Move the filter implementation to a separate directory

17 files changed, 266 insertions, 249 deletions

diff --git a/Alc/ALc.c b/Alc/ALc.c
index 8cfc7d25..597cc890 100644
--- a/Alc/ALc.c
+++ b/Alc/ALc.c
@@ -34,6 +34,8 @@
 #include "alListener.h"
 #include "alSource.h"
 #include "alBuffer.h"
+#include "alFilter.h"
+#include "alEffect.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "mastering.h"
diff --git a/Alc/effects/chorus.c b/Alc/effects/chorus.c
index 4f8c7ce3..eb0818e3 100644
--- a/Alc/effects/chorus.c
+++ b/Alc/effects/chorus.c
@@ -24,10 +24,10 @@
 #include <stdlib.h>
 
 #include "alMain.h"
-#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 static_assert(AL_CHORUS_WAVEFORM_SINUSOID == AL_FLANGER_WAVEFORM_SINUSOID, "Chorus/Flanger waveform value mismatch");
diff --git a/Alc/effects/dedicated.c b/Alc/effects/dedicated.c
index b9dd7db3..62a3894f 100644
--- a/Alc/effects/dedicated.c
+++ b/Alc/effects/dedicated.c
@@ -23,10 +23,10 @@
 #include <stdlib.h>
 
 #include "alMain.h"
-#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 typedef struct ALdedicatedState {
diff --git a/Alc/effects/distortion.c b/Alc/effects/distortion.c
index b112032c..09289175 100644
--- a/Alc/effects/distortion.c
+++ b/Alc/effects/distortion.c
@@ -24,10 +24,10 @@
 #include <stdlib.h>
 
 #include "alMain.h"
-#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 typedef struct ALdistortionState {
diff --git a/Alc/effects/echo.c b/Alc/effects/echo.c
index 34010c47..10e00f39 100644
--- a/Alc/effects/echo.c
+++ b/Alc/effects/echo.c
@@ -28,6 +28,7 @@
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 typedef struct ALechoState {
diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c
index 22db5d54..a5c65cee 100644
--- a/Alc/effects/equalizer.c
+++ b/Alc/effects/equalizer.c
@@ -24,10 +24,10 @@
 #include <stdlib.h>
 
 #include "alMain.h"
-#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 /*  The document  "Effects Extension Guide.pdf"  says that low and high  *
diff --git a/Alc/effects/modulator.c b/Alc/effects/modulator.c
index 9ec0d47e..bf8e8fd7 100644
--- a/Alc/effects/modulator.c
+++ b/Alc/effects/modulator.c
@@ -24,10 +24,10 @@
 #include <stdlib.h>
 
 #include "alMain.h"
-#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 #define MAX_UPDATE_SAMPLES 128
diff --git a/Alc/effects/pshifter.c b/Alc/effects/pshifter.c
index d36597ae..6542be56 100644
--- a/Alc/effects/pshifter.c
+++ b/Alc/effects/pshifter.c
@@ -24,10 +24,10 @@
 #include <stdlib.h>
 
 #include "alMain.h"
-#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "alError.h"
 #include "alu.h"
+#include "filters/defs.h"
 
 
 #define MAX_SIZE 2048
diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c
index ff1ee143..cd38b492 100644
--- a/Alc/effects/reverb.c
+++ b/Alc/effects/reverb.c
@@ -27,10 +27,9 @@
 #include "alMain.h"
 #include "alu.h"
 #include "alAuxEffectSlot.h"
-#include "alEffect.h"
-#include "alFilter.h"
 #include "alListener.h"
 #include "alError.h"
+#include "filters/defs.h"
 
