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-rw-r--r--Alc/effects/autowah.c2
-rw-r--r--Alc/effects/chorus.c7
-rw-r--r--Alc/effects/distortion.c2
-rw-r--r--Alc/effects/flanger.c7
-rw-r--r--Alc/effects/modulator.c5
-rw-r--r--Alc/effects/reverb.c4
-rw-r--r--Alc/hrtf.c2
-rw-r--r--Alc/panning.c26
-rw-r--r--OpenAL32/Include/alu.h1
-rw-r--r--OpenAL32/alFilter.c2
10 files changed, 28 insertions, 30 deletions
diff --git a/Alc/effects/autowah.c b/Alc/effects/autowah.c
index 9a3f8b94..29ec02af 100644
--- a/Alc/effects/autowah.c
+++ b/Alc/effects/autowah.c
@@ -111,7 +111,7 @@ static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo,
if((state->lfo++) % 30 == 0)
{
/* Using custom low-pass filter coefficients, to handle the resonance and peak-gain properties. */
- frequency = (1.0f + cosf(state->lfo * (1.0f / lerp(1.0f, 4.0f, state->AttackTime * state->ReleaseTime)) * 2.0f * F_PI / state->Frequency)) / OCTAVE;
+ frequency = (1.0f + cosf(state->lfo * (1.0f / lerp(1.0f, 4.0f, state->AttackTime * state->ReleaseTime)) * F_2PI / state->Frequency)) / OCTAVE;
frequency = expf((frequency - 1.0f) * 6.0f);
/* computing cutoff frequency and peak gain */
diff --git a/Alc/effects/chorus.c b/Alc/effects/chorus.c
index 9e74f8c8..4dbfc93d 100644
--- a/Alc/effects/chorus.c
+++ b/Alc/effects/chorus.c
@@ -122,7 +122,7 @@ static ALvoid ALchorusState_update(ALchorusState *state, ALCdevice *Device, cons
if(rate == 0.0f)
state->lfo_coeff = 0.0f;
else
- state->lfo_coeff = F_PI*2.0f / (frequency / rate);
+ state->lfo_coeff = F_2PI / (frequency / rate);
break;
}
@@ -152,12 +152,11 @@ static inline void Sinusoid(ALint *delay_left, ALint *delay_right, ALint offset,
{
ALfloat lfo_value;
- lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff*offset, 2.0f*F_PI));
+ lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff*offset, F_2PI));
lfo_value *= state->depth * state->delay;
*delay_left = fastf2i(lfo_value) + state->delay;
- lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff*(offset+state->lfo_disp),
- 2.0f*F_PI));
+ lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff*(offset+state->lfo_disp), F_2PI));
lfo_value *= state->depth * state->delay;
*delay_right = fastf2i(lfo_value) + state->delay;
}
diff --git a/Alc/effects/distortion.c b/Alc/effects/distortion.c
index 7402f0af..e39a8197 100644
--- a/Alc/effects/distortion.c
+++ b/Alc/effects/distortion.c
@@ -64,7 +64,7 @@ static ALvoid ALdistortionState_update(ALdistortionState *state, ALCdevice *Devi
state->attenuation = Slot->EffectProps.Distortion.Gain;
/* Store waveshaper edge settings */
- edge = sinf(Slot->EffectProps.Distortion.Edge * (F_PI/2.0f));
+ edge = sinf(Slot->EffectProps.Distortion.Edge * (F_PI_2));
edge = minf(edge, 0.99f);
state->edge_coeff = 2.0f * edge / (1.0f-edge);
diff --git a/Alc/effects/flanger.c b/Alc/effects/flanger.c
index d3cdd250..78be5e42 100644
--- a/Alc/effects/flanger.c
+++ b/Alc/effects/flanger.c
@@ -122,7 +122,7 @@ static ALvoid ALflangerState_update(ALflangerState *state, ALCdevice *Device, co
if(rate == 0.0f)
state->lfo_coeff = 0.0f;
else
- state->lfo_coeff = F_PI * 2.0f / (frequency / rate);
+ state->lfo_coeff = F_2PI / (frequency / rate);
break;
}
@@ -152,12 +152,11 @@ static inline void Sinusoid(ALint *delay_left, ALint *delay_right, ALint offset,
