[SCM] calf/master: FuckYEAH! analyzer's working in every fuckin' mode - fuckmake -j8 && sudo make install && calfjackhost analyzereleven (with spines to come..)
js at users.alioth.debian.org
js at users.alioth.debian.org
Tue May 7 15:41:08 UTC 2013
The following commit has been merged in the master branch:
commit 4a4194ee72f1f87a25b21cc1bff15e8e793640be
Author: Markus Schmidt <schmidt at boomshop.net>
Date: Fri Sep 7 04:19:58 2012 +0200
FuckYEAH! analyzer's working in every fuckin' mode - fuckmake -j8 && sudo make install && calfjackhost analyzereleven (with spines to come..)
diff --git a/src/calf/modules.h b/src/calf/modules.h
index e55630e..5e68c76 100644
--- a/src/calf/modules.h
+++ b/src/calf/modules.h
@@ -328,6 +328,7 @@ public:
bool get_gridline(int index, int subindex, float &pos, bool &vertical, std::string &legend, cairo_iface *context) const;
bool get_clear_all(int index) const;
mutable int _mode_old;
+ mutable bool _falling;
protected:
static const int max_phase_buffer_size = 8192;
int phase_buffer_size;
@@ -346,6 +347,7 @@ protected:
fftw_real *fft_smoothL, *fft_smoothR;
float *fft_deltaL, *fft_deltaR;
float *fft_holdL, *fft_holdR;
+ float *fft_fallingL, *fft_fallingR;
float *fft_freezeL, *fft_freezeR;
mutable int lintrans;
mutable int ____analyzer_phase_was_drawn_here;
diff --git a/src/metadata.cpp b/src/metadata.cpp
index 8bdc717..2736f71 100644
--- a/src/metadata.cpp
+++ b/src/metadata.cpp
@@ -982,12 +982,12 @@ CALF_PLUGIN_INFO(stereo) = { 0x8588, "StereoTools", "Calf Stereo Tools", "Markus
////////////////////////////////////////////////////////////////////////////
CALF_PORT_NAMES(analyzer) = {"In L", "In R", "Out L", "Out R"};
-const char *gonio_mode_names[] = { "Small Dots", "Medium Dots", "Big Dots", "Fields", "Lines" };
+const char *gonio_mode_names[] = { "Small Dots", "Medium Dots", "Big Dots", "Fields", "Lines (High CPU)" };
const char *analyzer_mode_names[] = { "Analyzer Average", "Analyzer Left", "Analyzer Right", "Analyzer Stereo", "Stereo Image", "Stereo Difference" };
const char *analyzer_smooth_names[] = { "Off", "Falling", "Transition" };
-const char *analyzer_post_names[] = { "Normalized", "Additive", "Denoised Peaks" };
+const char *analyzer_post_names[] = { "Normalized", "Average", "Additive", "Denoised Peaks" };
const char *analyzer_view_names[] = { "Bars", "Lines", "Cubic Splines" };
-const char *analyzer_scale_names[] = { "Loagrithmic", "Linear" };
+const char *analyzer_scale_names[] = { "Logarithmic", "Linear" };
CALF_PORT_PROPS(analyzer) = {
{ 0, 0, 1, 0, PF_FLOAT | PF_SCALE_GAIN | PF_CTL_METER | PF_CTLO_LABEL | PF_UNIT_DB | PF_PROP_OUTPUT | PF_PROP_OPTIONAL, NULL, "meter_L", "Level L" },
{ 0, 0, 1, 0, PF_FLOAT | PF_SCALE_GAIN | PF_CTL_METER | PF_CTLO_LABEL | PF_UNIT_DB | PF_PROP_OUTPUT | PF_PROP_OPTIONAL, NULL, "meter_R", "Level R" },
@@ -997,8 +997,8 @@ CALF_PORT_PROPS(analyzer) = {
{ 1.25, 0.5, 2, 0, PF_FLOAT | PF_SCALE_PERC | PF_CTL_KNOB | PF_UNIT_COEF | PF_PROP_NOBOUNDS, NULL, "analyzer_level", "Analyzer Level" },
{ 0, 0, 5, 0, PF_ENUM | PF_CTL_COMBO, analyzer_mode_names, "analyzer_mode", "Analyzer Mode" },
