRate)
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resampler.init(buffer.storage->mSampleRate, device->Frequency);
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const auto resampledCount = static_cast<uint>(
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(uint64_t{buffer.storage->mSampleLen}*device->Frequency+(buffer.storage->mSampleRate-1)) /
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buffer.storage->mSampleRate);
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const BandSplitter splitter{device->mXOverFreq / static_cast<float>(device->Frequency)};
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for(auto &e : *mChans)
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e.mFilter = splitter;
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mFilter.resize(numChannels, {});
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mOutput.resize(numChannels, {});
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/* Calculate the number of segments needed to hold the impulse response and
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* the input history (rounded up), and allocate them. Exclude one segment
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* which gets applied as a time-domain FIR filter. Make sure at least one
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* segment is allocated to simplify handling.
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*/
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mNumConvolveSegs = (resampledCount+(ConvolveUpdateSamples-1)) / ConvolveUpdateSamples;
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mNumConvolveSegs = maxz(mNumConvolveSegs, 2) - 1;
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const size_t complex_length{mNumConvolveSegs * m * (numChannels+1)};
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mComplexData = std::make_unique<complex_d[]>(complex_length);
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std::fill_n(mComplexData.get(), complex_length, complex_d{});
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mChannels = buffer.storage->mChannels;
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mAmbiLayout = buffer.storage->mAmbiLayout;
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mAmbiScaling = buffer.storage->mAmbiScaling;
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mAmbiOrder = minu(buffer.storage->mAmbiOrder, MaxConvolveAmbiOrder);
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auto srcsamples = std::make_unique<double[]>(maxz(buffer.storage->mSampleLen, resampledCount));
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complex_d *filteriter = mComplexData.get() + mNumConvolveSegs*m;
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for(size_t c{0};c < numChannels;++c)
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{
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/* Load the samples from the buffer, and resample to match the device. */
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LoadSamples(srcsamples.get(), buffer.samples.data() + bytesPerSample*c, realChannels,
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buffer.storage->mType, buffer.storage->mSampleLen);
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if(device->Frequency != buffer.storage->mSampleRate)
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resampler.process(buffer.storage->mSampleLen, srcsamples.get(), resampledCount,
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srcsamples.get());
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/* Store the first segment's samples in reverse in the time-domain, to
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* apply as a FIR filter.
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*/
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const size_t first_size{minz(resampledCount, ConvolveUpdateSamples)};
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std::transform(srcsamples.get(), srcsamples.get()+first_size, mFilter[c].rbegin(),
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[](const double d) noexcept -> float { return static_cast<float>(d); });
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size_t done{first_size};
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for(size_t s{0};s < mNumConvolveSegs;++s)
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{
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const size_t todo{minz(resampledCount-done, ConvolveUpdateSamples)};
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auto iter = std::copy_n(&srcsamples[done], todo, mFftBuffer.begin());
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done += todo;
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std::fill(iter, mFftBuffer.end(), complex_d{});
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forward_fft(mFftBuffer);
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filteriter = std::copy_n(mFftBuffer.cbegin(), m, filteriter);
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}
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}
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}
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void ConvolutionState::update(const ALCcontext *context, const EffectSlot *slot,
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const EffectProps* /*props*/, const EffectTarget target)
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{
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/* NOTE: Stereo and Rear are slightly different from normal mixing (as
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* defined in alu.cpp). These are 45 degrees from center, rather than the
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* 30 degrees used there.
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*
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* TODO: LFE is not mixed to output. This will require each buffer channel
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* to have its own output target since the main mixing buffer won't have an
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* LFE channel (due to being B-Format).
