IEEE SIGNAL PROCESSING LETTERS, VOL. 26, NO.

12, DECEMBER 2019 1827

A Neural Network-Based Nonlinear Acoustic

Echo Canceller
Mhd Modar Halimeh , Student Member, IEEE, Christian Huemmer , and Walter Kellermann , Fellow, IEEE

Abstract—In this letter, we introduce a novel approach for non-

linear acoustic echo cancellation. The proposed approach uses
the principle of transfer learning to train a neural network that
approximates the nonlinear function responsible for the nonlinear
distortions and generalizes this network to different acoustic con-
ditions. The topology of the proposed network is inspired by the
conventional adaptive filtering approaches for nonlinear acoustic
echo cancellation. The network is trained to model the nonlinear
distortions using the conventional error backpropagation algo-
rithm. In deployment, and in order to account for any varia-
tion or discrepancy between training and deployment conditions,
only a subset of the network’s parameters is adapted using the
significance-aware elitist resampling particle filter. The proposed
approach is evaluated and verified using synthesized nonlinear Fig. 1. NLAEC scenario.
distortions and real nonlinear distortions recorded by a commercial
mobile phone.
continuously due to the time-varying nature of the acoustic prop-
Index Terms—Nonlinear Acoustic Echo Cancellation, Artificial
Neural Networks, Transfer Learning, Particle Filters.
agation path from the loudspeaker to the microphone [16]–[18].
Alternatively, one can focus on modeling only the nonlinear
loudspeaker signal distortions using a neural network, which is
I. INTRODUCTION challenged by the difficulty of applying inverse filtering tech-
niques for parameter estimation, see, e.g., [19].
HE task of eliminating the acoustic coupling between a
T loudspeaker and a microphone has received significant
attention for decades [1], [2]. Nowadays, this attention is still
In this letter, a neural network-based acoustic echo canceller
is proposed which uses a network topology that is inspired
by conventional NLAEC approaches. The network models the
driven by the increasing need for full-duplex human-machine in- entire nonlinear echo path and is first trained using the error
terfaces in our everyday life. At the same time, such interfaces are backpropagation algorithm. Since the network models the entire
increasingly challenged by nonlinear distortions in miniaturized echo path, which is expected to be time-variant, the principle of
devices such as mobile phones or smart loudspeakers [3], [4]. transfer learning [20] is employed by using the part of the trained
A conventional approach to nonlinear acoustic echo cancel- network corresponding to the time-invariant components of the
lation (NLAEC) is to account for loudspeaker signal distortions echo path and generalizing it to the diverse acoustic conditions
by using a set of nonlinear basis functions [5]–[7], e.g., monomi- expected in testing. During deployment, a subset of the network
als [8], [9], Legendre functions [10], Volterra kernels [11], [12], parameters which corresponds to the time-varying characteris-
functional link expansions [13], or by using kernel methods, tics in the echo path, is adapted online via the Significance-
e.g., as in [14], [15]. An interesting alternative to model the Aware Elitist Resampling Particle Filter (SA-ERPF) [21]–[23]
nonlinear distortions is the use of an artificial neural network in order to account for potential smooth variations in the echo
(ANN), where several concepts are distinguished [16]–[19]: For path.
one, one can either model the entire nonlinear echo path using a
neural network, resulting in a network that needs to be adapted
II. PROBLEM FORMULATION
As depicted in Fig. 1, at time instant n, the far-end signal
Manuscript received July 18, 2019; revised October 18, 2019; ac-
cepted October 27, 2019. Date of publication November 4, 2019; xn = [x[n], x[n − 1], . . ., x[n − Lc + 1]]T is first nonlinearly
date of current version November 20, 2019. This work was sup- distorted by the loudspeaker and possibly other nonlinear com-
ported by the Deutsche Forschungsgemeinschaft (DFG) under contract ponents, resulting in the distorted signal d[n]. The nonlinearly
KE 890/9-1. The associate editor coordinating the review of this manuscript distorted signal d[n] propagates through the acoustic enclo-
and approving it for publication was Prof. Woon-Seng Gan. (Corresponding
author: Mhd Modar Halimeh.)
sure, characterized by the Room Impulse Response (RIR) hn .
M. M. Halimeh and W. Kellermann are with the Chair of Multime- The resulting signal y[n] is captured by the microphone. A
dia Communications and Signal Processing (LMS), University of Erlangen- Wiener-Hammerstein (WH)-type model [24] is used to model
Nuremberg, 91058 Erlangen, Germany (e-mail: mhd.m.halimeh@fau.de; walter. this nonlinear system. The WH model is a cascade of two
kellermann@fau.de).
C. Huemmer is with Siemens Healthineers, 91052 Erlangen, Germany (e-
linear Finite Impulse Response (FIR) filters, ĉ and ĥ, with a
mail: huemmer@lnt.de). memoryless nonlinear preprocessor f (·) in between. The filter
Digital Object Identifier 10.1109/LSP.2019.2951311 ĉ = [ĉ0 , . . ., ĉLc −1 ]T , will be called the prefilter in the following,

