This document discusses adaptive notch filtering algorithms for acoustic feedback control. It begins with an introduction to sound reinforcement systems and the acoustic feedback problem. When the forward path gain exceeds the maximum stable gain, howling can occur due to peaks in the frequency response of the room. Adaptive notch filtering is then presented as a solution, where notch filters are adapted using algorithms like the LMS to suppress problematic room modes. The document concludes by covering the ANF-LMS adaptive notch filtering algorithm and its application to acoustic feedback control.
NOISE CANCELLATION USING LMS ALGORITHM
OBJECTIVE
• INTRODUCTION
• ADAPTIVE FILTER
• BLOCK DIAGRAM
• LEAST MEAN SQUARE - LMS
• ADVANTAGES AND DISADVANTAGES
• MATLAB CODE
• CONCLUSION
ADAPTIVE NOISE CANCELLATION
➢ Adaptive noise cancellation is the approach used for estimating a desired
signal d(n) from a noise-corrupted observation.
x(n) = d(n) + v1(n)
➢ Usually the method uses a primary input containing the corrupted signal
and a reference input containing noise correlated in some unknown way
with the primary noise.
➢ The reference input v1(n) can be filtered and subtracted from the primary
input to obtain the signal estimate 𝑑 ̂(n).
➢ As the measurement system is a black box, no reference signal that is
correlated with the noise is available.
An adaptive filter is composed of two parts, the digital filter and the
adaptive algorithm.
• A digital filter with adjustable coefficients wn(z) and an adaptive algorithm
which is used to adjust or modify the coefficients of the filter.
• The adaptive filter can be a Finite Impulse Response FIR filter or an
Infinite Impulse Response IIR filter.
ALGORITHMS FOR ADAPTIVE EQUALIZATION
• There are three different types of adaptive filtering algorithms.
➢ Zero forcing (ZF)
➢ least mean square (LMS)
➢ Recursive least square filter (RLS)
• Recursive least square is an adaptive filter algorithm that recursively finds the coefficients
that minimize a weighted linear least squares cost function relating to the input signals.
• This approach is different from the least mean-square algorithm that aim to reduce the
mean-square error.
Least Mean Square - LMS
• The LMS algorithm in general, consists of two basics procedure:
1. Filtering process, which involve, computing the output (d(n - d)) of a linear filter in
response to the input signal and generating an estimation error by comparing this
output with a desired response as follows:
y(n) is filter output and is the desired response at time n
2. Adaptive process, which involves the automatics adjustment of the parameter of the
filter in accordance with the estimation error.
➢ where wn is the estimate of the weight value vector at time n, x(n) is the input
signal vector.
➢ e(n) is the filter error vector and μ is the step-size, which determines the filter
convergence rate and overall behavior.
➢ One of the difficulties in the design and implementation of the LMS adaptive
filter is the selection of the step-size μ. This parameter must lie in a specific
range, so that the LMS algorithm converges.
➢ LMS algorithm, aims to reduce the mean-square error.
The convergence characteristics of the LMS adaptive algorithm depends on two
factors: the step-size μ and the eigenvalue spread of the autocorrelation matrix .
The step-size μ must lie in a specific range
where 𝜆𝑚𝑎𝑥 is the largest eigenvalue of the autocorrelation matrix Rx.
• A large value of the step-size μ will lead to a faster convergence but may be less
stable around the minimum value. T
NOISE CANCELLATION USING LMS ALGORITHM
OBJECTIVE
• INTRODUCTION
• ADAPTIVE FILTER
• BLOCK DIAGRAM
• LEAST MEAN SQUARE - LMS
• ADVANTAGES AND DISADVANTAGES
• MATLAB CODE
• CONCLUSION
ADAPTIVE NOISE CANCELLATION
➢ Adaptive noise cancellation is the approach used for estimating a desired
signal d(n) from a noise-corrupted observation.
x(n) = d(n) + v1(n)
➢ Usually the method uses a primary input containing the corrupted signal
and a reference input containing noise correlated in some unknown way
with the primary noise.
➢ The reference input v1(n) can be filtered and subtracted from the primary
input to obtain the signal estimate 𝑑 ̂(n).
➢ As the measurement system is a black box, no reference signal that is
correlated with the noise is available.
An adaptive filter is composed of two parts, the digital filter and the
adaptive algorithm.
• A digital filter with adjustable coefficients wn(z) and an adaptive algorithm
which is used to adjust or modify the coefficients of the filter.
• The adaptive filter can be a Finite Impulse Response FIR filter or an
Infinite Impulse Response IIR filter.
ALGORITHMS FOR ADAPTIVE EQUALIZATION
• There are three different types of adaptive filtering algorithms.