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*/
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static const ChanMap MonoMap[1]{
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{ FrontCenter, 0.0f, 0.0f }
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}, StereoMap[2]{
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{ FrontLeft, Deg2Rad(-45.0f), Deg2Rad(0.0f) },
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{ FrontRight, Deg2Rad( 45.0f), Deg2Rad(0.0f) }
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}, RearMap[2]{
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{ BackLeft, Deg2Rad(-135.0f), Deg2Rad(0.0f) },
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{ BackRight, Deg2Rad( 135.0f), Deg2Rad(0.0f) }
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}, QuadMap[4]{
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{ FrontLeft, Deg2Rad( -45.0f), Deg2Rad(0.0f) },
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{ FrontRight, Deg2Rad( 45.0f), Deg2Rad(0.0f) },
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{ BackLeft, Deg2Rad(-135.0f), Deg2Rad(0.0f) },
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{ BackRight, Deg2Rad( 135.0f), Deg2Rad(0.0f) }
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}, X51Map[6]{
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{ FrontLeft, Deg2Rad( -30.0f), Deg2Rad(0.0f) },
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{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
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{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
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{ LFE, 0.0f, 0.0f },
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{ SideLeft, Deg2Rad(-110.0f), Deg2Rad(0.0f) },
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{ SideRight, Deg2Rad( 110.0f), Deg2Rad(0.0f) }
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}, X61Map[7]{
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{ FrontLeft, Deg2Rad(-30.0f), Deg2Rad(0.0f) },
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{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
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{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
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{ LFE, 0.0f, 0.0f },
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{ BackCenter, Deg2Rad(180.0f), Deg2Rad(0.0f) },
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{ SideLeft, Deg2Rad(-90.0f), Deg2Rad(0.0f) },
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{ SideRight, Deg2Rad( 90.0f), Deg2Rad(0.0f) }
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}, X71Map[8]{
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{ FrontLeft, Deg2Rad( -30.0f), Deg2Rad(0.0f) },
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{ FrontRight, Deg2Rad( 30.0f), Deg2Rad(0.0f) },
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{ FrontCenter, Deg2Rad( 0.0f), Deg2Rad(0.0f) },
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{ LFE, 0.0f, 0.0f },
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{ BackLeft, Deg2Rad(-150.0f), Deg2Rad(0.0f) },
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{ BackRight, Deg2Rad( 150.0f), Deg2Rad(0.0f) },
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{ SideLeft, Deg2Rad( -90.0f), Deg2Rad(0.0f) },
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{ SideRight, Deg2Rad( 90.0f), Deg2Rad(0.0f) }
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};
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if(mNumConvolveSegs < 1)
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return;
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mMix = &ConvolutionState::NormalMix;
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for(auto &chan : *mChans)
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std::fill(std::begin(chan.Target), std::end(chan.Target), 0.0f);
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const float gain{slot->Gain};
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if(mChannels == FmtBFormat3D || mChannels == FmtBFormat2D)
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{
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ALCdevice *device{context->mDevice.get()};
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if(device->mAmbiOrder > mAmbiOrder)
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{
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mMix = &ConvolutionState::UpsampleMix;
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const auto scales = BFormatDec::GetHFOrderScales(mAmbiOrder, device->mAmbiOrder);
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(*mChans)[0].mHfScale = scales[0];
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for(size_t i{1};i < mChans->size();++i)
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(*mChans)[i].mHfScale = scales[1];
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}
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mOutTarget = target.Main->Buffer;
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auto&& scales = GetAmbiScales(mAmbiScaling);
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const uint8_t *index_map{(mChannels == FmtBFormat2D) ?
