First work on SLM. Seems to be working properly without pre-filtering and bandpass bank

lasp/c/CMakeLists.txt

@@ -19,7 +19,7 @@ add_library(lasp_lib
   lasp_firfilterbank.c
   lasp_sosfilterbank.c
   lasp_decimation.c
-  lasp_sp_lowpass.c
+  lasp_slm.c
   )
 
 

lasp/c/lasp_math_raw.h

@@ -33,6 +33,7 @@
 #define d_sin sin
 #define d_cos cos
 #define d_pow pow
+#define d_log10 log10
 
 #else  // LASP_DOUBLE_PRECISION not defined
 #define c_conj conjf
@@ -47,7 +48,7 @@
 #define d_sin sinf
 #define d_cos cosf
 #define d_pow powf
-
+#define d_log10 log10f
 #endif // LASP_DOUBLE_PRECISION
 
 #ifdef M_PI

lasp/c/lasp_slm.c

@@ -0,0 +1,185 @@
+#define TRACERPLUS (10)
+#include "lasp_slm.h"
+#include "lasp_assert.h"
+#include "lasp_tracer.h"
+
+typedef struct Slm {
+    Sosfilterbank *prefilter;  /// Pre-filter, A, or C. If NULL, not used.
+    Sosfilterbank *bandpass;   /// Filterbank. If NULL, not used
+    Sosfilterbank **splowpass; /// Used for time-weighting of the squared signal
+    d ref_level;               /// Reference value for computing decibels
+    us downsampling_fac;       /// Every x'th sample is returned.
+    us cur_offset;             /// Storage for offset point in input arrays
+    vd Leq; /// Storage for the computed equivalent levels so far.
+
+} Slm;
+
+Slm *Slm_create(Sosfilterbank *prefilter, Sosfilterbank *bandpass, const d fs,
+        const d tau, const d ref_level, us *downsampling_fac) {
+    fsTRACE(15);
+    assertvalidptr(downsampling_fac);
+
+    Slm *slm = NULL;
+    if (tau <= 0) {
+        WARN("Invalid time constant");
+        return NULL;
+    } else if (ref_level <= 0) {
+        WARN("Invalid reference level");
+        return NULL;
+    } else if (fs <= 0) {
+        WARN("Invalid sampling frequency");
+        return NULL;
+    }
+
+    slm = (Slm *)a_malloc(sizeof(Slm));
+    slm->ref_level = ref_level;
+    slm->prefilter = prefilter;
+    slm->bandpass = bandpass;
+
+    /// Compute the downsampling factor. This one is chosen based on the
+    /// lowpass filter. Which has a -3 dB point of f = 1/(tau*2*pi). See LASP
+    /// documentation for the computation of its minus 20 dB point. We set the
+    /// reduction in its 'sampling frequency' such that its noise is at a level
+    /// of 20 dB less than its 'signal'.
+    const d fs_slm = 1 / (2 * number_pi * tau) * (1 - 0.01) / 0.01;
+    slm->downsampling_fac = (us)(fs / fs_slm);
+    slm->cur_offset = 0;
+    *downsampling_fac = slm->downsampling_fac;
+
+    /// Create the single pole lowpass
+    us filterbank_size;
+    if (bandpass) {
+        filterbank_size = Sosfilterbank_getFilterbankSize(bandpass);
+    } else {
+        filterbank_size = 1;
+    }
+
+    vd lowpass_sos = vd_alloc(6);
+    d b0 = 1.0 / (1 + 2 * tau * fs);
+    *getvdval(&lowpass_sos, 0) = b0;
+    *getvdval(&lowpass_sos, 1) = b0;
+    *getvdval(&lowpass_sos, 2) = 0;
+    *getvdval(&lowpass_sos, 3) = 1;
+    *getvdval(&lowpass_sos, 4) = (1 - 2 * tau * fs) * b0;
+    *getvdval(&lowpass_sos, 5) = 0;
+
+    slm->splowpass = a_malloc(filterbank_size * sizeof(Sosfilterbank *));
+    for (us ch = 0; ch < filterbank_size; ch++) {
+        /// Allocate a filterbank with one channel and one section.
+        slm->splowpass[ch] = Sosfilterbank_create(1, 1);
+        Sosfilterbank_setFilter(slm->splowpass[ch], 0, lowpass_sos);
+    }
+    vd_free(&lowpass_sos);
+
+    feTRACE(15);
+    return slm;
+}
+
