#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 LeftOut, ALfloat *restrict RightOut, 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++) *(LeftOut++) += (S[i] + D[i]) * 0.5f; /* Right = (S - D)/2.0 */ for(i = 0;i < todo;i++) *(RightOut++) += (S[i] - D[i]) * 0.5f; base += todo; } }