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GetAmbi2DLayout(mAmbiLayout).data() :
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GetAmbiLayout(mAmbiLayout).data()};
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std::array<float,MaxAmbiChannels> coeffs{};
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for(size_t c{0u};c < mChans->size();++c)
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{
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const size_t acn{index_map[c]};
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coeffs[acn] = scales[acn];
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ComputePanGains(target.Main, coeffs.data(), gain, (*mChans)[c].Target);
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coeffs[acn] = 0.0f;
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}
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}
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else
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{
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ALCdevice *device{context->mDevice.get()};
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al::span<const ChanMap> chanmap{};
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switch(mChannels)
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{
|
||||
case FmtMono: chanmap = MonoMap; break;
|
||||
case FmtStereo: chanmap = StereoMap; break;
|
||||
case FmtRear: chanmap = RearMap; break;
|
||||
case FmtQuad: chanmap = QuadMap; break;
|
||||
case FmtX51: chanmap = X51Map; break;
|
||||
case FmtX61: chanmap = X61Map; break;
|
||||
case FmtX71: chanmap = X71Map; break;
|
||||
case FmtBFormat2D:
|
||||
case FmtBFormat3D:
|
||||
break;
|
||||
}
|
||||
|
||||
mOutTarget = target.Main->Buffer;
|
||||
if(device->mRenderMode == RenderMode::Pairwise)
|
||||
{
|
||||
auto ScaleAzimuthFront = [](float azimuth, float scale) -> float
|
||||
{
|
||||
const float abs_azi{std::fabs(azimuth)};
|
||||
if(!(abs_azi >= al::MathDefs<float>::Pi()*0.5f))
|
||||
return std::copysign(minf(abs_azi*scale, al::MathDefs<float>::Pi()*0.5f), azimuth);
|
||||
return azimuth;
|
||||
};
|
||||
|
||||
for(size_t i{0};i < chanmap.size();++i)
|
||||
{
|
||||
if(chanmap[i].channel == LFE) continue;
|
||||
const auto coeffs = CalcAngleCoeffs(ScaleAzimuthFront(chanmap[i].angle, 2.0f),
|
||||
chanmap[i].elevation, 0.0f);
|
||||
ComputePanGains(target.Main, coeffs.data(), gain, (*mChans)[i].Target);
|
||||
}
|
||||
}
|
||||
else for(size_t i{0};i < chanmap.size();++i)
|
||||
{
|
||||
if(chanmap[i].channel == LFE) continue;
|
||||
const auto coeffs = CalcAngleCoeffs(chanmap[i].angle, chanmap[i].elevation, 0.0f);
|
||||
ComputePanGains(target.Main, coeffs.data(), gain, (*mChans)[i].Target);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void ConvolutionState::process(const size_t samplesToDo,
|
||||
const al::span<const FloatBufferLine> samplesIn, const al::span<FloatBufferLine> samplesOut)
|
||||
{
|
||||
if(mNumConvolveSegs < 1)
|
||||
return;
|
||||
|
||||
constexpr size_t m{ConvolveUpdateSize/2 + 1};
|
||||
size_t curseg{mCurrentSegment};
|
||||
auto &chans = *mChans;
|
||||
|
||||
for(size_t base{0u};base < samplesToDo;)
|
||||
{
|
||||
const size_t todo{minz(ConvolveUpdateSamples-mFifoPos, samplesToDo-base)};
|
||||
|
||||
std::copy_n(samplesIn[0].begin() + base, todo,
|
||||
mInput.begin()+ConvolveUpdateSamples+mFifoPos);
|
||||
|
||||
/* Apply the FIR for the newly retrieved input samples, and combine it
|
||||
* with the inverse FFT'd output samples.
|
||||
*/
|
||||
for(size_t c{0};c < chans.size();++c)
|
||||
{
|
||||
auto buf_iter = chans[c].mBuffer.begin() + base;
|
||||
apply_fir({std::addressof(*buf_iter), todo}, mInput.data()+1 + mFifoPos,
|
||||
mFilter[c].data());
|
||||
|
||||
auto fifo_iter = mOutput[c].begin() + mFifoPos;
|
||||
std::transform(fifo_iter, fifo_iter+todo, buf_iter, buf_iter, std::plus<>{});
|
||||
}
|
||||
|
||||
mFifoPos += todo;
|
||||
base += todo;
|
||||
|
||||
/* Check whether the input buffer is filled with new samples. */
|
||||
if(mFifoPos < ConvolveUpdateSamples) break;
|
||||
mFifoPos = 0;
|
||||
|
||||
/* Move the newest input to the front for the next iteration's history. */
|
||||
std::copy(mInput.cbegin()+ConvolveUpdateSamples, mInput.cend(), mInput.begin());
|
||||
|
||||
/* Calculate the frequency domain response and add the relevant
|
||||
* frequency bins to the FFT history.
