Add the Bauer stereophonic-to-binaural DSP (bs2b) code and hooks

1 files changed, 201 insertions, 0 deletions

diff --git a/Alc/bs2b.c b/Alc/bs2b.c
new file mode 100644
index 00000000..069ed614
--- /dev/null
+++ b/Alc/bs2b.c
@@ -0,0 +1,201 @@
+/*-

+ * Copyright (c) 2005 Boris Mikhaylov

+ *

+ * Permission is hereby granted, free of charge, to any person obtaining

+ * a copy of this software and associated documentation files (the

+ * "Software"), to deal in the Software without restriction, including

+ * without limitation the rights to use, copy, modify, merge, publish,

+ * distribute, sublicense, and/or sell copies of the Software, and to

+ * permit persons to whom the Software is furnished to do so, subject to

+ * the following conditions:

+ *

+ * The above copyright notice and this permission notice shall be

+ * included in all copies or substantial portions of the Software.

+ *

+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,

+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF

+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.

+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY

+ * CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,

+ * TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE

+ * SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.

+ */

+

+#include <math.h>

+

+#include "bs2b.h"

+

+#ifndef M_PI

+#define M_PI  3.14159265358979323846

+#endif

+

+/* Single pole IIR filter.

+ * O[n] = a0*I[n] + a1*I[n-1] + b1*O[n-1]

+ */

+

+/* Lowpass filter */

+#define lo_filter(in, out_1) (bs2b->a0_lo*(in) + bs2b->b1_lo*(out_1))

+

+/* Highboost filter */

+#define hi_filter(in, in_1, out_1) (bs2b->a0_hi*(in) + bs2b->a1_hi*(in_1) + bs2b->b1_hi*(out_1))

+

+/* Set up all data. */

+static void init(struct bs2b *bs2b)

+{

+    double Fc_lo, Fc_hi;

+    double G_lo,  G_hi;

+    double x;

+

+    if ((bs2b->srate > 192000) || (bs2b->srate < 2000))

+        bs2b->srate = BS2B_DEFAULT_SRATE;

+

+    switch(bs2b->level)

+    {

+    case BS2B_LOW_CLEVEL: /* Low crossfeed level */

+        Fc_lo = 360.0;

+        Fc_hi = 501.0;

+        G_lo  = 0.398107170553497;

+        G_hi  = 0.205671765275719;

+        break;

+

+    case BS2B_MIDDLE_CLEVEL: /* Middle crossfeed level */

+        Fc_lo = 500.0;

+        Fc_hi = 711.0;

+        G_lo  = 0.459726988530872;

+        G_hi  = 0.228208484414988;

+        break;

+

+    case BS2B_HIGH_CLEVEL: /* High crossfeed level (virtual speakers are closer to itself) */

+        Fc_lo = 700.0;

+        Fc_hi = 1021.0;

+        G_lo  = 0.530884444230988;

+        G_hi  = 0.250105790667544;

+        break;

+

+    case BS2B_LOW_ECLEVEL: /* Low easy crossfeed level */

+        Fc_lo = 360.0;

+        Fc_hi = 494.0;

+        G_lo  = 0.316227766016838;

+        G_hi  = 0.168236228897329;

+        break;

+

+    case BS2B_MIDDLE_ECLEVEL: /* Middle easy crossfeed level */

+        Fc_lo = 500.0;

+        Fc_hi = 689.0;

+        G_lo  = 0.354813389233575;

+        G_hi  = 0.187169483835901;

+        break;

+

+    default: /* High easy crossfeed level */

+        bs2b->level = BS2B_HIGH_ECLEVEL;

+

+        Fc_lo = 700.0;

+        Fc_hi = 975.0;

+        G_lo  = 0.398107170553497;

+        G_hi  = 0.205671765275719;

+        break;

+    } /* switch */

+

+    /* $fc = $Fc / $s;

+     * $d  = 1 / 2 / pi / $fc;

+     * $x  = exp(-1 / $d);

+     */

+

+    x           = exp(-2.0 * M_PI * Fc_lo / bs2b->srate);

+    bs2b->b1_lo = x;

+    bs2b->a0_lo = G_lo * (1.0 - x);

+

+    x           = exp(-2.0 * M_PI * Fc_hi / bs2b->srate);

+    bs2b->b1_hi = x;

+    bs2b->a0_hi = 1.0 - G_hi * (1.0 - x);

+    bs2b->a1_hi = -x;

+

+    bs2b->gain  = 1.0 / (1.0 - G_hi + G_lo);

+

+    bs2b_clear(bs2b);

+} /* init */

+

+/* Exported functions.

+ * See descriptions in "bs2b.h"

+ */

+

+void bs2b_set_level(struct bs2b *bs2b, int level)

+{

+    if(level == bs2b->level)

+        return;

+    bs2b->level = level;

+    init(bs2b);

+} /* bs2b_set_level */

+

+int bs2b_get_level(struct bs2b *bs2b)

+{

+    return bs2b->level;

+} /* bs2b_get_level */

+

+void bs2b_set_srate(struct bs2b *bs2b, int srate)

+{

+    if (srate == bs2b->srate)

+        return;

+    bs2b->srate = srate;

+    init(bs2b);

+} /* bs2b_set_srate */

+

+int bs2b_get_srate(struct bs2b *bs2b)

+{

+    return bs2b->srate;

+} /* bs2b_get_srate */

+

+void bs2b_clear(struct bs2b *bs2b)

+{

+    int loopv = sizeof(bs2b->last_sample);

+

+    while (loopv)

+    {

+        ((char *)&bs2b->last_sample)[--loopv] = 0;

+    }

+} /* bs2b_clear */

+

+int bs2b_is_clear(struct bs2b *bs2b)

+{

+    int loopv = sizeof(bs2b->last_sample);

+

+    while (loopv)

+    {

+        if (((char *)&bs2b->last_sample)[--loopv] != 0)

+            return 0;

+    }

+    return 1;

+} /* bs2b_is_clear */

+

+void bs2b_cross_feed(struct bs2b *bs2b, float *sample)

+{

+    /* Lowpass filter */

+    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(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 */

+    sample[0] = bs2b->last_sample.hi[0] + bs2b->last_sample.lo[1];

+    sample[1] = bs2b->last_sample.hi[1] + bs2b->last_sample.lo[0];

+

+    /* Bass boost cause allpass attenuation */

+    sample[0] *= bs2b->gain;

+    sample[1] *= bs2b->gain;

+

+    /* Clipping of overloaded samples */

+#if 0

+    if (sample[0] > 1.0)

+        sample[0] = 1.0;

+    if (sample[0] < -1.0)

+        sample[0] = -1.0;

+    if (sample[1] > 1.0)

+        sample[1] = 1.0;

+    if (sample[1] < -1.0)

+        sample[1] = -1.0;

+#endif

+} /* bs2b_cross_feed */