A New Biologically Inspired Fuzzy Expert

System-Based Voiced/Unvoiced Decision

Algorithm for Speech Enhancement

M. A. Ben Messaoud, A. Bouzid &

N. Ellouze

Cognitive Computation

ISSN 1866-9956

Cogn Comput
DOI 10.1007/s12559-015-9376-2

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Cogn Comput
DOI 10.1007/s12559-015-9376-2
A New Biologically Inspired Fuzzy Expert System-Based Voiced/
Unvoiced Decision Algorithm for Speech Enhancement
M. A. Ben Messaoud1 • A. Bouzid1 • N. Ellouze1
Received: 7 March 2015 / Accepted: 28 December 2015
 Springer Science+Business Media New York 2016
Abstract In this paper, we propose a speech enhance-
ment approach for a single-microphone system. The main
idea is to apply a specific transformation on the speech
signal depending on the voicing state of the signal. We
apply a voiced/unvoiced algorithm based on the multi-scale
product analysis with the use of fuzzy logic to make more
cognitively inspired use of speech information. A comb
filtering is applied on the voiced frames of the noisy speech
signal, and a spectral subtraction is operated on the
unvoiced frames of the same signal. Further, the harmonics
are enhanced by performing a designed comb filtering
using an adjustable bandwidth. The comb filter is tuned by
an accurate fundamental frequency estimation method. The
fundamental frequency estimation method is based on
computing the multi-scale product analysis of the noisy
speech. Experimental results show that the proposed
approach is capable of reducing noise in adverse noise
environments with little speech degradation and outper-
forms several competitive methods.
Keywords Speech enhancement  Multi-scale product 
Voiced decision  Fundamental frequency estimation 
Comb filter  Fuzzy logic
& M. A. Ben Messaoud
anouar.benmessaoud@yahoo.fr
1
National School of Engineers of Tunis, University of Tunis
El Manar, Tunis, Tunisia
Introduction
Speech enhancement methods currently available in the
literature can be subdivided into multi-channels and mono-
channel approaches [1]. Mono-channel is much more
arduous, as the mutual relation between channels cannot be
analysed.
The enhancement of a mono-channel speech signal
degraded by additive background noise has been largely
studied. Many methods in the field of speech enhancement
have been proposed to improve the quality, increase the
intelligibility of degraded speech signals and reduce the
listener’s fatigue.
Speech enhancement methods can be generally divided
into several categories: spectral subtractive algorithms,
wavelet transform, statistical-model-based algorithms,
subspace methods, harmonic structure of speech and cog-
nitive and brain-inspired intelligent.
Spectral subtraction (SS) [2, 3] is one of the first algo-
rithms applied to the problem of speech enhancement. It is
simple and provides a trade-off between speech distortion
and residual noise to some extent, and it suffers from an
artefact known as ‘‘musical noise’’ having an unnatural
structure that is perceptually annoying, composed of tones
random frequencies and has an increased variance. Many
methods have been proposed to eliminate this phenomenon
including perceptually motivated approaches [4, 5].
Wavelet transforms (WT) have been applied to various
speech applications [6]. The main challenge in such
denoising approaches based on the thresholding of the
wavelet coefficients of the noisy speech is the estimation of
a threshold value that marks a difference between the
wavelet coefficients of noise and that of clean speech.
Then, by using the threshold, the designing of a thresh-
olding scheme to minimize the effect of wavelet
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coefficients corresponding to the noise is another difficult
task considering the fact that the conventional discrete WT-
based denoising approaches exhibit a satisfactory perfor-
mance only at a relatively high signal-to-noise ratio.
Statistical model-based algorithms are one of the most
commonly used classes of speech denoising methods. In
the minimum mean-square error (MMSE estimator) [7, 8],
the frequency spectrum of the noisy speech is modified to
reduce the noise from noisy speech in the frequency
domain. A relatively large variance of spectral coefficients
is the problem of such an estimator. While adapting filter
gains of the MMSE estimator, spectral outliers may emerge
that is especially difficult to avoid under noisy conditions.
One of the major problems of Wiener filter-based methods
[9–11] is the requirement of obtaining clean speech
statistics necessary for their implementation. Both the
MMSE and the Wiener estimators have a moderate com-
putation load, but they offer no mechanism to control
trade-off between speech distortion and residual noise.
The signal subspace approach for speech enhancement
is an interesting generalization of the spectral weighting
methods. This technique has been originally proposed in
[12]. The speech estimation is considered there as a con-
strained optimization problem, where the speech distor-
tions are minimized subject to the residual noise power
level. Two linear estimators have been proposed: time-
domain-constrained (TDC) and spectral domain-con-
strained (SDC). Unlike DFT-based methods, signal sub-
space approaches decompose a signal space into speech
subspace and noise subspace using Karhunen–Loeve
transform (KLT). Then, the spectral weighting is per-
formed in the signal subspace only. The components pro-
jected onto the noise subspace are simply nulled which
results in a significantly better performance when com-
pared to the conventional frequency domain methods
where a full-band spectrum must be processed.
Other popular approach is the harmonic structure. It
consists to use the harmonic structure of voiced speech for
enhancing the speech quality [13–16]. In the work of [13,
14], the signal is modelled as harmonic plus noise com-
ponents. Then, the voiced signal is enhanced by estimating
the harmonic components. In [15], the authors use an
extension of the hidden Markov model (HMM)-based
MMSE estimator to enhance the harmonics for voiced
speech by incorporating a ternary voicing state. In [16], the
authors have adopted here sinusoidal model with the
speech enhancement context.
The perceptual aspects of speech are considerably more
complicated and less well understood. However, there are a
number of commonly accepted aspects of speech percep-
tion which play an important role in speech enhancement
systems. Perceptual cues of highly degraded speech can be
thought of two levels, namely cognitive and acoustic levels
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[17]. At the cognitive level, perception of degraded speech
is aided by the knowledge of context of conversation, the
syntax and semantics of the context and high-level features
like intonation and duration. At the acoustic level, sound
perception in degraded conditions happens mostly by
extrapolation of information from the high SNR regions to
the low SNR regions in the temporal domain. Furthermore,
it is generally understood that the cognitive and brain-in-
spired intelligent also plays a central importance in the
perception of speech [18, 19]. So, we may consider another
category of speech enhancement methods that consist to
use the cognitive and brain-inspired intelligent. Abel et al.
[20, 21] propose a multimodal speech enhancement sys-
tem, based on visual and audio extraction features to filter
the noisy speech in a cognitively inspired manner by using
a neuro-fuzzy system. The visual modality consists in
