aboutsummaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
Diffstat
-rw-r--r--Alc/effects/equalizer.c117
-rw-r--r--OpenAL32/Include/alFilter.h43
-rw-r--r--OpenAL32/alFilter.c78
3 files changed, 131 insertions, 107 deletions
diff --git a/Alc/effects/equalizer.c b/Alc/effects/equalizer.c
index 3a77c63f..9a1543e4 100644
--- a/Alc/effects/equalizer.c
+++ b/Alc/effects/equalizer.c
@@ -78,20 +78,6 @@ static ALequalizerStateFactory EqualizerFactory;
* filter coefficients" by Robert Bristow-Johnson *
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt */
-typedef enum ALEQFilterType {
- LOW_SHELF,
- HIGH_SHELF,
- PEAKING
-} ALEQFilterType;
-
-typedef struct ALEQFilter {
- ALEQFilterType type;
- ALfloat x[2]; /* History of two last input samples */
- ALfloat y[2]; /* History of two last output samples */
- ALfloat a[3]; /* Transfer function coefficients "a" */
- ALfloat b[3]; /* Transfer function coefficients "b" */
-} ALEQFilter;
-
typedef struct ALequalizerState {
DERIVE_FROM_TYPE(ALeffectState);
@@ -99,7 +85,7 @@ typedef struct ALequalizerState {
ALfloat Gain[MaxChannels];
/* Effect parameters */
- ALEQFilter bandfilter[4];
+ ALfilterState filter[4];
} ALequalizerState;
static ALvoid ALequalizerState_Destruct(ALequalizerState *state)
@@ -131,11 +117,10 @@ static ALvoid ALequalizerState_Update(ALequalizerState *state, ALCdevice *device
/* Calculate coefficients for the each type of filter */
for(it = 0; it < 4; it++)
{
- ALfloat gain;
+ ALfilterType type = ALfilterType_Peaking;
ALfloat filter_frequency;
ALfloat bandwidth = 0.0f;
- ALfloat w0;
- ALfloat alpha = 0.0f;
+ ALfloat gain;
/* convert linear gains to filter gains */
switch (it)
@@ -143,85 +128,28 @@ static ALvoid ALequalizerState_Update(ALequalizerState *state, ALCdevice *device
case 0: /* Low Shelf */
gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.LowGain)) / 40.0f);
filter_frequency = slot->EffectProps.Equalizer.LowCutoff;
+ type = ALfilterType_LowShelf;
break;
case 1: /* Peaking */
gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.Mid1Gain)) / 40.0f);
filter_frequency = slot->EffectProps.Equalizer.Mid1Center;
bandwidth = slot->EffectProps.Equalizer.Mid1Width;
+ type = ALfilterType_Peaking;
break;
case 2: /* Peaking */
gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.Mid2Gain)) / 40.0f);
filter_frequency = slot->EffectProps.Equalizer.Mid2Center;
bandwidth = slot->EffectProps.Equalizer.Mid2Width;
+ type = ALfilterType_Peaking;
break;
case 3: /* High Shelf */
gain = powf(10.0f, (20.0f * log10f(slot->EffectProps.Equalizer.HighGain)) / 40.0f);
filter_frequency = slot->EffectProps.Equalizer.HighCutoff;
+ type = ALfilterType_HighShelf;
break;
}
- w0 = 2.0f*F_PI * filter_frequency / frequency;
-
- /* Calculate filter coefficients depending on filter type */
- switch(state->bandfilter[it].type)
- {
- case LOW_SHELF:
- alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
- (1.0f / 0.75f - 1.0f) + 2.0f);
- state->bandfilter[it].b[0] = gain * ((gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].b[1] = 2.0f * gain * ((gain - 1.0f) -
- (gain + 1.0f) * cosf(w0));
- state->bandfilter[it].b[2] = gain * ((gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].a[0] = (gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha;
- state->bandfilter[it].a[1] = -2.0f * ((gain - 1.0f) +
- (gain + 1.0f) * cosf(w0));
- state->bandfilter[it].a[2] = (gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha;
- break;
- case HIGH_SHELF:
- alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
- (1.0f / 0.75f - 1.0f) + 2.0f);
- state->bandfilter[it].b[0] = gain * ((gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].b[1] = -2.0f * gain * ((gain - 1.0f) +
- (gain + 1.0f) *
- cosf(w0));
- state->bandfilter[it].b[2] = gain * ((gain + 1.0f) +
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha);
- state->bandfilter[it].a[0] = (gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) +
- 2.0f * sqrtf(gain) * alpha;
