Convert the filters to C++

1 files changed, 0 insertions, 123 deletions

diff --git a/Alc/filters/filter.c b/Alc/filters/filter.c
deleted file mode 100644
index 838a9a5d..00000000
--- a/Alc/filters/filter.c
+++ /dev/null
@@ -1,123 +0,0 @@
-
-#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;
-}