employing a visually derived Wiener filtering based on lip
tracking technique. Then, the authors use the beamforming
approach to exploiting spatial diversity of source and noise.
In [22, 23], DNNs (deep neural network) were used to
estimate a smoothed ideal ratio mask in the mel-frequency
domain for robust automatic speech recognition. Xu et al.
[24] propose to learn the complex mapping function from
noisy to clean speech with nonlinear DNN-based regres-
sion models using multi-condition training data encom-
passing different key factors in noisy speech, including
speakers, noise types and signal-to-noise ratios (SNRs).
During voiced speech segments, the regular glottal exci-
tation of the vocal tract produces energy at the fundamental
frequency F0 and its multiples. An alternative strategy of
speech enhancement comprising all those approaches is to
process the speech based on the fundamental frequency F0
estimation. The voiced noisy speech is improved by
enhancing the harmonic structures [25]. This harmonicity-
based enhancement is performed by extracting the spectral
components at integer multiples of fundamental frequency
(F0) with sinusoidal modelling [26] or adaptive comb filter
[27–30]. The F0 is used to pick out the voiced speech from
noise. In this strategy of methods, robust and accurate F0
estimation and voicing detection are essential issues, because
any errors in these values cause a severe deterioration in the
achieved speech quality. For example, errors in F0 estimation
may distort the target speech signal, while inaccuracies in
voicing detection may cause the loss of voiced segments.
Most voicing decision algorithms exploit almost any
elementary speech signal parameter that may be computed
independently of the type of input signal: energy, ampli-
tude, short-term autocorrelation coefficients, zero-crossings
count, ratio of signal amplitudes in different sub-bands or
after pre-processing as linear prediction error, or the sal-
ience of a pitch estimate. Voicing decision algorithms can
be grouped into three essential steps. The first has been
designed specifically to solve the voiced/unvoiced
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problem, using statistical inference on the training data [31,
32]. Most methods in this category use a statistical para-
metric method, which exploit only a few basic parameters
(a fixed threshold) by examining whether the value of a
certain feature exceeds a predetermined threshold. Inap-
propriate selection of the threshold, regardless of input
signal characteristics, results in performance degradation.
The second category that has attempted to solve this
problem is fundamental frequency estimators that, as an
aside, make voiced/unvoiced decisions based on the degree
of periodicity of the signal [33, 34]. In this category,
nonparametric methods based on linear discrimination
functions, multi-layer feed forward and recurrent neural
networks were adopted. The third category is more com-
plex algorithms based on acoustical features, and pattern
recognition techniques were used to separate the speech
segments into voiced/unvoiced [35, 36].
In this paper, we propose an approach for speech
enhancement based on voiced/unvoiced algorithm classi-
fication that can work in low SNR environments and under
varying noise conditions. The decision to classify a seg-
ment into voiced or unvoiced is based on the values of the
number of group, energy and zero-crossings. Since these
values are not precise, particularly in the presence of
background noise, it will be helpful to use fuzzy logic, in
order to improve the performance of our method based on
automatic cognitive classification and to increase the
robustness of the approach under noisy conditions. So in
this paper, we present a method that classifies the speech
based on fuzzy expert system of the spectral multi-scale
product (SMP) peaks, short-time energy of multi-scale
product (MP) and zero-crossings of MP.
For noisy voiced speech frames, the harmonic speech
structure obtained through short-time Fourier analysis is
enhanced by applying an adaptive comb filter. The quasi-
periodic signal produced by the interaction between the
glottal excitation and the vocal tract is not totally station-
ary. As a consequence, the spectral peaks belonging to
higher harmonics are more shifted in frequency than those
corresponding to lower harmonics. This inharmonicity
provokes that high-order harmonics are partially or totally
removed when applying a purely harmonic comb filter. To
palliate this problem, we have implemented a comb filter
with rectangular pass bands structure in the spectral
domain to enhance the harmonic structure. In unvoiced
frames, a spectral subtraction technique with geometric
advent of spectral subtraction using connected time–fre-
quency regions noise estimation is applied. The experi-
mental results indicate the superiority of our approach in
terms of the speech quality, and intelligibility under a wide
range of noisy environments, such as white, and real
acoustic interferences.
The remainder of this article is structured as follows. A
detailed description of our approach is provided in Sect. 2.
In Sect. 3, we describe the evaluation results. Section 4
concludes this work.
Our Proposed Approach
This section describes the principle components of our
approach for speech enhancement. It works as follows:
first, the signal is divided into 25.6 ms windows with 50 %
overlap between frames. This frame length was chosen
empirically, as it is enough long for good spectral esti-
mation of fundamental frequency value in each frame but
not so long as to modify temporal change in the speech
signals. Then, frames of original speech signals are clas-
sified to voiced/unvoiced frames by a fuzzy logic decision
algorithm. Second, we apply a designed comb filter on
voiced frames and a spectral subtraction method using
connected time–frequency regions noise estimation on
unvoiced frames to remove the residual noise. The comb
filter is tuned by the estimated fundamental frequency. The
fundamental frequency is estimated in the voiced frames
using the spectral compression multi-scale product method.
Finally, we use the overlap-add procedure to reconstruct
the enhanced speech signal.
The various steps implied in the proposed approach are
illustrated in Fig. 1. Each one will be detailed in the fol-
lowing subsections.
The voiced/unvoiced (V/UV) decision is at the heart of
our suggested approach as the processing for the speech
enhancement depends on the voicing state using the fuzzy
expert system. Accordingly, the voiced/unvoiced (V/UV)
decision algorithm will be closely analysed.
Voiced/Unvoiced Classification Algorithm
The voiced/unvoiced algorithm is based essentially on the
multi-scale product (MP) characteristics. In fact, the MP
attempts to enhance the peaks of the gradients caused by
true edges, while suppressing false peaks caused by noise.
It also permits to obtain a more simplified signal having a
periodic structure in the voiced frames and a quasi-null
signal in unvoiced frames [37].
Our voiced/unvoiced decision algorithm is given in the
block diagram depicted in the Fig. 2.
The WT is a multi-scale analysis which has been shown
to be very well suited for speech processing as glottal
closure instant (GCI) detection, pitch estimation, speech
enhancement and recognition and so on. Using the WT, a
speech signal can be analysed at specific scales corre-
sponding to the range of human speech.
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Voiced
frame Comb
filtering

