Split timeweighting into different types for each of the possible use cases.

.gitignore

@@ -20,3 +20,4 @@ doc
 .spyproject
 .cache
 _skbuild
+acme_log.log

src/lasp/dsp/lasp_slm.cpp

@@ -21,7 +21,7 @@ SLM::SLM(const d fs, const d Lref, const us downsampling_fac, const d tau,
       // components of
       // single pole low pass
       // filter
-  Lrefsq(Lref*Lref),                                       // Reference level
+      Lrefsq(Lref * Lref), // Reference level
       downsampling_fac(downsampling_fac),
 
       // Initalize mean square
@@ -35,6 +35,7 @@ SLM::SLM(const d fs, const d Lref, const us downsampling_fac, const d tau,
 
 {
   DEBUGTRACE_ENTER;
+  DEBUGTRACE_PRINT(_alpha);
 
   // Make sure thread pool is running
   getPool();
@@ -42,7 +43,7 @@ SLM::SLM(const d fs, const d Lref, const us downsampling_fac, const d tau,
   if (Lref <= 0) {
     throw rte("Invalid reference level");
   }
-  if (tau <= 0) {
+  if (tau < 0) {
     throw rte("Invalid time constant for Single pole lowpass filter");
   }
   if (fs <= 0) {
@@ -66,17 +67,17 @@ std::vector<unique_ptr<Filter>> createBandPass(const dmat &coefs) {
   }
   return bf;
 }
-us SLM::suggestedDownSamplingFac(const d fs,const d tau) {
+us SLM::suggestedDownSamplingFac(const d fs, const d tau) {
-  if(tau<0) throw rte("Invalid time weighting time constant");
+  if (fs <= 0)
-  if(fs<=0) throw rte("Invalid sampling frequency");
+    throw rte("Invalid sampling frequency");
   // A reasonable 'framerate' for the sound level meter, based on the
   // filtering time constant.
   if (tau > 0) {
     d fs_slm = 10 / tau;
-    if(fs_slm < 30) {
+    if (fs_slm < 30) {
       fs_slm = 30;
     }
-    return std::max((us) 1, static_cast<us>(fs / fs_slm));
+    return std::max((us)1, static_cast<us>(fs / fs_slm));
   } else {
     return 1;
   }
@@ -102,7 +103,7 @@ SLM SLM::fromBiquads(const d fs, const d Lref, const us downsampling_fac,
   );
 }
 
-vd SLM::run_single(vd work,const us i) {
+vd SLM::run_single(vd work, const us i) {
 
   // Filter input in-place
   _bandpass[i]->filter(work);
@@ -142,7 +143,7 @@ vd SLM::run_single(vd work,const us i) {
   Pmax(i) = std::max(Pmax(i), arma::max(work));
 
   // Convert to levels in dB
-  work = 10*arma::log10((work+arma::datum::eps)/Lrefsq);
+  work = 10 * arma::log10((work + arma::datum::eps) / Lrefsq);
 
   return work;
 }
@@ -170,8 +171,8 @@ dmat SLM::run(const vd &input_orig) {
     /* DEBUGTRACE_PRINT(res.n_rows); */
     /* DEBUGTRACE_PRINT(futs.size()); */
 
-    // Update the total number of samples harvested so far. NOTE: *This should be
+    // Update the total number of samples harvested so far. NOTE: *This should
-    // done AFTER the threads are done!!!*
+    // be done AFTER the threads are done!!!*
   }
   N += input.n_rows;
 

src/lasp/lasp_common.py

@@ -286,46 +286,57 @@ class this_lasp_shelve(Shelve):
         return os.path.join(lasp_appdir, f'{node}_config.shelve')
 
 class TimeWeighting:
-    none = (-1, 'Raw (no time weighting)')
+    # This is not actually a time weighting
+    none = (0, 'Raw (no time weighting)')
+
     uufast = (1e-4, '0.1 ms')
     ufast = (35e-3, 'Impulse (35 ms)')
     fast = (0.125, 'Fast (0.125 s)')
     slow = (1.0, 'Slow (1.0 s)')
     tens = (10., '10 s')
-    infinite = (0, 'Infinite')
+
-    types_realtime = (ufast, fast, slow, tens, infinite)
+    # All-averaging: compute the current average of all history. Kind of the
-    types_all = (none, uufast, ufast, fast, slow, tens, infinite)
+    # equivalent level. Only useful for power spectra. For Sound Level Meter,
+    # equivalent levels does that thing.
+    averaging = (-1, 'All-averaging')
+
+    types_all = (none, uufast, ufast, fast, slow, tens, averaging)
+
+    # Used for combobox of realt time power spectra
+    types_realtimeaps = (ufast, fast, slow, tens, averaging)
+    # Used for combobox of realt time sound level meter
+    types_realtimeslm = (ufast, fast, slow, tens)
+
+    # Used for combobox of sound level meter - time dependent
+    types_slmt = (none, uufast, ufast, fast, slow, tens)
+
+    # Used for combobox of sound level meter - Lmax statistics
+    types_slmstats = (uufast, ufast, fast, slow, tens)
 
