biocybernetics and biomedical engineering 42 (2022) 463– 480

Available at www.sciencedirect.com

ScienceDirect

journal homepage: www.elsevier.com/locate/bbe

Original Research Article

Voice disorder classification using speech

enhancement and deep learning models

Mounira Chaiani a, Sid Ahmed Selouani b,*, Malika Boudraa a, Mohammed Sidi Yakoub b
a
Laboratory of Speech Communication and Signal Processing, University of Sciences and Technology Houari Boumediene, Algiers, Algeria
b
Research Laboratory in Human-System Interaction, Université de Moncton, Shippagan Campus, New Brunswick E8S 1P6, Canada

A R T I C L E I N F O A B S T R A C T

Article history: With the recent development of speech-enabled interactive systems using artificial
Received 12 June 2021 agents, there has been substantial interest in the analysis and classification of voice dis-
Received in revised form orders to provide more inclusive systems for people living with specific speech and lan-
28 February 2022 guage impairments. In this paper, a two-stage framework is proposed to perform an
Accepted 4 March 2022 accurate classification of diverse voice pathologies. The first stage consists of speech
Available online 17 March 2022 enhancement processing based on the original premise, which considers impaired voice
as a noisy signal. To put this hypothesis into practice, the noise lestral harmonic-to-
noise ratio (CHNR). The second stage consists of a convolutional neural network with
Keywords:
long short-term memory (CNN-LSTM) architecture designed to learn complex features
Voice disorder classification
from spectrograms of the first-stage enhanced signals. A new sinusoidal rectified unit
CHNR
(SinRU) is proposed to be used as an activation function by the CNN-LSTM network.
Speech enhancement
The experiments are carried out by using two subsets of the Saarbruecken voice data-
CNN-LSTM
base (SVD) with different etiologies covering eight pathologies. The first subset contains
Trigonometric activation function
voice recordings of patients with vocal cordectomy, psychogenic dysphonia, pachyder-
mia laryngis and frontolateral partial laryngectomy, and the second subset contains
voice recordings of patients with vocal fold polyp, chronic laryngitis, functional dyspho-
nia, and vocal cord paresis. Dysarthria severity levels identification in Nemours and
Torgo databases is also carried out. The experimental results showed that using the
minimum mean square error (MMSE)-based signal enhancer prior to the CNN-LSTM net-
work using SinRU, led to a significant improvement in the automatic classification of the
investigated voice disorders and dysarhtria severity levels. These findings support the
hypothesis that using an appropriate speech enhancement preprocessing has positive
effects on the accuracy of the automatic classification of voice pathologies thanks to
the reduction of the intrinsic noise induced by the voice impairment.
Ó 2022 Nalecz Institute of Biocybernetics and Biomedical Engineering of the Polish Academy
of Sciences. Published by Elsevier B.V. All rights reserved.

