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-rw-r--r--Alc/ALc.c2
-rw-r--r--Alc/ALu.c28
-rw-r--r--Alc/alcDedicated.c2
-rw-r--r--Alc/alcEcho.c8
-rw-r--r--Alc/alcModulator.c10
-rw-r--r--Alc/alcReverb.c44
-rw-r--r--Alc/hrtf.c12
-rw-r--r--Alc/panning.c12
8 files changed, 59 insertions, 59 deletions
diff --git a/Alc/ALc.c b/Alc/ALc.c
index 0807dae9..54eb5ecf 100644
--- a/Alc/ALc.c
+++ b/Alc/ALc.c
@@ -839,7 +839,7 @@ static void alc_initconfig(void)
}
if(ConfigValueFloat("reverb", "boost", &valf))
- ReverbBoost *= aluPow(10.0f, valf / 20.0f);
+ ReverbBoost *= powf(10.0f, valf / 20.0f);
EmulateEAXReverb = GetConfigValueBool("reverb", "emulate-eax", AL_FALSE);
diff --git a/Alc/ALu.c b/Alc/ALu.c
index 581355f1..5ee170a0 100644
--- a/Alc/ALu.c
+++ b/Alc/ALu.c
@@ -300,7 +300,7 @@ ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
/* Update filter coefficients. Calculations based on the I3DL2
* spec. */
- cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
+ cw = cosf(F_PI*2.0f * LOWPASSFREQREF / Frequency);
/* We use two chained one-pole filters, so we need to take the
* square root of the squared gain, which is the same as the base
@@ -466,7 +466,7 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
aluNormalize(Direction);
/* Calculate distance attenuation */
- Distance = aluSqrt(aluDotproduct(Position, Position));
+ Distance = sqrtf(aluDotproduct(Position, Position));
ClampedDist = Distance;
Attenuation = 1.0f;
@@ -519,9 +519,9 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
case ExponentDistance:
if(ClampedDist > 0.0f && MinDist > 0.0f)
{
- Attenuation = aluPow(ClampedDist/MinDist, -Rolloff);
+ Attenuation = powf(ClampedDist/MinDist, -Rolloff);
for(i = 0;i < NumSends;i++)
- RoomAttenuation[i] = aluPow(ClampedDist/MinDist, -RoomRolloff[i]);
+ RoomAttenuation[i] = powf(ClampedDist/MinDist, -RoomRolloff[i]);
}
break;
@@ -539,9 +539,9 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
if(AirAbsorptionFactor > 0.0f && ClampedDist > MinDist)
{
ALfloat meters = maxf(ClampedDist-MinDist, 0.0f) * MetersPerUnit;
- DryGainHF *= aluPow(AIRABSORBGAINHF, AirAbsorptionFactor*meters);
+ DryGainHF *= powf(AIRABSORBGAINHF, AirAbsorptionFactor*meters);
for(i = 0;i < NumSends;i++)
- WetGainHF[i] *= aluPow(RoomAirAbsorption[i], AirAbsorptionFactor*meters);
+ WetGainHF[i] *= powf(RoomAirAbsorption[i], AirAbsorptionFactor*meters);
}
if(WetGainAuto)
@@ -558,12 +558,12 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
for(i = 0;i < NumSends;i++)
{
if(DecayDistance[i] > 0.0f)
- WetGain[i] *= aluPow(0.001f/*-60dB*/, ApparentDist/DecayDistance[i]);
+ WetGain[i] *= powf(0.001f/*-60dB*/, ApparentDist/DecayDistance[i]);
}
}
/* Calculate directional soundcones */
- Angle = aluAcos(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI);
+ Angle = acosf(aluDotproduct(Direction,SourceToListener)) * (180.0f/F_PI);
if(Angle > InnerAngle && Angle <= OuterAngle)
{
ALfloat scale = (Angle-InnerAngle) / (OuterAngle-InnerAngle);
@@ -679,8 +679,8 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
* the listener. This prevents +0 and -0 Z from producing
* inconsistent panning. Also, clamp Y in case FP precision errors
* cause it to land outside of -1..+1. */
- ev = aluAsin(clampf(Position[1], -1.0f, 1.0f));
- az = aluAtan2(Position[0], -Position[2]*ZScale);
+ ev = asinf(clampf(Position[1], -1.0f, 1.0f));
+ az = atan2f(Position[0], -Position[2]*ZScale);
}
/* Check to see if the HRIR is already moving. */
@@ -739,14 +739,14 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
Position[1] *= invlen;
Position[2] *= invlen;
- DirGain = aluSqrt(Position[0]*Position[0] + Position[2]*Position[2]);
- ComputeAngleGains(Device, aluAtan2(Position[0], -Position[2]*ZScale), 0.0f,
+ DirGain = sqrtf(Position[0]*Position[0] + Position[2]*Position[2]);
+ ComputeAngleGains(Device, atan2f(Position[0], -Position[2]*ZScale), 0.0f,
DryGain*DirGain, Matrix[0]);
}
/* Adjustment for vertical offsets. Not the greatest, but simple
* enough. */
- AmbientGain = DryGain * aluSqrt(1.0f/Device->NumChan) * (1.0f-DirGain);