V/UV Reconstruction Enhanced

Noisy
Framing decision of the speech
speech speech
algorithm
Spectral
subtraction
Unvoiced
frame

Fig. 1 Block diagram of the proposed speech enhancement system

Voiced
Group /Unvoiced
FFT classification decision
s1=1/2
Multi-Scale Product
Wavelet transform

Short time Rules of Fuzzy

Noisy
(MP)

s2=1 Energy ZCR, En, expert

speech system
Eni nbr group
frame

Short time
s3=2
Zero-crossings
rate Zci

Fig. 2 Block diagram of the voiced/unvoiced classification algorithm

One of the most important WT applications is the signal
singularity detection. In fact, continuous WT produce
modulus maxima at signal singularities allowing their
localization. These maxima appear at specific singularity
points, depending on wavelet vanishing moments.
The choice of the mother wavelet is crucial to detect
discontinuities. To choose the adequate wavelet, we must
take into account two considerable characteristics. The first
one is the number of the vanishing moments, and the
second one is relied on the wavelet support. WT with
n vanishing moments can be interpreted as a multi-scale
differential operator of nth order of a smoothed signal [38].
This provides a relationship between the differentiability of
the signal and wavelet modulus maxima decay at fine
scales.
However, one-scale analysis does not give a good pre-
cision. So, decision algorithm using multiple scales is
proposed by different works to circumvent this problem
[38, 39].
The MP was first used in image processing. It is based
on the multiplication of wavelet transform coefficients
(WTC) at some scales. It attempts to enhance the peaks of
the gradients caused by true edges, while suppressing
false peaks caused by noise. Xu and al. [40] rely on the
variations in scale of the WT. They use multiplication of
WT of the image at adjacent scales to distinguish
important edges from noise. Sadler and Swami [41] have
studied MP method of signal in the presence of noise. In
wavelet domain, it is well known that edge structures are
present at each sub-band, while noise decreases rapidly
along the scales. It has been observed that multiplying the
adjacent scales could sharpen edges while smoothing out
noise [42].
The MP consists of making the product of WTC of the
noisy speech frame x(n) at three successive dyadic scales as
follows:
Y
j¼1
pð nÞ ¼ W2 j xðnÞ: ð1Þ
j¼1
where W2 j xðnÞ is the wavelet transform of the noisy speech
frame x(n) at scale 2j.
The wavelet used in this analysis is the quadratic spline
function. The quadratic spline function has a compact
support and is continuously differentiable. It is the first
derivative of the cubic spline function. Thus, it has only
one vanishing moment. Wavelet decompositions is made at
scales s1 = 2-1, s2 = 20 and s3 = 21. Odd number of
terms in p(n) preserves the sign of maxima. Choosing the
product of three levels of wavelet decomposition is gen-
erally optimal and allows detection of small peaks.
Figure 3 depicts the speech signal (Fig. 3a) added to a
Gaussian noise with an SNR equal to -5 dB (Fig. 3b)
followed by the MP of the noisy speech.