* Corresponding author at: Information Management dpt, Université de Moncton, 218 Boul. J.-D. Gauthier, Shippagan, New Brunswick
E8S 1P6, Canada.
E-mail addresses: mchaiani@usthb.dz (M. Chaiani), sid-ahmed.selouani@umoncton.ca (S.A. Selouani), mboudraa@usthb.dz
(M. Boudraa), mohammed.sidi.yakoub@umoncton.ca (M. Sidi Yakoub).
https://doi.org/10.1016/j.bbe.2022.03.002
0168-8227/Ó 2022 Nalecz Institute of Biocybernetics and Biomedical Engineering of the Polish Academy of Sciences. Published by Elsevier
B.V. All rights reserved.
464 diabetes research and clinical practice
1. Introduction
The production of speech is a very complex process that
involves the coordination of multiple neuromotor systems
including respiration, phonation and articulation. Any dam-
age to the above systems or any part of them results in artic-
ulation, fluency, or voice disorders [1]. Therefore, commonly
used taxonomy classifies voice disorders according to their
origin, which can be organic, functional or psychogenic in
nature. While functional voice disorders are idiopathic,
organic voice disorders result from underlying structural or
neurological causes. Psychogenic voice disorders are a mani-
festation of one or more types of psychological problems [2].
1.1. Overview of voice disorders
Functional voice disorders usually relate to a misuse of the
voice resulting in an altered voice quality without an affected
physiology of the voice production organs [3].
Organic voice disorders occur when voice production
organs are impacted, and are divided into structural and neu-
rogenic. The structural abnormalities are due to physical defor-
mation of the voice production organs, such as vocal folds
lesions. The neurogenic abnormalities are due to problems
with the nervous system, altering the voice production [3].
Psychogenic voice disorders can originate from one or
more types of psychological processes. These processes can
include depression, the emotional state following a traumatic
or stressful event and anxiety. These psychological states
may induce a partial loss of control of some speech produc-
tion components, leading to disturbances in fluency, stam-
mering and voice changes [4].
The ultimate goal of speech therapists is to improve speech
communication for the benefit of people living with voice dis-
orders who often experience poor self-esteem and embarrass-
ment that can lead to many negative psychological impacts.
The therapist assesses the patients condition and quality of
their voice to accurately identify their voice disorder. This eval-
uation is carried out through auditory perception or by using
medical instruments. Auditory perception is based on the
speech therapist’s analysis of the patient’s voice quality during
a conversation. This subjective description depends on the
experience of the speech therapist and may differ from one
speech therapist to another. However, assessment using
instruments such as a rigid endoscope introduced through
the mouth or a flexible endoscope inserted through the nose
allows visual perception and objective assessment. Another
objective evaluation is the calculation and analysis of acoustic
measurements of a voice signal picked up by a microphone
using a computer. The automation of this method is attracting
increasing attention from researchers due to its noninvasive
nature, which is more comfortable for the patient and offers
a fast detection tool for speech therapists.
1.2. Related work
Several studies address the processing of pathological speech
in various aspects that include speech recognition for people
suffering from speech pathologies. To cope with the high
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0
variability of impaired speech, some recognizers perform
implicit modeling of phonetic variations. An effective
approach consists of using adaptation and representation
learning to improve the recognition rate of dysarthric speech
[5]. However, to overcome the phonetic modeling and adapta-
tion that often require a priori knowledge or transcribed data,
end-to-end machine learning techniques have been devel-
oped in various areas of pathological speech processing
including voice pathology detection and classification.
Detection is a classification task involving two classes. The
first class contains individuals with voice pathology and the
second one contains healthy individuals. Voice pathology
detection systems typically combine several features
extracted from the voice signal and use different machine
learning algorithms to perform the detection. By using a sup-
port vector machine (SVM) as a classifier, [6] evaluated the