+ AmbientGain = DryGain * sqrtf(1.0f/Device->NumChan) * (1.0f-DirGain);
for(i = 0;i < (ALint)Device->NumChan;i++)
{
enum Channel chan = Device->Speaker2Chan[i];
@@ -757,7 +757,7 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
ALSource->Params.Send[i].Gain = WetGain[i];
/* Update filter coefficients. */
- cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / Frequency);
+ cw = cosf(F_PI*2.0f * LOWPASSFREQREF / Frequency);
ALSource->Params.Direct.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
for(i = 0;i < NumSends;i++)
diff --git a/Alc/alcDedicated.c b/Alc/alcDedicated.c
index 60460962..64c2910b 100644
--- a/Alc/alcDedicated.c
+++ b/Alc/alcDedicated.c
@@ -61,7 +61,7 @@ static ALvoid DedicatedUpdate(ALeffectState *effect, ALCdevice *device, const AL
state->gains[s] = 0.0f;
if(Slot->effect.type == AL_EFFECT_DEDICATED_DIALOGUE)
- ComputeAngleGains(device, aluAtan2(0.0f, 1.0f), 0.0f, Gain, state->gains);
+ ComputeAngleGains(device, atan2f(0.0f, 1.0f), 0.0f, Gain, state->gains);
else if(Slot->effect.type == AL_EFFECT_DEDICATED_LOW_FREQUENCY_EFFECT)
state->gains[LFE] = Gain;
}
diff --git a/Alc/alcEcho.c b/Alc/alcEcho.c
index 7d050716..e3c3df96 100644
--- a/Alc/alcEcho.c
+++ b/Alc/alcEcho.c
@@ -106,7 +106,7 @@ static ALvoid EchoUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffec
state->FeedGain = Slot->effect.Echo.Feedback;
- cw = aluCos(F_PI*2.0f * LOWPASSFREQREF / frequency);
+ cw = cosf(F_PI*2.0f * LOWPASSFREQREF / frequency);
g = 1.0f - Slot->effect.Echo.Damping;
state->iirFilter.coeff = lpCoeffCalc(g, cw);
@@ -117,13 +117,13 @@ static ALvoid EchoUpdate(ALeffectState *effect, ALCdevice *Device, const ALeffec
state->Gain[1][i] = 0.0f;
}
- dirGain = aluFabs(lrpan);
+ dirGain = fabsf(lrpan);
/* First tap panning */
- ComputeAngleGains(Device, aluAtan2(-lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[0]);
+ ComputeAngleGains(Device, atan2f(-lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[0]);
/* Second tap panning */
- ComputeAngleGains(Device, aluAtan2(+lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[1]);
+ ComputeAngleGains(Device, atan2f(+lrpan, 0.0f), (1.0f-dirGain)*F_PI, gain, state->Gain[1]);
}
static ALvoid EchoProcess(ALeffectState *effect, ALuint SamplesToDo, const ALfloat *SamplesIn, ALfloat (*SamplesOut)[MaxChannels])
diff --git a/Alc/alcModulator.c b/Alc/alcModulator.c
index 5bfcdb22..ce91e95a 100644
--- a/Alc/alcModulator.c
+++ b/Alc/alcModulator.c
@@ -55,7 +55,7 @@ typedef struct ALmodulatorState {
static __inline ALfloat Sin(ALuint index)
{
- return aluSin(index * (F_PI*2.0f / WAVEFORM_FRACONE));
+ return sinf(index * (F_PI*2.0f / WAVEFORM_FRACONE));
}
static __inline ALfloat Saw(ALuint index)
@@ -144,12 +144,12 @@ static ALvoid ModulatorUpdate(ALeffectState *effect, ALCdevice *Device, const AL
Device->Frequency);
if(state->step == 0) state->step = 1;
- cw = aluCos(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
- Device->Frequency);
- a = (2.0f-cw) - aluSqrt(aluPow(2.0f-cw, 2.0f) - 1.0f);
+ cw = cosf(F_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
+ Device->Frequency);
+ a = (2.0f-cw) - sqrtf(powf(2.0f-cw, 2.0f) - 1.0f);
state->iirFilter.coeff = a;
- gain = aluSqrt(1.0f/Device->NumChan);
+ gain = sqrtf(1.0f/Device->NumChan);
gain *= Slot->Gain;
for(index = 0;index < MaxChannels;index++)
state->Gain[index] = 0.0f;
diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c
index 8f0b65ad..13f90511 100644
--- a/Alc/alcReverb.c
+++ b/Alc/alcReverb.c
@@ -261,7 +261,7 @@ static __inline ALfloat EAXModulation(ALverbState *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 - aluCos(F_PI*2.0f * State->Mod.Index / State->Mod.Range);
+ sinus = 1.0f - cosf(F_PI*2.0f * 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
@@ -720,8 +720,8 @@ static ALboolean ReverbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
// is calculated given the current sample rate. This ensures that the
// resulting filter response over time is consistent across all sample
// rates.