{
ALfloat lfo_value;
- lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff * offset, 2.0f*F_PI));
+ lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff * offset, F_2PI));
lfo_value *= state->depth * state->delay;
*delay_left = fastf2i(lfo_value) + state->delay;
- lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff * (offset+state->lfo_disp),
- 2.0f*F_PI));
+ lfo_value = 1.0f + sinf(fmodf(state->lfo_coeff * (offset+state->lfo_disp), F_2PI));
lfo_value *= state->depth * state->delay;
*delay_right = fastf2i(lfo_value) + state->delay;
}
diff --git a/Alc/effects/modulator.c b/Alc/effects/modulator.c
index 4fdd1261..17ca7e17 100644
--- a/Alc/effects/modulator.c
+++ b/Alc/effects/modulator.c
@@ -53,7 +53,7 @@ typedef struct ALmodulatorState {
static inline ALfloat Sin(ALuint index)
{
- return sinf(index * (F_PI*2.0f / WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f;
+ return sinf(index*(F_2PI/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f;
}
static inline ALfloat Saw(ALuint index)
@@ -139,8 +139,7 @@ static ALvoid ALmodulatorState_update(ALmodulatorState *state, ALCdevice *Device
if(state->step == 0) state->step = 1;
/* Custom filter coeffs, which match the old version instead of a low-shelf. */
- cw = cosf(F_PI*2.0f * Slot->EffectProps.Modulator.HighPassCutoff /
- Device->Frequency);
+ cw = cosf(F_2PI * Slot->EffectProps.Modulator.HighPassCutoff / Device->Frequency);
a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
state->Filter.b[0] = a;
diff --git a/Alc/effects/reverb.c b/Alc/effects/reverb.c
index 77bf4a16..75ec76e3 100644
--- a/Alc/effects/reverb.c
+++ b/Alc/effects/reverb.c
@@ -266,7 +266,7 @@ static inline ALfloat EAXModulation(ALreverbState *State, ALfloat in)
// Calculate the sinus rythm (dependent on modulation time and the
// sampling rate). The center of the sinus is moved to reduce the delay
// of the effect when the time or depth are low.
- sinus = 1.0f - cosf(F_PI*2.0f * State->Mod.Index / State->Mod.Range);
+ sinus = 1.0f - cosf(F_2PI * State->Mod.Index / State->Mod.Range);
// The depth determines the range over which to read the input samples
// from, so it must be filtered to reduce the distortion caused by even
@@ -1137,7 +1137,7 @@ static ALvoid ALreverbState_update(ALreverbState *State, ALCdevice *Device, cons
Slot->EffectProps.Reverb.AirAbsorptionGainHF,
Slot->EffectProps.Reverb.DecayTime);
- cw = cosf(F_PI*2.0f * hfscale);
+ cw = cosf(F_2PI * hfscale);
// Update the late lines.
UpdateLateLines(Slot->EffectProps.Reverb.Gain, Slot->EffectProps.Reverb.LateReverbGain,
x, Slot->EffectProps.Reverb.Density, Slot->EffectProps.Reverb.DecayTime,
diff --git a/Alc/hrtf.c b/Alc/hrtf.c
index e772062a..49bb97a3 100644
--- a/Alc/hrtf.c
+++ b/Alc/hrtf.c
@@ -92,7 +92,7 @@ static void CalcEvIndices(const struct Hrtf *Hrtf, ALfloat ev, ALuint *evidx, AL
*/
static void CalcAzIndices(const struct Hrtf *Hrtf, ALuint evidx, ALfloat az, ALuint *azidx, ALfloat *azmu)
{
- az = (F_PI*2.0f + az) * Hrtf->azCount[evidx] / (F_PI*2.0f);
+ az = (F_2PI + az) * Hrtf->azCount[evidx] / (F_2PI);
azidx[0] = fastf2u(az) % Hrtf->azCount[evidx];
azidx[1] = (azidx[0] + 1) % Hrtf->azCount[evidx];
*azmu = az - floorf(az);
diff --git a/Alc/panning.c b/Alc/panning.c
