Convert the filters to C++

1 files changed, 123 insertions, 0 deletions

diff --git a/Alc/filters/filter.cpp b/Alc/filters/filter.cpp
new file mode 100644
index 00000000..838a9a5d
--- /dev/null
+++ b/Alc/filters/filter.cpp
@@ -0,0 +1,123 @@
+
+#include "config.h"
+
+#include "AL/alc.h"
+#include "AL/al.h"
+
+#include "alMain.h"
+#include "defs.h"
+
+
+void BiquadFilter_setParams(BiquadFilter *filter, BiquadType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ)
+{
+    ALfloat alpha, sqrtgain_alpha_2;
+    ALfloat w0, sin_w0, cos_w0;
+    ALfloat a[3] = { 1.0f, 0.0f, 0.0f };
+    ALfloat b[3] = { 1.0f, 0.0f, 0.0f };
+
+    // Limit gain to -100dB
+    assert(gain > 0.00001f);
+
+    w0 = F_TAU * f0norm;
+    sin_w0 = sinf(w0);
+    cos_w0 = cosf(w0);
+    alpha = sin_w0/2.0f * rcpQ;
+
+    /* Calculate filter coefficients depending on filter type */
+    switch(type)
+    {
+        case BiquadType_HighShelf:
+            sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
+            b[0] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
+            b[1] = -2.0f*gain*((gain-1.0f) + (gain+1.0f)*cos_w0                   );
+            b[2] =       gain*((gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
+            a[0] =             (gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
+            a[1] =  2.0f*     ((gain-1.0f) - (gain+1.0f)*cos_w0                   );
+            a[2] =             (gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
+            break;
+        case BiquadType_LowShelf:
+            sqrtgain_alpha_2 = 2.0f * sqrtf(gain) * alpha;
+            b[0] =       gain*((gain+1.0f) - (gain-1.0f)*cos_w0 + sqrtgain_alpha_2);
+            b[1] =  2.0f*gain*((gain-1.0f) - (gain+1.0f)*cos_w0                   );
+            b[2] =       gain*((gain+1.0f) - (gain-1.0f)*cos_w0 - sqrtgain_alpha_2);
+            a[0] =             (gain+1.0f) + (gain-1.0f)*cos_w0 + sqrtgain_alpha_2;
+            a[1] = -2.0f*     ((gain-1.0f) + (gain+1.0f)*cos_w0                   );
+            a[2] =             (gain+1.0f) + (gain-1.0f)*cos_w0 - sqrtgain_alpha_2;
+            break;
+        case BiquadType_Peaking:
+            gain = sqrtf(gain);
+            b[0] =  1.0f + alpha * gain;
+            b[1] = -2.0f * cos_w0;
+            b[2] =  1.0f - alpha * gain;
+            a[0] =  1.0f + alpha / gain;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha / gain;
+            break;
+
+        case BiquadType_LowPass:
+            b[0] = (1.0f - cos_w0) / 2.0f;
+            b[1] =  1.0f - cos_w0;
+            b[2] = (1.0f - cos_w0) / 2.0f;
+            a[0] =  1.0f + alpha;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha;
+            break;
+        case BiquadType_HighPass:
+            b[0] =  (1.0f + cos_w0) / 2.0f;
+            b[1] = -(1.0f + cos_w0);
+            b[2] =  (1.0f + cos_w0) / 2.0f;
+            a[0] =   1.0f + alpha;
+            a[1] =  -2.0f * cos_w0;
+            a[2] =   1.0f - alpha;
+            break;
+        case BiquadType_BandPass:
+            b[0] =  alpha;
+            b[1] =  0;
+            b[2] = -alpha;
+            a[0] =  1.0f + alpha;
+            a[1] = -2.0f * cos_w0;
+            a[2] =  1.0f - alpha;
+            break;
+    }
+
+    filter->a1 = a[1] / a[0];
+    filter->a2 = a[2] / a[0];
+    filter->b0 = b[0] / a[0];
+    filter->b1 = b[1] / a[0];
+    filter->b2 = b[2] / a[0];
+}
+
+
+void BiquadFilter_processC(BiquadFilter *filter, ALfloat *RESTRICT dst, const ALfloat *RESTRICT src, ALsizei numsamples)
+{
+    const ALfloat a1 = filter->a1;
+    const ALfloat a2 = filter->a2;
+    const ALfloat b0 = filter->b0;
+    const ALfloat b1 = filter->b1;
+    const ALfloat b2 = filter->b2;
+    ALfloat z1 = filter->z1;
+    ALfloat z2 = filter->z2;
+    ALsizei i;
+
+    ASSUME(numsamples > 0);
+
+    /* Processing loop is Transposed Direct Form II. This requires less storage
+     * compared to Direct Form I (only two delay components, instead of a four-
+     * sample history; the last two inputs and outputs), and works better for
+     * floating-point which favors summing similarly-sized values while being
+     * less bothered by overflow.
+     *
+     * See: http://www.earlevel.com/main/2003/02/28/biquads/
+     */
+    for(i = 0;i < numsamples;i++)
+    {
+        ALfloat input = src[i];
+        ALfloat output = input*b0 + z1;
+        z1 = input*b1 - output*a1 + z2;
+        z2 = input*b2 - output*a2;
+        dst[i] = output;
+    }
+
+    filter->z1 = z1;
+    filter->z2 = z2;
+}