Update SLM API. Now works properly with different start and stop bands, and only overall band

This commit is contained in:
Anne de Jong 2020-01-22 21:11:20 +01:00
parent 685329d307
commit 22f1cb59ff
3 changed files with 215 additions and 191 deletions

View File

@ -4,206 +4,207 @@
#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.
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);
const d tau, const d ref_level, us *downsampling_fac) {
fsTRACE(15);
assertvalidptr(downsampling_fac);
Slm *slm = NULL;
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'.
if (tau > 0) {
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;
} else {
*downsampling_fac = 1;
slm->downsampling_fac = 1;
}
/// Create the single pole lowpass
us filterbank_size;
if (bandpass) {
filterbank_size = Sosfilterbank_getFilterbankSize(bandpass);
} else {
filterbank_size = 1;
}
if (tau > 0) {
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);
Slm *slm = NULL;
if (ref_level <= 0) {
WARN("Invalid reference level");
return NULL;
} else if (fs <= 0) {
WARN("Invalid sampling frequency");
return NULL;
}
vd_free(&lowpass_sos);
} else {
/// No low-pass filtering. Tau set to zero
slm->splowpass = NULL;
}
feTRACE(15);
return slm;
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'.
us ds_fac;
if (tau > 0) {
const d fs_slm = 1 / (2 * number_pi * tau) * (1 - 0.01) / 0.01;
dVARTRACE(15, fs_slm);
ds_fac = (us) (fs / fs_slm);
// If we get 0, it should be 1
if(ds_fac == 0) ds_fac++;
} else {
ds_fac = 1;
}
slm->downsampling_fac = ds_fac;
*downsampling_fac = ds_fac;
slm->cur_offset = 0;
/// Create the single pole lowpass
us filterbank_size;
if (bandpass) {
filterbank_size = Sosfilterbank_getFilterbankSize(bandpass);
} else {
filterbank_size = 1;
}
if (tau > 0) {
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);
} else {
/// No low-pass filtering. Tau set to zero
slm->splowpass = NULL;
}
feTRACE(15);
return slm;
}
dmat Slm_run(Slm *slm, vd *input_data) {
fsTRACE(15);
assertvalidptr(slm);
assert_vx(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;
/// 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;
/// 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;
/// Pre-calculate the size of the output data
us downsampling_fac = slm->downsampling_fac;
us samples_bandpassed = bandpassed.n_rows;
iVARTRACE(15, samples_bandpassed);
iVARTRACE(15, downsampling_fac);
us cur_offset = slm->cur_offset;
/// Compute the number of samples output
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;
if (slm->splowpass) {
levels = dmat_alloc(nsamples_output, filterbank_size);
} else {
levels = dmat_alloc(samples_bandpassed, filterbank_size);
}
for (us ch = 0; ch < bandpassed.n_cols; ch++) {
iVARTRACE(15, ch);
vd chan = dmat_column(&bandpassed, ch);
/// Inplace squaring of the signal
for (us sample = 0; sample < bandpassed.n_rows; sample++) {
tmp = getdmatval(&bandpassed, sample, ch);
*tmp = *tmp * *tmp;
/// Compute the number of samples output
int nsamples_output = samples_bandpassed;
if(downsampling_fac > 1) {
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;
}
// Now that all data for the channel is squared, we can run it through
// the low-pass filter
if (slm->splowpass) {
cur_offset = slm->cur_offset;
iVARTRACE(15, nsamples_output);
iVARTRACE(15, cur_offset);
dmat levels = dmat_alloc(nsamples_output, filterbank_size);
/// 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");
for (us ch = 0; ch < bandpassed.n_cols; ch++) {
iVARTRACE(15, ch);
vd chan = dmat_column(&bandpassed, ch);
/// Inplace squaring of the signal
for (us sample = 0; sample < bandpassed.n_rows; sample++) {
tmp = getdmatval(&bandpassed, sample, ch);
*tmp = *tmp * *tmp;
}
/// Output resulting levels at a lower interval
us i = 0;
while (cur_offset < samples_bandpassed) {
iVARTRACE(10, i);
iVARTRACE(10, cur_offset);
/// Compute level
d level = 10 * d_log10(*getvdval(&power_filtered, cur_offset) /
ref_level / ref_level);
// Now that all data for the channel is squared, we can run it through
// the low-pass filter
cur_offset = slm->cur_offset;
*getdmatval(&levels, i++, ch) = level;
cur_offset = cur_offset + downsampling_fac;
}
iVARTRACE(15, cur_offset);
iVARTRACE(15, i);
dbgassert(i == (int) nsamples_output, "BUG");
/// Apply single-pole lowpass filter for current filterbank channel
TRACE(15, "Start filtering");
vd power_filtered;
if(slm->splowpass) {
power_filtered = Sosfilterbank_filter(slm->splowpass[ch], &chan);
} else {
power_filtered = dmat_foreign(&chan);
}
TRACE(15, "Filtering done");
dbgassert(chan.n_rows == power_filtered.n_rows, "BUG");
vd_free(&chan);
vd_free(&power_filtered);
/// Output resulting levels at a lower interval
us i = 0;
while (cur_offset < samples_bandpassed) {
iVARTRACE(10, i);
iVARTRACE(10, cur_offset);
/// Compute level
d 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 == (int) nsamples_output, "BUG");
vd_free(&power_filtered);
vd_free(&chan);
}
}
slm->cur_offset = cur_offset - samples_bandpassed;
slm->cur_offset = cur_offset - samples_bandpassed;
if (!slm->splowpass) {
/// Raw copy of to levels. Happens only when the low-pass filter does not
/// have to come into action.
dmat_copy(&levels, &bandpassed);
}
vd_free(&prefiltered);
dmat_free(&bandpassed);
feTRACE(15);
return levels;
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);
}
us filterbank_size;
if (slm->bandpass) {
filterbank_size = Sosfilterbank_getFilterbankSize(slm->bandpass);
Sosfilterbank_free(slm->bandpass);
} else {
filterbank_size = 1;
}
if (slm->splowpass) {
for (us ch = 0; ch < filterbank_size; ch++) {
Sosfilterbank_free(slm->splowpass[ch]);
fsTRACE(15);
assertvalidptr(slm);
if (slm->prefilter) {
Sosfilterbank_free(slm->prefilter);
}
a_free(slm->splowpass);
}
a_free(slm);
feTRACE(15);
us filterbank_size;
if (slm->bandpass) {
filterbank_size = Sosfilterbank_getFilterbankSize(slm->bandpass);
Sosfilterbank_free(slm->bandpass);
} else {
filterbank_size = 1;
}
if (slm->splowpass) {
for (us ch = 0; ch < filterbank_size; ch++) {
Sosfilterbank_free(slm->splowpass[ch]);
}
a_free(slm->splowpass);
}
a_free(slm);
feTRACE(15);
}

