Merge branch 'develop' of ssh://code.ascee.nl:12001/ASCEE/lasp into develop

This commit is contained in:
Casper Jansen 2021-05-07 11:22:39 +02:00
commit fa6fbbe12d
4 changed files with 199 additions and 25 deletions

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@ -67,7 +67,7 @@ endif(LASP_FLOAT STREQUAL "double")
# ##################### END Cmake variables converted to a macro
set(Python_ADDITIONAL_VERSIONS "3.8 3.9")
set(Python_ADDITIONAL_VERSIONS "3.8")
# #################### Setting definitions and debug-specific compilation flags
# General make flags

176
lasp/filter/biquad.py Normal file
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@ -0,0 +1,176 @@
#!/usr/bin/env python3
# -*- coding: utf-8 -*-
"""!
Author: J.A. de Jong - ASCEE V.O.F.
Description: Filter design implementation of common biquad filters that are
often used in parametric equalizers.
Major source is Audio EQ Cookbook:
https://archive.is/20121220231853/http://www.musicdsp.org/
files/Audio-EQ-Cookbook.txt
The definition of the BiQuad filter coefficients as coming out of these
functions defines the filter as:
y[n] = 1/ba[3] * ( ba[0] * x[n] + ba[1] * x[n-1] + ba[2] * x[n-2] +
+ ba[4] * y[n-1] + ba[5] * y[n-2]
)
*Note that all filters are normalized such that ba[3] is by definition equal to
1.0!*
"""
__all__ = ['peaking', 'biquadTF', 'notch', 'lowpass', 'highpass',
'highshelve', 'lowshelve']
from numpy import array, cos, pi, sin, sqrt
from scipy.interpolate import interp1d
from scipy.signal import sosfreqz
def peaking(fs, f0, Q, gain):
"""
Design of peaking biquad filter
Args:
fs: Sampling frequency [Hz]
f0: Center frequency
Q: Quality factor (~ inverse of bandwidth)
gain: Increase in level at the center frequency [dB]
"""
A = sqrt(10**(gain/20))
omg0 = 2*pi*f0/fs
alpha = sin(omg0)/Q/2
b0 = 1+alpha*A
b1 = -2*cos(omg0)
b2 = 1-alpha*A
a0 = 1 + alpha/A
a1 = -2*cos(omg0)
a2 = 1-alpha/A
return array([b0/a0, b1/a0, b2/a0, a0/a0, a1/a0, a2/a0])
def notch(fs, f0, Q):
"""
Notch filter
Args:
fs: Sampling frequency [Hz]
f0: Center frequency [Hz]
Q: Quality factor (~ inverse of bandwidth)
"""
omg0 = 2*pi*f0/fs
alpha = sin(omg0)/Q/2
b0 = 1
b1 = -2*cos(omg0)
b2 = 1
a0 = 1 + alpha
a1 = -2*cos(omg0)
a2 = 1 - alpha
return array([b0/a0, b1/a0, b2/a0, a0/a0, a1/a0, a2/a0])
def lowpass(fs, f0, Q):
"""
Second order low pass filter
Args:
fs: Sampling frequency [Hz]
f0: Cut-off frequency [Hz]
Q: Quality factor (~ inverse of bandwidth)
"""
w0 = 2*pi*f0/fs
alpha = sin(w0)/Q/2
b0 = (1 - cos(w0))/2
b1 = 1 - cos(w0)
b2 = (1 - cos(w0))/2
a0 = 1 + alpha
a1 = -2*cos(w0)
a2 = 1 - alpha
return array([b0/a0, b1/a0, b2/a0, a0/a0, a1/a0, a2/a0])
def highpass(fs, f0, Q):
"""
Second order high pass filter
Args:
fs: Sampling frequency [Hz]
f0: Cut-on frequency [Hz]
Q: Quality factor (~ inverse of bandwidth)
"""
w0 = 2*pi*f0/fs
alpha = sin(w0)/Q/2
b0 = (1 + cos(w0))/2
b1 = -(1 + cos(w0))
b2 = (1 + cos(w0))/2
a0 = 1 + alpha
a1 = -2*cos(w0)
a2 = 1 - alpha
return array([b0/a0, b1/a0, b2/a0, a0/a0, a1/a0, a2/a0])
def highshelve(fs, f0, Q, gain):
"""
High shelving filter
Args:
fs: Sampling frequency [Hz]
f0: Cut-on frequency [Hz]
Q: Quality factor (~ inverse of bandwidth)
gain: Increase in level w.r.t. "wire" [dB]
"""
w0 = 2*pi*f0/fs
alpha = sin(w0)/Q/2
A = 10**(gain/40)
b0 = A*((A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha)
b1 = -2*A*((A-1) + (A+1)*cos(w0))
b2 = A*((A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha)
a0 = (A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha
a1 = 2*((A-1) - (A+1)*cos(w0))
a2 = (A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha
return array([b0/a0, b1/a0, b2/a0, a0/a0, a1/a0, a2/a0])
def lowshelve(fs, f0, Q, gain):
"""
Low shelving filter
Args:
fs: Sampling frequency [Hz]
f0: Cut-on frequency [Hz]
Q: Quality factor (~ inverse of bandwidth)
gain: Increase in level w.r.t. "wire" [dB]
"""
w0 = 2*pi*f0/fs
alpha = sin(w0)/Q/2
A = 10**(gain/40)
b0 = A*((A+1) - (A-1)*cos(w0) + 2*sqrt(A)*alpha)
b1 = 2*A*((A-1) - (A+1)*cos(w0))
b2 = A*((A+1) - (A-1)*cos(w0) - 2*sqrt(A)*alpha)
a0 = (A+1) + (A-1)*cos(w0) + 2*sqrt(A)*alpha
a1 = -2*((A-1) + (A+1)*cos(w0))
a2 = (A+1) + (A-1)*cos(w0) - 2*sqrt(A)*alpha
return array([b0/a0, b1/a0, b2/a0, a0/a0, a1/a0, a2/a0])
def biquadTF(fs, freq, ba):
"""
Computes the transfer function of the biquad.
Interpolates the frequency response to `freq`
Args:
fs: Sampling frequency [Hz]
freq: Frequency array to compute the
ba: Biquad filter coefficients in common form.
TODO: This code is not yet tested
"""
freq2, h = sosfreqz(ba, worN=freq, fs=fs)
interpolator = interp1d(freq2, h, kind='quadratic')
return interpolator(freq)