 /* This is a user config option for modifying the overall output of the reverb
  * effect.
diff --git a/Alc/filters/defs.h b/Alc/filters/defs.h
new file mode 100644
index 00000000..f4e1e62c
--- /dev/null
+++ b/Alc/filters/defs.h
@@ -0,0 +1,118 @@
+#ifndef ALC_FILTER_H
+#define ALC_FILTER_H
+
+#include "AL/al.h"
+#include "math_defs.h"
+
+/* 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
+ */
+/* Implementation note: For the shelf filters, the specified gain is for the
+ * reference frequency, which is the centerpoint of the transition band. This
+ * better matches EFX filter design. To set the gain for the shelf itself, use
+ * the square root of the desired linear gain (or halve the dB gain).
+ */
+
+typedef enum ALfilterType {
+    /** EFX-style low-pass filter, specifying a gain and reference frequency. */
+    ALfilterType_HighShelf,
+    /** EFX-style high-pass filter, specifying a gain and reference frequency. */
+    ALfilterType_LowShelf,
+    /** Peaking filter, specifying a gain and reference frequency. */
+    ALfilterType_Peaking,
+
+    /** Low-pass cut-off filter, specifying a cut-off frequency. */
+    ALfilterType_LowPass,
+    /** High-pass cut-off filter, specifying a cut-off frequency. */
+    ALfilterType_HighPass,
+    /** Band-pass filter, specifying a center frequency. */
+    ALfilterType_BandPass,
+} ALfilterType;
+
+typedef struct ALfilterState {
+    ALfloat x[2]; /* History of two last input samples  */
+    ALfloat y[2]; /* History of two last output samples */
+    ALfloat b0, b1, b2; /* Transfer function coefficients "b" */
+    ALfloat a1, a2; /* Transfer function coefficients "a" (a0 is pre-applied) */
+} ALfilterState;
+/* Currently only a C-based filter process method is implemented. */
+#define ALfilterState_process ALfilterState_processC
+
+/**
+ * Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
+ * reference gain and shelf slope parameter.
+ * \param gain 0 < gain
+ * \param slope 0 < slope <= 1
+ */
+inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
+{
+    return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
+}
+/**
+ * Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the normalized
+ * reference frequency and bandwidth.
+ * \param f0norm 0 < f0norm < 0.5.
+ * \param bandwidth 0 < bandwidth
+ */
+inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth)
+{
+    ALfloat w0 = F_TAU * f0norm;
+    return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
+}
+
+inline void ALfilterState_clear(ALfilterState *filter)
+{
+    filter->x[0] = 0.0f;
+    filter->x[1] = 0.0f;
+    filter->y[0] = 0.0f;
+    filter->y[1] = 0.0f;
+}
+
+/**
+ * Sets up the filter state for the specified filter type and its parameters.
+ *
+ * \param filter The filter object to prepare.
+ * \param type The type of filter for the object to apply.
+ * \param gain The gain for the reference frequency response. Only used by the
+ *             Shelf and Peaking filter types.
+ * \param f0norm The normalized reference frequency (ref_freq / sample_rate).
+ *               This is the center point for the Shelf, Peaking, and BandPass
+ *               filter types, or the cutoff frequency for the LowPass and
+ *               HighPass filter types.
+ * \param rcpQ The reciprocal of the Q coefficient for the filter's transition
+ *             band. Can be generated from calc_rcpQ_from_slope or
+ *             calc_rcpQ_from_bandwidth depending on the available data.
+ */
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ);
+
+inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src)
+{
+    dst->b0 = src->b0;
+    dst->b1 = src->b1;
+    dst->b2 = src->b2;
+    dst->a1 = src->a1;
+    dst->a2 = src->a2;
+}
+
+void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
+
+inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples)
+{
+    if(numsamples >= 2)
+    {
+        filter->x[1] = src[numsamples-2];
+        filter->x[0] = src[numsamples-1];
+        filter->y[1] = src[numsamples-2];
+        filter->y[0] = src[numsamples-1];
+    }
+    else if(numsamples == 1)
+    {
+        filter->x[1] = filter->x[0];
+        filter->x[0] = src[0];
+        filter->y[1] = filter->y[0];
+        filter->y[0] = src[0];
+    }
+}
+
+#endif /* ALC_FILTER_H */
diff --git a/Alc/filters/filter.c b/Alc/filters/filter.c
new file mode 100644
index 00000000..1cf18f08
--- /dev/null
+++ b/Alc/filters/filter.c
@@ -0,0 +1,133 @@
+
+#include "config.h"
+
+#include "AL/alc.h"
+#include "AL/al.h"
+
+#include "alMain.h"
+#include "defs.h"
+
+extern inline void ALfilterState_clear(ALfilterState *filter);
+extern inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src);
+extern inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples);
+extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope);
+extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth);
+
+
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType 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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
+{
+    ALsizei i;
+    if(LIKELY(numsamples > 1))
+    {
+        ALfloat x0 = filter->x[0];
+        ALfloat x1 = filter->x[1];
+        ALfloat y0 = filter->y[0];
+        ALfloat y1 = filter->y[1];
+
+        for(i = 0;i < numsamples;i++)
+        {
+            dst[i] = filter->b0* src[i] +
+                     filter->b1*x0 + filter->b2*x1 -
+                     filter->a1*y0 - filter->a2*y1;
+            y1 = y0; y0 = dst[i];
+            x1 = x0; x0 = src[i];
+        }
+
+        filter->x[0] = x0;
+        filter->x[1] = x1;
+        filter->y[0] = y0;
+        filter->y[1] = y1;
+    }
+    else if(numsamples == 1)
+    {
+        dst[0] = filter->b0 * src[0] +
+                 filter->b1 * filter->x[0] +
+                 filter->b2 * filter->x[1] -
+                 filter->a1 * filter->y[0] -
+                 filter->a2 * filter->y[1];
+        filter->x[1] = filter->x[0];
+        filter->x[0] = src[0];
+        filter->y[1] = filter->y[0];
+        filter->y[0] = dst[0];
+    }
+}
diff --git a/Alc/mixer/mixer_c.c b/Alc/mixer/mixer_c.c
index 5f913101..e40c2cad 100644
--- a/Alc/mixer/mixer_c.c
+++ b/Alc/mixer/mixer_c.c
@@ -93,45 +93,6 @@ const ALfloat *Resample_bsinc_C(const InterpState *state, const ALfloat *restric
 }
 