+dmat Slm_run(Slm *slm, vd *input_data) {
+    fsTRACE(15);
+    assertvalidptr(slm);
+    assert_vx(input_data);
+
+    /// First step: run the input data through the pre-filter
+    vd prefiltered;
+    if (slm->prefilter)
+        prefiltered = Sosfilterbank_filter(slm->prefilter, input_data);
+    else {
+        prefiltered = dmat_foreign(input_data);
+    }
+    dmat bandpassed;
+    if (slm->bandpass) {
+        bandpassed = Sosfilterbank_filter(slm->bandpass, &prefiltered);
+    } else {
+        bandpassed = dmat_foreign(&prefiltered);
+    }
+    us filterbank_size = bandpassed.n_cols;
+
+    /// Next step: square all values. We do this in-place. Then we filter for
+    /// each channel.
+    d ref_level = slm->ref_level;
+    d *tmp;
+
+    /// Pre-calculate the size of the output data
+    us downsampling_fac = slm->downsampling_fac;
+    us samples_bandpassed = bandpassed.n_rows;
+    iVARTRACE(15, samples_bandpassed);
+    us cur_offset = slm->cur_offset;
+
+    int nsamples_output = (samples_bandpassed - cur_offset) / downsampling_fac;
+    while(nsamples_output*downsampling_fac + cur_offset < samples_bandpassed)
+        nsamples_output++;
+    if(nsamples_output < 0) nsamples_output = 0;
+
+    iVARTRACE(15, nsamples_output);
+    iVARTRACE(15, cur_offset);
+
+    dmat levels = dmat_alloc(nsamples_output, filterbank_size);
+
+    for (us ch = 0; ch < bandpassed.n_cols; ch++) {
+        iVARTRACE(15, ch);
+        /// Inplace squaring of the signal
+        for (us sample = 0; sample < bandpassed.n_rows; sample++) {
+            tmp = getdmatval(&bandpassed, sample, ch);
+            *tmp = *tmp * *tmp;
+        }
+
+        // Now that all data for the channel is squared, we can run it through
+        // the low-pass filter
+        vd chan = dmat_column(&bandpassed, ch);
+
+        /// Apply single-pole lowpass filter for current filterbank channel
+        vd power_filtered = Sosfilterbank_filter(slm->splowpass[ch], &chan);
+        dbgassert(chan.n_rows == power_filtered.n_rows, "BUG");
+
+        /// Output resulting levels at a lower interval
+        us i = 0;
+        d level;
+        while (cur_offset < samples_bandpassed) {
+            iVARTRACE(10, i);
+            iVARTRACE(10, cur_offset);
+            level = 10 * d_log10(*getvdval(&power_filtered, cur_offset) / ref_level /
+                    ref_level);
+
+            *getdmatval(&levels, i++, ch) = level;
+            cur_offset = cur_offset + downsampling_fac;
+        }
+        iVARTRACE(15, cur_offset);
+        iVARTRACE(15, i);
+        dbgassert(i == nsamples_output, "BUG");
+        slm->cur_offset = cur_offset - samples_bandpassed;
+
+        vd_free(&chan);
+        vd_free(&power_filtered);
+    }
+
+    vd_free(&prefiltered);
+    dmat_free(&bandpassed);
+    feTRACE(15);
+    return levels;
+}
+
+void Slm_free(Slm *slm) {
+    fsTRACE(15);
+    assertvalidptr(slm);
+    if (slm->prefilter) {
+        Sosfilterbank_free(slm->prefilter);
+    }
+    assertvalidptr(slm->splowpass);
+
+    us filterbank_size;
+    if (slm->bandpass) {
+        filterbank_size = Sosfilterbank_getFilterbankSize(slm->bandpass);
+        Sosfilterbank_free(slm->bandpass);
+    } else {
+        filterbank_size = 1;
+    }
+    for (us ch = 0; ch < filterbank_size; ch++) {
+        Sosfilterbank_free(slm->splowpass[ch]);
+    }
+    a_free(slm->splowpass);
+
+    a_free(slm);
+
+    feTRACE(15);
+}