➢ Zero forcing (ZF)
➢ least mean square (LMS)
➢ Recursive least square filter (RLS)
• Recursive least square is an adaptive filter algorithm that recursively finds the coefficients
that minimize a weighted linear least squares cost function relating to the input signals.
• This approach is different from the least mean-square algorithm that aim to reduce the
mean-square error.
Least Mean Square - LMS
• The LMS algorithm in general, consists of two basics procedure:
1. Filtering process, which involve, computing the output (d(n - d)) of a linear filter in
response to the input signal and generating an estimation error by comparing this
output with a desired response as follows:
y(n) is filter output and is the desired response at time n
2. Adaptive process, which involves the automatics adjustment of the parameter of the
filter in accordance with the estimation error.
➢ where wn is the estimate of the weight value vector at time n, x(n) is the input
signal vector.
➢ e(n) is the filter error vector and μ is the step-size, which determines the filter
convergence rate and overall behavior.
➢ One of the difficulties in the design and implementation of the LMS adaptive
filter is the selection of the step-size μ. This parameter must lie in a specific
range, so that the LMS algorithm converges.
➢ LMS algorithm, aims to reduce the mean-square error.
The convergence characteristics of the LMS adaptive algorithm depends on two
factors: the step-size μ and the eigenvalue spread of the autocorrelation matrix .
The step-size μ must lie in a specific range
where 𝜆𝑚𝑎𝑥 is the largest eigenvalue of the autocorrelation matrix Rx.
• A large value of the step-size μ will lead to a faster convergence but may be less
stable around the minimum value. T
Low power vlsi implementation adaptive noise cancellor based on least means s...shaik chand basha
We are trying to implement an adaptive filter with input weights. The adaptive parameters are obtained by simulating noise canceller on MATLAB. Simulink model of adaptive Noise canceller was developed and Processed by FPGA.
A Decisive Filtering Selection Approach For Improved Performance Active Noise...IOSR Journals
Abstract : In this work we present a filtering selection approach for efficient ANC system. Active noise cancellation (ANC) has wide application in next generation human machine interaction to automobile Heating Ventilating and Air Conditioning (HVAC) devices. We compare conventional adaptive filters algorithms LMS, NLMS, VSLMS, VSNLMS, VSLSMS for a predefined input sound file, where various algorithms run and result in standard output and better performance. The wiener filter based on least means squared (LMS) algorithm family is most sought after solution of ANC. This family includes LMS, NLMS, VSLMS, VSNLMS, VFXLMS, FX-sLMS and many more. Some of these are nonlinear algorithm, which provides better solution for nonlinear noisy environment. The components of the ANC systems like microphones and loudspeaker exhibit nonlinearities themselves. The nonlinear transfer function create worse situation. This is a task which is some sort of a prediction of suitable solution to the problems. The Radial Basis Function of Neural Networks (RBF NN) has been known to be suitable for nonlinear function approximation [1]. The classical approach to RBF implementation is to fix the number of hidden neurons based on some property of the input data, and estimate the weights connecting the hidden and output neurons using linear least square method. So an efficient novel decisive approach for better performing ANC algorithms has been proposed. Keywords - Adaptive filters, Winner filter ANC, Least mean square, N LMS, VSNLMS, RBF.
Fpga implementation of optimal step size nlms algorithm and its performance a...eSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Performance of MMSE Denoise Signal Using LS-MMSE TechniqueIJMER
This paper presents performance of mmse denoises signal using consistent cycle spinning
(ccs) and least square (LS) techniques. In the past decade, TV denoise technique is used to reduced the
noisy signal. The main drawback is the low quality signal and high MMSE signal. Presently, we
proposed the CCS-MMSE and LS-MMSE technique .The CCS-MMSE technique consists of two steps.
They are wavelet based denoise and consistent cycle spinning. The wavelet denoise is powerful
decorrelating effect on many signal domains. The consistent cycle spinning is used to estimation the
MMSE in the signal domain. The LS-MMSE is better estimation of MMSE signal domain compare to
CCS-MMSE.The experimental result shows the average MMSE reduction using various techniques.