1070-9908 © 2019 IEEE. Personal use is permitted, but republication/redistribution requires IEEE permission.
See http://www.ieee.org/publications_standards/publications/rights/index.html for more information.

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1828
Fig. 2. The proposed ANN architecture.
is of length Lc and it introduces memory into the nonlinear
model. The nonlinear signal d[n] is then estimated by
ˆ = f (ân , ĉT xn ),
d[n]
where ân is a parameter vector that characterizes the function
f (·). By using ĉ = 1, the WH model becomes equivalent to a
Hammerstein model for memoryless nonlinearities.
The nonlinear preprocessor f (·) consists of a group of M basis
functions weighted by the vector ân = [â1 , . . ., âM ]T
M
 tions during testing, i.e., deployment. Therefore, a representative
f (ân , ĉT xn ) = âi fi (ĉT xn ).
i=1
The echo signal y[n] is estimated by ŷ[n], which is obtained
ˆ with the FIR
by convolving the estimated distorted signal d[n]
filter ĥn = [ĥn,1 , . . ., ĥn,Lh ] T
ŷ[n] = ĥTn dˆn ,
ˆ d[n
where dˆn = [d[n], ˆ − 1], . . ., d[n
ˆ − Lh + 1]]T .
III. A NEURAL NETWORK-BASED NLAEC
In this section, we describe a neural network for mimicking the
WH echo path model with the expectation that the performance
of the model will benefit from the superior modelling capability
of neural networks while keeping the number of parameters
minimum.
A. Topology and Training of the Network
The WH model discussed earlier can be represented equiva-
lently using an ANN. In this model, the nonlinearly distorted
signal is approximated by a neural network as illustrated in
Fig. 2. In this network, the input vector xn is first weighted
via the weight vector ĉ, representing the prefilter in the WH
model. The first layer L1 consists of a single neuron with a linear
activation function. This neuron’s output equals the output of
the prefilter ĉT xn , and is weighted by w = [w1 , . . ., wM ]T and
passed to the nonlinear layer L2 which consists of M neurons.
The output of the i-th neuron in this layer is
fi (ĉ xn ) = tanh(wi ĉ xn ).
T T
The outputs of the neurons are then weighted by the weight
vector ân = [â1 , . . ., âM ]T followed by the layer L3, consisting
of a single linear neuron with its output being obtained by (2).
Furthermore, the signal d[n] ˆ is passed to the linear layer
L4, consisting of Lh neurons and Lh − 1 delays, and is then
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IEEE SIGNAL PROCESSING LETTERS, VOL. 26, NO. 12, DECEMBER 2019
weighted by the weight vector ĥn representing the FIR fil-
ter, and the output of the network is again given by (3). Fi-
nally, the ANN uses unbiased neurons that are centered at
zero.
By highlighting the parts of the network that correspond to
WH model components in Fig. 2, a one-to-one relation between
the parameters of the network, except for the vector w, and
those of a WH model is established. The weight vector w is
viewed as an optimization parameter that controls the shape of
the basis functions of the nonlinear preprocessor in Fig. 1. This
is beneficial, since it allows the network to optimally adjust its
basis functions to model the nonlinear distortions, as opposed
to conventional approaches where a set of basis functions is
pre-selected which is not necessarily optimum.
(1) The network’s parameters are optimized by training the net-
work to model the entire echo path, including the linear part,
eliminating the need for inverse filtering techniques, which may
be expected to be an additional source of errors.
After being initialized by random weights sampled from a
standard normal density, the network is trained using the conven-
tional error backpropagation algorithm. Intuitively, the network
needs to be trained on conditions similar to the expected condi-
(2) selection of volume levels is needed for training to cover a wide
range of levels during deployment.
B. The Deployment Phase
By training the network to model the entire echo path, gener-
alization to time-varying acoustic conditions is not guaranteed.
(3) Hence, after stopping the training, we discard the last trained
layer (L4) and replace it by an adaptive FIR filter ĥn that is not
constrained to the length of the linear layer in training. The rest
of the network is then used to model the nonlinearities in the
testing signal. This procedure is similar to transfer learning in
the machine learning literature [20].
To account for unseen variations and discrepancy between the
training and testing conditions, we allow the weight vector ân
in L3, in addition to the FIR filter ĥ, to be adapted continuously
via the SA-ERPF algorithm. By doing so, we assume implicitly
that the memory characteristics and functional shape of the
basis functions needed for modelling the nonlinear distortions is
time-invariant and only the individual contributions of the basis
functions are time-varying. Consequently, adapting the network
during the deployment phase is computationally not more costly
than adapting a WH model with an equivalent number of basis
functions.
Finally, while the online adaptation of the prefilter ĉ is prob-
lematic for conventional adaptive filtering approaches [25], this
challenge is resolved in the ANN-based approach by training
the prefilter instead of adapting it continuously.
C. The SA Elitist Resampling Particle Filter
Due to its superior performance shown for Hammerstein
(4) model-based NLAEC, we choose to adapt the networks pa-
rameters using the SA-ERPF [21]–[23]. In the following, the
SA-ERPF is reviewed briefly. For a comprehensive description
of the algorithm, the reader is referred to [21]–[23].
Since the FIR filter ĥn is typically a long filter with many
coefficients of small values, the SA-ERPF performs a temporal
decomposition of ĥn into the direct-path component ĥdirect,n ,
HALIMEH et al.: NEURAL NETWORK-BASED NONLINEAR ACOUSTIC ECHO CANCELLER 1829

