#include "config.h" #include "uhjfilter.h" #include <algorithm> #include <iterator> #include "AL/al.h" #include "alnumeric.h" #include "opthelpers.h" namespace { /* This is the maximum number of samples processed for each inner loop * iteration. */ #define MAX_UPDATE_SAMPLES 128 constexpr std::array<float,4> Filter1CoeffSqr{{ 0.479400865589f, 0.876218493539f, 0.976597589508f, 0.997499255936f }}; constexpr std::array<float,4> Filter2CoeffSqr{{ 0.161758498368f, 0.733028932341f, 0.945349700329f, 0.990599156685f }}; void allpass_process(al::span<AllPassState,4> state, float *dst, const float *src, const std::array<float,4> &coeffs, const size_t todo) { const std::array<float,4> aa{coeffs}; std::array<std::array<float,2>,4> z{{state[0].z, state[1].z, state[2].z, state[3].z}}; auto proc_sample = [aa,&z](float sample) noexcept -> float { for(size_t i{0};i < 4;++i) { const float output{sample*aa[i] + z[i][0]}; z[i][0] = z[i][1]; z[i][1] = output*aa[i] - sample; sample = output; } return sample; }; std::transform(src, src+todo, dst, proc_sample); state[0].z = z[0]; state[1].z = z[1]; state[2].z = z[2]; state[3].z = z[3]; } } // namespace /* NOTE: There seems to be a bit of an inconsistency in how this encoding is * supposed to work. Some references, such as * * http://members.tripod.com/martin_leese/Ambisonic/UHJ_file_format.html * * specify a pre-scaling of sqrt(2) on the W channel input, while other * references, such as * * https://en.wikipedia.org/wiki/Ambisonic_UHJ_format#Encoding.5B1.5D * and * https://wiki.xiph.org/Ambisonics#UHJ_format * * do not. The sqrt(2) scaling is in line with B-Format decoder coefficients * which include such a scaling for the W channel input, however the original * source for this equation is a 1985 paper by Michael Gerzon, which does not * apparently include the scaling. Applying the extra scaling creates a louder * result with a narrower stereo image compared to not scaling, and I don't * know which is the intended result. */ void Uhj2Encoder::encode(FloatBufferLine &LeftOut, FloatBufferLine &RightOut, FloatBufferLine *InSamples, const size_t SamplesToDo) { ASSUME(SamplesToDo > 0); const auto winput = al::assume_aligned<16>(InSamples[0].cbegin()); const auto xinput = al::assume_aligned<16>(InSamples[1].cbegin()); const auto yinput = al::assume_aligned<16>(InSamples[2].cbegin()); /* D = 0.6554516*Y */ std::transform(yinput, yinput+SamplesToDo, mTemp.begin(), [](const float y) noexcept -> float { return 0.6554516f*y; }); /* NOTE: Filter1 requires a 1 sample delay for the final output, so take * the last processed sample from the previous run as the first output * sample. */ mSide[0] = mLastY; allpass_process(mFilter1_Y, mSide.data()+1, mTemp.data(), Filter1CoeffSqr, SamplesToDo); mLastY = mSide[SamplesToDo]; /* D += j(-0.3420201*W + 0.5098604*X) */ std::transform(winput, winput+SamplesToDo, xinput, mTemp.begin(), [](const float w, const float x) noexcept -> float { return -0.3420201f*w + 0.5098604f*x; }); allpass_process(mFilter2_WX, mTemp.data(), mTemp.data(), Filter2CoeffSqr, SamplesToDo); for(size_t i{0};i < SamplesToDo;++i) mSide[i] += mTemp[i]; /* S = 0.9396926*W + 0.1855740*X */ std::transform(winput, winput+SamplesToDo, xinput, mTemp.begin(), [](const float w, const float x) noexcept -> float { return 0.9396926f*w + 0.1855740f*x; }); mMid[0] = mLastWX; allpass_process(mFilter1_WX, mMid.data()+1, mTemp.data(), Filter1CoeffSqr, SamplesToDo); mLastWX = mMid[SamplesToDo]; /* Left = (S + D)/2.0 */ float *RESTRICT left{al::assume_aligned<16>(LeftOut.data())}; for(size_t i{0};i < SamplesToDo;i++) left[i] += (mMid[i] + mSide[i]) * 0.5f; /* Right = (S - D)/2.0 */ float *RESTRICT right{al::assume_aligned<16>(RightOut.data())}; for(size_t i{0};i < SamplesToDo;i++) right[i] += (mMid[i] - mSide[i]) * 0.5f; }