Fpga implementation of optimal step size nlms algorithm and its performance a...eSAT Journals
Abstract The Normalized Least Mean Square error (NLMS) algorithm is most popular due to its simplicity. The conflicts of fast convergence and low excess mean square error associated with a fixed step size NLMS are solved by using an optimal step size NLMS algorithm. The main objective of this paper is to derive a new nonparametric algorithm to control the step size and also the theoretical performance analysis of the steady state behavior is presented in the paper. The simulation experiments are performed in Matlab. The simulation results show that the proposed algorithm as superior performance in Fast convergence rate, low error rate, and has superior performance in noise cancellation. Index Terms: Least Mean square algorithm (LMS), Normalized least mean square algorithm (NLMS)
2013.06.18 Time Series Analysis Workshop ..Applications in Physiology, Climat...NUI Galway
Professor Dimitris Kugiumtzis, Aristotle University of Thessaloniki, Greece, presented this workshop on nonlinear analysis of time series as part of the Summer School on Modern Statisitical Analysis and Computational Methods hosted by the Social Sciences Compuing Hub at the Whitaker Institute, NUI Galway on 17th-19th June 2013.
Low power vlsi implementation adaptive noise cancellor based on least means s...shaik chand basha
We are trying to implement an adaptive filter with input weights. The adaptive parameters are obtained by simulating noise canceller on MATLAB. Simulink model of adaptive Noise canceller was developed and Processed by FPGA.
A Decisive Filtering Selection Approach For Improved Performance Active Noise...IOSR Journals
Abstract : In this work we present a filtering selection approach for efficient ANC system. Active noise cancellation (ANC) has wide application in next generation human machine interaction to automobile Heating Ventilating and Air Conditioning (HVAC) devices. We compare conventional adaptive filters algorithms LMS, NLMS, VSLMS, VSNLMS, VSLSMS for a predefined input sound file, where various algorithms run and result in standard output and better performance. The wiener filter based on least means squared (LMS) algorithm family is most sought after solution of ANC. This family includes LMS, NLMS, VSLMS, VSNLMS, VFXLMS, FX-sLMS and many more. Some of these are nonlinear algorithm, which provides better solution for nonlinear noisy environment. The components of the ANC systems like microphones and loudspeaker exhibit nonlinearities themselves. The nonlinear transfer function create worse situation. This is a task which is some sort of a prediction of suitable solution to the problems. The Radial Basis Function of Neural Networks (RBF NN) has been known to be suitable for nonlinear function approximation [1]. The classical approach to RBF implementation is to fix the number of hidden neurons based on some property of the input data, and estimate the weights connecting the hidden and output neurons using linear least square method. So an efficient novel decisive approach for better performing ANC algorithms has been proposed. Keywords - Adaptive filters, Winner filter ANC, Least mean square, N LMS, VSNLMS, RBF.
Fpga implementation of optimal step size nlms algorithm and its performance a...eSAT Publishing House
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
Performance of MMSE Denoise Signal Using LS-MMSE TechniqueIJMER
This paper presents performance of mmse denoises signal using consistent cycle spinning
(ccs) and least square (LS) techniques. In the past decade, TV denoise technique is used to reduced the
noisy signal. The main drawback is the low quality signal and high MMSE signal. Presently, we
proposed the CCS-MMSE and LS-MMSE technique .The CCS-MMSE technique consists of two steps.
They are wavelet based denoise and consistent cycle spinning. The wavelet denoise is powerful
decorrelating effect on many signal domains. The consistent cycle spinning is used to estimation the
MMSE in the signal domain. The LS-MMSE is better estimation of MMSE signal domain compare to
CCS-MMSE.The experimental result shows the average MMSE reduction using various techniques.
Fpga implementation of optimal step size nlms algorithm and its performance a...eSAT Journals
Abstract The Normalized Least Mean Square error (NLMS) algorithm is most popular due to its simplicity. The conflicts of fast convergence and low excess mean square error associated with a fixed step size NLMS are solved by using an optimal step size NLMS algorithm. The main objective of this paper is to derive a new nonparametric algorithm to control the step size and also the theoretical performance analysis of the steady state behavior is presented in the paper. The simulation experiments are performed in Matlab. The simulation results show that the proposed algorithm as superior performance in Fast convergence rate, low error rate, and has superior performance in noise cancellation. Index Terms: Least Mean square algorithm (LMS), Normalized least mean square algorithm (NLMS)
2013.06.18 Time Series Analysis Workshop ..Applications in Physiology, Climat...NUI Galway
Professor Dimitris Kugiumtzis, Aristotle University of Thessaloniki, Greece, presented this workshop on nonlinear analysis of time series as part of the Summer School on Modern Statisitical Analysis and Computational Methods hosted by the Social Sciences Compuing Hub at the Whitaker Institute, NUI Galway on 17th-19th June 2013.
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About
Indigenized remote control interface card suitable for MAFI system CCR equipment. Compatible for IDM8000 CCR. Backplane mounted serial and TCP/Ethernet communication module for CCR remote access. IDM 8000 CCR remote control on serial and TCP protocol.