 
-void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples)
-{
-    ALsizei i;
-    if(LIKELY(numsamples > 1))
-    {
-        ALfloat x0 = filter->x[0];
-        ALfloat x1 = filter->x[1];
-        ALfloat y0 = filter->y[0];
-        ALfloat y1 = filter->y[1];
-
-        for(i = 0;i < numsamples;i++)
-        {
-            dst[i] = filter->b0* src[i] +
-                     filter->b1*x0 + filter->b2*x1 -
-                     filter->a1*y0 - filter->a2*y1;
-            y1 = y0; y0 = dst[i];
-            x1 = x0; x0 = src[i];
-        }
-
-        filter->x[0] = x0;
-        filter->x[1] = x1;
-        filter->y[0] = y0;
-        filter->y[1] = y1;
-    }
-    else if(numsamples == 1)
-    {
-        dst[0] = filter->b0 * src[0] +
-                 filter->b1 * filter->x[0] +
-                 filter->b2 * filter->x[1] -
-                 filter->a1 * filter->y[0] -
-                 filter->a2 * filter->y[1];
-        filter->x[1] = filter->x[0];
-        filter->x[0] = src[0];
-        filter->y[1] = filter->y[0];
-        filter->y[0] = dst[0];
-    }
-}
-
-
 static inline void ApplyCoeffs(ALsizei Offset, ALfloat (*restrict Values)[2],
                                const ALsizei IrSize,
                                const ALfloat (*restrict Coeffs)[2],
diff --git a/CMakeLists.txt b/CMakeLists.txt
index 646a437c..1be64433 100644
--- a/CMakeLists.txt
+++ b/CMakeLists.txt
@@ -728,6 +728,7 @@ SET(ALC_OBJS  Alc/ALc.c
               Alc/effects/null.c
               Alc/effects/pshifter.c
               Alc/effects/reverb.c
+              Alc/filters/filter.c
               Alc/helpers.c
               Alc/hrtf.c
               Alc/uhjfilter.c
diff --git a/OpenAL32/Include/alFilter.h b/OpenAL32/Include/alFilter.h
index 4e506722..2634d5e8 100644
--- a/OpenAL32/Include/alFilter.h
+++ b/OpenAL32/Include/alFilter.h
@@ -1,9 +1,8 @@
 #ifndef _AL_FILTER_H_
 #define _AL_FILTER_H_
 
-#include "alMain.h"
-
-#include "math_defs.h"
+#include "AL/alc.h"
+#include "AL/al.h"
 
 #ifdef __cplusplus
 extern "C" {
@@ -13,118 +12,6 @@ extern "C" {
 #define HIGHPASSFREQREF  (250.0f)
 
