commit c880db1a67e4af83bfe46257951840f7536015d1
parent 44511fa374847cabbc238b8a7bbc0179719020ad
Author: Jonathan Moore Liles <j.liles@unix.net>
Date: Tue, 15 Dec 2020 19:22:52 -0800
Clarity and formatting edits.
Diffstat:
8 files changed, 40 insertions(+), 42 deletions(-)
diff --git a/src/DSP/AnalogFilter.cpp b/src/DSP/AnalogFilter.cpp
@@ -376,7 +376,7 @@ void AnalogFilter::singlefilterout_freqbuf(float *smp, fstage &hist,
/* recompute coeffs for each 8 samples */
freq = freqbuf[i];
computefiltercoefs();
-
+
if(order == 1) { //First order filter
for ( int j = 0; j < 8; j++ )
{
@@ -390,7 +390,7 @@ void AnalogFilter::singlefilterout_freqbuf(float *smp, fstage &hist,
const float coeff_[5] = {coeff.c[0], coeff.c[1], coeff.c[2], coeff.d[1], coeff.d[2]};
float work[4] = {hist.x1, hist.x2, hist.y1, hist.y2};
-
+
AnalogBiquadFilterA(coeff_, smp[i + 0], work);
AnalogBiquadFilterB(coeff_, smp[i + 1], work);
AnalogBiquadFilterA(coeff_, smp[i + 2], work);
@@ -399,7 +399,7 @@ void AnalogFilter::singlefilterout_freqbuf(float *smp, fstage &hist,
AnalogBiquadFilterB(coeff_, smp[i + 5], work);
AnalogBiquadFilterA(coeff_, smp[i + 6], work);
AnalogBiquadFilterB(coeff_, smp[i + 7], work);
-
+
hist.x1 = work[0];
hist.x2 = work[1];
hist.y1 = work[2];
@@ -413,9 +413,8 @@ void AnalogFilter::singlefilterout_freqbuf(float *smp, fstage &hist,
void AnalogFilter::filterout(float *smp)
{
-
float freqbuf[buffersize];
-
+
if ( freq_smoothing.apply( freqbuf, buffersize, freq ) )
{
/* in transition, need to do fine grained interpolation */
@@ -428,7 +427,7 @@ void AnalogFilter::filterout(float *smp)
for(int i = 0; i < stages + 1; ++i)
singlefilterout(smp, history[i], coeff);
}
-
+
for(int i = 0; i < buffersize; ++i)
smp[i] *= outgain;
}
diff --git a/src/DSP/AnalogFilter.h b/src/DSP/AnalogFilter.h
@@ -74,7 +74,7 @@ class AnalogFilter:public Filter
int order; //the order of the filter (number of poles)
- Value_Smoothing_Filter freq_smoothing; /* for smoothing freq modulations to avoid zipper effect */
+ Value_Smoothing_Filter freq_smoothing; /* for smoothing freq modulations to avoid zipper effect */
};
}
diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp
@@ -44,7 +44,7 @@ FormantFilter::FormantFilter(const FilterParams *pars, Allocator *alloc, unsigne
formant_amp_smoothing[i].sample_rate(srate);
formant_amp_smoothing[i].reset(1.0f);
}
-
+
for(int i = 0; i < numformants; ++i) {
currentformants[i].freq = 1000.0f;
currentformants[i].amp = 1.0f;
@@ -202,16 +202,16 @@ void FormantFilter::filterout(float *smp)
memset(smp, 0, bufferbytes);
float formantbuf[buffersize];
-
+
for(int j = 0; j < numformants; ++j) {
-
+
float tmpbuf[buffersize];
for(int i = 0; i < buffersize; ++i)
tmpbuf[i] = inbuffer[i] * outgain;
formant[j]->filterout(tmpbuf);
-
+
if ( formant_amp_smoothing[j].apply( formantbuf, buffersize, currentformants[j].amp ) )
{
for(int i = 0; i < buffersize; ++i)
diff --git a/src/DSP/FormantFilter.h b/src/DSP/FormantFilter.h
@@ -54,8 +54,8 @@ class FormantFilter:public Filter
float Qfactor, formantslowness, oldQfactor;
float vowelclearness, sequencestretch;
Allocator &memory;
-
- Value_Smoothing_Filter formant_amp_smoothing[FF_MAX_FORMANTS];
+
+ Value_Smoothing_Filter formant_amp_smoothing[FF_MAX_FORMANTS];
};
}
diff --git a/src/DSP/Value_Smoothing_Filter.cpp b/src/DSP/Value_Smoothing_Filter.cpp
@@ -20,15 +20,18 @@
#include "Value_Smoothing_Filter.h"
#include <math.h>
+/* compensate for missing nonlib macro */
+#define assume_aligned(x) (x)