statistical measurements of the vocal tract area, [7] examined
the multidimensional voice program parameters and [8]
reported that the 1 to 8 kHz frequency bands contain relevant
information using the entropy and the autocorrelation peak
and its corresponding delay. [9] statistically measured the dis-
crete wavelet transform signals after an empirical mode
decomposition (EMD). [10] also used EMD to propose intrinsic
mode function related features including cepstral coefficient
(IMFCC). [11] performed a fusion of the decisions of an
extreme learning machine, Gaussian mixture model (GMM)
and SVM, which had as input the fusion of MPEG-7 audio
parameters and interlaced derivative pattern parameters.
[12] opted for GMM and processed the spectrum based on
the psychophysical conditions of hearing. The performance
of extreme gradient boosting (XGBoost), DenseNet and Isola-
tion Forest were compared by [13] using at the input a combi-
nation of a raw audio signal, spectrogram, Mel-frequency
cepstral coefficients (MFCC) and acoustic parameters. [14]
compared the performance of classification algorithms SVM,
decision tree, Bayesian classification, logistic model tree, k-
nearest neighbor one (k-NN) and K* with a vector input com-
posed of MFCC, fundamental frequency, perturbation-related
measures and noise-related measures.
The increasing interest and dramatic development of
deep-learning-based approaches have enabled practical
applications in pathological speech processing, achieving
major breakthroughs and greatly improved performance
compared to traditional signal processing methods. For
instance, deep neural network architectures are used to per-
form dysarthric speech analysis and recognition [15]16. In
the voice pathology detection field, the transfer learning
approach was adopted by [17] using VGG16 and CaffeNet as
feature extractors with the SVM as a classifier, by [18] using
ResNet34 and by [19], who performed a connectionist fusion
of the features obtained from three pretrained AlexNet net-
works arranged in parallel.
To train deep neural network (DNN) architectures from
scratch, [20] opted for a CNN-LSTM with a raw speech signal
as input to detect voice pathology and achieved 68.08% accu-
racy on the SVD database. In [21], 71% accuracy was obtained
for the detection of 6 organic pathologies of the SVD database
using a convolutional deep belief network (CDBN) with spec-
trograms as inputs. The authors in [22] chose MFCC as a
 diabetes research and clinical practice
descriptive vector to compare the performance of SVM, GMM
and DNN with three hidden layers, which gave the best accu-
racy. The authors in [23] opted for MFCCs as inputs for the
DNN, which essentially consists of two sparse autoencoders.
A comparison of objective and subjective detection of laryn-
geal cancer was conducted in [24]. The subjective approach
consisted of playing back the voice recordings to 4 volunteers
including 2 trained laryngologists. The objective method was
based on algorithms such as SVM, XGBoost, light gradient
boosted machine (LGBM), artificial neural network (ANN),
one-dimensional convolutional neural network (1D-CNN)
and two-dimensional convolutional neural network (2D-
CNN). The feature extraction provided MFCCs, short-time
Fourier transform (STFT), jitter, shimmer, harmonic-to-noise
ratio (HNR), fundamental frequency and the raw voice signal.
According to their evaluation on a database of 50 males with
laryngeal cancer and 45 healthy males, most of the automatic
methods surpassed the expert diagnosis, which was 69.9%
accurate. The raw signal processed by a 1D-CNN gave the best
result of 85.2%.
It is worth mentioning that most recent studies opted for
pairwise classification, i.e., the classification between two
voice pathologies at the same time (one vs. one or one vs. all
configurations). This pairwise classification was carried out
by the systems presented in [6–8,19] on three common voice
pathologies (cysts, polyps and paralysis) extracted from the
SVD, Arabic voice pathology database (AVDP) and Mas-
sachusetts eye and ear infirmary (MEEI) databases. In [12], the
pairwise approach was applied to classify 5 pathologies (ad-
ductor spasmodic dysphonia, keratosis, vocal fold nodules,
vocal fold polyp and paralysis) of the MEEI database. The
pairwise-based system presented in [9] classified 5 pathologies
(dysphonia, laryngitis, funktionelle dysphonia, rekurrenspars-
ese and hyperfunktionelle dysphonia) of the SVD and 4
pathologies (Alzheimer, Parkinson, chronic laryngitis and
paralysis) of a private database.
In real-life conditions, the pairwise approach is not helpful
in regard to the time to perform an accurate diagnosis. A pre-
