Use cosf and sinf when available

Also clear away a few more MSVC precision warnings

5 files changed, 24 insertions, 24 deletions

diff --git a/Alc/ALu.c b/Alc/ALu.c
index 670df970..a3e121e0 100644
--- a/Alc/ALu.c
+++ b/Alc/ALu.c
@@ -211,8 +211,8 @@ ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
             DryGain *= aluSqrt(2.0f/4.0f);
             for(c = 0;c < 2;c++)
             {
-                pos = aluCart2LUTpos(cos(angles_Rear[c] * (M_PI/180.0)),
-                                     sin(angles_Rear[c] * (M_PI/180.0)));
+                pos = aluCart2LUTpos(aluCos((ALfloat)M_PI/180.0f * angles_Rear[c]),
+                                     aluSin((ALfloat)M_PI/180.0f * angles_Rear[c]));
                 SpeakerGain = Device->PanningLUT[pos];
 
                 for(i = 0;i < (ALint)Device->NumChan;i++)
@@ -284,7 +284,7 @@ ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
                 /* Get the static HRIR coefficients and delays for this
                  * channel. */
                 GetLerpedHrtfCoeffs(ALContext->Device->Hrtf,
-                                    0.0, angles[c] * (M_PI/180.0),
+                                    0.0f, (ALfloat)M_PI/180.0f * angles[c],
                                     DryGain*ListenerGain,
                                     ALSource->Params.HrtfCoeffs[c],
                                     ALSource->Params.HrtfDelay[c]);
@@ -301,8 +301,8 @@ ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
                 SrcMatrix[c][LFE] += DryGain * ListenerGain;
                 continue;
             }
-            pos = aluCart2LUTpos(cos(angles[c] * (M_PI/180.0)),
-                                 sin(angles[c] * (M_PI/180.0)));
+            pos = aluCart2LUTpos(aluCos((ALfloat)M_PI/180.0f * angles[c]),
+                                 aluSin((ALfloat)M_PI/180.0f * angles[c]));
             SpeakerGain = Device->PanningLUT[pos];
 
             for(i = 0;i < (ALint)Device->NumChan;i++)
@@ -321,7 +321,7 @@ ALvoid CalcNonAttnSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
 
     /* Update filter coefficients. Calculations based on the I3DL2
      * spec. */
-    cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
+    cw = aluCos((ALfloat)M_PI*2.0f * LOWPASSFREQCUTOFF / 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
@@ -790,7 +790,7 @@ ALvoid CalcSourceParams(ALsource *ALSource, const ALCcontext *ALContext)
         ALSource->Params.Send[i].WetGain = WetGain[i];
 
     /* Update filter coefficients. */
-    cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / Frequency);
+    cw = aluCos((ALfloat)M_PI*2.0f * LOWPASSFREQCUTOFF / Frequency);
 
     ALSource->Params.iirFilter.coeff = lpCoeffCalc(DryGainHF, cw);
     for(i = 0;i < NumSends;i++)
diff --git a/Alc/alcEcho.c b/Alc/alcEcho.c
index 76249d87..8caeb7ca 100644
--- a/Alc/alcEcho.c
+++ b/Alc/alcEcho.c
@@ -111,7 +111,7 @@ static ALvoid EchoUpdate(ALeffectState *effect, ALCcontext *Context, const ALeff
 
     state->FeedGain = Slot->effect.Echo.Feedback;
 
-    cw = cos(2.0*M_PI * LOWPASSFREQCUTOFF / frequency);
+    cw = aluCos((ALfloat)M_PI*2.0f * LOWPASSFREQCUTOFF / frequency);
     g = 1.0f - Slot->effect.Echo.Damping;
     state->iirFilter.coeff = lpCoeffCalc(g, cw);
 
diff --git a/Alc/alcModulator.c b/Alc/alcModulator.c
index 1a2ffd64..e6f94737 100644
--- a/Alc/alcModulator.c
+++ b/Alc/alcModulator.c
@@ -52,19 +52,19 @@ typedef struct ALmodulatorState {
 #define WAVEFORM_FRACBITS  16
 #define WAVEFORM_FRACMASK  ((1<<WAVEFORM_FRACBITS)-1)
 
