Use a helper macro for pi*2

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)