• Remote control: Parallel or serial interface.
• Compatible with MAFI CCR system.
• Compatible with IDM8000 CCR.
• Compatible with Backplane mount serial communication.
• Compatible with commercial and Defence aviation CCR system.
• Remote control system for accessing CCR and allied system over serial or TCP.
• Indigenized local Support/presence in India.
• Easy in configuration using DIP switches.
Technical Specifications
Indigenized remote control interface card suitable for MAFI system CCR equipment. Compatible for IDM8000 CCR. Backplane mounted serial and TCP/Ethernet communication module for CCR remote access. IDM 8000 CCR remote control on serial and TCP protocol.
Key Features
Indigenized remote control interface card suitable for MAFI system CCR equipment. Compatible for IDM8000 CCR. Backplane mounted serial and TCP/Ethernet communication module for CCR remote access. IDM 8000 CCR remote control on serial and TCP protocol.
• Remote control: Parallel or serial interface
• Compatible with MAFI CCR system
• Copatiable with IDM8000 CCR
• Compatible with Backplane mount serial communication.
• Compatible with commercial and Defence aviation CCR system.
• Remote control system for accessing CCR and allied system over serial or TCP.
• Indigenized local Support/presence in India.
Application
• Remote control: Parallel or serial interface.
• Compatible with MAFI CCR system.
• Compatible with IDM8000 CCR.
• Compatible with Backplane mount serial communication.
• Compatible with commercial and Defence aviation CCR system.
• Remote control system for accessing CCR and allied system over serial or TCP.
• Indigenized local Support/presence in India.
• Easy in configuration using DIP switches.
1. Digital Signal Processing 2
Les 4: Adaptieve filtering
Prof. dr. ir. Toon van Waterschoot
Faculteit Industriële Ingenieurswetenschappen
ESAT – Departement Elektrotechniek
KU Leuven, Belgium
2. Digital Signal Processing 2: Vakinhoud
• Les 1: Eindige woordlengte
• Les 2: Lineaire predictie
• Les 3: Optimale filtering
• Les 4: Adaptieve filtering
• Les 5: Detectieproblemen
• Les 6: Spectrale signaalanalyse
• Les 7: Schattingsproblemen 1
• Les 8: Schattingsproblemen 2
• Les 9: Sigma-Deltamodulatie
• Les 10: Transformatiecodering
3. Les 4: Adaptieve filtering
• Linear adaptive filtering algorithms
RLS, steepest descent, LMS, …
• Case study: Adaptive notch filters for acoustic
feedback control
acoustic feedback problem, acoustic feedback control, adaptive
notch filters, ANF-LMS algorithm …
4. Les 4: Adaptieve filtering
• Linear adaptive filtering algorithms
S. V. Vaseghi, Multimedia Signal Processing
- Ch. 9, “Adaptive Filters: Kalman, RLS, LMS”
• Section 9.3, “Sample Adaptive Filters”
• Section 9.4, “RLS Adaptive Filters”
• Section 9.5, “The Steepest-Descent Method”
• Section 9.6, “LMS Filter”
• Case study: Adaptive notch filters for acoustic
feedback control
Course notes:
T. van Waterschoot, “Adaptive notch filters for acoustic feedback
control”, Course Notes Digital Signal Processing-2, KU Leuven,
Faculty of Engineering Technology, Dept. ESAT, Oct. 2014.
5. Les 4: Adaptieve filtering
• Linear adaptive filtering algorithms
RLS, steepest descent, LMS, …
• Case study: Adaptive notch filters for acoustic
feedback control
acoustic feedback problem, acoustic feedback control, adaptive
notch filters, ANF-LMS algorithm …
6. Linear adaptive filtering algorithms
• Adaptive filtering concept
• Recursive Least Squares (RLS) adaptive filters
• Steepest Descent method
• Least Mean Squares (LMS) adaptive filters
• Computational complexity
7. Adaptive filtering concept (1)
• FIR adaptive filter
- signal flow graph
of the algorithm to initial parameter states.
Optimisation criteria: In this chapter two optimality criteria are used. One is based on the minim
ation of the mean of squared error (used in LMS, RLS and Kalman filter) and the other is bas
on constrained minimisation of the norm of the incremental change in the filter coefficients wh
esults in normalised LMS (NLMS). In Chapter 12, adaptive filters with non-linear object
unctions are considered for independent component analysis.