{ 0, 0, 1, 2, PF_ENUM | PF_CTL_COMBO, analyzer_scale_names, "analyzer_scale", "Analyzer Scale" },
- { 1, 0, 2, 0, PF_ENUM | PF_CTL_COMBO, analyzer_post_names, "analyzer_post", "Analyzer Post FFT" },
- { 1, 0, 2, 2, PF_ENUM | PF_CTL_COMBO , analyzer_view_names, "analyzer_view", "Analyzer View" },
+ { 2, 0, 3, 0, PF_ENUM | PF_CTL_COMBO, analyzer_post_names, "analyzer_post", "Analyzer Post FFT" },
+ { 1, 0, 1, 2, PF_ENUM | PF_CTL_COMBO , analyzer_view_names, "analyzer_view", "Analyzer View" },
{ 1, 0, 2, 0, PF_ENUM | PF_CTL_COMBO, analyzer_smooth_names, "analyzer_smoothing", "Analyzer Smoothing" },
{ 6, 2, 8, 0, PF_INT | PF_SCALE_LINEAR | PF_CTL_KNOB | PF_UNIT_COEF | PF_PROP_GRAPH, NULL, "analyzer_accuracy", "Analyzer Accuracy" },
{ 13, 1, 15, 0, PF_INT | PF_SCALE_LINEAR | PF_CTL_KNOB | PF_UNIT_COEF | PF_PROP_GRAPH, NULL, "analyzer_speed", "Analyzer Speed" },
diff --git a/src/modules.cpp b/src/modules.cpp
index 4417214..d98e845 100644
--- a/src/modules.cpp
+++ b/src/modules.cpp
@@ -889,6 +889,11 @@ analyzer_audio_module::analyzer_audio_module() {
memset(fft_smoothL, 0, max_fft_cache_size * sizeof(fftw_real));
memset(fft_smoothR, 0, max_fft_cache_size * sizeof(fftw_real));
+ fft_fallingL = (float*) calloc(max_fft_cache_size, sizeof(float));
+ fft_fallingR = (float*) calloc(max_fft_cache_size, sizeof(float));
+ memset(fft_fallingL, false, max_fft_cache_size * sizeof(float));
+ memset(fft_fallingR, false, max_fft_cache_size * sizeof(float));
+
fft_deltaL = (float*) calloc(max_fft_cache_size, sizeof(float));
fft_deltaR = (float*) calloc(max_fft_cache_size, sizeof(float));
memset(fft_deltaL, 0, max_fft_cache_size * sizeof(float));
@@ -934,8 +939,8 @@ void analyzer_audio_module::params_changed() {
___sanitize = true;
}
if(*params[param_analyzer_smoothing] != _smooth_old) {
- memset(fft_deltaL, 0, max_fft_cache_size * sizeof(float));
- memset(fft_deltaR, 0, max_fft_cache_size * sizeof(float));
+// memset(fft_deltaL, 0, max_fft_cache_size * sizeof(float));
+// memset(fft_deltaR, 0, max_fft_cache_size * sizeof(float));
_smooth_old = *params[param_analyzer_smoothing];
___sanitize = true;
}
@@ -963,6 +968,8 @@ void analyzer_audio_module::params_changed() {
memset(fft_smoothR, 1e-20, max_fft_cache_size * sizeof(float));
memset(fft_deltaL, 1e-20, max_fft_cache_size * sizeof(float));
memset(fft_deltaR, 1e-20, max_fft_cache_size * sizeof(float));
+ memset(fft_fallingL, false, max_fft_cache_size * sizeof(float));
+ memset(fft_fallingR, false, max_fft_cache_size * sizeof(float));
memset(spline_buffer, 0, 200 * sizeof(int));
____analyzer_phase_was_drawn_here = 0;
}
@@ -1093,9 +1100,7 @@ bool analyzer_audio_module::get_graph(int index, int subindex, float *data, int
// we want to remember old fft_out values for smoothing as well
// and we fill the hold buffer in this (extra) cycle
for(int i = 0; i < _accuracy; i++) {
- // fill smoothing buffer
- fft_smoothL[i] = fft_outL[i];
- fft_smoothR[i] = fft_outR[i]; // go to the right position back in time according to accuracy
+ // go to the right position back in time according to accuracy
