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#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;
}
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