mirror of https://github.com/axmolengine/axmol.git
666 lines
24 KiB
C++
666 lines
24 KiB
C++
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#include "config.h"
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#include <algorithm>
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#include <array>
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#include <complex>
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#include <cstddef>
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#include <functional>
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#include <iterator>
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#include <memory>
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#include <stdint.h>
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#include <utility>
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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 "alcomplex.h"
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#include "almalloc.h"
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#include "alnumbers.h"
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#include "alnumeric.h"
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#include "alspan.h"
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#include "base.h"
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#include "core/ambidefs.h"
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#include "core/bufferline.h"
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#include "core/buffer_storage.h"
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#include "core/context.h"
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#include "core/devformat.h"
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#include "core/device.h"
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#include "core/effectslot.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/mixer.h"
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#include "intrusive_ptr.h"
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#include "polyphase_resampler.h"
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#include "vector.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(float *RESTRICT dst, const std::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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/* FIXME: Handle ADPCM decoding here. */
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case FmtIMA4:
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case FmtMSADPCM:
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std::fill_n(dst, samples, 0.0f);
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break;
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}
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#undef HANDLE_FMT
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}
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constexpr auto GetAmbiScales(AmbiScaling scaletype) noexcept
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{
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switch(scaletype)
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{
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case AmbiScaling::FuMa: return al::span{AmbiScale::FromFuMa};
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case AmbiScaling::SN3D: return al::span{AmbiScale::FromSN3D};
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case AmbiScaling::UHJ: return al::span{AmbiScale::FromUHJ};
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case AmbiScaling::N3D: break;
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}
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return al::span{AmbiScale::FromN3D};
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}
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constexpr auto GetAmbiLayout(AmbiLayout layouttype) noexcept
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{
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if(layouttype == AmbiLayout::FuMa) return al::span{AmbiIndex::FromFuMa};
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return al::span{AmbiIndex::FromACN};
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}
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constexpr auto GetAmbi2DLayout(AmbiLayout layouttype) noexcept
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{
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if(layouttype == AmbiLayout::FuMa) return al::span{AmbiIndex::FromFuMa2D};
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return al::span{AmbiIndex::FromACN2D};
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}
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constexpr float sin30{0.5f};
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constexpr float cos30{0.866025403785f};
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constexpr float sin45{al::numbers::sqrt2_v<float>*0.5f};
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constexpr float cos45{al::numbers::sqrt2_v<float>*0.5f};
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constexpr float sin110{ 0.939692620786f};
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constexpr float cos110{-0.342020143326f};
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struct ChanPosMap {
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Channel channel;
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std::array<float,3> pos;
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};
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using complex_f = std::complex<float>;
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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_f,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{}, mLfScale{};
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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_f[]> 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 DeviceBase *device, const BufferStorage *buffer) override;
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void update(const ContextBase *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.processScale(src, chan.mHfScale, chan.mLfScale);
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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 DeviceBase *device, const BufferStorage *buffer)
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{
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using UhjDecoderType = UhjDecoder<512>;
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static constexpr auto DecoderPadding = UhjDecoderType::sInputPadding;
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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_f{});
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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 || buffer->mSampleLen < 1) return;
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mChannels = buffer->mChannels;
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mAmbiLayout = IsUHJ(mChannels) ? AmbiLayout::FuMa : buffer->mAmbiLayout;
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mAmbiScaling = IsUHJ(mChannels) ? AmbiScaling::UHJ : buffer->mAmbiScaling;
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mAmbiOrder = minu(buffer->mAmbiOrder, MaxConvolveAmbiOrder);
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constexpr size_t m{ConvolveUpdateSize/2 + 1};
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const auto bytesPerSample = BytesFromFmt(buffer->mType);
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const auto realChannels = buffer->channelsFromFmt();
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const auto numChannels = (mChannels == FmtUHJ2) ? 3u : ChannelsFromFmt(mChannels, mAmbiOrder);
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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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* attenuation 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->mSampleRate)
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resampler.init(buffer->mSampleRate, device->Frequency);
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const auto resampledCount = static_cast<uint>(
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(uint64_t{buffer->mSampleLen}*device->Frequency+(buffer->mSampleRate-1)) /
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buffer->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_f[]>(complex_length);
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std::fill_n(mComplexData.get(), complex_length, complex_f{});
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/* Load the samples from the buffer. */
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const size_t srclinelength{RoundUp(buffer->mSampleLen+DecoderPadding, 16)};