lasp/c/lasp_slm.h

@@ -2,60 +2,71 @@
 //
 // Author: J.A. de Jong - ASCEE
 //
-// Description: Implementation of a real time Sound Level Meter, based on
+// Description: Multi-purpose implementation of a real time Sound Level Meter,
-// given slm.
+// can be used for full-signal filtering, (fractional) octave band filtering,
+// etc.
 // //////////////////////////////////////////////////////////////////////
 #pragma once
-#ifndef LASP_slm_H
+#ifndef LASP_SLM_H
-#define LASP_slm_H
+#define LASP_SLM_H
 #include "lasp_types.h"
 #include "lasp_mat.h"
 #include "lasp_sosfilterbank.h"
 
 typedef struct Slm Slm;
 
+#define TAU_FAST (1.0/8.0)
+#define TAU_SLOW (1.0)
+#define TAU_IMPULSE (35e-3)
+
 /** 
- * Initializes a Sound level meter.
+ * Initializes a Sound level meter. NOTE: Sound level meter takes over
+ * ownership of pointers to the filterbanks for prefiltering and band-pass
+ * filtering! After passing them to the constructor of the Slm, they should not
+ * be touched anymore.
  *
- * @param fb: Sosfiterbank handle, used to pre-filter data
+ * @param[in] weighting: Sosfiterbank handle, used to pre-filter data. This is
- * @param T:  Single pole low pass filter time constant to use.
+ * in most cases an A-weighting filter, or C-weighting. If NULL, no
+ * pre-filtering is done. That can be the case in situations of Z-weighting.
+ * @param[in] fb: Sosfiterbank handle, bandpass filters. 
+ * @param[in] fs: Sampling frequency in [Hz], used for computing the
+ * downsampling factor and size of output arrays.
+ * @param[in] tau: Time constant of single pole low pass filter for squared data.
+ * Three pre-defined values can be used: TAU_FAST, for fast filtering,
+ * TAU_SLOW, for slow filtering, TAU_IMPULSE for impulse filtering
+ * @param[in] ref_level: Reference level when computing dB's. I.e. P_REF_AIR for
+ * sound pressure levels in air
+ * @param[out] downsampling_fac: Here, the used downsampling factor is stored
+ * which is used on returning data. If the value is for example 10, the
+ * 'sampling' frequency of output data from `Slm_run` is 4800 is fs is set to
+ * 48000. This downsampling factor is a function of the used time weighting.
+ *
+ * @return Slm: Handle of the sound level meter, NULL on error.
  */
-slm* slm_create(Sosfilterbank* fb);
+Slm* Slm_create(Sosfilterbank* weighting,Sosfilterbank* bandpass,
+                const d fs,
+                const d tau,
+                const d ref_level,
+                us* downsampling_fac);
 
 /**
- * Initialize the filter coeficients in the slm
+ * Run the sound level meter on a piece of time data.
  *
- * @param fb: slm handle
+ * @param[in] slm: Slm handle
- * @param filter_no: Filter number in the bank
+ * @param[in] input_data: Vector of input data samples.
- * @param coefss: Array of filter coefficients. Should have a length of 
- * nsections x 6, for each of the sections, it contains (b0, b1, b2, a0, 
- * a1, a2), where a are the numerator coefficients and b are the denominator 
- * coefficients. 
  *
-*/
+ * @return Output result of Sound Level values in [dB], for each bank in the filter
-void slm_setFilter(slm* fb,const us filter_no,
+ * bank.
-                             const vd coefs);
-
-/** 
- * Filters x using h, returns y
- *
- * @param x Input time sequence block. Should have at least one sample.
-
- * @return Filtered output in an allocated array. The number of
- * columns in this array equals the number of filters in the
- * slm. The number of output samples is equal to the number of
- * input samples in x.
  */
-dmat slm_filter(slm* fb,
+dmat Slm_run(Slm* slm,
-                          const vd* x);
+             vd* input_data);
-
 /** 
- * Cleans up an existing filter bank.
+ * Cleans up an existing Slm
  *
  * @param f slm handle
  */
-void slm_free(slm* f);
+void Slm_free(Slm* f);
 