// settings and cycling in the main buffer
int _fpos = (fpos - _accuracy * 2 \
+ (i * 2)) % max_fft_buffer_size;
@@ -1133,10 +1138,21 @@ bool analyzer_audio_module::get_graph(int index, int subindex, float *data, int
fft_inL[i] = valL;
fft_inR[i] = valR;
- // fill smoothing buffer with the last out values
- // before fft is recalced
- fft_smoothL[i] = fft_outL[i];
- fft_smoothR[i] = fft_outR[i];
+ // fill smoothing & falling buffer
+ if(_param_smooth == 2) {
+ fft_smoothL[i] = fft_outL[i];
+ fft_smoothR[i] = fft_outR[i];
+ }
+ if(_param_smooth == 1) {
+ if(fft_smoothL[i] < fabs(fft_outL[i])) {
+ fft_smoothL[i] = fabs(fft_outL[i]);
+ fft_deltaL[i] = 1.f;
+ }
+ if(fft_smoothR[i] < fabs(fft_outR[i])) {
+ fft_smoothR[i] = fabs(fft_outR[i]);
+ fft_deltaR[i] = 1.f;
+ }
+ }
// fill hold buffer with last out values
// before fft is recalced
@@ -1166,7 +1182,7 @@ bool analyzer_audio_module::get_graph(int index, int subindex, float *data, int
int _lintrans = (int)((float)points * log((20.f + 2.f * \
(float)srate / (float)_accuracy) / 20.f) / log(1000.f));
lintrans = (int)(_lintrans + points % _lintrans / \
- floor(points / _lintrans)) / 4; // / 4 was added to see finer bars but breaks low end
+ floor(points / _lintrans)) / 2; // / 4 was added to see finer bars but breaks low end
}
for (int i = 0; i <= points; i++)
{
@@ -1223,8 +1239,20 @@ bool analyzer_audio_module::get_graph(int index, int subindex, float *data, int
fft_outL[_iter] += fabs(fft_outL[j]);
fft_outR[_iter] += fabs(fft_outR[j]);
}
+ fft_outL[_iter] /= (_iter - iter);
+ fft_outR[_iter] /= (_iter - iter);
break;
case 2:
+ // Analyzer Additive - cycle through skipped values and
+ // add them
+ // if fft was renewed, recalc the absolute values if
+ // frequencies are skipped
+ for(int j = iter + 1; j < _iter; j++) {
+ fft_outL[_iter] += fabs(fft_outL[j]);
+ fft_outR[_iter] += fabs(fft_outR[j]);
+ }
+ break;
+ case 3:
// Analyzer Denoised Peaks - filter out unwanted noise
for(int k = 0; k < std::max(10 , std::min(400 ,\
(int)(2.f*(float)((_iter - iter))))); k++) {
@@ -1288,40 +1316,69 @@ bool analyzer_audio_module::get_graph(int index, int subindex, float *data, int
// was done above
// #######################################
if(subindex == 0) {
- float _frate = 0.1;
- // ############################################################################################################################################
- if(_param_smooth == 1 and fabs(fft_smoothL[iter]) - fabs(fft_outL[iter]) > _frate * _param_speed) {
- // falling
- if(fftdone) {
- // rebuild delta values after fft was done
- fft_deltaL[iter] = pow(fabs(fft_outL[iter]) / (fabs(fft_outL[iter]) - _frate * _param_speed), 1.f / _param_speed);
- fft_deltaR[iter] = pow(fabs(fft_outR[iter]) / (fabs(fft_outL[iter]) - _frate * _param_speed), 1.f / _param_speed);
- } else {
- // change fft_smooth according to delta
- fft_smoothL[iter] *= fft_deltaL[iter];
- fft_smoothR[iter] *= fft_deltaR[iter];
- }
- } else if(_param_smooth >= 1) {
+ float _fdelta = 0.91;
+ if(_param_smooth == 2) {
// smoothing
if(fftdone) {
// rebuild delta values after fft was done
