commit 0abeffe03c799329dbd19e1a98957b1f7f3607a4
parent 0d2ef00cd5164b623648aabc3c0efc32650baa52
Author: Jonathan Moore Liles <[email protected]>
Date: Sat, 9 Jan 2021 15:17:07 -0800
Replace tabs with spaces to comply with style guidelines.
Diffstat:
7 files changed, 167 insertions(+), 167 deletions(-)
diff --git a/src/DSP/AnalogFilter.cpp b/src/DSP/AnalogFilter.cpp
@@ -274,22 +274,22 @@ void AnalogFilter::setfreq(float frequency)
if(frequency < 0.1f)
frequency = 0.1f;
else if ( frequency > MAX_FREQ )
- frequency = MAX_FREQ;
+ frequency = MAX_FREQ;
float rap = freq / frequency;
if(rap < 1.0f)
rap = 1.0f / rap;
frequency = ceilf(frequency);/* fractional Hz changes are not
- * likely to be audible and waste CPU,
- * esp since we're already smoothing
- * changes, so round it */
+ * likely to be audible and waste CPU,
+ * esp since we're already smoothing
+ * changes, so round it */
if ( fabsf( frequency - freq ) >= 1.0f )
{
- /* only perform computation if absolutely necessary */
- freq = frequency;
- recompute = true;
+ /* only perform computation if absolutely necessary */
+ freq = frequency;
+ recompute = true;
}
}
@@ -356,8 +356,8 @@ void AnalogFilter::singlefilterout(float *smp, fstage &hist)
if ( recompute )
{
- computefiltercoefs(freq,q);
- recompute = false;
+ computefiltercoefs(freq,q);
+ recompute = false;
}
if(order == 1) { //First order filter
@@ -397,45 +397,45 @@ void AnalogFilter::singlefilterout_freqbuf(float *smp, fstage &hist,
for ( int i = 0; i < buffersize; i += 8 )
{
- /* recompute coeffs for each 8 samples */
-
- const float f = ceilf(freqbuf[i] * MAX_FREQ);
-
- if ( fabsf( f - frequency ) >= 1.0f )
- {
- /* don't perform computation more often than necessary */
- computefiltercoefs(f,q);
- frequency = f;
- }
-
- if(order == 1) { //First order filter
- for ( int j = 0; j < 8; j++ )
- {
- float y0 = smp[i+j] * coeff.c[0] + hist.x1 * coeff.c[1]
- + hist.y1 * coeff.d[1];
- hist.y1 = y0;
- hist.x1 = smp[i+j];
- smp[i+j] = y0;
- }
- } else if(order == 2) {//Second order filter
-
- 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);
- AnalogBiquadFilterB(coeff_, smp[i + 3], work);
- AnalogBiquadFilterA(coeff_, smp[i + 4], work);
- 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];
- hist.y2 = work[3];
- }
+ /* recompute coeffs for each 8 samples */
+
+ const float f = ceilf(freqbuf[i] * MAX_FREQ);
+
+ if ( fabsf( f - frequency ) >= 1.0f )
+ {
+ /* don't perform computation more often than necessary */
+ computefiltercoefs(f,q);
+ frequency = f;
+ }
+
+ if(order == 1) { //First order filter
+ for ( int j = 0; j < 8; j++ )
+ {
+ float y0 = smp[i+j] * coeff.c[0] + hist.x1 * coeff.c[1]
+ + hist.y1 * coeff.d[1];
+ hist.y1 = y0;
+ hist.x1 = smp[i+j];
+ smp[i+j] = y0;
+ }
+ } else if(order == 2) {//Second order filter
+
+ 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);
+ AnalogBiquadFilterB(coeff_, smp[i + 3], work);
+ AnalogBiquadFilterA(coeff_, smp[i + 4], work);
+ 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];
+ hist.y2 = work[3];
+ }
}
recompute = true;
@@ -447,15 +447,15 @@ void AnalogFilter::filterout(float *smp)
if ( freq_smoothing.apply( freqbuf, buffersize, freq * MAX_FREQ_CO ) )
{
- /* in transition, need to do fine grained interpolation */
- for(int i = 0; i < stages + 1; ++i)
- singlefilterout_freqbuf(smp, history[i], freqbuf);