-static __inline ALdouble Sin(ALuint index)
+static __inline ALfloat Sin(ALuint index)
 {
-    return sin(index * (M_PI*2.0 / (1<<WAVEFORM_FRACBITS)));
+    return aluSin(index * ((ALfloat)M_PI*2.0f / (1<<WAVEFORM_FRACBITS)));
 }
 
-static __inline ALdouble Saw(ALuint index)
+static __inline ALfloat Saw(ALuint index)
 {
-    return index*(2.0/(1<<WAVEFORM_FRACBITS)) - 1.0;
+    return index*(2.0f/(1<<WAVEFORM_FRACBITS)) - 1.0f;
 }
 
-static __inline ALdouble Square(ALuint index)
+static __inline ALfloat Square(ALuint index)
 {
-    return (index&(1<<(WAVEFORM_FRACBITS-1))) ? -1.0 : 1.0;
+    return (index&(1<<(WAVEFORM_FRACBITS-1))) ? -1.0f : 1.0f;
 }
 
 
@@ -151,8 +151,8 @@ static ALvoid ModulatorUpdate(ALeffectState *effect, ALCcontext *Context, const
     if(!state->step)
         state->step = 1;
 
-    cw = cos(2.0*M_PI * Slot->effect.Modulator.HighPassCutoff /
-                        Device->Frequency);
+    cw = aluCos((ALfloat)M_PI*2.0f * Slot->effect.Modulator.HighPassCutoff /
+                                     Device->Frequency);
     a = (2.0f-cw) - aluSqrt(aluPow(2.0f-cw, 2.0f) - 1.0f);
     state->iirFilter.coeff = a;
 
diff --git a/Alc/alcReverb.c b/Alc/alcReverb.c
index 425b7052..0555db64 100644
--- a/Alc/alcReverb.c
+++ b/Alc/alcReverb.c
@@ -237,7 +237,7 @@ 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 - cos(2.0f * M_PI * State->Mod.Index / State->Mod.Range);
+    sinus = 1.0f - aluCos((ALfloat)M_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
@@ -764,7 +764,7 @@ static __inline ALfloat CalcDecayLength(ALfloat coeff, ALfloat decayTime)
 // calculation.
 static __inline ALfloat CalcI3DL2HFreq(ALfloat hfRef, ALuint frequency)
 {
-    return cos(2.0f * M_PI * hfRef / frequency);
+    return aluCos((ALfloat)M_PI*2.0f * hfRef / frequency);
 }
 
 // Calculate an attenuation to be applied to the input of any echo models to
@@ -797,9 +797,9 @@ static __inline ALvoid CalcMatrixCoeffs(ALfloat diffusion, ALfloat *x, ALfloat *
     t = diffusion * atan(n);
 
     // Calculate the first mixing matrix coefficient.
-    *x = cos(t);
+    *x = aluCos(t);
     // Calculate the second mixing matrix coefficient.
-    *y = sin(t) / n;
+    *y = aluSin(t) / n;
 }
 
 // Calculate the limited HF ratio for use with the late reverb low-pass
diff --git a/Alc/panning.c b/Alc/panning.c
index 983d3ef6..f600284d 100644
--- a/Alc/panning.c
+++ b/Alc/panning.c
@@ -295,8 +295,8 @@ ALvoid aluInitPanning(ALCdevice *Device)
                 /* source between speaker s and speaker s+1 */
                 Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
                                  (SpeakerAngle[s+1]-SpeakerAngle[s]);
-                PanningLUT[Speaker2Chan[s]]   = cos(Alpha);
-                PanningLUT[Speaker2Chan[s+1]] = sin(Alpha);
+                PanningLUT[Speaker2Chan[s]]   = aluCos(Alpha);
+                PanningLUT[Speaker2Chan[s+1]] = aluSin(Alpha);
                 break;
             }
         }
@@ -307,8 +307,8 @@ ALvoid aluInitPanning(ALCdevice *Device)
                 Theta += 2.0f * M_PI;
             Alpha = M_PI_2 * (Theta-SpeakerAngle[s]) /
                              (2.0f * M_PI + SpeakerAngle[0]-SpeakerAngle[s]);
-            PanningLUT[Speaker2Chan[s]] = cos(Alpha);
-            PanningLUT[Speaker2Chan[0]] = sin(Alpha);
+            PanningLUT[Speaker2Chan[s]] = aluCos(Alpha);
+            PanningLUT[Speaker2Chan[0]] = aluSin(Alpha);
         }
     }
 }