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@ -67,6 +67,9 @@ class FilterBankDesigner:
return bool(np.all(llim_full <= h_dB) and
np.all(ulim_full >= h_dB))
def band_limits(self, x, filter_class):
raise NotImplementedError()
def fm(self, x):
"""Returns the exact midband frequency of the bandpass filter.
@ -569,3 +572,4 @@ class ThirdOctaveBankDesigner(FilterBankDesigner):
else:
raise ValueError('Unimplemented sampling frequency for SOS'
'filter design')

View File

@ -30,32 +30,42 @@ class SLM:
"""
def __init__(self,
fs,
fbdesigner,
tw=TimeWeighting.fast,
fw=FreqWeighting.A,
xmin = None,
xmax = None,
include_overall=True):
"""
Initialize a sound level meter object.
Args:
fs: Sampling frequency of input data [Hz]
fbdesigner: FilterBankDesigner to use for creating the
(fractional) octave bank filters
(fractional) octave bank filters. Set this one to None to only do
overalls
fs: Sampling frequency [Hz]
tw: Time Weighting to apply
fw: Frequency weighting to apply
xmin: Filter designator of lowest band
xmax: Filter designator of highest band
include_overall: If true, a non-functioning filter is added which
is used to compute the overall level.
"""
self.fbdesigner = fbdesigner
self.xs = fbdesigner.xs[:]
if xmin is None:
xmin = fbdesigner.xs[0]
if xmax is None:
xmax = fbdesigner.xs[-1]
self.xs = list(range(xmin, xmax + 1))
nfilters = len(self.xs)
if include_overall: nfilters +=1
self.include_overall = include_overall
fs = fbdesigner.fs
spld = SPLFilterDesigner(fs)
if fw == FreqWeighting.A:
prefilter = spld.A_Sos_design().flatten()
@ -64,24 +74,33 @@ class SLM:
elif fw == FreqWeighting.Z:
prefilter = None
else:
raise ValueError('Not implemented prefilter')
raise ValueError(f'Not implemented prefilter {fw}')
# 'Probe' size of filter coefficients
self.nom_txt = []
sos0 = fbdesigner.createSOSFilter(self.xs[0]).flatten()
sos = np.empty((nfilters, sos0.size), dtype=float, order='C')
sos[0, :] = sos0
if fbdesigner is not None:
assert fbdesigner.fs == fs
sos0 = fbdesigner.createSOSFilter(self.xs[0]).flatten()
sos = np.empty((nfilters, sos0.size), dtype=float, order='C')
sos[0, :] = sos0
for i, x in enumerate(self.xs[1:]):
sos[i, :] = fbdesigner.createSOSFilter(x).flatten()
self.nom_txt.append(fbdesigner.nominal_txt(x))
for i, x in enumerate(self.xs[1:]):
sos[i, :] = fbdesigner.createSOSFilter(x).flatten()
self.nom_txt.append(fbdesigner.nominal_txt(x))
if include_overall:
# Create a unit impulse response filter, every third index equals
# 1, so b0 = 1 and a0 is 1 (by definition)
sos[-1,:] = 0
sos[-1,::3] = 1
if include_overall:
# Create a unit impulse response filter, every third index equals
# 1, so b0 = 1 and a0 is 1 (by definition)
sos[-1,:] = 0
sos[-1,::3] = 1
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.
assert include_overall == False
sos = None
self.nom_txt.append('overall')
self.slm = pyxSlm(prefilter, sos,
@ -142,7 +161,7 @@ class SLM:
output[self.nom_txt[i]] = {'t': t,
'data': levels[:, i],
'x': x}
if self.include_overall:
if self.include_overall and self.fbdesigner is not None:
output['overall'] = {'t': t, 'data': levels[:, i+1], 'x': 0}
return output