- State->Mod.Coeff = aluPow(MODULATION_FILTER_COEFF,
- MODULATION_FILTER_CONST / frequency);
+ State->Mod.Coeff = powf(MODULATION_FILTER_COEFF,
+ MODULATION_FILTER_CONST / frequency);
// The early reflection and late all-pass filter line lengths are static,
// so their offsets only need to be calculated once.
@@ -744,21 +744,21 @@ static ALboolean ReverbDeviceUpdate(ALeffectState *effect, ALCdevice *Device)
// until the decay reaches -60 dB.
static __inline ALfloat CalcDecayCoeff(ALfloat length, ALfloat decayTime)
{
- return aluPow(0.001f/*-60 dB*/, length/decayTime);
+ return powf(0.001f/*-60 dB*/, length/decayTime);
}
// Calculate a decay length from a coefficient and the time until the decay
// reaches -60 dB.
static __inline ALfloat CalcDecayLength(ALfloat coeff, ALfloat decayTime)
{
- return aluLog10(coeff) * decayTime / aluLog10(0.001f)/*-60 dB*/;
+ return log10f(coeff) * decayTime / log10f(0.001f)/*-60 dB*/;
}
// Calculate the high frequency parameter for the I3DL2 coefficient
// calculation.
static __inline ALfloat CalcI3DL2HFreq(ALfloat hfRef, ALuint frequency)
{
- return aluCos(F_PI*2.0f * hfRef / frequency);
+ return cosf(F_PI*2.0f * hfRef / frequency);
}
// Calculate an attenuation to be applied to the input of any echo models to
@@ -778,7 +778,7 @@ static __inline ALfloat CalcDensityGain(ALfloat a)
* calculated by inverting the square root of this approximation,
* yielding: 1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2).
*/
- return aluSqrt(1.0f - (a * a));
+ return sqrtf(1.0f - (a * a));
}
// Calculate the mixing matrix coefficients given a diffusion factor.
@@ -787,13 +787,13 @@ static __inline ALvoid CalcMatrixCoeffs(ALfloat diffusion, ALfloat *x, ALfloat *
ALfloat n, t;
// The matrix is of order 4, so n is sqrt (4 - 1).
- n = aluSqrt(3.0f);
- t = diffusion * aluAtan(n);
+ n = sqrtf(3.0f);
+ t = diffusion * atanf(n);
// Calculate the first mixing matrix coefficient.
- *x = aluCos(t);
+ *x = cosf(t);
// Calculate the second mixing matrix coefficient.
- *y = aluSin(t) / n;
+ *y = sinf(t) / n;
}
// Calculate the limited HF ratio for use with the late reverb low-pass
@@ -913,7 +913,7 @@ static ALvoid UpdateDecorrelator(ALfloat density, ALuint frequency, ALverbState
*/
for(index = 0;index < 3;index++)
{
- length = (DECO_FRACTION * aluPow(DECO_MULTIPLIER, (ALfloat)index)) *
+ length = (DECO_FRACTION * powf(DECO_MULTIPLIER, (ALfloat)index)) *
LATE_LINE_LENGTH[0] * (1.0f + (density * LATE_LINE_MULTIPLIER));
State->DecoTap[index] = fastf2u(length * frequency);
}
@@ -947,7 +947,7 @@ static ALvoid UpdateLateLines(ALfloat reverbGain, ALfloat lateGain, ALfloat xMix
decayTime));
// Calculate the all-pass feed-back and feed-forward coefficient.