+ /* in transition, need to do fine grained interpolation */
+ for(int i = 0; i < stages + 1; ++i)
+ singlefilterout_freqbuf(smp, history[i], freqbuf);
}
else
{
- /* stable state, just use one coeff */
- for(int i = 0; i < stages + 1; ++i)
- singlefilterout(smp, history[i]);
+ /* stable state, just use one coeff */
+ for(int i = 0; i < stages + 1; ++i)
+ singlefilterout(smp, history[i]);
}
for(int i = 0; i < buffersize; ++i)
diff --git a/src/DSP/FormantFilter.cpp b/src/DSP/FormantFilter.cpp
@@ -41,8 +41,8 @@ FormantFilter::FormantFilter(const FilterParams *pars, Allocator *alloc, unsigne
for(int i = 0; i < FF_MAX_FORMANTS; ++i)
{
- formant_amp_smoothing[i].sample_rate(srate);
- formant_amp_smoothing[i].reset(1.0f);
+ formant_amp_smoothing[i].sample_rate(srate);
+ formant_amp_smoothing[i].reset(1.0f);
}
for(int i = 0; i < numformants; ++i) {
@@ -105,7 +105,7 @@ void FormantFilter::setpos(float frequency)
if((fabsf(oldinput - input) < 0.001f) && (fabsf(slowinput - input) < 0.001f)
&& (fabsf(Qfactor - oldQfactor) < 0.001f)) {
- // oldinput=input; setting this will cause problems at very slow changes
+ // oldinput=input; setting this will cause problems at very slow changes
firsttime = false;
return;
}
@@ -207,21 +207,21 @@ void FormantFilter::filterout(float *smp)
float tmpbuf[buffersize];
- for(int i = 0; i < buffersize; ++i)
+ for(int i = 0; i < buffersize; ++i)
tmpbuf[i] = inbuffer[i] * outgain;
- formant[j]->filterout(tmpbuf);
+ formant[j]->filterout(tmpbuf);
- if ( formant_amp_smoothing[j].apply( formantbuf, buffersize, currentformants[j].amp ) )
- {
- for(int i = 0; i < buffersize; ++i)
- smp[i] += tmpbuf[i] * formantbuf[i];
- }
- else
- {
- for(int i = 0; i < buffersize; ++i)
+ if ( formant_amp_smoothing[j].apply( formantbuf, buffersize, currentformants[j].amp ) )
+ {
+ for(int i = 0; i < buffersize; ++i)
+ smp[i] += tmpbuf[i] * formantbuf[i];
+ }
+ else
+ {
+ for(int i = 0; i < buffersize; ++i)
smp[i] += tmpbuf[i] * currentformants[j].amp;
- }
+ }
}
}
diff --git a/src/DSP/SVFilter.cpp b/src/DSP/SVFilter.cpp
@@ -211,21 +211,21 @@ void SVFilter::filterout(float *smp)
if ( freq_smoothing.apply( freqbuf, buffersize, freq ) )
{
- /* 8 sample chunks seems to work OK for AnalogFilter, so do that here too. */
- for ( int i = 0; i < buffersize; i += 8 )
- {
- freq = freqbuf[i];
- computefiltercoefs();
-
- for(int j = 0; j < stages + 1; ++j)
- singlefilterout(smp + i, st[j], par, 8 );
- }
-
- freq = freqbuf[buffersize - 1];
- computefiltercoefs();
+ /* 8 sample chunks seems to work OK for AnalogFilter, so do that here too. */
+ for ( int i = 0; i < buffersize; i += 8 )
+ {
+ freq = freqbuf[i];
+ computefiltercoefs();
+
+ for(int j = 0; j < stages + 1; ++j)
+ singlefilterout(smp + i, st[j], par, 8 );
+ }
+
+ freq = freqbuf[buffersize - 1];
+ computefiltercoefs();
}
else
- for(int i = 0; i < stages + 1; ++i)
+ for(int i = 0; i < stages + 1; ++i)
singlefilterout(smp, st[i], par, buffersize );
for(int i = 0; i < buffersize; ++i)
diff --git a/src/DSP/Value_Smoothing_Filter.cpp b/src/DSP/Value_Smoothing_Filter.cpp
@@ -37,13 +37,13 @@ Value_Smoothing_Filter::apply( sample_t * __restrict__ dst, nframes_t nframes, f
{
if ( _reset_on_next_apply )
{
- reset( gt );
- _reset_on_next_apply = false;
- return false;
+ reset( gt );
+ _reset_on_next_apply = false;
+ return false;