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auto srcsamples = std::make_unique<float[]>(srclinelength * numChannels);
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std::fill_n(srcsamples.get(), srclinelength * numChannels, 0.0f);
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for(size_t c{0};c < numChannels && c < realChannels;++c)
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LoadSamples(srcsamples.get() + srclinelength*c, buffer->mData.data() + bytesPerSample*c,
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realChannels, buffer->mType, buffer->mSampleLen);
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if(IsUHJ(mChannels))
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{
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auto decoder = std::make_unique<UhjDecoderType>();
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std::array<float*,4> samples{};
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for(size_t c{0};c < numChannels;++c)
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samples[c] = srcsamples.get() + srclinelength*c;
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decoder->decode({samples.data(), numChannels}, buffer->mSampleLen, buffer->mSampleLen);
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}
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auto ressamples = std::make_unique<double[]>(buffer->mSampleLen +
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(resampler ? resampledCount : 0));
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complex_f *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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/* Resample to match the device. */
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if(resampler)
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{
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std::copy_n(srcsamples.get() + srclinelength*c, buffer->mSampleLen,
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ressamples.get() + resampledCount);
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resampler.process(buffer->mSampleLen, ressamples.get()+resampledCount,
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resampledCount, ressamples.get());
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}
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else
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std::copy_n(srcsamples.get() + srclinelength*c, buffer->mSampleLen, ressamples.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(ressamples.get(), ressamples.get()+first_size, mFilter[c].rbegin(),
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[](const double d) noexcept -> float { return static_cast<float>(d); });
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auto fftbuffer = std::vector<std::complex<double>>(ConvolveUpdateSize);
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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(&ressamples[done], todo, fftbuffer.begin());
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done += todo;
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std::fill(iter, fftbuffer.end(), std::complex<double>{});
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forward_fft(al::span{fftbuffer});
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filteriter = std::copy_n(fftbuffer.cbegin(), m, filteriter);
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}
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}
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}
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void ConvolutionState::update(const ContextBase *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 constexpr ChanPosMap MonoMap[1]{
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{ FrontCenter, std::array{0.0f, 0.0f, -1.0f} }
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}, StereoMap[2]{
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{ FrontLeft, std::array{-sin45, 0.0f, -cos45} },
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{ FrontRight, std::array{ sin45, 0.0f, -cos45} },
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}, RearMap[2]{
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{ BackLeft, std::array{-sin45, 0.0f, cos45} },
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{ BackRight, std::array{ sin45, 0.0f, cos45} },
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}, QuadMap[4]{
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{ FrontLeft, std::array{-sin45, 0.0f, -cos45} },
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{ FrontRight, std::array{ sin45, 0.0f, -cos45} },
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{ BackLeft, std::array{-sin45, 0.0f, cos45} },
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{ BackRight, std::array{ sin45, 0.0f, cos45} },
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}, X51Map[6]{
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{ FrontLeft, std::array{-sin30, 0.0f, -cos30} },
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{ FrontRight, std::array{ sin30, 0.0f, -cos30} },
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{ FrontCenter, std::array{ 0.0f, 0.0f, -1.0f} },
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{ LFE, {} },
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{ SideLeft, std::array{-sin110, 0.0f, -cos110} },
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{ SideRight, std::array{ sin110, 0.0f, -cos110} },
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}, X61Map[7]{
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{ FrontLeft, std::array{-sin30, 0.0f, -cos30} },
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{ FrontRight, std::array{ sin30, 0.0f, -cos30} },
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{ FrontCenter, std::array{ 0.0f, 0.0f, -1.0f} },
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{ LFE, {} },
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{ BackCenter, std::array{ 0.0f, 0.0f, 1.0f} },
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{ SideLeft, std::array{-1.0f, 0.0f, 0.0f} },
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{ SideRight, std::array{ 1.0f, 0.0f, 0.0f} },
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}, X71Map[8]{
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{ FrontLeft, std::array{-sin30, 0.0f, -cos30} },
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{ FrontRight, std::array{ sin30, 0.0f, -cos30} },
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{ FrontCenter, std::array{ 0.0f, 0.0f, -1.0f} },
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{ LFE, {} },
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{ BackLeft, std::array{-sin30, 0.0f, cos30} },
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{ BackRight, std::array{ sin30, 0.0f, cos30} },
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{ SideLeft, std::array{ -1.0f, 0.0f, 0.0f} },
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{ SideRight, std::array{ 1.0f, 0.0f, 0.0f} },
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};
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if(mNumConvolveSegs < 1) UNLIKELY
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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(IsAmbisonic(mChannels))
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{
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DeviceBase *device{context->mDevice};
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if(mChannels == FmtUHJ2 && !device->mUhjEncoder)
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{
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mMix = &ConvolutionState::UpsampleMix;
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(*mChans)[0].mHfScale = 1.0f;
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(*mChans)[0].mLfScale = DecoderBase::sWLFScale;
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(*mChans)[1].mHfScale = 1.0f;
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(*mChans)[1].mLfScale = DecoderBase::sXYLFScale;
|
|
(*mChans)[2].mHfScale = 1.0f;
|
|
(*mChans)[2].mLfScale = DecoderBase::sXYLFScale;
|
|
}
|
|
else if(device->mAmbiOrder > mAmbiOrder)
|
|
{
|
|
mMix = &ConvolutionState::UpsampleMix;
|
|
const auto scales = AmbiScale::GetHFOrderScales(mAmbiOrder, device->mAmbiOrder,
|
|
device->m2DMixing);
|
|
(*mChans)[0].mHfScale = scales[0];
|
|
(*mChans)[0].mLfScale = 1.0f;
|
|
for(size_t i{1};i < mChans->size();++i)
|
|
{
|
|
(*mChans)[i].mHfScale = scales[1];
|
|
(*mChans)[i].mLfScale = 1.0f;
|
|
}
|
|
}
|
|
mOutTarget = target.Main->Buffer;
|
|
|
|
const auto scales = GetAmbiScales(mAmbiScaling);
|
|
const uint8_t *index_map{Is2DAmbisonic(mChannels) ?