- state->bandfilter[it].a[1] = 2.0f * ((gain - 1.0f) -
- (gain + 1.0f) * cosf(w0));
- state->bandfilter[it].a[2] = (gain + 1.0f) -
- (gain - 1.0f) * cosf(w0) -
- 2.0f * sqrtf(gain) * alpha;
- break;
- case PEAKING:
- alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
- state->bandfilter[it].b[0] = 1.0f + alpha * gain;
- state->bandfilter[it].b[1] = -2.0f * cosf(w0);
- state->bandfilter[it].b[2] = 1.0f - alpha * gain;
- state->bandfilter[it].a[0] = 1.0f + alpha / gain;
- state->bandfilter[it].a[1] = -2.0f * cosf(w0);
- state->bandfilter[it].a[2] = 1.0f - alpha / gain;
- break;
- }
- state->bandfilter[it].b[0] /= state->bandfilter[it].a[0];
- state->bandfilter[it].b[1] /= state->bandfilter[it].a[0];
- state->bandfilter[it].b[2] /= state->bandfilter[it].a[0];
- state->bandfilter[it].a[0] /= state->bandfilter[it].a[0];
- state->bandfilter[it].a[1] /= state->bandfilter[it].a[0];
- state->bandfilter[it].a[2] /= state->bandfilter[it].a[0];
+ ALfilterState_setParams(&state->filter[it], type, gain, filter_frequency/frequency, bandwidth);
}
}
@@ -242,22 +170,7 @@ static ALvoid ALequalizerState_Process(ALequalizerState *state, ALuint SamplesTo
ALfloat smp = SamplesIn[base+it];
for(ft = 0;ft < 4;ft++)
- {
- ALEQFilter *filter = &state->bandfilter[ft];
- ALfloat outsmp;
-
- outsmp = filter->b[0] * smp +
- filter->b[1] * filter->x[0] +
- filter->b[2] * filter->x[1] -
- filter->a[1] * filter->y[0] -
- filter->a[2] * filter->y[1];
-
- filter->x[1] = filter->x[0];
- filter->x[0] = smp;
- filter->y[1] = filter->y[0];
- filter->y[0] = outsmp;
- smp = outsmp;
- }
+ smp = ALfilterState_processSingle(&state->filter[ft], smp);
temps[it] = smp;
}
@@ -294,20 +207,10 @@ ALeffectState *ALequalizerStateFactory_create(ALequalizerStateFactory *factory)
if(!state) return NULL;
SET_VTABLE2(ALequalizerState, ALeffectState, state);
- state->bandfilter[0].type = LOW_SHELF;
- state->bandfilter[1].type = PEAKING;
- state->bandfilter[2].type = PEAKING;
- state->bandfilter[3].type = HIGH_SHELF;
-
/* Initialize sample history only on filter creation to avoid */
/* sound clicks if filter settings were changed in runtime. */
for(it = 0; it < 4; it++)
- {
- state->bandfilter[it].x[0] = 0.0f;
- state->bandfilter[it].x[1] = 0.0f;
- state->bandfilter[it].y[0] = 0.0f;
- state->bandfilter[it].y[1] = 0.0f;
- }
+ ALfilterState_clear(&state->filter[it]);
return STATIC_CAST(ALeffectState, state);
}
diff --git a/OpenAL32/Include/alFilter.h b/OpenAL32/Include/alFilter.h
index 94feb3ee..f70b839e 100644
--- a/OpenAL32/Include/alFilter.h
+++ b/OpenAL32/Include/alFilter.h
@@ -46,6 +46,49 @@ static __inline ALfloat lpFilter2PC(const FILTER *iir, ALfloat input)
ALfloat lpCoeffCalc(ALfloat g, ALfloat cw);
+typedef enum ALfilterType {
+ ALfilterType_HighShelf,
+ ALfilterType_LowShelf,
+ ALfilterType_Peaking,
+} ALfilterType;
+
+typedef struct ALfilterState {
+ ALfloat x[2]; /* History of two last input samples */
+ ALfloat y[2]; /* History of two last output samples */
+ ALfloat a[3]; /* Transfer function coefficients "a" */
+ ALfloat b[3]; /* Transfer function coefficients "b" */
+} ALfilterState;
+
+void ALfilterState_clear(ALfilterState *filter);
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_scale, ALfloat bandwidth);
+
+static __inline ALfloat ALfilterState_processSingle(ALfilterState *filter, ALfloat sample)
+{
+ ALfloat outsmp;
+
+ outsmp = filter->b[0] * sample +
+ filter->b[1] * filter->x[0] +
+ filter->b[2] * filter->x[1] -
+ filter->a[1] * filter->y[0] -
+ filter->a[2] * filter->y[1];
+ filter->x[1] = filter->x[0];
+ filter->x[0] = sample;
+ filter->y[1] = filter->y[0];
+ filter->y[0] = outsmp;
+
+ return outsmp;
+}
+
+static __inline ALfloat ALfilterState_processSingleC(const ALfilterState *filter, ALfloat sample)
+{
+ return filter->b[0] * sample +
+ filter->b[1] * filter->x[0] +
+ filter->b[2] * filter->x[1] -
+ filter->a[1] * filter->y[0] -
+ filter->a[2] * filter->y[1];
+}
+
+
typedef struct ALfilter {
// Filter type (AL_FILTER_NULL, ...)
