diff options
| -rw-r--r-- | Alc/effects/autowah.c | 2 | ||||
| -rw-r--r-- | Alc/effects/chorus.c | 2 | ||||
| -rw-r--r-- | Alc/effects/flanger.c | 2 | ||||
| -rw-r--r-- | Alc/effects/modulator.c | 4 | ||||
| -rw-r--r-- | Alc/effects/reverb.c | 4 | ||||
| -rw-r--r-- | Alc/hrtf.c | 6 | ||||
| -rw-r--r-- | OpenAL32/Include/alu.h | 2 | ||||
| -rw-r--r-- | OpenAL32/alFilter.c | 2 |
8 files changed, 12 insertions, 12 deletions
diff --git a/Alc/effects/autowah.c b/Alc/effects/autowah.c index a444ace3..6770f719 100644 --- a/Alc/effects/autowah.c +++ b/Alc/effects/autowah.c @@ -112,7 +112,7 @@ static ALvoid ALautowahState_process(ALautowahState *state, ALuint SamplesToDo, * ALfilterType_LowPass. However, instead of passing a bandwidth, * we use the resonance property for Q. This also inlines the call. */ - w0 = F_2PI * cutoff / state->Frequency; + w0 = F_TAU * cutoff / state->Frequency; /* FIXME: Resonance controls the resonant peak, or Q. How? Not sure * that Q = resonance*0.1. */ diff --git a/Alc/effects/chorus.c b/Alc/effects/chorus.c index aef12647..7aa5898b 100644 --- a/Alc/effects/chorus.c +++ b/Alc/effects/chorus.c @@ -134,7 +134,7 @@ static ALvoid ALchorusState_update(ALchorusState *state, ALCdevice *Device, cons state->lfo_scale = 4.0f / state->lfo_range; break; case CWF_Sinusoid: - state->lfo_scale = F_2PI / state->lfo_range; + state->lfo_scale = F_TAU / state->lfo_range; break; } diff --git a/Alc/effects/flanger.c b/Alc/effects/flanger.c index 1af56102..f6191abd 100644 --- a/Alc/effects/flanger.c +++ b/Alc/effects/flanger.c @@ -134,7 +134,7 @@ static ALvoid ALflangerState_update(ALflangerState *state, ALCdevice *Device, co state->lfo_scale = 4.0f / state->lfo_range; break; case FWF_Sinusoid: - state->lfo_scale = F_2PI / state->lfo_range; + state->lfo_scale = F_TAU / state->lfo_range; break; } diff --git a/Alc/effects/modulator.c b/Alc/effects/modulator.c index d76fddad..dceb408e 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_2PI/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f; + return sinf(index*(F_TAU/WAVEFORM_FRACONE) - F_PI)*0.5f + 0.5f; } static inline ALfloat Saw(ALuint index) @@ -139,7 +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_2PI * Slot->EffectProps.Modulator.HighPassCutoff / Device->Frequency); + cw = cosf(F_TAU * 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 9170e1c9..1b9c37d9 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_2PI * State->Mod.Index / State->Mod.Range); + sinus = 1.0f - cosf(F_TAU * 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 @@ -1145,7 +1145,7 @@ static ALvoid ALreverbState_update(ALreverbState *State, ALCdevice *Device, cons Slot->EffectProps.Reverb.AirAbsorptionGainHF, Slot->EffectProps.Reverb.DecayTime); - cw = cosf(F_2PI * hfscale); + cw = cosf(F_TAU * hfscale); // Update the late lines. UpdateLateLines(Slot->EffectProps.Reverb.Gain, Slot->EffectProps.Reverb.LateReverbGain, x, Slot->EffectProps.Reverb.Density, Slot->EffectProps.Reverb.DecayTime, @@ -82,7 +82,7 @@ static void CalcEvIndices(ALuint evcount, ALfloat ev, ALuint *evidx, ALfloat *ev */ static void CalcAzIndices(ALuint azcount, ALfloat az, ALuint *azidx, ALfloat *azmu) { - az = (F_2PI + az) * azcount / F_2PI; + az = (F_TAU + az) * azcount / F_TAU; azidx[0] = fastf2u(az) % azcount; azidx[1] = (azidx[0] + 1) % azcount; *azmu = az - floorf(az); @@ -359,8 +359,8 @@ void GetBFormatHrtfCoeffs(const struct Hrtf *Hrtf, const ALuint num_chans, ALflo lidx = evoffset + azi_idx; ridx = evoffset + ((azcount-azi_idx) % azcount); - az = (ALfloat)azi_idx / (ALfloat)azcount * F_2PI; - if(az > F_PI) az -= F_2PI; + az = (ALfloat)azi_idx / (ALfloat)azcount * F_TAU; + if(az > F_PI) az -= F_TAU; x = cosf(-az) * cosf(elev); y = sinf(-az) * cosf(elev); diff --git a/OpenAL32/Include/alu.h b/OpenAL32/Include/alu.h index 50cde1bf..e167979b 100644 --- a/OpenAL32/Include/alu.h +++ b/OpenAL32/Include/alu.h @@ -20,7 +20,7 @@ #define F_PI (3.14159265358979323846f) #define F_PI_2 (1.57079632679489661923f) -#define F_2PI (6.28318530717958647692f) +#define F_TAU (6.28318530717958647692f) #ifndef FLT_EPSILON #define FLT_EPSILON (1.19209290e-07f) diff --git a/OpenAL32/alFilter.c b/OpenAL32/alFilter.c index 72eff9bb..ccf256e1 100644 --- a/OpenAL32/alFilter.c +++ b/OpenAL32/alFilter.c @@ -344,7 +344,7 @@ void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat g // Limit gain to -100dB gain = maxf(gain, 0.00001f); - w0 = F_2PI * freq_mult; + w0 = F_TAU * freq_mult; /* Calculate filter coefficients depending on filter type */ switch(type) |