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#include "config.h"
#include "alu.h"
#include "uhjfilter.h"
/* This is the maximum number of samples processed for each inner loop
* iteration. */
#define MAX_UPDATE_SAMPLES 256
static const ALfloat Filter1Coeff[4] = {
0.6923878f, 0.9360654322959f, 0.9882295226860f, 0.9987488452737f
};
static const ALfloat Filter2Coeff[4] = {
0.4021921162426f, 0.8561710882420f, 0.9722909545651f, 0.9952884791278f
};
/* 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 EncodeUhj2(Uhj2Encoder *enc, ALfloat (*restrict OutBuffer)[BUFFERSIZE], ALfloat (*restrict InSamples)[BUFFERSIZE], ALuint SamplesToDo)
{
ALuint base, i, c;
for(base = 0;base < SamplesToDo;)
{
ALfloat D[MAX_UPDATE_SAMPLES/2], S[MAX_UPDATE_SAMPLES/2];
ALuint todo = minu(SamplesToDo - base, MAX_UPDATE_SAMPLES/2);
/* D = 0.6554516*Y */
for(i = 0;i < todo;i++)
{
ALfloat in = 0.6554516f*InSamples[2][base+i];
for(c = 0;c < 4;c++)
{
ALfloat aa = Filter1Coeff[c]*Filter1Coeff[c];
ALfloat out = aa*(in + enc->Filter1_Y[c].y[1]) - enc->Filter1_Y[c].x[1];
enc->Filter1_Y[c].x[1] = enc->Filter1_Y[c].x[0];
enc->Filter1_Y[c].x[0] = in;
enc->Filter1_Y[c].y[1] = enc->Filter1_Y[c].y[0];
enc->Filter1_Y[c].y[0] = out;
in = out;
}
/* NOTE: Filter1 requires a 1 sample delay for the base output, so
* take the sample before the last for output.
*/
D[i] = enc->Filter1_Y[3].y[1];
}
/* D += j(-0.3420201*W + 0.5098604*X) */
for(i = 0;i < todo;i++)
{
ALfloat in = -0.3420201f*InSamples[0][base+i] +
0.5098604f*InSamples[1][base+i];
for(c = 0;c < 4;c++)
{
ALfloat aa = Filter2Coeff[c]*Filter2Coeff[c];
ALfloat out = aa*(in + enc->Filter2_WX[c].y[1]) - enc->Filter2_WX[c].x[1];
enc->Filter2_WX[c].x[1] = enc->Filter2_WX[c].x[0];
enc->Filter2_WX[c].x[0] = in;
enc->Filter2_WX[c].y[1] = enc->Filter2_WX[c].y[0];
enc->Filter2_WX[c].y[0] = out;
in = out;
}
D[i] += enc->Filter2_WX[3].y[0];
}
/* S = 0.9396926*W + 0.1855740*X */
for(i = 0;i < todo;i++)
{
ALfloat in = 0.9396926f*InSamples[0][base+i] +
0.1855740f*InSamples[1][base+i];
for(c = 0;c < 4;c++)
{
ALfloat aa = Filter1Coeff[c]*Filter1Coeff[c];
ALfloat out = aa*(in + enc->Filter1_WX[c].y[1]) - enc->Filter1_WX[c].x[1];
enc->Filter1_WX[c].x[1] = enc->Filter1_WX[c].x[0];
enc->Filter1_WX[c].x[0] = in;
enc->Filter1_WX[c].y[1] = enc->Filter1_WX[c].y[0];
enc->Filter1_WX[c].y[0] = out;
in = out;
}
S[i] = enc->Filter1_WX[3].y[1];
}
/* Left = (S + D)/2.0 */
for(i = 0;i < todo;i++)
OutBuffer[0][base + i] += (S[i] + D[i]) * 0.5f;
/* Right = (S - D)/2.0 */
for(i = 0;i < todo;i++)
OutBuffer[1][base + i] += (S[i] - D[i]) * 0.5f;
base += todo;
}
}
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