cise identification of one pathology that has to be recognized
among many others leads to a multiclass identification,
which is by far a more complicated task.
1.3. Goal and main contributions
In this work, a new approach is proposed to provide an effi-
cient solution that allows the precise multiclass identification
of vocal pathologies. For this purpose, various configurations
are investigated to assess the relevance of a speech enhance-
ment pretreatment and the optimal structure based on deep
neural networks for a multiclass recognition of voice disor-
ders. This is far more complex than the widely used pairwise
approaches. Besides this, we have extended the application of
the proposed system to dysarthria severity levels’ classifica-
tion. The goal is to evaluate the effectiveness of the enhance-
ment approach within a specific speech pathology, namely
dysarthria. Our original contributions are threefold:
(i)to design a two-stage multiclassifier of voice disorders
where the first stage is a speech enhancement module
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 465
and the second stage is a CNN-LSTM that performs the
identification of voice impairments by using two subsets
of the Saarbruecken voice database (SVD) [25] with differ-
ent etiologies covering the following eight pathologies:
vocal cordectomy, psychogenic dysphonia, pachydermia
laryngis, frontolateral partial laryngectomy, vocal fold
polyp, chronic laryngitis, functional dysphonia, and vocal
cord paresis; an automatic classification of dysarthria
severity levels is also performed by using Nemours[26]
and Torgo [27] databases;
(ii)to propose the cepstral harmonic-to-noise ratio (CHNR)
as an innovative estimator of pathological noise to enable
proper functioning of the enhancement module;
(iii)to propose the sinusoidal rectified unit (SinRU) as a new
activation function that is expected to improve the perfor-
mance of the CNN-LSTM classifier.
The remainder of this paper is structured as follows. Sec-
tion 2 introduces the two-stage voice pathology classification,
which is composed of a voice enhancement module based on
the CHNR for noise estimation and a deep neural network.
Section 3 presents the database, evaluation metrics, evalu-
ated speech enhancement algorithms, proposed activation
function, as well as the results of the evaluation on two sets
of pathological voices and dysarthria severity levels. We dis-
cuss the obtained experimental results in Section 4. Finally,
conclusions are outlined in Section 5.
2. Proposed approach: Deep learning of
pathologically enhanced speech
In this section, the proposed voice pathology classification
system is presented. As illustrated by the block diagram
in Fig. 1, there are two main stages that cooperate to
achieve the classification process. The first stage focuses
on enhancing the pathological voice signal based on the
premise that there are some similarities between disordered
voice and noisy speech uttered in adverse conditions.
Unlike the studies dealing with adverse conditions such
those under the channel effect [28] or dealing with an
external source of noise [29], our approach make the
hypothesis that a pathological voice is intrinsically noisy.
This assumption means that the voice impairment induces
a ‘‘noise effect” on the voice produced by a person suffering
from various voice disorders. Based on preliminary observa-
tions and experimental evidence presented in [30,31], the
use of speech enhancement techniques to improve quality
and intelligibility of pathological speech is logically adopted
to reduce the effect of its induced noise. Although the ori-
gin of the degradation differs, we can formulate the
hypothesis that conventional noisy speech results from
the degradation of the environment, while pathological
voice results from the degradation of the speech production
system. During the second stage, the enhanced signal is
projected into the time–frequency domain thanks to a spec-
trogram representation. This 2D representation is used by
the deep learning-based feature learning module to extract
the relevant characteristics allowing the classification of
voice pathologies.
466 diabetes research and clinical practice
Fig. 1 – Classification system for two-stage voice pathologies.
2.1. Enhancement of pathological voice
Speech enhancement aims at reducing the impact of noise
without deteriorating the quality and intelligibility of speech.
Recently, the application of speech enhancement was
extended to improve the robustness of dysarthric speech
recognition systems [30,31] since a positive correlation is
