#include "config.h" #include "alu.h" #include "uhjfilter.h" namespace { /* This is the maximum number of samples processed for each inner loop * iteration. */ #define MAX_UPDATE_SAMPLES 128 constexpr ALfloat Filter1CoeffSqr[4] = { 0.479400865589f, 0.876218493539f, 0.976597589508f, 0.997499255936f }; constexpr ALfloat Filter2CoeffSqr[4] = { 0.161758498368f, 0.733028932341f, 0.945349700329f, 0.990599156685f }; void allpass_process(AllPassState *state, ALfloat *RESTRICT dst, const ALfloat *RESTRICT src, const ALfloat aa, ALsizei todo) { ALfloat z1 = state->z[0]; ALfloat z2 = state->z[1]; ALsizei i; for(i = 0;i < todo;i++) { ALfloat input = src[i]; ALfloat output = input*aa + z1; z1 = z2; z2 = output*aa - input; dst[i] = output; } state->z[0] = z1; state->z[1] = z2; } } // 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 EncodeUhj2(Uhj2Encoder *enc, ALfloat *RESTRICT LeftOut, ALfloat *RESTRICT RightOut, ALfloat (*RESTRICT InSamples)[BUFFERSIZE], ALsizei SamplesToDo) { ALfloat D[MAX_UPDATE_SAMPLES], S[MAX_UPDATE_SAMPLES]; ALfloat temp[2][MAX_UPDATE_SAMPLES]; ALsizei base, i; ASSUME(SamplesToDo > 0); for(base = 0;base < SamplesToDo;) { ALsizei todo = mini(SamplesToDo - base, MAX_UPDATE_SAMPLES); ASSUME(todo > 0); /* D = 0.6554516*Y */ for(i = 0;i < todo;i++) temp[0][i] = 0.6554516f*InSamples[2][base+i]; allpass_process(&enc->Filter1_Y[0], temp[1], temp[0], Filter1CoeffSqr[0], todo); allpass_process(&enc->Filter1_Y[1], temp[0], temp[1], Filter1CoeffSqr[1], todo); allpass_process(&enc->Filter1_Y[2], temp[1], temp[0], Filter1CoeffSqr[2], todo); allpass_process(&enc->Filter1_Y[3], temp[0], temp[1], Filter1CoeffSqr[3], todo); /* 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. */ D[0] = enc->LastY; for(i = 1;i < todo;i++) D[i] = temp[0][i-1]; enc->LastY = temp[0][i-1]; /* D += j(-0.3420201*W + 0.5098604*X) */ for(i = 0;i < todo;i++) temp[0][i] = -0.3420201f*InSamples[0][base+i] + 0.5098604f*InSamples[1][base+i]; allpass_process(&enc->Filter2_WX[0], temp[1], temp[0], Filter2CoeffSqr[0], todo); allpass_process(&enc->Filter2_WX[1], temp[0], temp[1], Filter2CoeffSqr[1], todo); allpass_process(&enc->Filter2_WX[2], temp[1], temp[0], Filter2CoeffSqr[2], todo); allpass_process(&enc->Filter2_WX[3], temp[0], temp[1], Filter2CoeffSqr[3], todo); for(i = 0;i < todo;i++) D[i] += temp[0][i]; /* S = 0.9396926*W + 0.1855740*X */ for(i = 0;i < todo;i++) temp[0][i] = 0.9396926f*InSamples[0][base+i] + 0.1855740f*InSamples[1][base+i]; allpass_process(&enc->Filter1_WX[0], temp[1], temp[0], Filter1CoeffSqr[0], todo); allpass_process(&enc->Filter1_WX[1], temp[0], temp[1], Filter1CoeffSqr[1], todo); allpass_process(&enc->Filter1_WX[2], temp[1], temp[0], Filter1CoeffSqr[2], todo); allpass_process(&enc->Filter1_WX[3], temp[0], temp[1], Filter1CoeffSqr[3], todo); S[0] = enc->LastWX; for(i = 1;i < todo;i++) S[i] = temp[0][i-1]; enc->LastWX = temp[0][i-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; } }