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Pre-apply the crossfeed filter gain to the input sample coefficients

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This commit is contained in:
Chris Robinson
2014-06-12 09:40:30 -07:00
parent 4c706f59d9
commit 19ec7b2ad2
2 changed files with 16 additions and 26 deletions
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Alc/bs2b.c
+6 -7
View File
@@ -38,7 +38,7 @@ static void init(struct bs2b *bs2b)
{
float Fc_lo, Fc_hi;
float G_lo, G_hi;
float x;
float x, g;
bs2b->srate = clampi(bs2b->srate, 2000, 192000);
@@ -89,21 +89,20 @@ static void init(struct bs2b *bs2b)
break;
} /* switch */
g = 1.0f / (1.0f - G_hi + G_lo);
/* $fc = $Fc / $s;
* $d = 1 / 2 / pi / $fc;
* $x = exp(-1 / $d);
*/
x = expf(-2.0f * F_PI * Fc_lo / bs2b->srate);
bs2b->b1_lo = x;
bs2b->a0_lo = G_lo * (1.0f - x);
bs2b->a0_lo = G_lo * (1.0f - x) * g;
x = expf(-2.0f * F_PI * Fc_hi / bs2b->srate);
bs2b->b1_hi = x;
bs2b->a0_hi = 1.0f - G_hi * (1.0f - x);
bs2b->a1_hi = -x;
bs2b->gain = 1.0f / (1.0f - G_hi + G_lo);
bs2b->a0_hi = (1.0f - G_hi * (1.0f - x)) * g;
bs2b->a1_hi = -x * g;
} /* init */
/* Exported functions.
OpenAL32/Include/bs2b.h
+10 -19
View File
@@ -59,9 +59,6 @@ struct bs2b {
float a1_hi;
float b1_hi;
/* Global gain against overloading */
float gain;
/* Buffer of last filtered sample.
* [0] - first channel, [1] - second channel
*/
@@ -93,11 +90,9 @@ void bs2b_clear(struct bs2b *bs2b);
/* Crossfeeds one stereo sample that are pointed by sample.
* [0] - first channel, [1] - second channel.
* Returns crossfided samle by sample pointer.
* Returns crossfided sample by sample pointer.
*/
/* sample points to floats */
inline void bs2b_cross_feed(struct bs2b *bs2b, float *restrict samples)
inline void bs2b_cross_feed(struct bs2b *bs2b, float *restrict sample)
{
/* Single pole IIR filter.
* O[n] = a0*I[n] + a1*I[n-1] + b1*O[n-1]
@@ -110,22 +105,18 @@ inline void bs2b_cross_feed(struct bs2b *bs2b, float *restrict samples)
#define hi_filter(in, in_1, out_1) (bs2b->a0_hi*(in) + bs2b->a1_hi*(in_1) + bs2b->b1_hi*(out_1))
/* Lowpass filter */
bs2b->last_sample.lo[0] = lo_filter(samples[0], bs2b->last_sample.lo[0]);
bs2b->last_sample.lo[1] = lo_filter(samples[1], bs2b->last_sample.lo[1]);
bs2b->last_sample.lo[0] = lo_filter(sample[0], bs2b->last_sample.lo[0]);
bs2b->last_sample.lo[1] = lo_filter(sample[1], bs2b->last_sample.lo[1]);
/* Highboost filter */
bs2b->last_sample.hi[0] = hi_filter(samples[0], bs2b->last_sample.asis[0], bs2b->last_sample.hi[0]);
bs2b->last_sample.hi[1] = hi_filter(samples[1], bs2b->last_sample.asis[1], bs2b->last_sample.hi[1]);
bs2b->last_sample.asis[0] = samples[0];
bs2b->last_sample.asis[1] = samples[1];
bs2b->last_sample.hi[0] = hi_filter(sample[0], bs2b->last_sample.asis[0], bs2b->last_sample.hi[0]);
bs2b->last_sample.hi[1] = hi_filter(sample[1], bs2b->last_sample.asis[1], bs2b->last_sample.hi[1]);
bs2b->last_sample.asis[0] = sample[0];
bs2b->last_sample.asis[1] = sample[1];
/* Crossfeed */
samples[0] = bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1];
samples[1] = bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0];
/* Bass boost cause allpass attenuation */
samples[0] *= bs2b->gain;
samples[1] *= bs2b->gain;
sample[0] = bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1];
sample[1] = bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0];
#undef hi_filter
#undef lo_filter
} /* bs2b_cross_feed */