found between intelligibility scores of dysarthric and noisy
speech [32]. To assess the impact of using speech enhance-
ment techniques, our study focuses on dysphonia, which is
characterized by alterations in voice pitch, loudness and qual-
ity. Impaired voice quality can be described as hoarseness
[33]. Hoarseness, quantified by the harmonics-to-noise ratio
[34], is considered to be the class that encompasses breathi-
ness and roughness [35] and is thus technically perceived as
noise or strange tones [36]. The level of hoarseness can be
estimated by evaluating the extent to which noise replaces
the harmonic structure in the spectrogram. The relationship
between the harmonic component and noise was quantified
as the harmonics-to-noise ratio (HNR). This quantifiable indi-
cator can constitute an alternative to the conventional signal-
to-noise ratio (SNR) and therefore offers the possibility of
using speech enhancement techniques to improve the quality
of pathological voice. For the purposes of this study, we use
an adapted HNR developed in the cepstral domain. This cal-
culation is detailed in the context of the minimum-mean
square error (MMSE)-based signal enhancer [37].
Let us consider a noisy voice signal xðiÞ, which is the sum
of a clean voice signal sðiÞ and a noise nðiÞ:
xðiÞ ¼ sðiÞ þ nðiÞ: ð1Þ
The short-term Fourier transform of noisy signal is given by
Xðt; f Þ ¼ Sðt; f Þ þ Nðt; f Þ; ð2Þ
where 0 6 t 6 T and 0 6 f 6 F are the time frame and spectral
component, respectively. To estimate the clean voice signal b
S,
the noisy signal is attenuated by a gain function G.
b ð9Þ
S ðt; f Þ ¼ G  Xðt; f Þ; ð3Þ
The signals are enhanced by the log-MMSE estimator, which
minimizes the mean-square error between the log-
magnitude spectra of the clean and estimated signals, leading
to noise attenuation without distorting the signal too much
[38]. Its spectral gain is given by
" Z #
eðt; f Þ 1 1 ey
Gðe; mÞ ¼  exp dy ; ð4Þ
1 þ eðt; f Þ 2 mðt;f Þ y
eðt; f Þ
mðt; f Þ ¼ cðt; f Þ  ; ð5Þ
1 þ eðt; f Þ
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0
where c and e are the a posteriori and a priori SNRs, respec-
tively, defined by
j Xðt; f Þj2
cðt; f Þ ¼ h i; ð6Þ
E j Nðt; f Þj2
and
h i
E j Sðt; f Þj2
eðt; f Þ ¼ h i; ð7Þ
E j Nðt; f Þj2
Since e cannot be calculated in concrete terms, it is esti-
mated by the decision-directed approach.
2
eðt; f Þ ¼ a  cðt  1; f Þ  Gðt  1; f Þ þ ð1  aÞ
 maxðcðt; f Þ  1; 0Þ; ð8Þ
h i
where a is a smoothing coefficient. The noise E j Nðt; f Þj2 is
estimated from the silent frames of the noisy signal.
The data used throughout our experiments are the sus-
tained vowels/a/ that are extracted from the SVD corpus.
The vowels are characterized by their quasiperiodicity, which
results from vocal cord vibration at a fundamental frequency
F0 . Since these signals do not contain any silence, we replaced
the a posteriori SNR c with a CHNR whose calculation steps,
illustrated by Fig. 2, are detailed below.
The CHNR is a ratio of overall energy to noise energy (in
dB) and is used in evaluating voice quality [34]. It measures
the noise amount in a voice signal, and its calculation is based
on the cepstrum [39]. Segments are first framed with an adap-
tive window of a length five times the fundamental period F10 ,
the nonlinear least squares (NLS) estimator of F0 [40] is used.
Based on the hypothesis that the voice signal can be
expressed as a source signal filtered by the vocal tract, the
log magnitude spectrum of the windowed signal results in
the summation of the logarithms of the source spectra
logXð f Þ and the spectral envelope logHð f Þ, as shown by
Yð f Þ ¼ log j Sð f Þ j
¼ logXð f Þ þ logHð f Þ
Fig. 2 – CHNR calculation steps.
 diabetes research and clinical practice
To separate the noise in the source signal from the har-
monic part and the spectral envelope, a transition to the real
cepstral domain Eq. (10) is required, and a liftering of the cep-
strum is processed.
yðsÞ ¼ IFFTðlog j Sð f Þ jÞ
with s the quefrency.
The lifter consists of the following steps:
 detection of the rahmonics peaks in the cepstrum;
 first derivation of the cepstrum and detection of the sign
change on the two sides of the peak to determine the peak
width;
 zeroing of the rahmonic peak to obtain the comb-liftered
part CðsÞ.
The Fourier transform of the comb-liftered part is an
approximation of the noise spectrum ApNoiseð f Þ, which is
corrected Eq. (13) by subtracting the deviance Dð f Þ Eq. (12)
so that the noise spectrum drops below the harmonic
minima.
ApNoiseð f Þ ¼ FFTðCðsÞÞ
Dð f Þ ¼j minðY ð f Þ  ApNoiseð f ÞÞ j
with ði  1Þ:F0 < f < i:F0 and i P 1
Noiseð f Þ ¼ ApNoiseð f Þ  Dð f Þ
The CHNR is then given by:
 