Adaptation
algorithm
“Desired” or “target”
signal x(m)
Input y(m)
z–1 . . .
y(m – 1) y(m – 2) y(m – P – 1)
x(m)
ˆ
w0 w1
Transversal filter
e(m)
z–1 z–1
w2
wP – 1
this part is different
from Wiener filter
8. Adaptive filtering concept (2)
• Adaptive filtering concept
- adaptive filter = time-varying optimal filter
- filter coefficients are updated whenever new input/desired
signal sample (or block of samples) is provided
- general updating scheme:
optimal filter (time t) = optimal filter (time t-1) + adaptation gain * error
• Design choices:
- FIR/IIR structure
- filter order
- cost function
- adaptation algorithm
9. Linear adaptive filtering algorithms
• Adaptive filtering concept
• Recursive Least Squares (RLS) adaptive filters
• Steepest Descent method
• Least Mean Squares (LMS) adaptive filters
• Computational complexity
10. Recursive Least Squares (RLS) algorithm (1)
• Online Wiener/LS filter implementation
- starting point: Wiener filter or least squares estimate
- how can we implement this filter in online applications?
• at time m, only data {x(0),y(0),…,x(m),y(m)} are available
• optimal filter coefficients w might be time-varying
w = R̂ 1
yy r̂yx = (YT
Y) 1
YT
x
w =
2
6
6
6
4
w0
w1
.
.
.
wP 1
3
7
7
7
5
, Y =
2
6
6
6
4
yT
(0)
yT
(1)
.
.
.
yT
(N 1)
3
7
7
7
5
, y(m) =
2
6
6
6
4
y(m)
y(m 1)
.
.
.
y(m P + 1)
3
7
7
7
5
, x =
2
6
6
6
4
x(0)
x(1)
.
.
.
x(N 1)
3
7
7
7
5
11. Recursive Least Squares (RLS) algorithm (2)
• Recursive time update of correlation matrix/vector
- consider the LS estimate at time m:
- the correlation matrix/vector can be computed recursively as
- if optimal filter w is time-varying, use “forgetting” mechanism:
w(m) = R̂ 1
yy (m)r̂yx(m)
R̂yy(m) =
m
X
n=0
y(n)yT
(n) = R̂yy(m 1) + y(m)yT
(m)
r̂yx(m) =
m
X
n=0
y(n)x(n) = r̂yx(m 1) + y(m)x(m)
r̂yx(m) =
m
X
n=0
m n
y(n)x(n) = r̂yx(m 1) + y(m)x(m)
R̂yy(m) =
m
X
n=0
m n
y(n)yT
(n) = R̂yy(m 1) + y(m)yT
(m)
0 ⌧ < 1
12. Recursive Least Squares (RLS) algorithm (3)
• Matrix inversion lemma (MIL)
- since autocorrelation matrix needs to be inverted at each
time m, recursive computation of inverse matrix is desired
- define inverse autocorrelation matrix
- Matrix inversion lemma:
with gain vector
yy(m) = R̂ 1
yy (m)
k(m) =
1
yy(m 1)y(m)
1 + 1yT (m) yy(m 1)y(m)
R̂yy(m) = R̂yy(m 1) + y(m)yT
(m)
m
yy(m) = 1
yy(m 1) 1
k(m)yT
(m) yy(m 1)
13. Recursive Least Squares (RLS) algorithm (4)
• Recursive time update of filter coefficients
- plug recursive update for into optimal filter solution:
- replacing inverse autocorrelation matrix by
results in a recursive time update of filter coefficients
r̂yx(m)
w(m) = yy(m)[ ryx(m 1) + y(m)x(m)]
yy(m) = 1
yy(m 1) 1
k(m)yT
(m) yy(m 1)
w(m) = yy(m 1)ryx(m 1)
| {z }
w(m 1)
+k(m) [x(m) yT
(m)w(m 1)]
| {z }
e(m)
27. Introduction (1): Sound reinforcement (1)
Goal: to deliver sufficiently high sound level
and best possible sound quality to audience
• sound sources
• microphones
• mixer & amp
• loudspeakers
• monitors
• room
• audience
28. • We will restrict ourselves to the single-channel case
(= single loudspeaker, single microphone)
Introduction (2): Sound reinforcement (2)
29. Introduction (3): Sound reinforcement (3)
• Assumptions:
– loudspeaker has linear & flat response
– microphone has linear & flat response
– forward path (amp) has linear & flat response
– acoustic feedback path has linear response
• But: acoustic feedback path has non-flat response
30. • Acoustic feedback path response: example room (36 m3)
impulse response frequency magnitude response
Introduction (4): Sound reinforcement (4)
direct
coupling
early
reflections
diffuse
sound field
peaks/dips = anti-nodes/nodes of standing waves
peaks ~10 dB above average, and separated by ~10 Hz
31. • “Desired” system transfer function:
• Closed-loop system transfer function:
– spectral coloration
– acoustic echoes
– risk of instability
• “Loop response”:
– loop gain
– loop phase
Introduction (5): Acoustic feedback (1)
U(z)
V (z)
= G(z)
U(z)
V (z)
=
G(z)
1 G(z)F(z)
|G(ei!