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Fig. 3 a Clean speech, b speech corrupted by -5 dB white noise. c Multi-scale product (MP) of noisy speech
We can clearly see the effect of the multi-scale product
in eliminating the noise sounds Fig. 3c. The MP has the
same behaviour regarding the noise sounds.
In the voiced frames, the MP has a periodic structure
with maxima corresponding to the glottal opening instant
(GOI) and minima corresponding to the glottal closing
instant (GCI) unlike the unvoiced speech case [38], so a
voicing detection algorithm can be derived.
Figure 4 depicts a voiced region of speech signal pro-
nounced by a women speaker. The speech signal is fol-
lowed with its electroglottographic (EGG) signal, and at
the bottom, we find the multi-scale product (MP) of the
voiced speech signal. The EGG signal was the easiest way
to measure the GCI and GOI as it is a direct representation
of the glottal activity. On the MP signal, Fig. 4c shows
minima corresponding to GCIs and maxima representing
the GOIs crop up.
The proposed voiced/unvoiced algorithm uses three
criteria to decide the voicing nature. It is about the group
classification of the spectral multi-scale product peaks
(SMP), the short-time energy of MP and the short-time
zero-crossings of MP.
The reason of using three parameters is that each
parameter will perform well and accurately, just when the
value of that parameter is very high or very low. So the use
of three parameters will help us in this matter. Also to
complete the decision, it is necessary to use more than one
parameter. In this case, each parameter will correct some of
the other parameters mistakes, because each parameter’s
base is different from the other one.
The MP of a voiced frame gives a low zero-crossings
rate, a high energy because of the periodic structure of the
signal and a number of group of SMP equal to 1 or to 2,
whereas an unvoiced frame of the MP gives a high zero-
crossings count, a low energy near to zero and a number of
SMP group greater than 2.
Spectral MP group
The first parameter for the determination of the V/UV
decisions is the group classification. We operate the spec-
trum of the signal multi-scale product (SMP) by using the
fast Fourier transform (FFT) algorithm. We define the
number of group constituted by computing the distance
separating two successive peak positions of the MP spec-
trum. Then, we rank this distance in the growing order to
compose the Ei vector with i corresponding to the ith
frame. If the difference between two successive elements
of the vector is less or equal to (THDIST), the two elements
belong to the same group. If the number of group is equal
to 1 or to 2, the frame is voiced; else, the frame is declared
unvoiced.
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Fig. 4 Speech signal pronounced by a women speaker; a speech signal, b EGG signal and c the multi-scale product
Short-time energy of MP
The second parameter for the determination of the V/UV
decisions is the energy. The threshold of energy (THE)
must be higher to decide that the frame is voiced. For the
energy parameter, the voiced frame has high energy
because of its periodicity.
Short-time zero-crossings of MP
The third parameter for the determination of the V/UV
decisions is the zero-crossings rate (ZCR) of the MP. It can
be defined as:
, speech will have relatively high zero-crossings rate, rela-
PN
ZCRðnÞ ¼ absðsgn½pðiÞ  sgn½pði  1Þ
i¼1 2N
( ð2Þ
1; pðiÞ  0
with sgn½pðiÞ ¼
0; pðiÞ\0
where p(n) is the multi-scale product of the speech frame
and sgn½ is the sign function. The threshold of ZCR
(THZCR) should be lower to decide that the frame is voiced.
The choice of the three thresholds is experimentally
determined using the NOIZEUS database. The voiced/un-
voiced decision is related to the optimal choice of the
thresholds of the three parameters. The optimal thresholds
are obtained after the computation of the requested feature
for each frame. The median value is taken as threshold.
This procedure is used to extract the THDIST, THE and
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THZCR which are the median values of three parameters for
each frame in the speech database.
Energy of multi-scale product (MP) for unvoiced speech
is significantly smaller than for voiced speech; hence, the
short-time energy can be used to distinguish voice and
unvoiced speech. The short-time energy of MP can also be
used to distinguish speech from silence. But the use of
short-time energy of MP is not sufficient alone; hence, it is
used coupled with the zero-crossings of MP and the
spectrum of MP in the classification of speech. Hence, we
state that voice speech should be characterized by rela-
tively high energy, relatively low zero-crossings rate, and a
number of group are lower or equal to 2, while unvoiced
tively low energy, and a number of group are higher than 2.
Also, we have not said what we mean by high and low
values of short-time energy, short-time zero-crossings rate,
and it is really not possible to be precise. Hence, we see
that this is a problem that fit to be solved by fuzzy logic. In
this paper, we propose a method to classify the speech into
voiced and unvoiced detection using short-time energy,
zero-crossings and number of group in a fuzzy logic sys-
tem. The voiced/unvoiced algorithm uses the fuzzy logic.
Fuzzy logic (FL) is a method to computing based on
‘‘degrees of truth’’ rather than the usual Boolean logic. FL
includes 0 and 1 as extreme cases of truth but also includes
the various states of truth in between. FL seems closer to
the way our brains work. We aggregate data and form a
number of partial truths which we aggregate further into
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higher truths which in turn, when certain thresholds are
exceeded, cause certain further results such as motor
reaction. A similar kind of process is used in artificial
computer neural network and expert systems. Fuzzy sets
are a super-set of classical sets. Each element in a fuzzy set
is associated with real number which represents the degree
of membership of the element in the set. FL allows highly
nonlinear, poorly understood or mathematically complex
systems to be modelled reliably and efficiently. And it
deals well with noise data. These characteristics show that
FL might be an effective tool for speech enhancement.
Most applications of fuzzy logic use it as the underlying
logic system for fuzzy expert systems (FES). A FES is an
expert system that uses a collection of fuzzy membership
functions and rules, instead of Boolean logic, to reason
about data. The rules in a fuzzy expert system are usually
of a form similar to the following:
if x is low and y is high, then z = medium.
Where x and y are input variables, z is an output vari-
able, low is a membership function (fuzzy subset) defined
on x, high is a membership function defined on y, and
medium is a membership function defined on z. The part of
the rule between the ‘‘if’’ and ‘‘then’’ is the rule’s premise.
This is a fuzzy logic expression that describes to what
degree the rule is applicable. The part of the rule following
the ‘‘then’’ is the rule’s consequent. This part of the rule
assigns a membership function to each of one or more
output variables.
Based on our work [43], we use an expert system gen-
erator called CLIPS. It is a productive development and
delivery expert system tool which provides a complete
environment for the construction of rule-based expert
systems.
We apply the following rules:
R1: If a frame has a low zero-crossings rate, then it is
voiced.
R2: If a frame has a high zero-crossings rate, then it is
unvoiced.
R3: If a frame has a high energy, then it is voiced.
R4: If a frame has a low energy, then it is unvoiced.
R5: If a frame has a number of group less than three,
then it is voiced.
R6: If a frame has a number of group more than two,
then it is unvoiced.
We incorporate fuzzy logic with the rules to improve the
decision-making capability of the expert system.
The priorities of rules are given by the fuzzy CLIPS
expert system. First, for each rule, the system calculated
the error probability, and the one with the least error
probability was chosen as the first priority. Then, to choose
the second priority, the system calculated the conditional
error probability for the rest of the rules.
Enhancement of Voiced Frames
To enhance the speech, a designed comb filter is applied on
the voiced frames. The idea consists of taking the voiced
speech frame and eliminates any frequency components
that do not accord with the fundamental frequency and its
harmonics.
The different steps are illustrated in Fig. 5. It consists of
multiplying the frequency response of our designed filter
by the magnitude spectrum of the voiced speech frame.
Then, the inverse fast Fourier transform (IFFT) is used to
obtain the final enhanced voiced frame. The phase must be
staying the same after comb filtering.
Comb Filtering Description
A comb filter is a filter with multiple pass bands and stop
bands. For transmitting only the harmonic components of
the speech signal, the pass bands must be centred at mul-
tiples of the speech fundamental frequency, i.e. the fre-
quency response of the comb filter has to be a periodic
function with period equal to the fundamental frequency.
Because voiced speech signals have time-varying funda-
mental frequency, the comb filter for enhancement of
voiced speech has to be an adaptive filter tuned by the
instantaneous fundamental frequency of the speech. It
means that the comb filter varies from frame to frame.
We use a rectangular comb filter whose pass bands have
a fix width of 10 Hz and tuned by the fundamental fre-
quency F0. This comb filter is a feedforward filter with a
number of delays equal to fs/F0 samples, where fs is the
sampling frequency and F0 is the fundamental frequency.
The harmonic transfer function of the designed comb filter
is presented in Fig. 6. The spacing between the pass bands
is defined by the fundamental frequency F0. The spectral
maxima Spmax and the spectral minima Spmin are given by
the required attenuation for the SNR. The frequency
response of our designed comb filter is as follows:
Hðf Þ ¼ Spmax for iF0  Bd=2  f  iF0 þ Bd=2
¼ Spmin for ði  1ÞF0  Bd=2  f  iF0 þ Bd=2
ð3Þ
where Bd\F0=2 is the bandwidth of the comb filter,
h i
i = 1,2,… fix fs=2F0 . The frequency 0  f  fs=2 is
F0
Noisy voiced Harmonic Enhanced
frame FFT comb filter IFFT voiced frame
Fig. 5 Block diagram of the noisy voiced frame enhancement
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Fig. 6 Harmonic transfer function of a designed comb filter, where Bd = 10 Hz and F0 = 88 Hz
sampled with the frequency step fs=Nf , where Nf is the
dimension of the applied FFT.
Figure 6 illustrates the transfer function of our designed
comb filter with adjustable bandwidth.
When a frame is considered as voiced, the signal is
filtered by our comb filter tuned to the estimated funda-
mental frequency F0 and its integer multiples. The esti-
mation of the fundamental frequency F0 is detailed in the
subsection of fundamental frequency estimation based on
the spectrum compression of the multi-scale product
(SCMP). The use of this designed comb filter involves
some implementations issues that can be manifested by the
problem of the number of harmonics. The number of har-
monics of the voiced speech is difficult to estimate. In
general, this number depends on the speaker and the fun-
damental frequency estimation, but is usually assumed
fixed in advance, as in our approach. Therefore, if the
number of harmonics is different to the chosen number of
bands, either noise components will be captured or high-
frequency speech harmonics will be eliminated.
Our method for estimating the fundamental frequency
will be described in the following subsection.
Fundamental Frequency Estimation Based on SCMP
The comb filtering step requires an accurate estimation of
the fundamental frequency. In this work, we use the
spectrum compression (SC) of the multi-scale product
(MP) to estimate F0 inspired by our work in [37]. It can be
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decomposed into three steps, as shown in Fig. 7. The first
step consists of computing the MP of the voiced sound at
successive scales. The second step consists of calculating
the spectrum function (SF) of the obtained signal. These
two steps were described by our work reported in [44]. In
the third step, the spectrum of the MP is compressed by
integer factors (R = 2, 3, 4) and the obtained functions are
multiplied.
The first peak in the spectrum multi-scale product
(SMP) coincides with the second peak in the SMP com-
pressed by a factor of two, which coincides with the third
peak in the SMP compressed by a factor of three. So, when
Noisy Speech
Frame
Multi-Scale Product
(MP)
FFT
Product of Compression
Spectrum (SCMP)
Pitch Estimation
Fig. 7 Block diagram of the proposed method for fundamental
frequency estimation
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the various spectrums are multiplied together, the har- Time-frequency