+
void
Value_Smoothing_Filter::sample_rate ( nframes_t n )
{
const float FS = n;
const float T = 0.05f;
-
+
w = _cutoff / (FS * T);
}
-
+
bool
Value_Smoothing_Filter::apply( sample_t * __restrict__ dst, nframes_t nframes, float gt )
{
@@ -41,32 +44,31 @@ Value_Smoothing_Filter::apply( sample_t * __restrict__ dst, nframes_t nframes, f
if ( target_reached(gt) )
return false;
-
- /* sample_t * dst_ = (sample_t*) assume_aligned(dst); */
- sample_t * dst_ = (sample_t*)(dst);
-
+
+ sample_t * dst_ = (sample_t*) assume_aligned(dst);
+
const float a = 0.07f;
const float b = 1 + a;
-
+
const float gm = b * gt;
-
+
float g1 = this->g1;
float g2 = this->g2;
-
+
for (nframes_t i = 0; i < nframes; i++)
{
g1 += w * (gm - g1 - a * g2);
g2 += w * (g1 - g2);
dst_[i] = g2;
}
-
+
g2 += 1e-10f; /* denormal protection */
-
+
if ( fabsf( gt - g2 ) < 0.0001f )
g2 = gt;
-
+
this->g1 = g1;
this->g2 = g2;
-
+
return true;
}
diff --git a/src/DSP/Value_Smoothing_Filter.h b/src/DSP/Value_Smoothing_Filter.h
@@ -26,11 +26,11 @@ typedef float sample_t;
class Value_Smoothing_Filter
{
float w, g1, g2;
-
+
float _cutoff;
bool _reset_on_next_apply;
-
+
public:
Value_Smoothing_Filter ( )
@@ -40,9 +40,8 @@ public:
_reset_on_next_apply = false;
}
-
void reset_on_next_apply ( bool v ) { _reset_on_next_apply = v; }
-
+
void cutoff ( float v ) { _cutoff = v; }
void reset ( float v ) { g2 = g1 = v; }
@@ -50,7 +49,7 @@ public:
inline bool target_reached ( float gt ) const { return gt == g2; }
void sample_rate ( nframes_t n );
-
+
bool apply( sample_t * __restrict__ dst, nframes_t nframes, float gt );
};
diff --git a/src/Misc/Master.cpp b/src/Misc/Master.cpp
@@ -777,7 +777,7 @@ Master::Master(const SYNTH_T &synth_, Config* config)
fakepeakpart[npart] = 0;
}
-
+
ScratchString ss;
for(int npart = 0; npart < NUM_MIDI_PARTS; ++npart)
{
@@ -792,7 +792,7 @@ Master::Master(const SYNTH_T &synth_, Config* config)
smoothing.sample_rate( synth.samplerate );
smoothing.reset_on_next_apply( true ); /* necessary to make CI tests happy, otherwise of no practical use */
-
+
//Insertion Effects init
for(int nefx = 0; nefx < NUM_INS_EFX; ++nefx)
insefx[nefx] = new EffectMgr(*memory, synth, 1, &time);
@@ -1269,7 +1269,7 @@ bool Master::AudioOut(float *outr, float *outl)
continue;
Stereo<float> newvol(part[npart]->volume);
-
+
float pan = part[npart]->panning;
if(pan < 0.5f)
newvol.l *= pan * 2.0f;
@@ -1291,7 +1291,7 @@ bool Master::AudioOut(float *outr, float *outl)
for ( int i = 0; i < synth.buffersize; ++i )
part[npart]->partoutl[i] *= newvol.l;
}
-
+
if ( smoothing_part_r[npart].apply( gainbuf, synth.buffersize, newvol.r ) )
{
for ( int i = 0; i < synth.buffersize; ++i )
@@ -1302,8 +1302,8 @@ bool Master::AudioOut(float *outr, float *outl)
for ( int i = 0; i < synth.buffersize; ++i )
part[npart]->partoutr[i] *= newvol.r;
}
- }
-
+ }
+
//System effects
for(int nefx = 0; nefx < NUM_SYS_EFX; ++nefx) {
if(sysefx[nefx]->geteffect() == 0)
diff --git a/src/Misc/Master.h b/src/Misc/Master.h
@@ -252,12 +252,10 @@ class Master
class DataObj& d, int msg_id = -1,
Master* master_from_mw = nullptr);
- Value_Smoothing_Filter smoothing;
+ Value_Smoothing_Filter smoothing;
- Value_Smoothing_Filter smoothing_part_l[NUM_MIDI_PARTS];
- Value_Smoothing_Filter smoothing_part_r[NUM_MIDI_PARTS];
-
-
+ Value_Smoothing_Filter smoothing_part_l[NUM_MIDI_PARTS];
+ Value_Smoothing_Filter smoothing_part_r[NUM_MIDI_PARTS];
};
class master_dispatcher_t : public rtosc::savefile_dispatcher_t
