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Move the filter implementation to a separate directory

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This commit is contained in:
Chris Robinson
2018-03-22 07:05:40 -07:00
parent 6ea3b5445f
commit 7a23330ffe
17 changed files with 266 additions and 249 deletions
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Alc/ALc.c
+2
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@@ -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"
Alc/effects/chorus.c
+1 -1
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@@ -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");
Alc/effects/dedicated.c
+1 -1
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@@ -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 {
Alc/effects/distortion.c
+1 -1
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@@ -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 {
Alc/effects/echo.c
+1
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@@ -28,6 +28,7 @@
#include "alAuxEffectSlot.h"
#include "alError.h"
#include "alu.h"
#include "filters/defs.h"
typedef struct ALechoState {
Alc/effects/equalizer.c
+1 -1
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@@ -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 *
Alc/effects/modulator.c
+1 -1
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@@ -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
Alc/effects/pshifter.c
+1 -1
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@@ -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
Alc/effects/reverb.c
+1 -2
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@@ -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.
Alc/filters/defs.h
+118
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@@ -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 */
Alc/filters/filter.c
+133
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@@ -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];
}
}
Alc/mixer/mixer_c.c
-39
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@@ -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],
CMakeLists.txt
+1
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@@ -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
OpenAL32/Include/alFilter.h
+2 -115
View File
@@ -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 {
OpenAL32/Include/alu.h
+1 -2
View File
@@ -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)
OpenAL32/alFilter.c
-85
View File
@@ -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))
OpenAL32/alSource.c
+1
View File
@@ -31,6 +31,7 @@
#include "alError.h"
#include "alSource.h"
#include "alBuffer.h"
#include "alFilter.h"
#include "alAuxEffectSlot.h"
#include "ringbuffer.h"