consisting of only the LSA significant coefficients centered For adapting the network, the SA-ERPF is configured
around the main peak, and a complementary component ĥcomp,n , with Np = 80 particles and a Gaussian likelihood density of
which captures all other coefficients [26]. Similarly, the distorted N (y[n], σv2 ), with σv2 = 10−4 . In concatenating the RIR, LSA =
signal dˆn is also decomposed into dˆdirect,n and dˆcomp,n . Finally, 11 is used. Moreover, to provide a reference, a linear FIR filter
the direct-path signal is obtained by of 512-taps adapted via the NLMS algorithm with a step size
0.2 is also evaluated.
ydirect [n] = y[n] − ĥTcomp,n dˆcomp,n . (5) As a measure of performance, the Echo Return Loss Enhance-
ment (ERLE) is used
The decomposition allows the SA-ERPF to concentrate the
E{y[n]2 }
computational power on the estimation of the short but signifi- ERLEn = 10log10 ; e[n] = y[n] − ĥTn dˆn , (8)
cant parameter vector ẑ n = [ĥTdirect,n , âTn ]T rather than the long E{e[n]2 }
and sparse vector [ĥTn , âTn ]T . The parameter vector’s posterior where E{·} denotes the expectation operator.
density p(ẑ n |ŷ1:n ) is approximated by a set of Np particles and
(m)
their associated weights {ẑ (m) n , qn ; m = 1 : Np }, where the
A. Synthesized Nonlinear Distortions
Np (m)
weights are normalized such that m= qm = 1. In this experiment, the AFI ANN’s capability to learn under-
The SA-ERPF starts with an initial population of particles lying nonlinear functions is examined. Specifically, we consider
drawn from an initial distribution p(ẑ 0 ). Unlike other particle four types of nonlinearities, i.e., the Fourier Series [29]
filters, e.g., [27], the SA-ERPF operates on two populations of
3  
particles, an elitist set ΦE,n and a non-elitist set ΦNE,n . At each cT xn
d1 [n] = a1,i sin iπ , (9)
time instant n, the particles’ weights are updated according
i=1
1.5
to [23]. Afterwards, the particles are divided into ΦE,n and
ΦNE,n depending on their weights as the odd-order Legendre polynomials of the first kind [10]
 (m) 
2
ΦE,n , if qn ≥ N1p d2 [n] = a2,i L2i+1 (cT xn ), (10)
ẑ n ∈
(m)
(m) (6)
ΦNE,n , if qn < N1p . i=0