)F(ei!
)|
G(ei!
)F(ei!
)
32. • Nyquist stability criterion:
– if there exists a radial frequency ω for which
then the closed-loop system is unstable
– if the unstable system is excited at the critical frequency ω,
then an oscillation at this frequency will occur = howling
• Maximum stable gain (MSG):
– maximum forward path gain before instability
– primarily determined by peaks in frequency magnitude
response of the room
– 2-3 dB gain margin is desirable to avoid ringing
Introduction (6): Acoustic feedback (2)
⇢
|G(ei!
)F(ei!
)| 1
G(ei!
)F(ei!
) = n2⇡, n 2 Z
|F(ei!
)|
33. • Example of closed-loop system instability:
loop gain loudspeaker spectrogram
Introduction (7): Acoustic feedback (3)
|G(ei!
)F(ei!
)| U(ei!
, t)
34. Case study: Adaptive notch filters
• Introduction
• Acoustic feedback control
• Adaptive notch filtering
• ANF-LMS algorithm
35. Acoustic feedback control (1)
• Goal of acoustic feedback control
= to solve the acoustic feedback problem
– either completely (to remove acoustic coupling)
– or partially (to remove howling from loudspeaker signal)
• Manual acoustic feedback control:
– proper microphone/loudspeaker selection & positioning
– a priori room equalization using 1/3 octave graphic EQ filters
– ad-hoc discrete room modes suppression using notch filters
• Automatic acoustic feedback control:
– no intervention of sound engineer required
– different approaches can be classified into four categories
36. Acoustic feedback control (2)
1. phase modulation (PM) methods
– smoothing of “loop gain” (= closed-loop magnitude response)
– phase/frequency/delay modulation, frequency shifting
– well suited for reverberation enhancement systems (low gain)
2. spatial filtering methods
– (adaptive) microphone beamforming for reducing direct coupling
3. gain reduction methods
– (frequency-dependent) gain reduction after howling detection
– most popular method for sound reinforcement applications
4. room modeling methods
– adaptive inverse filtering (AIF): adaptive equalization of acoustic
feedback path response
– adaptive feedback cancellation (AFC): adaptive prediction and
subtraction of feedback (≠howling) component in microphone signal
37. Case study: Adaptive notch filters
• Introduction
• Acoustic feedback control
• Adaptive notch filtering
• ANF-LMS algorithm
38. Adaptive notch filtering (1)
• Gain reduction methods
– automation of the actions a sound engineer would undertake
• Classification of gain reduction methods
– automatic gain control (full-band gain reduction)
– automatic equalization (1/3 octave bandstop filters)
– notch filtering (NF) (1/10-1/60 octave filters)
gain margin
G’(z)F(z)
f
Winst in stabiele versterking
door notch filter
Notch−filterkarakteristiek
gain margin
G(z)F(z)
H(z)
f
f
Notch filter characteristic
Increase of stable gain due to notch filter
G F
x(t)
v(t)
y(t)
e(t)
u(t)
Notch Filter
39. Adaptive notch filtering (2)
• Adaptive notch filter
– filter that automatically finds & removes narrowband signals
– based on constrained second-order pole-zero filter structure
– constraint 1: poles and zeros lie on same radial lines
−1 −0.5 0 0.5 1
−1
−0.8
−0.6
−0.4
−0.2
0
0.2
0.4
0.6
0.8
1
Real Part
Imaginary
Part
α ri
r
i
Bode Diagram
Frequency (rad/sec)
Phase
(deg)
Magnitude
(dB)
−15
−10
−5
0
5
10
3
10
4
−60
−30
0
30
60
Figure 3: Pole-zero map (left) and Bode plot (right) of an 2nd
order ANF with
eros z = r e±j!i
and poles p = ↵r e±j!i
−1 −0.5
−1
−0.8
−0.6
−0.4
−0.2
0
0.2
0.4
0.6
Imaginary
Part
Figure 3: Pole-zero
zeros zi = rie±j!i
an
3.1 Adaptive
3.1.1 Overview
The ANF filter s
conceived by Rao et
signals buried in bro
−1 −0.5 0 0.5 1
−1
−0.8
−0.6
−0.4
−0.2
0
0.2
0.4
0.6
0.8
1
Real Part
Imaginary
Part
α ri
ri
Phase
(deg)
Magnitude
(dB)
−15
−10
−5
0
10
−60
−30
0
30
60
Figure 3: Pole-zero map (left) and Bode pl
zeros zi = rie±j!i
and poles pi = ↵rie±j!i
3.1 Adaptive Notch Filter (AN
3.1.1 Overview of the literature
The ANF filter structure. The Adap
conceived by Rao et al. [15] as a means for
40. Adaptive notch filtering (3)
• ANF transfer function
– cascade of constrained second-order pole-zero filters:
– constraint 2: zeros are forced to lie on unit circle
– “pole radius” ρ = “debiasing parameter” α
ANF transfer function in the z-domain looks like
H(z 1
) =
2n
Y
i=1
(1 ziz 1
)
2n
Y
i=1
(1 ↵ziz 1
)
where 0 ↵ < 1 (3)
=
1 + a1z 1
+ a2z 2
+ ... + a2nz 2n
1 + ↵a1z 1 + ↵2a2z 2 + ... + ↵2na2nz 2n
(4)
=
A(z 1
)
A(↵z 1)
(5)
With this structure a filter of order 2n has 2n unknown parameters and may
suppress at most n narrow-band components. A Bode plot of a 2nd
order ANF
is shown on the right in Figure 3.