connected speech
monics line up and reinforce the fundamental frequency F0. regions estimation
The harmonic product measures the maximum coincidence
of harmonics for each fast Fourier transform of the multi- Noisy Smoothing
Noisy
scale product. FFT estimation
unvoiced frame estimate
Figure 8 illustrates the SCMP method. The signals of
the Fig. 8a–c represent, respectively, the SMP compressed Enhanced Enhanced Gain
with a factor R = 2, 3 and 4 of the first voiced noisy speech unvoiced frame IFFT spectrum
function
region from Fig. 3b. The signal illustrated in the Fig. 8d
shows the multiplication of the functions issued from the Fig. 9 Block diagram of the noisy unvoiced frame enhancement
compression of the SMP with one peak giving the funda-
mental frequency estimation.
uxðkÞ ¼ usðkÞ þ nðkÞ ð4Þ
Enhancement of Unvoiced Frames by Spectral First, we apply the FFT to compute periodogram of
Subtraction noisy speech, that is PUX (j,k) = |UX (j,k)|2. After com-
puting periodograms, they are under process of smoothing.
For unvoiced frames, we employ an improved method based The smoothed periodograms are temporally minimum
on the psycho-acoustical model of the spectral density sub- tracked and are used for purpose of speech presence
traction called ‘‘geometric approach for spectral subtraction’’ detection. This detection is utilized to attain low biased
(GA) proposed by [4] which is robust under various types of noise power spectral density (PSD) estimates PN0 (j,k) and
interferences. The unvoiced noisy speech spectrum UX(jxn) for noise periodogram estimates PN (j,k) which is equal to
at frequency xn is calculated by the summation of two PUX (j,k) in speech absence condition. But if speech is
complex-valued spectra. The spectral subtraction (SS) with present, noise periodogram estimate is equal to noise PSD
GA is described in the Fig. 9. estimation. In later case, recursive smoothed bias com-
Consider usðkÞ is the unvoiced clean speech and n(k) is pensation parameter is put on minimum tracked values.
the noise signal and uxðkÞ is the noisy unvoiced speech The bias compensation factor is updated during absence of
frame: speech in frames while remains unchanged during speech

Fig. 8 SCMP of a voiced noisy speech. a Spectrum compression of MP with R = 2. b Spectrum compression of MP with R = 3. c Spectrum
compression of MP with R = 4. d Spectrum functions multiplication

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presence. The noise magnitude periodogram estimation aUX ejuUX ¼ aUS ejuUS þ aN ejuN ð5Þ
|N(j,k)| is computed from noise PSD estimation, and on the
basis of these information, decision of speech presence is where aUX, uUX, aUS, uUS and aN, uN are, respectively, the
made and used in speech enhancement method. The noisy magnitudes, and phase angles of unvoiced noisy speech,
speech periodograms PUX (j,k) are spectrally smoothed. unvoiced clean speech and noise.
The PUX (j,k) bands are composed of weighted sum of Unlike the traditional SS, the GA consider phase during
2X ? 1 band. computations. We apply the sinus function:
The spectrally smoothed periodograms are temporally G ¼ aUX sinðuN  uUX Þ ¼ aUS sinðuN  uUS Þ ð6Þ
smoothed recursively with time–frequency changing
jGj ¼ a2UX sin2 ðuN  uUX Þ ¼ a2US sin2 ðuN  uUS Þ ð7Þ
smoothing factor to create the temporally spectrally
 