a mixture of the two nonlinear functions

Partial resampling is applied by discarding the non-elitist
particles while keeping the elitist particles. To maintain the
number of particles Np fixed, the elitist particles are used to
construct a Gaussian density N (µn , Σn ) where the mean vector
µn and the covariance matrix Σn are estimated as in [27]. This
density is then used to draw new particles that substitute the
discarded ones.
As an estimate of the parameter vector, the SA-ERPF extracts
the Minimum Mean Square Estimate (MMSE) via
Np
 The signal is distorted using (9)–(12) to generate the dif-
ẑ MMSE
n = qn(m) ẑ (m)
n .
m=1
In order to account for variations in the complementary
component of the RIR ĥcomp,n , we adapt this vector through
a conventional Normalized Mean Square (NLMS) algorithm
as in [26]. Finally, we refer to the proposed algorithm as the
Adaptive Filtering-Inspired (AFI) network.
IV. EXPERIMENTAL RESULTS
In this Section, the AFI ANN is evaluated in two experiments,
the first uses synthesized nonlinearities while the second uses
real distortions recorded by a mobile phone. The speech ut-
terances used in generating training and testing data in both
experiments are taken from the TIMIT corpus [28].
In all experiments, the network consists of a prefilter ĉ of
length Lc = 4, followed by a nonlinear layer with M = 9 neu-
rons. The FIR filters ĥn , or equivalently the linear layer, has
Lh = 512 taps. All speech signals are recorded at fs = 16 kHz.
The network is trained using a training step size of 10−4 . The
various hyperparameters, i.e., the training step size, the number
of neurons and the length of the training signals, were selected
empirically.
 
cT xn
d3 [n] = a3,1 cT xn +a3,2 (cT xn )3 +a3,3 sin 3π , (11)
1.5
and a completely different nonlinearity modelling the soft-
clipping behavior observed in loudspeakers,
2
d4 [n] = arctan(5cT xn ). (12)
π
As a training sequences, a one-minute-long speech signal
concatenating multiple speakers from both genders is used.
(7) ferent signals {di [n]|i = 1, . . ., 4} using the prefilter c =
[1, 0.6, 0.2, 0.1]T and parameter vectors a1 = [0.8, 0.4, −0.2]T ,
a2 = [0.1, 0.1, 0.1]T , and a3 = [0.5, 0.2, 0.2]T . Afterwards,
each signal is convolved with a recorded RIR h1,n , and corrupted
by additive, normally-distributed, white noise v[n], resulting in
four different training sequences given by
yi [n] = hT1,n di,n + v[n]; i = {1, 2, 3, 4}, (13)
where the variance σv2 is chosen such that the logarithmic Signal-
to-Noise Ratio (SNR) is 30 dB.
The testing sequences are generated by using a different
speech signal of one minute duration, consisting of different
speakers and recorded at fs = 16 kHz. Similarly to the training
sequence, the signal is also distorted using (9)–(12) and the
resulting signals are then convolved with a recorded impulse
response ĥ2,n , which represents a different acoustic enclosure
than that of the training sequences but of equal length. Finally,
the four testing sequences are corrupted by normally distributed
or Laplace-distributed additive white noise at various SNR
levels.
The AFI ANN is compared to two different WH models
which use a prefilter ĉ with 4 taps and use either the Legendre
polynomials (10) or the Fourier series (9) as basis functions