Nehorai [17] proposed an adaptive notch filter with half as much parameters
by imposing a second constraint: the zeros zi should lie on the unit circle. A
necessary condition to meet this constraint is that the numerator coefficients
1
2n
Y
i=1
(1 ↵ziz 1
)
=
1 + a1z 1
+ a2z 2
+ ... + a2nz 2n
1 + ↵a1z 1 + ↵2a2z 2 + ... + ↵2na2nz 2n
(4)
=
A(z 1
)
A(↵z 1)
(5)
With this structure a filter of order 2n has 2n unknown parameters and may
suppress at most n narrow-band components. A Bode plot of a 2nd
order ANF
is shown on the right in Figure 3.
Nehorai [17] proposed an adaptive notch filter with half as much parameters
by imposing a second constraint: the zeros zi should lie on the unit circle. A
necessary condition to meet this constraint is that the numerator coefficients
have a mirror symmetric form (i.e. when zi is a zero, 1
zi
will also be a zero). A
2nth
order ANF with n unknown parameters thus has a transfer function
H(z 1
) =
1 + a1z 1
+ ... + anz n
+ ... + a1z 2n+1
+ z 2n
1 + ⇢a1z 1 + ... + ⇢nanz n + ... + ⇢2n 1a1z 2n+1 + ⇢2nz 2n
(6)
=
A(z 1
)
A(⇢z 1)
(7)
where the debiasing parameter ↵ has been replaced by the “pole radius ⇢”.
Estimating the filter coefficients. Including an adaptive notch filter in the
41. Adaptive notch filtering (4)
• ANF coefficient estimation
– coefficient vector
– least squares (LS) cost function:
=
A(z )
A(⇢z 1)
(7)
the debiasing parameter ↵ has been replaced by the “pole radius ⇢”.
ating the filter coefficients. Including an adaptive notch filter in the
d path of the 1-microphone/1-loudspeaker setup results in the scheme
ed in Figure 4. An estimate of parameter vector ✓ = [a1 a2 ... an]T
is
ed by minimizing the cost function VN (✓):
ˆ
✓ = arg min
✓
VN (✓) (8)
= arg min
✓
N
X
t=1
e2
(✓, t) (9)
= arg min
✓
N
X
t=1
A(✓, z 1
)
A(✓, ⇢z 1)
y(t)
2
(10)
the ANF order is chosen approximately twice the number of expected
w-band components, minimizing the square of the filter output e(✓, t) with
er stucture as defined in (6) results in a filter with n notches at the desired
ncies. As for the bandwidth of the notches, the pole radius ⇢ plays a
role: the closer ⇢ is to 1, the narrower the notches will be. Choosing ⇢
ter ↵ has been replaced by the “pole radius ⇢”.