CHNRb ¼ 20 Y b  Noiseb ;
fs
where 0 6 b 6 2
is used to define a frequency band with fs as
the sampling frequency.
It can be seen that CHNRb is the a posteriori SNR in dB. We
CHNRb
replaced c in Eq. (8) with 10 10 . Once the signals are
enhanced, their representation in the time–frequency domain
is calculated thanks to the spectrogram that is used through-
out all of the experiments. The use of spectrograms is moti-
vated by the two-dimensional nature of this feature
representation and is therefore more suitable for convolu-
tional networks. This representation visualizes the entire
spectral decomposition of the voice signal on a single graph
and is obtained from the squared magnitude of the short-
lived Fourier transform. This latter is the Fourier transform
of a windowed signal by an N-length window shifted
wðn  tÞ across the signal xðnÞ as given by
X
N1
j2nf p
XSTFT ðt; f Þ ¼ xðnÞwðn  tÞe N :
n¼0
The resulting spectrograms are fed into the feature
learner.
2.2. CNN-LSTM architecture for voice
classification
The proposed network, illustrated in Fig. 3, learns to
extract local features from the spectrogram by one 1D con-
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 467
volutional layer with a rectified linear unit (ReLU) activa-
tion function and one maximum pooling layer. Moreover,
it learns to extract the global features using one LSTM
layer. The circled unit in Fig. 3 is magnified in Fig. 4.
ð10Þ The CNN-LSTM combination is then used to classify patho-
logical voices. The application of the 1D convolution on
the spectrogram was not previously used except by [41]
for seizure detection on electroencephalogram signals.
The convolution layer, which aims to learn to extract
local features from its input, includes several filters, also
known as convolution kernels. As shown in Fig. 3, given a
spectrogram I 2 RTF with T frames and F frequency bins,
the convolution layer convolves I with L kernels Kl 2 RMF
Eq. (16). The multitude of kernels allows the creation of
multiple preactivated feature maps Cl 2 RTc 1 , with
Tc ¼ T  M þ 1.
X
M X
F
Cl ðuÞ ¼ convðI; KÞu ¼ Kl ði; jÞIðuþi;jÞ ð16Þ
i¼1 j¼1
Each of the L preactivated feature maps is obtained once the
kernel has been shared by all temporal components of the
spectrogram.
To allow the network to learn complex data, an element-
wise nonlinear function is applied to the preactivated feature
ð11Þ maps.
ð12Þ Al ¼ rðCl Þ ð17Þ
The ReLU function [42], which sets its negative inputs to zero
and preserves the positive input values, allows rapid training
ð13Þ
of the network thanks to its simple function rðCl Þ ¼ maxðCl ; 0Þ
and performs very well in several contexts [43].
To reduce the resolution of the features and increase their
ð14Þ
robustness, the data must pass through a pooling layer. The
max pooling layer [44] performs a nonlinear subsampling of
its input. It calculates the maximum of each nonoverlapped
region of length p along the time axis.
p1 Tc
Pl ðsÞ ¼ maxi¼0 fAl ðsp  iÞg; s ¼ 1; 2;    ; ð18Þ
p
Features that have a very low probability of being active
can be separated using this layer. The resulting feature
maps are fed into an LSTM layer to capture their short
and long-term contextual dependencies. LSTM is a recur-
rent neural network that learns the short and long-term
dependencies of its input sequences. It is composed of G
memory blocks which in turn consist of an activation cell
and an input, a forget and an output gate, as shown in
Fig. 4.
At time t, the current input data Xt and the data from pre-
vious hidden state ht1 are fed to the memory block to update
ð15Þ its state defined by Eq. (23). This is accomplished using Eqs.
(19)–(22).
 
