Use ALfilterState for the distortion effect filters

2 files changed, 27 insertions, 0 deletions

diff --git a/OpenAL32/Include/alFilter.h b/OpenAL32/Include/alFilter.h
index 6e9abd6a..690a496b 100644
--- a/OpenAL32/Include/alFilter.h
+++ b/OpenAL32/Include/alFilter.h
@@ -9,10 +9,18 @@ extern "C" {
 
 #define LOWPASSFREQREF  (5000)
 
+
+/* Filters implementation is based on the "Cookbook formulae for audio   *
+ * EQ biquad filter coefficients" by Robert Bristow-Johnson              *
+ * http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt                   */
+
 typedef enum ALfilterType {
     ALfilterType_HighShelf,
     ALfilterType_LowShelf,
     ALfilterType_Peaking,
+
+    ALfilterType_LowPass,
+    ALfilterType_BandPass,
 } ALfilterType;
 
 typedef struct ALfilterState {
diff --git a/OpenAL32/alFilter.c b/OpenAL32/alFilter.c
index abfcbdd4..f15a2800 100644
--- a/OpenAL32/alFilter.c
+++ b/OpenAL32/alFilter.c
@@ -392,6 +392,25 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
             filter->a[1] = -2.0f * cosf(w0);
             filter->a[2] =  1.0f - alpha / gain;
             break;
+
+        case ALfilterType_LowPass:
+            alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
+            filter->b[0] = (1.0f - cosf(w0)) / 2.0f;
+            filter->b[1] =  1.0f - cosf(w0);
+            filter->b[2] = (1.0f - cosf(w0)) / 2.0f;
+            filter->a[0] =  1.0f + alpha;
+            filter->a[1] = -2.0f * cosf(w0);
+            filter->a[2] =  1.0f - alpha;
+            break;
+        case ALfilterType_BandPass:
+            alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
+            filter->b[0] =  alpha;
+            filter->b[1] =  0;
+            filter->b[2] = -alpha;
+            filter->a[0] =  1.0f + alpha;
+            filter->a[1] = -2.0f * cosf(w0);
+            filter->a[2] =  1.0f - alpha;
+            break;
     }
 
     filter->b[2] /= filter->a[0];