ALenum type;
diff --git a/OpenAL32/alFilter.c b/OpenAL32/alFilter.c
index 312cf681..6b4ce633 100644
--- a/OpenAL32/alFilter.c
+++ b/OpenAL32/alFilter.c
@@ -325,6 +325,84 @@ AL_API ALvoid AL_APIENTRY alGetFilterfv(ALuint filter, ALenum param, ALfloat *va
}
+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;
+}
+
+void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat freq_scale, ALfloat bandwidth)
+{
+ ALfloat alpha;
+ ALfloat w0;
+
+ w0 = 2.0f*F_PI * freq_scale;
+
+ /* Calculate filter coefficients depending on filter type */
+ switch(type)
+ {
+ case ALfilterType_HighShelf:
+ alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f/gain) *
+ (1.0f/0.75f - 1.0f) + 2.0f);
+ filter->b[0] = gain * ((gain + 1.0f) +
+ (gain - 1.0f) * cosf(w0) +
+ 2.0f * sqrtf(gain) * alpha);
+ filter->b[1] = -2.0f * gain * ((gain - 1.0f) +
+ (gain + 1.0f) * cosf(w0));
+ filter->b[2] = gain * ((gain + 1.0f) +
+ (gain - 1.0f) * cosf(w0) -
+ 2.0f * sqrtf(gain) * alpha);
+ filter->a[0] = (gain + 1.0f) -
+ (gain - 1.0f) * cosf(w0) +
+ 2.0f * sqrtf(gain) * alpha;
+ filter->a[1] = 2.0f * ((gain - 1.0f) -
+ (gain + 1.0f) * cosf(w0));
+ filter->a[2] = (gain + 1.0f) -
+ (gain - 1.0f) * cosf(w0) -
+ 2.0f * sqrtf(gain) * alpha;
+ break;
+ case ALfilterType_LowShelf:
+ alpha = sinf(w0) / 2.0f * sqrtf((gain + 1.0f / gain) *
+ (1.0f / 0.75f - 1.0f) + 2.0f);
+ filter->b[0] = gain * ((gain + 1.0f) -
+ (gain - 1.0f) * cosf(w0) +
+ 2.0f * sqrtf(gain) * alpha);
+ filter->b[1] = 2.0f * gain * ((gain - 1.0f) -
+ (gain + 1.0f) * cosf(w0));
+ filter->b[2] = gain * ((gain + 1.0f) -
+ (gain - 1.0f) * cosf(w0) -
+ 2.0f * sqrtf(gain) * alpha);
+ filter->a[0] = (gain + 1.0f) +
+ (gain - 1.0f) * cosf(w0) +
+ 2.0f * sqrtf(gain) * alpha;
+ filter->a[1] = -2.0f * ((gain - 1.0f) +
+ (gain + 1.0f) * cosf(w0));
+ filter->a[2] = (gain + 1.0f) +
+ (gain - 1.0f) * cosf(w0) -
+ 2.0f * sqrtf(gain) * alpha;
+ break;
+ case ALfilterType_Peaking:
+ alpha = sinf(w0) * sinhf(logf(2.0f) / 2.0f * bandwidth * w0 / sinf(w0));
+ filter->b[0] = 1.0f + alpha * gain;
+ filter->b[1] = -2.0f * cosf(w0);
+ filter->b[2] = 1.0f - alpha * gain;
+ filter->a[0] = 1.0f + alpha / gain;
+ filter->a[1] = -2.0f * cosf(w0);
+ filter->a[2] = 1.0f - alpha / gain;
+ break;
+ }
+
+ filter->b[0] /= filter->a[0];
+ filter->b[1] /= filter->a[0];
+ filter->b[2] /= filter->a[0];
+ filter->a[0] /= filter->a[0];
+ filter->a[1] /= filter->a[0];
+ filter->a[2] /= filter->a[0];
+}
+
+
ALfloat lpCoeffCalc(ALfloat g, ALfloat cw)
{
ALfloat a = 0.0f;
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-29 19:46:07 +0000