if(_param_mode < 5) {
fft_deltaL[iter] = pow(fabs(fft_outL[iter]) / fabs(fft_smoothL[iter]), 1.f / _param_speed);
- fft_deltaR[iter] = pow(fabs(fft_outR[iter]) / fabs(fft_smoothR[iter]), 1.f / _param_speed);
} else {
fft_deltaL[iter] = (posneg * fabs(fft_outL[iter]) - fft_smoothL[iter]) / _param_speed;
- fft_deltaR[iter] = (posneg * fabs(fft_outR[iter]) - fft_smoothR[iter]) / _param_speed;
}
} else {
// change fft_smooth according to delta
if(_param_mode < 5) {
fft_smoothL[iter] *= fft_deltaL[iter];
- fft_smoothR[iter] *= fft_deltaR[iter];
} else {
fft_smoothL[iter] += fft_deltaL[iter];
- fft_smoothR[iter] += fft_deltaR[iter];
}
}
+ } else if(_param_smooth == 1) {
+ // falling
+ if(fftdone) {
+ // rebuild delta values after fft was done
+ //fft_deltaL[iter] = _fdelta;
+ }
+ // change fft_smooth according to delta
+ fft_smoothL[iter] *= fft_deltaL[iter];
+
+ if(fft_deltaL[iter] > _fdelta) {
+ fft_deltaL[iter] *= 1.f - (16.f - _param_speed) / 2000.f;
+ }
+ }
+
+ if(_param_mode > 2 and _param_mode < 5) {
+ // we need right buffers, too for stereo image and
+ // stereo analyzer
+ if(_param_smooth == 2) {
+ // smoothing
+ if(fftdone) {
+ // rebuild delta values after fft was done
+ if(_param_mode < 5) {
+ fft_deltaR[iter] = pow(fabs(fft_outR[iter]) / fabs(fft_smoothR[iter]), 1.f / _param_speed);
+ } else {
+ fft_deltaR[iter] = (posneg * fabs(fft_outR[iter]) - fft_smoothR[iter]) / _param_speed;
+ }
+ } else {
+ // change fft_smooth according to delta
+ if(_param_mode < 5) {
+ fft_smoothR[iter] *= fft_deltaR[iter];
+ } else {
+ fft_smoothR[iter] += fft_deltaR[iter];
+ }
+ }
+ } else if(_param_smooth == 1) {
+ // falling
+ if(fftdone) {
+ // rebuild delta values after fft was done
+ //fft_deltaR[iter] = _fdelta;
+ }
+ // change fft_smooth according to delta
+ fft_smoothR[iter] *= fft_deltaR[iter];
+ if(fft_deltaR[iter] > _fdelta)
+ fft_deltaR[iter] *= 1.f - (16.f - _param_speed) / 2000.f;
+ }
}
}
// #######################################
@@ -1381,16 +1438,18 @@ bool analyzer_audio_module::get_graph(int index, int subindex, float *data, int
// we want to draw Stereo Image
if(subindex == 0 or subindex == 2) {
// Left channel signal
- data[i] = (0.2f * (1.f - stereo_coeff) * pow(fabs(valL), 5.f) + 0.8f * ( 1.f - stereo_coeff) * pow(valL, 3.f) + valL * stereo_coeff);
+ data[i] = fabs(valL) * stereo_coeff * 0.0001;
} else if (subindex == 1 or subindex == 3) {
// Right channel signal
- data[i] = (0.2f * (1.f - stereo_coeff) * pow(fabs(valR), 5.f) + 0.8f * ( 1.f - stereo_coeff) * pow(valR, 3.f) + valR * stereo_coeff) * -1.f;
+ data[i] = fabs(valR) * stereo_coeff * -0.0001f;
}
break;
case 5:
// We want to draw Stereo Difference
if(i) {
- data[i] = 0.2f * (1.f - stereo_coeff) * pow(valL, 5.f) + 0.8f * ( 1.f - stereo_coeff) * pow(valL, 3.f) + valL * stereo_coeff;
+
+ //data[i] = stereo_coeff * valL * 2.f;
+ data[i] = (0.2f * (1.f - stereo_coeff) * pow(valL, 5.f) + 0.8f * ( 1.f - stereo_coeff) * pow(valL, 3.f) + valL * stereo_coeff) * 1.8f;
}
else data[i] = 0.f;
break;
--
calf audio plugins packaging
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