index 10615570..80041696 100644
--- a/Alc/panning.c
+++ b/Alc/panning.c
@@ -186,9 +186,9 @@ void ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, A
}
/* Sound is between last and first speakers */
if(angle < SpeakerAngle[0])
- angle += F_PI*2.0f;
- a = (angle-SpeakerAngle[i]) /
- (F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
+ angle += F_2PI;
+ a = (angle-SpeakerAngle[i]) /
+ (F_2PI + SpeakerAngle[0]-SpeakerAngle[i]);
gains[Speaker2Chan[i]] = sqrtf(1.0f-a) * ingain;
gains[Speaker2Chan[0]] = sqrtf( a) * ingain;
return;
@@ -213,7 +213,7 @@ void ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, A
}
for(i = 0;done < device->NumChan;i++)
{
- SpeakerAngle[done] = device->SpeakerAngle[i]-angle + F_PI*2.0f;
+ SpeakerAngle[done] = device->SpeakerAngle[i]-angle + F_2PI;
Speaker2Chan[done] = device->Speaker2Chan[i];
done++;
}
@@ -236,7 +236,7 @@ void ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, A
}
for(i = device->NumChan-1;done < device->NumChan;i--)
{
- SpeakerAngle[done] = device->SpeakerAngle[i]-angle - F_PI*2.0f;
+ SpeakerAngle[done] = device->SpeakerAngle[i]-angle - F_2PI;
Speaker2Chan[done] = device->Speaker2Chan[i];
done--;
}
@@ -266,14 +266,14 @@ void ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, A
}
if(SpeakerAngle[i] > rangle)
{
- a = (F_PI*2.0f + rangle-SpeakerAngle[last]) /
- (F_PI*2.0f + SpeakerAngle[i]-SpeakerAngle[last]);
+ a = (F_2PI + rangle-SpeakerAngle[last]) /
+ (F_2PI + SpeakerAngle[i]-SpeakerAngle[last]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, a);
}
else if(SpeakerAngle[last] < rangle)
{
- a = (rangle-SpeakerAngle[last]) /
- (F_PI*2.0f + SpeakerAngle[i]-SpeakerAngle[last]);
+ a = (rangle-SpeakerAngle[last]) /
+ (F_2PI + SpeakerAngle[i]-SpeakerAngle[last]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, a);
}
} while(0);
@@ -318,14 +318,14 @@ void ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth, A
}
if(SpeakerAngle[i] < langle)
{
- a = (langle-SpeakerAngle[i]) /
- (F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
+ a = (langle-SpeakerAngle[i]) /
+ (F_2PI + SpeakerAngle[0]-SpeakerAngle[i]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, 1.0f-a);
}
else if(SpeakerAngle[0] > langle)
{
- a = (F_PI*2.0f + langle-SpeakerAngle[i]) /
- (F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
+ a = (F_2PI + langle-SpeakerAngle[i]) /
+ (F_2PI + SpeakerAngle[0]-SpeakerAngle[i]);
tmpgains[chan] = lerp(tmpgains[chan], 1.0f, 1.0f-a);
}
} while(0);
diff --git a/OpenAL32/Include/alu.h b/OpenAL32/Include/alu.h
index d27301b3..55b8edd1 100644
--- a/OpenAL32/Include/alu.h
+++ b/OpenAL32/Include/alu.h
@@ -15,6 +15,7 @@
#define F_PI (3.14159265358979323846f)
#define F_PI_2 (1.57079632679489661923f)
+#define F_2PI (6.28318530717958647692f)
#ifndef FLT_EPSILON
#define FLT_EPSILON (1.19209290e-07f)
diff --git a/OpenAL32/alFilter.c b/OpenAL32/alFilter.c
index 026a5814..899ff5f0 100644
--- a/OpenAL32/alFilter.c
+++ b/OpenAL32/alFilter.c
@@ -339,7 +339,7 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g
// Limit gain to -100dB
gain = maxf(gain, 0.00001f);
- w0 = 2.0f*F_PI * freq_scale;
+ w0 = F_2PI * freq_scale;
/* Calculate filter coefficients depending on filter type */
switch(type)
generated by cgit v1.2.3 (git 2.39.1) at 2024-11-29 09:40:08 +0000