     default = fast
-    default_index = 3
-    default_index_realtime = 1
 
     @staticmethod
-    def fillComboBox(cb, realtime=False):
+    def fillComboBox(cb, types):
         """
         Fill TimeWeightings to a combobox
 
         Args:
             cb: QComboBox to fill
+            types: The types to fill it with
         """
         cb.clear()
-        if realtime:
+
-            types = TimeWeighting.types_realtime
+        logging.debug(f'{types}')
-            defindex = TimeWeighting.default_index_realtime
-        else:
-            types = TimeWeighting.types_all
-            defindex = TimeWeighting.default_index
         for tw in types:
             cb.addItem(tw[1], tw)
-
+            if TimeWeighting.default == tw:
-        cb.setCurrentIndex(defindex)
+                cb.setCurrentIndex(cb.count()-1)
 
     @staticmethod
     def getCurrent(cb):
-        if cb.count() == len(TimeWeighting.types_realtime):
+        for type in TimeWeighting.types_all:
-            return TimeWeighting.types_realtime[cb.currentIndex()]
+            if cb.currentText() == type[1]:
-        else:
+                return type
-            return TimeWeighting.types_all[cb.currentIndex()]
 
 
 class FreqWeighting:

src/lasp/lasp_slm.py

@@ -32,10 +32,10 @@ class SLM:
     def __init__(self,
                  fs,
                  fbdesigner=None,
-                 tw: TimeWeighting =TimeWeighting.fast,
+                 tw: TimeWeighting = TimeWeighting.fast,
-                 fw: FreqWeighting =FreqWeighting.A,
+                 fw: FreqWeighting = FreqWeighting.A,
-                 xmin = None,
+                 xmin=None,
-                 xmax = None,
+                 xmax=None,
                  include_overall=True,
                  level_ref_value=P_REF,
                  offset_t=0):
@@ -72,7 +72,8 @@ class SLM:
         self.xs = list(range(xmin, xmax + 1))
 
         nfilters = len(self.xs)
-        if include_overall: nfilters +=1
+        if include_overall:
+            nfilters += 1
         self.include_overall = include_overall
         self.offset_t = offset_t
 
@@ -91,7 +92,7 @@ class SLM:
 
         # This is a bit of a hack, as the 5 is hard-encoded here, but should in
         # fact be coming from somewhere else..
-        sos_overall = np.array([1,0,0,1,0,0]*5, dtype=float)
+        sos_overall = np.array([1, 0, 0, 1, 0, 0]*5, dtype=float)
 
         if fbdesigner is not None:
             assert fbdesigner.fs == fs
@@ -108,14 +109,14 @@ class SLM:
                 # Create a unit impulse response filter, every third index equals
                 # 1, so b0 = 1 and a0 is 1 (by definition)
                 # a0 = 1, b0 = 1, rest is zero
-                sos[:,-1] = sos_overall
+                sos[:, -1] = sos_overall
                 self.nom_txt.append('overall')
 
         else:
             # No filterbank, means we do only compute the overall values. This
             # means that in case of include_overall, it creates two overall
             # channels. That would be confusing, so we do not allow it.
-            sos = sos_overall[:,np.newaxis]
+            sos = sos_overall[:, np.newaxis]
             self.nom_txt.append('overall')
 
         # Downsampling factor, determine from single pole low pass filter time
@@ -137,7 +138,6 @@ class SLM:
         # Initialize counter to 0
         self.N = 0
 
-
     def run(self, data):
         """
         Add new fresh timedata to the Sound Level Meter
@@ -169,7 +169,6 @@ class SLM:
 
         return output
 
-
     def return_as_dict(self, dat):
         """
         Helper function used to return resulting data in a proper way.