smoothed periodogram P(j,k). jGj ¼ a2UX 1  cos2 ðuN  uUX Þ ð8Þ
The temporal minimum values Pmin(j,k) are computed  
from P(j,k) by tracing within minimum search window. jGj ¼ a2US 1  cos2 ðuN  uUS Þ ð9Þ
The speech presence results in increase of power in We obtain a new version of the gain function G:
temporally smoothed spectrum because of the additive 2 
noise at particular time–frequency regions. As a result, G2 ¼ aUS a2 ð10Þ
UX
ratio of temporally smoothed spectrum to noise PSD esti- rffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
2 
mate becomes more robust to estimate the SNR and noise- G ¼ aUS a2
UX
to-noise ratio at specific time–frequency regions. rffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffiffi
ffi
2
The smoothing phenomenon ensures the speech pres- ¼ 1  cos ð u N  u UX = Þ ð11Þ
1  cos2 ðuN  uUX Þ
ence detection even in conditions where noisy speech
power is unstable. As a result, connected speech presence Finally, the estimated signal spectrum is passed through
and absence regions can be achieved. IFFT which utilize phase of noisy signal, and we obtain
Here we have computed two different noise estimations: enhanced speech.
one is noise PSD estimation and second is noise spectrum The GA based on power SS technique has the advantage
estimation. The PSD estimation is used in speech of minimizing the musical noise and incorporating the
enhancement algorithm, while noise spectrum estimation cross terms implying phase differences between the noisy
shows the properties of residual noise from speech speech and the noise.
enhancement algorithm. The speech enhancement algo-
rithm for this noise estimation is spectral subtraction with
geometric approach. The block diagram for the connected Experimental Results
time–frequency region noise estimation algorithm is illus-
trated in Fig. 10. In this section, the proposed approach for speech
Then, we compute the gain function by transforming the enhancement is evaluated using objective and subjective
equation above to polar form: tests.

Fig. 10 Block diagram of the Spectrum of noisy Spectral PUX(j,k)) Temporal

time–frequency connected unvoiced frame smoothing smoothing
speech regions PUX(j,k)
P(j,k)

Speech
Maximum presence
tracking

Noise Speech
N(j,k))
estimation enhancement

Spectrum of
Enhanced
unvoiced frame

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Simulation Conditions As seen, the most problematic noise is the car interfer-
ence, with a GPE of 7.34 % at -5 dB. In this case, the
To evaluate and compare the performance of our proposed SWIPE method gives the best result because it considers
approach, we carry out simulations with the real noisy only the highly voiced frames.
speech sentences from the NOIZEUS database [45], which To evaluate our voicing decision algorithm, we deter-
is composed of phonetically balanced utterances. The noisy mine the UV/V error defined as the unvoiced frames con-
speech database contains 30 sentences belonging to three sidered as voiced, and the V/UV error corresponding to the
male and three female speakers. The speech signal is percentage of voiced frames considered as unvoiced.
sampled at 8 kHz and is composed of 30 sentences Table 2 presents the performances of our voiced/un-
belonging to three male and three female speakers, and was voiced classification algorithm with fuzzy expert system
corrupted by four background noises (airport, car, train and (proposed algo_MP ? FES), and without FES (proposed
babble) at four SNR levels (-5, 0, 5 and 10 dB). To algo_MP) and SWIPE methods in a noisy environment.
simulate the receiving frequency characteristics of tele- As depicted in Table 2, the proposed voiced/unvoiced
phone handsets, the speech and noise signals were filtered decision algorithm that used the fuzzy expert system with
by the modified intermediate reference system (IRS) filters MP in this work appears the most robust algorithm for
used in [46] for evaluation of the PESQ measure. V/UV errors. In the case of UV/V error, the SWIPE method
For the experiments, the length of the signal frames is set shows the lowest values. This can be explained by the fact
to 25.6 ms. The hop size between frames is set to 32 ms. In that this algorithm does not consider the weak voicing state
the blocks devoted to enhance voiced and unvoiced frames, like in the beginning and the end of any voiced sound.
windowing with a Hanning window is applied, and the order However, our proposed algorithm has the highest perfor-
of the FFT is set to 8192 frequency bins. This FFT size mances in all cases, depending on the fuzzy logic expert
provides a high enough resolution for the proper enhance- system.
ment of low frequencies and was chosen empirically as a In the Table 2, it can be noticed that the UV/V error
trade-off between achieved quality and complexity. decreases when the V/UV error increases. In fact, for
example under the car noise, our voiced/unvoiced classi-
Evaluations of Fundamental Frequency Estimation fication algorithm, respectively, SWIPE in car noise for
and Voiced/Unvoiced Classification Algorithms example, when the UV/V decreases from 1.70 to 1.28 %
(1.03–0.09 %), the V/UV increases from 5.21 to 10.70 %
Here, we have only analysed the influence of the voicing (8.37–15.56 %). It is obvious that our proposed algorithm
detector over the enhancement process. In order to illus-
trate the impact of F0 determination in the system, we
Table 1 Fundamental frequency estimation performance of GPE in a
measure the gross pitch error (GPE) of the SCMP algo-
noisy environment
rithm and we compare it to the spectral multi-scale product
method (SMP) [39] and the sawtooth waveform inspired Type of noise SNR level (dB) SCMP SMP [39] SWIPE [47]
pitch estimator (SWIPE) [47]. The SMP and SWIPE GPE (%)
methods were configured with the same settings (frame Airport 10 2.28 2.45 2.56
size and hop size) as the proposed algorithm. 5 2.73 3.04 3.43
The GPE is defined as the percentage of the correctly 0 2.86 3.17 3.45
classified voiced frames which have an incorrect funda- -5 3.05 3.34 3.78
mental frequency.
Car 10 3.56 5.83 3.06
The V/UV error is defined as the percentage of voiced
5 4.30 7.11 3.94
frames that are classified as unvoiced, and the UV/V error
0 5.06 8.41 4.63
is defined as the percentage of unvoiced frames that are
-5 7.34 9.20 6.03
classified as voiced.
Train 10 1.09 1.43 1.17
The reference F0 contours for the thirty voiced utter-
5 1.25 1.63 1.38
ances of the NOIZEUS speech database and the reference
0 1.57 1.98 1.79
voiced/unvoiced classification were built using our algo-
-5 2.19 3.17 2.88
rithm autocorrelation of the multi-scale product analysis
Babble 10 0.86 1.37 1.21
[48] on the clean speech and are manually corrected.
5 1.30 2.85 2.40
Table 1 presents the GPE of the SCMP, SMP and
0 1.81 3.27 2.73
SWIPE methods in a noisy environment.
As depicted in Table 1, when the SNR level decreases, -5 2.93 4.58 3.46
the SCMP algorithm remains robust even at -5 dB. Bold values indicate the highest improvements

