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-rw-r--r--Alc/ALc.c2
-rw-r--r--Alc/effects/chorus.c2
-rw-r--r--Alc/effects/dedicated.c2
-rw-r--r--Alc/effects/distortion.c2
-rw-r--r--Alc/effects/echo.c1
-rw-r--r--Alc/effects/equalizer.c2
-rw-r--r--Alc/effects/modulator.c2
-rw-r--r--Alc/effects/pshifter.c2
-rw-r--r--Alc/effects/reverb.c3
-rw-r--r--Alc/filters/defs.h118
-rw-r--r--Alc/filters/filter.c133
-rw-r--r--Alc/mixer/mixer_c.c39
12 files changed, 261 insertions, 47 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],
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-11 19:48:56 +0000