 
-/* 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
- */
-/* Implementation note: For the shelf filters, the specified gain is for the
- * reference frequency, which is the centerpoint of the transition band. This
- * better matches EFX filter design. To set the gain for the shelf itself, use
- * the square root of the desired linear gain (or halve the dB gain).
- */
-
-typedef enum ALfilterType {
-    /** EFX-style low-pass filter, specifying a gain and reference frequency. */
-    ALfilterType_HighShelf,
-    /** EFX-style high-pass filter, specifying a gain and reference frequency. */
-    ALfilterType_LowShelf,
-    /** Peaking filter, specifying a gain and reference frequency. */
-    ALfilterType_Peaking,
-
-    /** Low-pass cut-off filter, specifying a cut-off frequency. */
-    ALfilterType_LowPass,
-    /** High-pass cut-off filter, specifying a cut-off frequency. */
-    ALfilterType_HighPass,
-    /** Band-pass filter, specifying a center frequency. */
-    ALfilterType_BandPass,
-} ALfilterType;
-
-typedef struct ALfilterState {
-    ALfloat x[2]; /* History of two last input samples  */
-    ALfloat y[2]; /* History of two last output samples */
-    ALfloat b0, b1, b2; /* Transfer function coefficients "b" */
-    ALfloat a1, a2; /* Transfer function coefficients "a" (a0 is pre-applied) */
-} ALfilterState;
-/* Currently only a C-based filter process method is implemented. */
-#define ALfilterState_process ALfilterState_processC
-
-/**
- * Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
- * reference gain and shelf slope parameter.
- * \param gain 0 < gain
- * \param slope 0 < slope <= 1
- */
-inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
-{
-    return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
-}
-/**
- * Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the normalized
- * reference frequency and bandwidth.
- * \param f0norm 0 < f0norm < 0.5.
- * \param bandwidth 0 < bandwidth
- */
-inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth)
-{
-    ALfloat w0 = F_TAU * f0norm;
-    return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
-}
-
-inline void ALfilterState_clear(ALfilterState *filter)
-{
-    filter->x[0] = 0.0f;
-    filter->x[1] = 0.0f;
-    filter->y[0] = 0.0f;
-    filter->y[1] = 0.0f;
-}
-
-/**
- * Sets up the filter state for the specified filter type and its parameters.
- *
- * \param filter The filter object to prepare.
- * \param type The type of filter for the object to apply.
- * \param gain The gain for the reference frequency response. Only used by the
- *             Shelf and Peaking filter types.
- * \param f0norm The normalized reference frequency (ref_freq / sample_rate).
- *               This is the center point for the Shelf, Peaking, and BandPass
- *               filter types, or the cutoff frequency for the LowPass and
- *               HighPass filter types.
- * \param rcpQ The reciprocal of the Q coefficient for the filter's transition
- *             band. Can be generated from calc_rcpQ_from_slope or
- *             calc_rcpQ_from_bandwidth depending on the available data.
- */
-void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ);
-
-inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src)
-{
-    dst->b0 = src->b0;
-    dst->b1 = src->b1;
-    dst->b2 = src->b2;
-    dst->a1 = src->a1;
-    dst->a2 = src->a2;
-}
-
-void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
-
-inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples)
-{
-    if(numsamples >= 2)
-    {
-        filter->x[1] = src[numsamples-2];
-        filter->x[0] = src[numsamples-1];
-        filter->y[1] = src[numsamples-2];
-        filter->y[0] = src[numsamples-1];
-    }
-    else if(numsamples == 1)
-    {
-        filter->x[1] = filter->x[0];
-        filter->x[0] = src[0];
-        filter->y[1] = filter->y[0];
-        filter->y[0] = src[0];
-    }
-}
-
-
 struct ALfilter;
 
 typedef struct ALfilterVtable {
diff --git a/OpenAL32/Include/alu.h b/OpenAL32/Include/alu.h
index 14262d9b..341b3760 100644
--- a/OpenAL32/Include/alu.h
+++ b/OpenAL32/Include/alu.h
@@ -12,13 +12,12 @@
 
 #include "alMain.h"
 #include "alBuffer.h"
-#include "alFilter.h"
-#include "alAuxEffectSlot.h"
 
 #include "hrtf.h"
 #include "align.h"
 #include "nfcfilter.h"
 #include "math_defs.h"
+#include "filters/defs.h"
 
 
 #define MAX_PITCH  (255)
diff --git a/OpenAL32/alFilter.c b/OpenAL32/alFilter.c
index 08f66d26..7d8a886c 100644
--- a/OpenAL32/alFilter.c
+++ b/OpenAL32/alFilter.c
@@ -30,11 +30,6 @@
 
 extern inline void LockFilterList(ALCdevice *device);
 extern inline void UnlockFilterList(ALCdevice *device);
-extern inline void ALfilterState_clear(ALfilterState *filter);
-extern inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src);
-extern inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples);
-extern inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope);
-extern inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth);
 
 static ALfilter *AllocFilter(ALCcontext *context);
 static void FreeFilter(ALCdevice *device, ALfilter *filter);
@@ -343,86 +338,6 @@ AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *va
 }
 
 
-void ALfilterState_setParams(ALfilterState *filter, ALfilterType 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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterType_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 ALfilterType_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];
-}
-
-
 static void ALlowpass_setParami(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, ALint UNUSED(val))
 { alSetError(context, AL_INVALID_ENUM, "Invalid low-pass integer property 0x%04x", param); }
 static void ALlowpass_setParamiv(ALfilter *UNUSED(filter), ALCcontext *context, ALenum param, const ALint *UNUSED(vals))
diff --git a/OpenAL32/alSource.c b/OpenAL32/alSource.c
index 40b2c494..e3e40166 100644
--- a/OpenAL32/alSource.c
+++ b/OpenAL32/alSource.c
@@ -31,6 +31,7 @@
 #include "alError.h"
 #include "alSource.h"
 #include "alBuffer.h"
+#include "alFilter.h"
 #include "alAuxEffectSlot.h"
 #include "ringbuffer.h"