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Table 2 Performances of the voiced/unvoiced classification

Type of SNR level V/UV (%) UV/V(%)
noise (dB)
Proposed Proposed SWIPE Proposed Proposed SWIPE
algo_MP ? FES algo_MP [47] algo_MP ? FES algo_MP [47]

Airport 10 3.47 4.02 4.73 1.52 2.33 1.09

5 4.09 4.56 5.25 1.44 1.89 0.71
0 4.63 5.73 6.14 1.59 3.46 0.49
-5 4.95 6.10 8.52 1.78 1.67 0.15
Car 10 5.21 7.35 8.37 1.70 3.51 1.03
5 7.38 9.81 11.56 1.64 4.28 0.48
0 9.86 11.57 13.89 1.03 5.43 0.24
-5 10.70 13.93 15.56 1.28 7.98 0.09
Train 10 4.36 5.10 5.01 5.26 5.73 1.95
5 3.28 5.42 6.29 6.91 7.27 1.55
0 3.52 7.53 8.15 4.70 4.85 0.74
-5 2.77 7.11 9.18 3.25 3.73 0.53
Babble 10 3.14 4.39 5.30 4.53 5.82 2.17
5 2.08 7.80 8.52 4.39 7.83 1.68
0 4.36 9.31 10.47 3.07 3.25 0.82
-5 7.95 9.84 11.90 2.18 4.08 0.44
Bold values indicate the highest improvements

based on fuzzy logic is more accurate than the other
algorithms in the voiced classification.
Objective Evaluation Measures
To test the performance of proposed speech enhancement
system, the objective quality measurement tests, signal-to-
noise ratio (SNR), segmental signal-to-noise ratio
(segSNR) and Itakura–Saito distance test were used.
The following equation was computed for evaluation of
SNR results of enhanced speech signals:
  
SNRoutput ¼ 10 log10 r2s r2n ð12Þ
where r2s and r2n are the variances of noise and clean
speech, respectively. We assume that due to the indepen-
dency of speech and noise, the variance of the noisy speech
signal is equal to the sum of the speech variance and noise
variance.
The frame-based segmental SNR is a reasonable mea-
sure of speech quality. It is formed by averaging frame
level estimates as follows:
! and a priori SNR estimator with Ephraim’s (MMSE-
X 1 PNmþN1 2
1M n¼Nm x ð nÞ
segSNR ¼ 10 log10 PNmþN1
M m¼0 ½x~ðnÞ  xðnÞ2
n¼Nm
ð13Þ
where x(n) denotes original speech signal, x~ðnÞ denotes
enhanced speech signal, M denotes number of frames, and
N denotes number of samples in each short time frame.
Itakura–Saito distance measure, based on the dissimilarity
between the original and the enhanced speech, is computed
between sets of linear prediction coefficients (LPC), esti-
mated over synchronous frames. This measure is heavily
influenced by spectral dissimilarity due to mismatch in for-
mant locations, with little contribution from errors in match-
ing spectral valleys. Such behaviour is desirable since the
auditory system is more sensitive to errors in formant location
and bandwidth than to spectral valleys between peaks.
In this paper, the average Itakura–Saito measure (as
defined by the following formula) across all speech frames
of the given sentence was computed to evaluate the speech
enhancement algorithm:
  
ISdða; bÞ ¼ ða  bÞT Rða  bÞ aT Ra ð14Þ
where a is the LPC of original speech signal x(n), b is the
LPC of enhanced speech signal x~ðnÞ, and R is the (Toe-
plitz) autocorrelation matrix.
Our approach is compared to two recent speech
enhancement methods: deep neural network (DNN) [24]
logSTSA) [6].
For noisy speech signals at SNRs (-5, 0, 5, 10 dB),
SNRoutput, segSNR and Itakura–Saito distance (ISd) results
of enhanced speech signals were presented as in Table 3.
Objective quality measurement test results proved the
superiority of the proposed speech enhancement system to the
other popular methods DNN and MMSE-logSTSA estimators.

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Table 3 Performance comparison of different methods in terms of SRNoutput, segSNR, ISd

Type of SNR level SNRoutput SegSNR ISd
noise (dB)
Prop- DNN MMSE- Prop- DNN MMSE- Prop- DNN MMSE-
App [24] logSTSA [6] App [24] logSTSA [6] App [24] logSTSA [6]

Airport 10 15.18 14.93 13.38 4.59 4.45 1.68 7.02 7.36 5.14
5 12.33 11.28 11.65 3.12 3.09 0.24 3.90 3.94 3.71
0 8.45 7.45 8.07 3.04 2.38 -1.45 2.59 2.47 2.50
-5 5.60 4.52 4.98 2.36 2.27 -2.41 1.10 0.86 0.48
Car 10 14.83 14.46 13.76 6.78 6.67 2.93 5.83 5.59 4.93
5 11.67 11.16 11.84 5.01 4.55 1.87 4.78 4.56 3.64
0 8.34 6.44 8.27 4.85 4.09 1.87 4.65 4.10 2.29
-5 5.23 4.07 5.09 2.14 2.33 -2.10 2.35 2.09 0.34
Train 10 13.52 13.81 13.29 7.94 8.18 4.74 5.09 5.35 4.48
5 11.66 12.07 11.54 5.08 5.57 3.43 3.82 4.10 3.23
0 7.75 7.14 7.02 3.71 3.69 2.93 3.54 3.23 2.68
-5 3.42 2.93 2.86 2.32 1.45 0.38 1.89 1.65 1.00
Babble 10 15.59 12.77 14.93 7.28 6.09 3.43 6.02 5.80 5.05
5 12.81 10.24 12.08 5.15 4.85 2.04 4.00 3.84 3.51
0 8.68 6.62 8.45 3.86 3.07 1.09 2.73 2.50 2.25
-5 6.22 4.02 5.83 1.33 0.76 -1.38 1.39 1.17 0.84
Bold values indicate the highest improvements