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1830
TABLE I
AVERAGE ERLE (dB) FOR EXPERIMENTS IN SECTION IV-A
for the memoryless nonlinear preprocessor, followed by an FIR
filter with 512 taps. Both models are adapted via the NLMS
algorithm. To compare the different approaches, the average
ERLE values achieved by each approach after convergence are
shown in Table I for the different nonlinearities.
It can be seen in Table I that using the matching basis functions
w.r.t. the underlying nonlinearity yields the best performance
compared to the other approaches. This is seen for, e.g., a
WH model using Legendre polynomials to identify a speech
signal distorted by a Legendre polynomial-based nonlinearity.
However, the AFI ANN achieves a performance close to the
optimum basis functions for the nonlinearities (9) and (10). This
indicates the ability to learn the underlying nonlinear function
to a high degree of accuracy. For distortions generated by a
mixture of the two basis functions in (11), the two WH models
based on the individual basis functions perform similarly. At the
same time, the AFI ANN is capable of outperforming both by
nearly 2 dB on average. For distortions generated by a function
that does not match any of the model basis functions, i.e.,
(12), the AFI ANN is outperforming the other approaches by
at least more than 2dB. In order to examine the generalization
capabilities of the AFI ANN, Table I shows the performance
of the AFI ANN when trained for normally-distributed white
noise (denoted by N in the table) at 30 dB SNR and tested for
different SNR levels and for Laplace-distributed white noise,
denoted by L in the table. As seen from the results, the AFI ANN
maintains its advantageous performance across the different
SNRs and noise types, confirming that the network is learning
the underlying nonlinear function rather than overfitting to the
seen data. Finally, small variations in the NLMS step size in the
AFI ANN adaptation were observed to lead to small variations
in the ERLE, suggesting that the proposed model is robust w.r.t.
the adaptation parameters.
B. Real Nonlinear Distortions
In this experiment, the training signal is a one-minute-long
speech signal representing multiple speakers from both genders.
It is emitted by a loudspeaker of a hand-held mobile phone,
which introduces unknown nonlinear distortions, and recorded
at an estimated SNR level of 20 dB, for which the noise power
was measured during speech pauses and the noisy signal’s power
was used instead of the clean signal’s power. To generate the test
signal, the same mobile phone was placed in a different acoustic
enclosure, and speech signals representing different speakers
from both genders, were emitted by the phone’s loudspeakers
using identical volume settings to those of the training signal,
and were recorded by its microphone at an estimated SNR
level of 20 dB as well. In this experiment, the AFI ANN is
compared to a Legendre polynomial-based and to a Fourier
series-based WH models with a prefilter ĉ of four taps each.
The Legendre-based WH uses the first four odd-order Legendre
polynomials as basis functions for the nonlinear preprocessor,
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IEEE SIGNAL PROCESSING LETTERS, VOL. 26, NO. 12, DECEMBER 2019
Fig. 3. ERLE in dB for the real nonlinear distortions.
while the Fourier series-based WH uses the first three
odd-order basis functions. Both are followed by an FIR filter of
length Lh = 512.
As can be seen in Fig. 3, the AFI network outperforms conven-
tional NLAEC approaches by a considerable margin, except for
a few instances, where the conventional approaches outperform
the AFI network slightly. Overall, the AFI network yields an
average ERLE of 11.1 dB, while the Legendre polynomial-based
approach yields an average of 8.9 dB, the Fourier series-based
model resulted in an average ERLE of 9.5 dB, and the purely
linear AEC achieves an average ERLE of only 7.2 dB. If we
assume that the modeling capacity of the AFI ANN covers
that of the conventional models, the time intervals where the
conventional approaches outperform the AFI ANN can be seen
as an effect of lacking training data for this situation. Moreover,
a tendency towards overfitting was observed when increasing
the number of neurons, above the number of neurons used in
this experiment, in either the nonlinear layer or the prefiltering
layer, motivating a careful network-designing phase. As shown
by the two experiments, a one-minute-long speech signal is
enough to capture the nonlinear distortions at a single volume
level. However, in order to model the distortions emitted at
different levels, one would need a longer training sequence that
covers the targeted volume levels. Alternatively, convex com-
bination approaches can be employed, similar to [12], to com-
bine several networks trained separately to model the nonlinear
distortions at different levels. Such techniques are especially
effective when abrupt and drastic changes in the volume levels
are present, where the AFI ANN’s performance is expected
to degrade.
V. CONCLUSION
In this letter, we introduced a novel, neural network-based
approach to NLAEC. The approach consists of training a neural
network to model the entire echo path, closely following a
WH model, using the conventional error backpropagation al-
gorithm. Then, the last linear layer of the network is replaced
by an adaptive FIR filter and the rest of the network is used
as a nonlinear preprocessor in the WH model. The proposed
approach is capable of modelling memoryless nonlinearities as
well as nonlinearities with memory using a linear FIR prefilter.
To account for variations in the nonlinear distortions, a single
weight vector is adapted during application of the network
through the SA-ERPF. This weight vector controls the individual
contributions of the learned basis functions. The AFI ANN
outperforms conventional approaches for real and synthesized
nonlinearities as demonstrated through experimental results.
HALIMEH et al.: NEURAL NETWORK-BASED NONLINEAR ACOUSTIC ECHO CANCELLER
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