|
||||
*/
|
||||
auto fftiter = std::copy_n(mInput.cbegin(), ConvolveUpdateSamples, mFftBuffer.begin());
|
||||
std::fill(fftiter, mFftBuffer.end(), complex_d{});
|
||||
forward_fft(mFftBuffer);
|
||||
|
||||
std::copy_n(mFftBuffer.cbegin(), m, &mComplexData[curseg*m]);
|
||||
|
||||
const complex_d *RESTRICT filter{mComplexData.get() + mNumConvolveSegs*m};
|
||||
for(size_t c{0};c < chans.size();++c)
|
||||
{
|
||||
std::fill_n(mFftBuffer.begin(), m, complex_d{});
|
||||
|
||||
/* Convolve each input segment with its IR filter counterpart
|
||||
* (aligned in time).
|
||||
*/
|
||||
const complex_d *RESTRICT input{&mComplexData[curseg*m]};
|
||||
for(size_t s{curseg};s < mNumConvolveSegs;++s)
|
||||
{
|
||||
for(size_t i{0};i < m;++i,++input,++filter)
|
||||
mFftBuffer[i] += *input * *filter;
|
||||
}
|
||||
input = mComplexData.get();
|
||||
for(size_t s{0};s < curseg;++s)
|
||||
{
|
||||
for(size_t i{0};i < m;++i,++input,++filter)
|
||||
mFftBuffer[i] += *input * *filter;
|
||||
}
|
||||
|
||||
/* Reconstruct the mirrored/negative frequencies to do a proper
|
||||
* inverse FFT.
|
||||
*/
|
||||
for(size_t i{m};i < ConvolveUpdateSize;++i)
|
||||
mFftBuffer[i] = std::conj(mFftBuffer[ConvolveUpdateSize-i]);
|
||||
|
||||
/* Apply iFFT to get the 256 (really 255) samples for output. The
|
||||
* 128 output samples are combined with the last output's 127
|
||||
* second-half samples (and this output's second half is
|
||||
* subsequently saved for next time).
|
||||
*/
|
||||
inverse_fft(mFftBuffer);
|
||||
|
||||
/* The iFFT'd response is scaled up by the number of bins, so apply
|
||||
* the inverse to normalize the output.
|
||||
*/
|
||||
for(size_t i{0};i < ConvolveUpdateSamples;++i)
|
||||
mOutput[c][i] =
|
||||
static_cast<float>(mFftBuffer[i].real() * (1.0/double{ConvolveUpdateSize})) +
|
||||
mOutput[c][ConvolveUpdateSamples+i];
|
||||
for(size_t i{0};i < ConvolveUpdateSamples;++i)
|
||||
mOutput[c][ConvolveUpdateSamples+i] =
|
||||
static_cast<float>(mFftBuffer[ConvolveUpdateSamples+i].real() *
|
||||
(1.0/double{ConvolveUpdateSize}));
|
||||
}
|
||||
|
||||
/* Shift the input history. */
|
||||
curseg = curseg ? (curseg-1) : (mNumConvolveSegs-1);
|
||||
}
|
||||
mCurrentSegment = curseg;
|
||||
|
||||
/* Finally, mix to the output. */
|
||||
(this->*mMix)(samplesOut, samplesToDo);
|
||||
}
|
||||
|
||||
|
||||
struct ConvolutionStateFactory final : public EffectStateFactory {
|
||||
al::intrusive_ptr<EffectState> create() override
|
||||
{ return al::intrusive_ptr<EffectState>{new ConvolutionState{}}; }
|
||||
};
|
||||
|
||||
} // namespace
|
||||
|
||||
EffectStateFactory *ConvolutionStateFactory_getFactory()
|
||||
{
|
||||
static ConvolutionStateFactory ConvolutionFactory{};
|
||||
return &ConvolutionFactory;
|
||||
}
|
||||
Loading…
Add table
Add a link
Reference in a new issue