- State->Late.ApFeedCoeff = 0.5f * aluPow(diffusion, 2.0f);
+ State->Late.ApFeedCoeff = 0.5f * powf(diffusion, 2.0f);
for(index = 0;index < 4;index++)
{
@@ -991,7 +991,7 @@ static ALvoid UpdateEchoLine(ALfloat reverbGain, ALfloat lateGain, ALfloat echoT
State->Echo.DensityGain = CalcDensityGain(State->Echo.Coeff);
// Calculate the echo all-pass feed coefficient.
- State->Echo.ApFeedCoeff = 0.5f * aluPow(diffusion, 2.0f);
+ State->Echo.ApFeedCoeff = 0.5f * powf(diffusion, 2.0f);
// Calculate the echo all-pass attenuation coefficient.
State->Echo.ApCoeff = CalcDecayCoeff(ECHO_ALLPASS_LENGTH, decayTime);
@@ -1025,12 +1025,12 @@ static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *Reflection
/* Attenuate reverb according to its coverage (dirGain=0 will give
* Gain*ambientGain, and dirGain=1 will give Gain). */
- ambientGain = minf(aluSqrt(2.0f/Device->NumChan), 1.0f);
+ ambientGain = minf(sqrtf(2.0f/Device->NumChan), 1.0f);
length = earlyPan[0]*earlyPan[0] + earlyPan[1]*earlyPan[1] + earlyPan[2]*earlyPan[2];
if(length > 1.0f)
{
- length = 1.0f / aluSqrt(length);
+ length = 1.0f / sqrtf(length);
earlyPan[0] *= length;
earlyPan[1] *= length;
earlyPan[2] *= length;
@@ -1038,22 +1038,22 @@ static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *Reflection
length = latePan[0]*latePan[0] + latePan[1]*latePan[1] + latePan[2]*latePan[2];
if(length > 1.0f)
{
- length = 1.0f / aluSqrt(length);
+ length = 1.0f / sqrtf(length);
latePan[0] *= length;
latePan[1] *= length;
latePan[2] *= length;
}
- dirGain = aluSqrt(earlyPan[0]*earlyPan[0] + earlyPan[2]*earlyPan[2]);
+ dirGain = sqrtf(earlyPan[0]*earlyPan[0] + earlyPan[2]*earlyPan[2]);
for(index = 0;index < MaxChannels;index++)
State->Early.PanGain[index] = 0.0f;
- ComputeAngleGains(Device, aluAtan2(earlyPan[0], earlyPan[2]), (1.0f-dirGain)*F_PI,
+ ComputeAngleGains(Device, atan2f(earlyPan[0], earlyPan[2]), (1.0f-dirGain)*F_PI,
lerp(ambientGain, 1.0f, dirGain) * Gain, State->Early.PanGain);
- dirGain = aluSqrt(latePan[0]*latePan[0] + latePan[2]*latePan[2]);
+ dirGain = sqrtf(latePan[0]*latePan[0] + latePan[2]*latePan[2]);
for(index = 0;index < MaxChannels;index++)
State->Late.PanGain[index] = 0.0f;
- ComputeAngleGains(Device, aluAtan2(latePan[0], latePan[2]), (1.0f-dirGain)*F_PI,
+ ComputeAngleGains(Device, atan2f(latePan[0], latePan[2]), (1.0f-dirGain)*F_PI,
lerp(ambientGain, 1.0f, dirGain) * Gain, State->Late.PanGain);
}
@@ -1141,7 +1141,7 @@ static ALvoid ReverbUpdate(ALeffectState *effect, ALCdevice *Device, const ALeff
ALuint index;
/* Update channel gains */
- gain *= aluSqrt(2.0f/Device->NumChan) * ReverbBoost;
+ gain *= sqrtf(2.0f/Device->NumChan) * ReverbBoost;
for(index = 0;index < MaxChannels;index++)
State->Gain[index] = 0.0f;
for(index = 0;index < Device->NumChan;index++)
diff --git a/Alc/hrtf.c b/Alc/hrtf.c
index 0185a113..cd413ccd 100644
--- a/Alc/hrtf.c
+++ b/Alc/hrtf.c
@@ -70,7 +70,7 @@ static void CalcAzIndices(ALuint evidx, ALfloat az, ALuint *azidx, ALfloat *azmu
az = (F_PI*2.0f + az) * azCount[evidx] / (F_PI*2.0f);
azidx[0] = fastf2u(az) % azCount[evidx];
azidx[1] = (azidx[0] + 1) % azCount[evidx];
- *azmu = az - aluFloor(az);
+ *azmu = az - floorf(az);
}
// Calculates the normalized HRTF transition factor (delta) from the changes
@@ -84,7 +84,7 @@ ALfloat CalcHrtfDelta(ALfloat oldGain, ALfloat newGain, const ALfloat olddir[3],
// Calculate the normalized dB gain change.