}
if ( target_reached(gt) )
- return false;
+ return false;
sample_t * dst_ = (sample_t*) assume_aligned(dst);
@@ -57,15 +57,15 @@ Value_Smoothing_Filter::apply( sample_t * __restrict__ dst, nframes_t nframes, f
for (nframes_t i = 0; i < nframes; i++)
{
- g1 += w * (gm - g1 - a * g2);
- g2 += w * (g1 - g2);
- dst_[i] = g2;
+ g1 += w * (gm - g1 - a * g2);
+ g2 += w * (g1 - g2);
+ dst_[i] = g2;
}
- g2 += 1e-10f; /* denormal protection */
+ g2 += 1e-10f; /* denormal protection */
if ( fabsf( gt - g2 ) < 0.0001f )
- g2 = gt;
+ g2 = gt;
this->g1 = g1;
this->g2 = g2;
diff --git a/src/DSP/Value_Smoothing_Filter.h b/src/DSP/Value_Smoothing_Filter.h
@@ -34,11 +34,11 @@ class Value_Smoothing_Filter
public:
Value_Smoothing_Filter ( )
- {
- g1 = g2 = 0;
- _cutoff = 10.0f;
- _reset_on_next_apply = false;
- }
+ {
+ g1 = g2 = 0;
+ _cutoff = 10.0f;
+ _reset_on_next_apply = false;
+ }
void reset_on_next_apply ( bool v ) { _reset_on_next_apply = v; }
diff --git a/src/Misc/Master.cpp b/src/Misc/Master.cpp
@@ -189,7 +189,7 @@ static const Ports auto_param_ports = {
d.broadcast(d.loc, "s", a.slots[slot].automations[param].param_path);
}
else
- d.reply(d.loc, "s", a.slots[slot].automations[param].param_path);
+ d.reply(d.loc, "s", a.slots[slot].automations[param].param_path);
rEnd},
{"clear:", rDoc("Clear automation param"), 0,
rBegin;
@@ -784,10 +784,10 @@ Master::Master(const SYNTH_T &synth_, Config* config)
part[npart] = new Part(*memory, synth, time, config->cfg.GzipCompression,
config->cfg.Interpolation, µtonal, fft, &watcher,
(ss+"/part"+npart+"/").c_str);
- smoothing_part_l[npart].sample_rate( synth.samplerate );
- smoothing_part_l[npart].reset_on_next_apply( true ); /* necessary to make CI tests happy, otherwise of no practical use */
- smoothing_part_r[npart].sample_rate( synth.samplerate );
- smoothing_part_r[npart].reset_on_next_apply( true ); /* necessary to make CI tests happy, otherwise of no practical use */
+ smoothing_part_l[npart].sample_rate( synth.samplerate );
+ smoothing_part_l[npart].reset_on_next_apply( true ); /* necessary to make CI tests happy, otherwise of no practical use */
+ smoothing_part_r[npart].sample_rate( synth.samplerate );
+ smoothing_part_r[npart].reset_on_next_apply( true ); /* necessary to make CI tests happy, otherwise of no practical use */
}
smoothing.sample_rate( synth.samplerate );
@@ -1119,52 +1119,52 @@ bool Master::hasMasterCb() const
template <class T>
struct def_skip
{
- static void skip(const char*& argptr) { argptr += sizeof(T); }
+ static void skip(const char*& argptr) { argptr += sizeof(T); }
};
template <class T>
struct str_skip
{
- static void skip(const char*& argptr) { while(argptr++); /*TODO: 4 padding */ }
+ static void skip(const char*& argptr) { while(argptr++); /*TODO: 4 padding */ }
};
template<class T, class Display = T, template<class TMP> class SkipsizeFunc = def_skip>
void _dump_prim_arg(const char*& argptr, std::ostream& os)
{
- os << ' ' << (Display)*(const T*)argptr;
- SkipsizeFunc<T>::skip(argptr);
+ os << ' ' << (Display)*(const T*)argptr;
+ SkipsizeFunc<T>::skip(argptr);
}
void dump_msg(const char* ptr, std::ostream& os = std::cerr)
{
- assert(*ptr == '/');
- os << ptr;
-
- while(*++ptr) ; // skip address
- while(!*++ptr) ; // skip 0s
-
- assert(*ptr == ',');
- os << ' ' << (ptr + 1);
-
- const char* argptr = ptr;