Table 4 Performance comparison of different methods in terms of PESQ, WSS and LLR measures
Type of SNR PESQ WSS LLR
noise level (dB)
Prop-App DNN MMSE- Prop-App DNN MMSE- Prop-App DNN MMSE-
[24] logSTSA [6] [24] logSTSA [6] [24] logSTSA [6]

Airport 10 2.69 2.60 2.54 52.45 53.50 56.18 0.59 0.83 1.18
5 2.39 2.34 2.33 63.79 67.33 71.02 1.13 1.25 1.36
0 1.93 1.85 1.88 74.36 72.91 82.87 1.27 1.39 1.48
-5 1.56 1.42 1.45 83.22 80.58 89.53 1.59 1.73 1.91
Car 10 2.51 2.46 2.53 45.18 47.61 51.35 0.56 0.53 0.61
5 2.28 2.23 2.25 57.45 59.83 60.79 0.71 0.75 0.78
0 1.98 1.85 1.97 65.28 70.75 73.63 0.88 0.93 1.01
-5 1.76 1.61 1.71 79.52 82.61 87.45 1.12 1.21 1.23
Train 10 2.67 2.43 2.25 49.37 51.53 54.35 0.55 0.59 0.61
5 2.48 2.39 1.94 67.28 69.40 71.22 0.67 0.72 0.79
0 2.23 2.10 1.81 79.16 82.25 85.25 0.81 0.89 1.01
-5 1.97 1.68 1.45 91.36 92.78 97.44 1.02 1.13 1.19
Babble 10 2.64 2.54 2.47 48.43 51.34 57.18 0.98 1.02 1.10
5 2.31 2.27 2.09 66.45 69.57 72.55 1.16 1.29 1.31
0 2.11 1.85 1.78 81.07 83.26 88.34 1.31 1.52 1.59
-5 1.87 1.68 1.32 97.58 99.08 103.56 1.77 1.77 1.82
Bold values indicate the highest improvements

Subjective Evaluation Measures • The perceptual evaluation of speech quality (PESQ)

score is a mean opinion score, showing high
To measure the enhancement quality of the noisy speech, correlations with subjective listening tests. It ranges
we calculate three parameters: from 1 to 5. The higher PESQ score indicates the

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Table 5 Subjective quality test

Type of noise SNR level (dB) Prop-App DNN [24] MMSE-logSTSA [6]
(MOS) results
Airport 10 3.05 2.98 2.78
5 2.39 2.27 2.11
0 2.06 1.96 1.59
Car 10 3.44 3.35 3.06
5 3.32 3.10 2.81
0 2.56 2.43 2.17
Train 10 2.93 2.81 2.72
5 2.56 2.39 2.21
0 2.18 1.87 1.67
Babble 10 3.85 3.78 3.31
5 3.68 3.49 3.04
0 3.03 2.90 2.13
Bold values indicate the highest improvements

higher perceptual quality and the weaker speech
distortions.
• The weighted spectral slope (WSS) is found by
calculating the weighted difference between the adja-
cent spectral magnitudes in each frequency band. The
spectral slope is obtained as the difference between
adjacent spectral magnitudes.
• The log-likelihood ratio (LLR) measure is based on the
difference between the all-pole models of the enhanced
and clean speech. It is mainly related to the similarity of
spectral envelope.
The selection of these measures is based on their cor-
relation with the subjective quality assessments. The defi-
nition of all these measures can be found in [49].
The results of the PESQ scores, WSS and LLR values
are detailed in Table 4.
As it can be seen in Table 4, the proposed approach is
characterized by the highest PESQ scores showing that our
approach has a better perceived quality in the most cases.
However, the MMSE-logSTSA method performs best in
the case of stationary noise. This can be explained by the
fact that the MMSE estimator can provide an accurate
estimation of the noise.
Our approach presents the lowest WSS scores confirm-
ing its performance.
Table 4 shows that our approach provides the best LLR
values. This parameter confirms that the results for the two
methods: DNN and MMSE-logSTSA, are quite similar at
low-input SNR values.
We can conclude again that our approach overcomes the
disadvantage of the DNN, the MMSE or the spectral sub-
traction methods used alone. The three parameters perfor-
mance evaluation shows that our approach is globally the
most suitable for noise elimination.
Informal listening tests are also realized, where a group
of ten listeners (seven women and three men) are asked to
perceptually evaluate six enhanced speech signals from
the NOIZEUS database with three background noises
(white, car and babble) at three SNR levels (0, 5 and
10 dB). The listeners have used the mean opinion score
(MOS) to evaluate the difference between the residual
noise characteristics of the enhanced speech from 1 to 5.
Table 5 presents the statistic results of subjective eval-
uation for four speech enhancement methods.
Conclusions
This paper concerns an approach for speech enhancement
that relies on the harmonic structure of the underlying
speech. The main idea is to apply convenient transforma-
tion on voiced sounds and other specific processing on
unvoiced sounds to eliminate noise. Our V/UV decision
algorithm is based essentially on the multi-scale product
(MP) characteristics. Then, we apply a fuzzy expert system
to determine the most suitable decision to use for a frame
of speech. The voiced frames are filtered by a comb filter,
and the unvoiced sounds undergo a spectral subtraction.
The comb filter is designed with a controllable bandwidth
and is tuned by the estimated fundamental frequency (F0).
The F0 is determined in the voiced frames using our
spectral compression multi-scale product method.
Simulation results show that the proposed approach
yields better results in terms of higher PESQ scores and
lower WSS values than those of compared results. Informal
listening tests also confirms the efficiency of the proposed
approach that results in a better enhanced speech than
obtained by recent methods.

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Future work concerns the addition of fuzzy logic rules to
filter the noisy speech signal in a more cognitive manner.
Acknowledgments The authors would like to thank Dr. A. Abel for
his help throughout the revision of paper by her thesis.
Compliance with Ethical Standards
Conflict of Interest M. A. Ben Messaoud, A. Bouzid and N.
Ellouze declare that they have no conflict of interest.
Informed Consent All procedures followed were in accordance
with the ethical standards of the responsible committee on human
experimentation (institutional and national) and with the Helsinki
Declaration of 1975, as revised in 2008 (5). Additional informed
consent was obtained from all patients for which identifying infor-
mation is included in this article.
Human and Animal Rights This article does not contain any
studies with human or animal subjects performed by the any of the
authors.
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