 
-#endif // LASP_slm_H
+#endif // LASP_SLM_H
 //////////////////////////////////////////////////////////////////////

lasp/lasp_slm.py

@@ -4,7 +4,7 @@
 Sound level meter implementation
 @author: J.A. de Jong - ASCEE
 """
-from .wrappers import SPLowpass
+from .wrappers import Slm as pyxSlm
 import numpy as np
 from .lasp_common import (TimeWeighting, P_REF)
 
@@ -24,11 +24,11 @@ class SLM:
     """
     Multi-channel Sound Level Meter. Input data: time data with a certain
     sampling frequency. Output: time-weighted (fast/slow) sound pressure
-    levels in dB(A/C/Z).
+    levels in dB(A/C/Z). Possibly in octave bands.
 
     """
 
-    def __init__(self, fs, tw=TimeWeighting.default):
+    def __init__(self, fs, tw=TimeWeighting.default, ):
         """
         Initialize a sound level meter object.
         Number of channels comes from the weighcal object.

lasp/wrappers.pyx

@@ -455,6 +455,76 @@ cdef extern from "lasp_decimation.h":
     dmat Decimator_decimate(c_Decimator* dec,const dmat* samples) nogil
     void Decimator_free(c_Decimator* dec) nogil
 
+
+cdef extern from "lasp_slm.h":
+    ctypedef struct c_Slm "Slm"
+    d TAU_FAST, TAU_SLOW, TAU_IMPULSE
+    c_Slm* Slm_create(c_Sosfilterbank* prefilter,
+                    c_Sosfilterbank* bandpass,
+                    d fs, d tau, d ref_level,
+                    us* downsampling_fac)
+    dmat Slm_run(c_Slm* slm,vd* input_data)
+    void Slm_free(c_Slm* slm)
+
+
+tau_fast = TAU_FAST
+tau_slow = TAU_SLOW
+tau_impulse = TAU_IMPULSE
+
+cdef class Slm:
+    cdef:
+       c_Slm* c_slm 
+       us downsampling_fac
+
+    def __cinit__(self, d[:, ::1] sos_prefilter,
+                        d[:, ::1] sos_bandpass,
+                        d fs, d tau, d ref_level):
+
+        cdef:
+            us prefilter_nsections = sos_prefilter.shape[1] // 6
+            us bandpass_nsections = sos_bandpass.shape[1] // 6
+            us bandpass_nchannels = sos_bandpass.shape[0]
+            c_Sosfilterbank* prefilter = NULL
+            c_Sosfilterbank* bandpass = NULL
+        
+        if sos_prefilter is not None:
+            if sos_prefilter.shape[0] != 1:
+                raise ValueError('Prefilter should have only one channel')
+            prefilter = Sosfilterbank_create(1,prefilter_nsections)
+            if prefilter is NULL:
+                raise RuntimeError('Error creating pre-filter')
+
+        if sos_bandpass is not None:
+            bandpass =  Sosfilterbank_create(bandpass_nchannels,
+                                             bandpass_nsections)
+            if bandpass == NULL:
+                if prefilter:
+                    Sosfilterbank_free(prefilter)
+                raise RuntimeError('Error creating bandpass filter')
+
+        self.c_slm = Slm_create(prefilter, bandpass,
+                                fs, tau, ref_level,
+                                &self.downsampling_fac)
+        if self.c_slm is NULL:
+            Sosfilterbank_free(prefilter)
+            Sosfilterbank_free(bandpass)
+            raise RuntimeError('Error creating sound level meter')
+
+    def run(self, d[::1] data):
+        cdef vd data_vd = dmat_foreign_data(data.shape[0], 1,
+                                      &data[0], False)
+        cdef dmat res = Slm_run(self.c_slm, &data_vd)
+        result = dmat_to_ndarray(&res,True)
+        return result
+        
+
+    def __dealloc__(self):
+        if self.c_slm:
+            Slm_free(self.c_slm)
+
+
+
+
 cdef class Decimator:
     cdef:
         c_Decimator* dec