fficients. Including an adaptive notch filter in the
ophone/1-loudspeaker setup results in the scheme
estimate of parameter vector ✓ = [a1 a2 ... an]T
is
cost function VN (✓):
arg min
✓
VN (✓) (8)
arg min
✓
N
X
t=1
e2
(✓, t) (9)
arg min
✓
N
X
t=1
A(✓, z 1
)
A(✓, ⇢z 1)
y(t)
2
(10)
osen approximately twice the number of expected
inimizing the square of the filter output e(✓, t) with
n (6) results in a filter with n notches at the desired
dwidth of the notches, the pole radius ⇢ plays a
42. Case study: Adaptive notch filters
• Introduction
• Acoustic feedback control
• Adaptive notch filtering
• ANF-LMS algorithm
43. ANF-LMS algorithm (1)
• 2nd order constrained ANF implementation
– Direct-Form II implementation of second-order ANF
A 2nd
order ANF of the type described above was applied to hearing aids
y Kates [20], only to detect oscillations due to acoustic feedback. Later on,
Maxwell et al. [21] employed Kates’ algorithm to suppress feedback oscillations
or comparison with adaptive feedback cancellation techniques. Their imple-
mentation in Direct Form II follows directly from the ANF tranfer function
6):
x(t) = y(t) + ⇢(t)a(t 1)x(t 1) ⇢2
(t)x(t 2) (12)
e(t) = x(t) a(t 1)x(t 1) + x(t 2) (13)
here y(t) and e(t) represent the ANF input resp. output as before, x(t) is
ntroduced as an auxiliary variable and some signs have been changed. This
nd
order ANF has only one parameter a(t) that appears in both transfer func-
on numerator and denominator. Instead of solving the nonlinear minimization
roblem (10) the filter coefficient update is done in an approximate way as sug-
ested by Travassos-Romano et al. [19]. Only the FIR portion of the filter is
dapted to track the frequency of the narrow-band components and the coef-
cients are then copied to the IIR portion of the filter. The filter coefficient
7
È
È
z 1
z 1
È
È
y(t) x(t)
x(t 1)
x(t 2)
e(t)
⇢(t)a(t 1)
⇢2
(t)
a(t 1)
44. ANF-LMS algorithm (2)
• ANF filter coefficient update
– adaptation strategy: only FIR portion of filter is adapted,
coefficients are then copied to IIR portion of filter
– LMS filter coefficient update:
y Kates [20], only to detect oscillations due to acoustic feedback. Later on,
Maxwell et al. [21] employed Kates’ algorithm to suppress feedback oscillations
or comparison with adaptive feedback cancellation techniques. Their imple-
mentation in Direct Form II follows directly from the ANF tranfer function
6):
x(t) = y(t) + ⇢(t)a(t 1)x(t 1) ⇢2
(t)x(t 2) (12)
e(t) = x(t) a(t 1)x(t 1) + x(t 2) (13)
here y(t) and e(t) represent the ANF input resp. output as before, x(t) is
ntroduced as an auxiliary variable and some signs have been changed. This
nd
order ANF has only one parameter a(t) that appears in both transfer func-
on numerator and denominator. Instead of solving the nonlinear minimization
roblem (10) the filter coefficient update is done in an approximate way as sug-
ested by Travassos-Romano et al. [19]. Only the FIR portion of the filter is
dapted to track the frequency of the narrow-band components and the coef-
cients are then copied to the IIR portion of the filter. The filter coefficient
7
ate can thus be calculated as follows:
aupd(t) = arg min
a
(e2
(t)) (14)
= arg
✓
d
da
e2
(t) = 0
◆
(15)
= arg
✓
2e(t)
d
da
e(t) = 0
◆
(16)
= arg
⇣
2e(t)( x(t 1)) = 0
⌘
(17)
re the last equality follows from (13). In this way we obtain an LMS filter
45. ANF-LMS algorithm (2)
• 2nd order ANF-LMS algorithm
= arg 2e(t)( x(t 1)) = 0 (17)
where the last equality follows from (13). In this way we obtain an LMS filter
update which completes the filter implementation given by (12)-(13):
a(t) = a(t 1) + 2µe(t)x(t 1) (18)
The 2nd
order ANF-LMS algorithm is summarized in Algorithm 1.
Algorithm 1 : 2nd
order ANF-LMS algorithm
Input step size µ, initial pole radius ⇢(0), final pole radius ⇢(1), expo-
nential decay time constant , input data {y(t)}N
t=1, initial conditions
x(0), x( 1), a(0)
Output 2nd
order ANF parameter {a(t)}N
t=1
1: for t = 1, . . . , N do
2: ⇢(t) = ⇢(t 1) + (1 )⇢(1)
3: x(t) = y(t) + ⇢(t)a(t 1)x(t 1) ⇢2
(t)x(t 2)
4: e(t) = x(t) a(t 1)x(t 1) + x(t 2)
5: a(t) = a(t 1) + 2µe(t)x(t 1)
6: end for
Higher order ANF’s can be implemented in a similar way. As an example
we give the di↵erence equations describing an 8th
order ANF with LMS update:
x(t) = y(t) + ⇢a1(t 1)x(t 1) ⇢2
a2(t 1)x(t 2) + ⇢3
a3(t 1)x(t 3)