newGain = maxf(newGain, 0.0001f);
oldGain = maxf(oldGain, 0.0001f);
- gainChange = aluFabs(aluLog10(newGain / oldGain) / aluLog10(0.0001f));
+ gainChange = fabsf(log10f(newGain / oldGain) / log10f(0.0001f));
// Calculate the normalized listener to source angle change when there is
// enough gain to notice it.
@@ -94,9 +94,9 @@ ALfloat CalcHrtfDelta(ALfloat oldGain, ALfloat newGain, const ALfloat olddir[3],
// No angle change when the directions are equal or degenerate (when
// both have zero length).
if(newdir[0]-olddir[0] || newdir[1]-olddir[1] || newdir[2]-olddir[2])
- angleChange = aluAcos(olddir[0]*newdir[0] +
- olddir[1]*newdir[1] +
- olddir[2]*newdir[2]) / F_PI;
+ angleChange = acosf(olddir[0]*newdir[0] +
+ olddir[1]*newdir[1] +
+ olddir[2]*newdir[2]) / F_PI;
}
@@ -217,7 +217,7 @@ ALuint GetMovingHrtfCoeffs(const struct Hrtf *Hrtf, ALfloat elevation, ALfloat a
ridx[3] = evOffset[evidx[1]] + ((azCount[evidx[1]]-azidx[1]) % azCount[evidx[1]]);
// Calculate the stepping parameters.
- delta = maxf(aluFloor(delta*(Hrtf->sampleRate*0.015f) + 0.5f), 1.0f);
+ delta = maxf(floorf(delta*(Hrtf->sampleRate*0.015f) + 0.5f), 1.0f);
step = 1.0f / delta;
// Calculate the normalized and attenuated target HRIR coefficients using
diff --git a/Alc/panning.c b/Alc/panning.c
index 4bdaf6b0..6b412f89 100644
--- a/Alc/panning.c
+++ b/Alc/panning.c
@@ -181,8 +181,8 @@ ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth,
/* Sound is between speakers i and i+1 */
a = (angle-SpeakerAngle[i]) /
(SpeakerAngle[i+1]-SpeakerAngle[i]);
- gains[Speaker2Chan[i]] = aluSqrt(1.0f-a) * ingain;
- gains[Speaker2Chan[i+1]] = aluSqrt( a) * ingain;
+ gains[Speaker2Chan[i]] = sqrtf(1.0f-a) * ingain;
+ gains[Speaker2Chan[i+1]] = sqrtf( a) * ingain;
return;
}
}
@@ -191,12 +191,12 @@ ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth,
angle += F_PI*2.0f;
a = (angle-SpeakerAngle[i]) /
(F_PI*2.0f + SpeakerAngle[0]-SpeakerAngle[i]);
- gains[Speaker2Chan[i]] = aluSqrt(1.0f-a) * ingain;
- gains[Speaker2Chan[0]] = aluSqrt( a) * ingain;
+ gains[Speaker2Chan[i]] = sqrtf(1.0f-a) * ingain;
+ gains[Speaker2Chan[0]] = sqrtf( a) * ingain;
return;
}
- if(aluFabs(angle)+hwidth > F_PI)
+ if(fabsf(angle)+hwidth > F_PI)
{
/* The coverage area would go outside of -pi...+pi. Instead, rotate the
* speaker angles so it would be as if angle=0, and keep them wrapped
@@ -329,7 +329,7 @@ ALvoid ComputeAngleGains(const ALCdevice *device, ALfloat angle, ALfloat hwidth,
for(i = 0;i < device->NumChan;i++)
{
enum Channel chan = device->Speaker2Chan[i];
- gains[chan] = aluSqrt(tmpgains[chan]) * ingain;
+ gains[chan] = sqrtf(tmpgains[chan]) * ingain;
}
}