|
|
GetAmbi2DLayout(mAmbiLayout).data() :
|
|
GetAmbiLayout(mAmbiLayout).data()};
|
|
|
|
std::array<float,MaxAmbiChannels> coeffs{};
|
|
for(size_t c{0u};c < mChans->size();++c)
|
|
{
|
|
const size_t acn{index_map[c]};
|
|
coeffs[acn] = scales[acn];
|
|
ComputePanGains(target.Main, coeffs.data(), gain, (*mChans)[c].Target);
|
|
coeffs[acn] = 0.0f;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
DeviceBase *device{context->mDevice};
|
|
al::span<const ChanPosMap> chanmap{};
|
|
switch(mChannels)
|
|
{
|
|
case FmtMono: chanmap = MonoMap; break;
|
|
case FmtSuperStereo:
|
|
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:
|
|
case FmtUHJ2:
|
|
case FmtUHJ3:
|
|
case FmtUHJ4:
|
|
break;
|
|
}
|
|
|
|
mOutTarget = target.Main->Buffer;
|
|
if(device->mRenderMode == RenderMode::Pairwise)
|
|
{
|
|
/* Scales the azimuth of the given vector by 3 if it's in front.
|
|
* Effectively scales +/-45 degrees to +/-90 degrees, leaving > +90
|
|
* and < -90 alone.
|
|
*/
|
|
auto ScaleAzimuthFront = [](std::array<float,3> pos) -> std::array<float,3>
|
|
{
|
|
if(pos[2] < 0.0f)
|
|
{
|
|
/* Normalize the length of the x,z components for a 2D
|
|
* vector of the azimuth angle. Negate Z since {0,0,-1} is
|
|
* angle 0.
|
|
*/
|
|
const float len2d{std::sqrt(pos[0]*pos[0] + pos[2]*pos[2])};
|
|
float x{pos[0] / len2d};
|
|
float z{-pos[2] / len2d};
|
|
|
|
/* Z > cos(pi/4) = -45 < azimuth < 45 degrees. */
|
|
if(z > cos45)
|
|
{
|
|
/* Double the angle represented by x,z. */
|
|
const float resx{2.0f*x * z};
|
|
const float resz{z*z - x*x};
|
|
|
|
/* Scale the vector back to fit in 3D. */
|
|
pos[0] = resx * len2d;
|
|
pos[2] = -resz * len2d;
|
|
}
|
|
else
|
|
{
|
|
/* If azimuth >= 45 degrees, clamp to 90 degrees. */
|
|
pos[0] = std::copysign(len2d, pos[0]);
|
|
pos[2] = 0.0f;
|
|
}
|
|
}
|
|
return pos;
|
|
};
|
|
|
|
for(size_t i{0};i < chanmap.size();++i)
|
|
{
|
|
if(chanmap[i].channel == LFE) continue;
|
|
const auto coeffs = CalcDirectionCoeffs(ScaleAzimuthFront(chanmap[i].pos), 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 = CalcDirectionCoeffs(chanmap[i].pos, 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) UNLIKELY
|
|
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({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_f{});
|
|
forward_fft(al::span{mFftBuffer});
|
|
|
|
std::copy_n(mFftBuffer.cbegin(), m, &mComplexData[curseg*m]);
|
|
|
|
const complex_f *RESTRICT filter{mComplexData.get() + mNumConvolveSegs*m};
|
|
for(size_t c{0};c < chans.size();++c)
|
|
{
|
|
std::fill_n(mFftBuffer.begin(), m, complex_f{});
|
|
|
|
/* Convolve each input segment with its IR filter counterpart
|
|
* (aligned in time).
|
|
*/
|
|
const complex_f *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(al::span{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] =
|
|
(mFftBuffer[i].real()+mOutput[c][ConvolveUpdateSamples+i]) *
|
|
(1.0f/float{ConvolveUpdateSize});
|
|
for(size_t i{0};i < ConvolveUpdateSamples;++i)
|
|
mOutput[c][ConvolveUpdateSamples+i] = mFftBuffer[ConvolveUpdateSamples+i].real();
|
|
}
|
|
|
|
/* 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;
|
|
}
|