- while(*++argptr) ; // skip type string
- while(!*++argptr) ; // skip 0s
-
- char c;
- while((c = *++ptr))
- {
- switch(c)
- {
- case 'i':
- _dump_prim_arg<int32_t>(argptr, os); break;
- case 'c':
- _dump_prim_arg<int32_t, char>(argptr, os); break;
- // case 's':
- // _dump_prim_arg<char, const char*>(argptr, os); break;
- default:
- exit(1);
- }
- }
+ assert(*ptr == '/');
+ os << ptr;
+
+ while(*++ptr) ; // skip address
+ while(!*++ptr) ; // skip 0s
+
+ assert(*ptr == ',');
+ os << ' ' << (ptr + 1);
+
+ const char* argptr = ptr;
+ while(*++argptr) ; // skip type string
+ while(!*++argptr) ; // skip 0s
+
+ char c;
+ while((c = *++ptr))
+ {
+ switch(c)
+ {
+ case 'i':
+ _dump_prim_arg<int32_t>(argptr, os); break;
+ case 'c':
+ _dump_prim_arg<int32_t, char>(argptr, os); break;
+ // case 's':
+ // _dump_prim_arg<char, const char*>(argptr, os); break;
+ default:
+ exit(1);
+ }
+ }
}
#endif
@@ -1280,28 +1280,28 @@ bool Master::AudioOut(float *outr, float *outl)
- /* This is where the part volume (and pan) smoothing and application happens */
- if ( smoothing_part_l[npart].apply( gainbuf, synth.buffersize, newvol.l ) )
- {
- for ( int i = 0; i < synth.buffersize; ++i )
- part[npart]->partoutl[i] *= gainbuf[i];
- }
- else
- {
- 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 )
- part[npart]->partoutr[i] *= gainbuf[i];
- }
- else
- {
- for ( int i = 0; i < synth.buffersize; ++i )
- part[npart]->partoutr[i] *= newvol.r;
- }
+ /* This is where the part volume (and pan) smoothing and application happens */
+ if ( smoothing_part_l[npart].apply( gainbuf, synth.buffersize, newvol.l ) )
+ {
+ for ( int i = 0; i < synth.buffersize; ++i )
+ part[npart]->partoutl[i] *= gainbuf[i];
+ }
+ else
+ {
+ 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 )
+ part[npart]->partoutr[i] *= gainbuf[i];
+ }
+ else
+ {
+ for ( int i = 0; i < synth.buffersize; ++i )
+ part[npart]->partoutr[i] *= newvol.r;
+ }
}
//System effects
@@ -1372,19 +1372,19 @@ bool Master::AudioOut(float *outr, float *outl)
/* this is where the master volume smoothing and application happens */
if ( smoothing.apply( gainbuf, synth.buffersize, vol ) )
{
- for ( int i = 0; i < synth.buffersize; ++i )
- {
- outl[i] *= gainbuf[i];
- outr[i] *= gainbuf[i];
- }
+ for ( int i = 0; i < synth.buffersize; ++i )
+ {
+ outl[i] *= gainbuf[i];
+ outr[i] *= gainbuf[i];
+ }
}
else
{
- for ( int i = 0; i < synth.buffersize; ++i )
- {
- outl[i] *= vol;
- outr[i] *= vol;
- }
+ for ( int i = 0; i < synth.buffersize; ++i )
+ {
+ outl[i] *= vol;
+ outr[i] *= vol;
+ }
}
vuUpdate(outl, outr);
diff --git a/src/Misc/Part.cpp b/src/Misc/Part.cpp
@@ -681,9 +681,9 @@ void Part::SetController(unsigned int type, int par)
case C_volume:
ctl.setvolume(par);
if(ctl.volume.receive != 0)
- setVolumedB(volume127TodB( ctl.volume.volume * 127.0f ) );
+ setVolumedB(volume127TodB( ctl.volume.volume * 127.0f ) );
else
- /* FIXME: why do this? */
+ /* FIXME: why do this? */
setVolumedB(Volume);
break;
case C_sustain:
@@ -698,7 +698,7 @@ void Part::SetController(unsigned int type, int par)
ctl.resetall();
ReleaseSustainedKeys();
if(ctl.volume.receive != 0)
- setVolumedB(volume127TodB( ctl.volume.volume * 127.0f ) );
+ setVolumedB(volume127TodB( ctl.volume.volume * 127.0f ) );
else
setVolumedB(Volume);
setPpanning(Ppanning); //update the panning