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@ -21,7 +21,6 @@ __all__ = ['AvStream']
video_x, video_y = 640, 480
class AvStream:
"""Audio and video data stream, to which callbacks can be added for
processing the data."""

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@ -96,6 +96,9 @@ class IterRawData:
Args:
f: Audio dataset in the h5 file, accessed as f['audio']
channels: list of channel indices to use
istart: index of first sample
istop: index of last sample (not including istop)
"""
assert isinstance(channels, list)
@ -173,22 +176,6 @@ class IterData(IterRawData):
return scaleBlockSens(nextraw, self.sens)
def exportAsWave(fn, fs, data, force=False):
if '.wav' not in fn[-4:]:
fn += '.wav'
nchannels = data.shape[1]
sampwidth = getSampWidth(data.dtype)
if os.path.exists(fn) and not force:
raise RuntimeError('File already exists: %s', fn)
with wave.open(fn, 'w') as wf:
wf.setparams((nchannels, sampwidth, fs, 0, 'NONE', 'NONE'))
wf.writeframes(np.asfortranarray(data).tobytes())
class Measurement:
"""Provides access to measurement data stored in the h5 measurement file
format."""
@ -593,7 +580,8 @@ class Measurement:
return False
def exportAsWave(self, fn=None, force=False, newsampwidth=None, normalize=True):
def exportAsWave(self, fn=None, force=False, newsampwidth=None,
normalize=True, **kwargs):
"""Export measurement file as wave. In case the measurement data is
stored as floats, the values are scaled to the proper integer (PCM)
data format.
@ -621,15 +609,25 @@ class Measurement:
if os.path.exists(fn) and not force:
raise RuntimeError(f'File already exists: {fn}')
data = self.rawData()
if not np.isclose(self.samplerate%1,0):
raise RuntimeError(f'Sample rates should be approximately integer for exporting to Wave to work')
# TODO: With VERY large measurment files, this is not possible! Is this
# a theoretical case?
data = self.rawData(**kwargs)
if np.issubdtype(data.dtype, np.floating) and newsampwidth is None:
raise ValueError('Newsampwidth parameter should be given for floating point raw data')
if normalize:
maxabs = np.max(np.abs(data), axis=0)
data /= maxabs[np.newaxis, :]
# Scale back to maximum of absolute value
maxabs = np.max(np.abs(data))
data /= maxabs
if newsampwidth is not None:
# Convert to floats, then to new sample width
data = scaleBlockSens(data, self.sensitivity**0)
sensone = np.ones_like(self.sensitivity)
data = scaleBlockSens(data, sensone)
if newsampwidth == 2:
newtype = np.int16
@ -638,11 +636,12 @@ class Measurement:
else:
raise ValueError('Invalid sample width, should be 2 or 4')
scalefac = 2**(8*(newsampwidth-1))-1
scalefac = 2**(8*newsampwidth-1)-1
# Scale data to integer range, and then convert to integers
data = (data*scalefac).astype(newtype)
wavfile.write(fn, self.samplerate, data)
wavfile.write(fn, int(self.samplerate), data)
@staticmethod
def fromtxt(fn,