zynaddsubfx

commit 89839176b4fea3301d493364ea026d902a47179d
parent d7ebd07043e46807e5a3a387296231077567d6f9
Author: Christopher A. Oliver <[email protected]>
Date:   Sun,  8 Nov 2015 01:46:29 -0500

Use x-floor(x) rather than fmodf(x,1.0) in audio.

This gives the positive fractional part more reliably.
fmodf() has a number of edge cases that floor avoids.

Diffstat:
M
src/DSP/FormantFilter.cpp
 | 
13
++++---------
M
src/Effects/Chorus.cpp
 | 
5
++---
M
src/Effects/EffectLFO.cpp
 | 
3
++-
M
src/Effects/Phaser.cpp
 | 
6
++++--
M
src/Misc/Microtonal.cpp
 | 
2
+-
M
src/Params/PADnoteParameters.cpp
 | 
5
+++--

6 files changed, 16 insertions(+), 18 deletions(-)
diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp
@@ -115,21 +115,16 @@ void FormantFilter::setpos(float frequency)
     else
         oldinput = input;
 
-    float pos = fmodf(input * sequencestretch, 1.0f);
-    if(pos < 0.0f)
-        pos += 1.0f;
+    float pos = input * sequencestretch;
+    pos -= floorf(pos);
 
     F2I(pos * sequencesize, p2);
     p1 = p2 - 1;
     if(p1 < 0)
         p1 += sequencesize;
 
-    pos = fmodf(pos * sequencesize, 1.0f);
-    if(pos < 0.0f)
-        pos = 0.0f;
-    else
-    if(pos > 1.0f)
-        pos = 1.0f;
+    pos = pos * sequencesize;
+    pos -= floorf(pos);
     pos =
         (atanf((pos * 2.0f
                 - 1.0f)
diff --git a/src/Effects/Chorus.cpp b/src/Effects/Chorus.cpp
@@ -69,7 +69,6 @@ float Chorus::getdelay(float xlfo)
 //Apply the effect
 void Chorus::out(const Stereo<float *> &input)
 {
-    const float one = 1.0f;
     dl1 = dl2;
     dr1 = dr2;
     lfo.effectlfoout(&lfol, &lfor);
@@ -98,7 +97,7 @@ void Chorus::out(const Stereo<float *> &input)
         dlhi %= maxdelay;
 
         float dlhi2 = (dlhi - 1 + maxdelay) % maxdelay;
-        float dllo  = 1.0f - fmod(tmp, one);
+        float dllo  = 1.0f + floorf(tmp) - tmp;
         efxoutl[i] = cinterpolate(delaySample.l, maxdelay, dlhi2) * dllo
                      + cinterpolate(delaySample.l, maxdelay,
                                     dlhi) * (1.0f - dllo);
@@ -116,7 +115,7 @@ void Chorus::out(const Stereo<float *> &input)
         dlhi %= maxdelay;
 
         dlhi2      = (dlhi - 1 + maxdelay) % maxdelay;
-        dllo       = 1.0f - fmodf(tmp, one);
+        dllo       = 1.0f + floorf(tmp) - tmp;
         efxoutr[i] = cinterpolate(delaySample.r, maxdelay, dlhi2) * dllo
                      + cinterpolate(delaySample.r, maxdelay,
                                     dlhi) * (1.0f - dllo);
diff --git a/src/Effects/EffectLFO.cpp b/src/Effects/EffectLFO.cpp
@@ -60,7 +60,8 @@ void EffectLFO::updateparams(void)
     if(PLFOtype > 1)
         PLFOtype = 1;  //this has to be updated if more lfo's are added
     lfotype = PLFOtype;
-    xr      = fmodf(xl + (Pstereo - 64.0f) / 127.0f + 1.0f, 1.0f);
+    xr      = xl + (Pstereo - 64.0f) / 127.0f + 1.0f;
+    xr      -= floorf(xr);
 }
 
 
diff --git a/src/Effects/Phaser.cpp b/src/Effects/Phaser.cpp
@@ -135,8 +135,10 @@ void Phaser::AnalogPhase(const Stereo<float *> &input)
         Stereo<float> xn(input.l[i] * pangainL, input.r[i] * pangainR);
 
         if(barber) {
-            g.l = fmodf((g.l + 0.25f), ONE_);
-            g.r = fmodf((g.r + 0.25f), ONE_);
+            g.l += 0.25;
+            g.l -= floorf(g.l);
+            g.r += 0.25;
+            g.r -= floorf(g.r);
         }
 
         xn.l = applyPhase(xn.l, g.l, fb.l, hpf.l, yn1.l, xn1.l);
diff --git a/src/Misc/Microtonal.cpp b/src/Misc/Microtonal.cpp
@@ -655,7 +655,7 @@ void Microtonal::getfromXML(XMLwrapper *xml)
                     //populate fields for display
                     float x = logf(octave[i].tuning) / LOG_2 * 1200.0f;
                     octave[i].x1 = (int) floor(x);
-                    octave[i].x2 = (int) (floor(fmodf(x, 1.0f) * 1e6));
+                    octave[i].x2 = (int) (floor((x-octave[i].x1) * 1.0e6));
                 }
 
 
diff --git a/src/Params/PADnoteParameters.cpp b/src/Params/PADnoteParameters.cpp
@@ -684,8 +684,9 @@ void PADnoteParameters::generatespectrum_bandwidthMode(float *spectrum,
             const float ibasefreq = realfreq / (synth.samplerate_f * 0.5f) * size;
             for(int i = 0; i < profilesize; ++i) {
                 const float idfreq = (i / (float)profilesize - 0.5f) * ibw;
-                const int   spfreq  = (int) (idfreq + ibasefreq);
-                const float fspfreq = fmodf((float)idfreq + ibasefreq, 1.0f);
+                const float freqsum = idfreq + ibasefreq;
+                const int   spfreq  = (int)freqsum;
+                const float fspfreq = freqsum - spfreq;
                 if(spfreq <= 0)
                     continue;
                 if(spfreq >= size - 1)