AuroraOpenALSoft/Alc/mixer_c.c
Chris Robinson 444e9563b3 Add a mixing function to blend HRIRs
This is a bit more efficient than calling the normal HRTF mixing function
twice, and helps solve the problem of the values generated from convolution not
being consistent with the new HRIR.
2017-05-03 03:29:21 -07:00

209 lines
7.2 KiB
C

#include "config.h"
#include <assert.h>
#include "alMain.h"
#include "alu.h"
#include "alSource.h"
#include "alAuxEffectSlot.h"
static inline ALfloat point32(const ALfloat *restrict vals, ALsizei UNUSED(frac))
{ return vals[0]; }
static inline ALfloat lerp32(const ALfloat *restrict vals, ALsizei frac)
{ return lerp(vals[0], vals[1], frac * (1.0f/FRACTIONONE)); }
static inline ALfloat fir4_32(const ALfloat *restrict vals, ALsizei frac)
{ return resample_fir4(vals[-1], vals[0], vals[1], vals[2], frac); }
const ALfloat *Resample_copy32_C(const InterpState* UNUSED(state),
const ALfloat *restrict src, ALsizei UNUSED(frac), ALint UNUSED(increment),
ALfloat *restrict dst, ALsizei numsamples)
{
#if defined(HAVE_SSE) || defined(HAVE_NEON)
/* Avoid copying the source data if it's aligned like the destination. */
if((((intptr_t)src)&15) == (((intptr_t)dst)&15))
return src;
#endif
memcpy(dst, src, numsamples*sizeof(ALfloat));
return dst;
}
#define DECL_TEMPLATE(Sampler) \
const ALfloat *Resample_##Sampler##_C(const InterpState* UNUSED(state), \
const ALfloat *restrict src, ALsizei frac, ALint increment, \
ALfloat *restrict dst, ALsizei numsamples) \
{ \
ALsizei i; \
for(i = 0;i < numsamples;i++) \
{ \
dst[i] = Sampler(src, frac); \
\
frac += increment; \
src += frac>>FRACTIONBITS; \
frac &= FRACTIONMASK; \
} \
return dst; \
}
DECL_TEMPLATE(point32)
DECL_TEMPLATE(lerp32)
DECL_TEMPLATE(fir4_32)
#undef DECL_TEMPLATE
const ALfloat *Resample_bsinc32_C(const InterpState *state, const ALfloat *restrict src,
ALsizei frac, ALint increment, ALfloat *restrict dst,
ALsizei dstlen)
{
const ALfloat *fil, *scd, *phd, *spd;
const ALfloat sf = state->bsinc.sf;
const ALsizei m = state->bsinc.m;
ALsizei j_f, pi, i;
ALfloat pf, r;
src += state->bsinc.l;
for(i = 0;i < dstlen;i++)
{
// Calculate the phase index and factor.
#define FRAC_PHASE_BITDIFF (FRACTIONBITS-BSINC_PHASE_BITS)
pi = frac >> FRAC_PHASE_BITDIFF;
pf = (frac & ((1<<FRAC_PHASE_BITDIFF)-1)) * (1.0f/(1<<FRAC_PHASE_BITDIFF));
#undef FRAC_PHASE_BITDIFF
fil = ASSUME_ALIGNED(state->bsinc.coeffs[pi].filter, 16);
scd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].scDelta, 16);
phd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].phDelta, 16);
spd = ASSUME_ALIGNED(state->bsinc.coeffs[pi].spDelta, 16);
// Apply the scale and phase interpolated filter.
r = 0.0f;
for(j_f = 0;j_f < m;j_f++)
r += (fil[j_f] + sf*scd[j_f] + pf*(phd[j_f] + sf*spd[j_f])) * src[j_f];
dst[i] = r;
frac += increment;
src += frac>>FRACTIONBITS;
frac &= FRACTIONMASK;
}
return dst;
}
void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
{
ALsizei i;
if(numsamples > 1)
{
dst[0] = filter->b0 * src[0] +
filter->b1 * filter->x[0] +
filter->b2 * filter->x[1] -
filter->a1 * filter->y[0] -
filter->a2 * filter->y[1];
dst[1] = filter->b0 * src[1] +
filter->b1 * src[0] +
filter->b2 * filter->x[0] -
filter->a1 * dst[0] -
filter->a2 * filter->y[0];
for(i = 2;i < numsamples;i++)
dst[i] = filter->b0 * src[i] +
filter->b1 * src[i-1] +
filter->b2 * src[i-2] -
filter->a1 * dst[i-1] -
filter->a2 * dst[i-2];
filter->x[0] = src[i-1];
filter->x[1] = src[i-2];
filter->y[0] = dst[i-1];
filter->y[1] = dst[i-2];
}
else if(numsamples == 1)
{
dst[0] = filter->b0 * src[0] +
filter->b1 * filter->x[0] +
filter->b2 * filter->x[1] -
filter->a1 * filter->y[0] -
filter->a2 * filter->y[1];
filter->x[1] = filter->x[0];
filter->x[0] = src[0];
filter->y[1] = filter->y[0];
filter->y[0] = dst[0];
}
}
static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
const ALsizei IrSize,
const ALfloat (*restrict Coeffs)[2],
ALfloat left, ALfloat right)
{
ALsizei c;
for(c = 0;c < IrSize;c++)
{
const ALsizei off = (Offset+c)&HRIR_MASK;
Values[off][0] += Coeffs[c][0] * left;
Values[off][1] += Coeffs[c][1] * right;
}
}
#define MixHrtf MixHrtf_C
#define MixHrtfBlend MixHrtfBlend_C
#define MixDirectHrtf MixDirectHrtf_C
#include "mixer_inc.c"
#undef MixHrtf
void Mix_C(const ALfloat *data, ALsizei OutChans, ALfloat (*restrict OutBuffer)[BUFFERSIZE],
ALfloat *CurrentGains, const ALfloat *TargetGains, ALsizei Counter, ALsizei OutPos,
ALsizei BufferSize)
{
ALfloat gain, delta, step;
ALsizei c;
delta = (Counter > 0) ? 1.0f/(ALfloat)Counter : 0.0f;
for(c = 0;c < OutChans;c++)
{
ALsizei pos = 0;
gain = CurrentGains[c];
step = (TargetGains[c] - gain) * delta;
if(fabsf(step) > FLT_EPSILON)
{
ALsizei minsize = mini(BufferSize, Counter);
for(;pos < minsize;pos++)
{
OutBuffer[c][OutPos+pos] += data[pos]*gain;
gain += step;
}
if(pos == Counter)
gain = TargetGains[c];
CurrentGains[c] = gain;
}
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(;pos < BufferSize;pos++)
OutBuffer[c][OutPos+pos] += data[pos]*gain;
}
}
/* Basically the inverse of the above. Rather than one input going to multiple
* outputs (each with its own gain), it's multiple inputs (each with its own
* gain) going to one output. This applies one row (vs one column) of a matrix
* transform. And as the matrices are more or less static once set up, no
* stepping is necessary.
*/
void MixRow_C(ALfloat *OutBuffer, const ALfloat *Gains, const ALfloat (*restrict data)[BUFFERSIZE], ALsizei InChans, ALsizei InPos, ALsizei BufferSize)
{
ALsizei c, i;
for(c = 0;c < InChans;c++)
{
ALfloat gain = Gains[c];
if(!(fabsf(gain) > GAIN_SILENCE_THRESHOLD))
continue;
for(i = 0;i < BufferSize;i++)
OutBuffer[i] += data[c][InPos+i] * gain;
}
}