f
Fg ¼ sigmoid Wgf  Xt þ Ugf  ht1 þ bg ð19Þ
 
i
disorder Ig ¼ sigmoid Wgi  Xt þ Ugi  ht1 þ bg ð20Þ
 c
Cg ¼ tanh Wgc  Xt þ Ugc  ht1 þ bg ð21Þ
 o
Og ¼ sigmoid Wgo  Xt þ Ugo  ht1 þ bg ð22Þ
468 diabetes research and clinical practice 4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0

Fig. 3 – Proposed feature learner architecture.

ct ¼ Ig  Cg þ Fg  ct1 ð23Þ
Algorithm 1: Proposed voice disorder classification system
ht ¼ Og  tanhðct Þ ð24Þ Input Pathological voice signals s
Output Optimized parameters of enhanced learner and
where Wg and Ug are weight matrices, bg are bias vectors, and
pathology label for input signal
 is the elementwise (Hadamard) product. Fg, the forget gate
senf = enframe(s, window-size = 32 ms,
output, determines which information from the previous cell frame-shift = 16 ms)
state ct1 to forget and retain. Ig, the input gate output, deter- for each frame of senf do
mines the important information of Cg, the input modulation, Calculate F0 .
after their multiplication. The activation cell, as given by Eq. end for
(23), updates the state of the cell by summing the important foreach 5 periods (F10 ) of s do
Calculate CHNR
information of the input data and the information retained
end for
from the previous cell state. Og, the output gate output, con- senh =logMMSE(s, CHNR)
trols the hidden state of the next cell ht defined by Eq. (24). spec = spectrogram(senh , window = 30 ms, shift = 15 ms)
Some outputs of the LSTM layer are dropped out to prevent Create a set of D mini spectrograms of 36 frames
overfitting [45]. for d 2 D do
To finally classify the pathologies, an output layer with a S = CNN-LSTM(d)
^
b=argmax(S)
number of neurons equal to the B number of classes is ^ = CrossEntropyLoss(b; S)
f(b; b)
required. Using the softmax activation function, the outputs
Backpropagate gradient and update learner parameters
of the output layer are end for
exi F0 is the fundamental frequency. b and b^ are the true label
Si ðxÞ ¼ B ; ð25Þ
X and the predicted one, respectively.
exj
j¼1

with 1 6 i 6 B. From these outputs, a cross-entropy loss given

by Eq. (26) is calculated between the true labels b of the input
data, presented by one-hot vectors, and the predictions S.
X
B
f ¼  bi logðSi Þ ð26Þ
i¼1
The gradient of the obtained loss is backpropagated through
the network to optimize the error by updating the learner’s
parameters.
Algorithm 1 summarizes the processing steps followed by
pathological voice until the classification decision is obtained.
3. Experimental setup and results
In this section, we describe the datasets used in this study,
and we present the unsupervised and supervised enhance-
ment algorithms used for comparison purposes. We also
detail the hyperparameters of the feature learner in addition
to the proposed activation function. Moreover, we present the
results of the pathology classification system on two sets of
pathologies.
 diabetes research and clinical practice
Fig. 4 – Architecture of LSTM memory block.
3.1. Data
The selected datasets used in this study are extracted from a
German voice database freely available online [25]. Developed
by the Phonetic Institute of Saarland University, the Saar-
bruecken Voice Database (SVD) contains approximately 2041
voice recording sessions of 681 healthy and 1019 pathological
individuals. Speakers who were recorded before their pathol-
ogy and those recorded after their recovery were counted in
both healthy and pathological classes. A pathological subject
suffers from at least one of the 71 pathologies mentioned in
the database. The voice recording session consists of the sen-
tence ‘Guten Morgen, wie geht es Ihnen?’ whose English
translation is ‘Good morning, how are you?’, and the vowels
‘a’, ‘i’, and ‘u’ pronounced at different pitches (normal, high,
low and rising-falling). Each voice file is recorded in a
sound-treated room using a Computerized Speech Lab station
(model 4300B) at a sampling frequency of 50 kHz and an
amplitude resolution of 16 bits. The distance between the lips
and the microphone was maintained constant through the
use of a headset condenser microphone (NEM 192.15, Beyer-
dynamic, Heilbronn, Germany) [46]. For a large part of the
voice files, an electroglottographic (EGG) signal is available.
In our experiments, we opted for the sustained vowel/a/ pro-
nounced with a neutral pitch due to its rich harmonic
content.
3.2. Description of selected pathologies
The first selected dataset, presented in Table 1, covers
pathologies based on the following etiology: psychogenic,
organic, and impairment following surgery on the vocal cords.
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 469
This pathology selection allows us to validate the hypothesis
which considers pathological voice as a noisy signal, espe-
cially on a disorder acquired after surgical intervention.
Indeed, in real contexts, an automatic system capable of dis-
tinguishing vocal pathologies of different etiologies is neces-
sary for the future integration of systems based on machine
learning in a medical environment.
Psychogenic dysphonia is a voice alteration of psychologi-
cal etiology such as depression and anxiety. A patient may
present it unconsciously as an organically caused disorder
to avoid confronting emotional conflict, stress or personal
failure [4].
Pachydermia laryngis occurs due to lesions of the ary-
tenoid cartilage, where the vocal cords are attached and
appear like ulcers. These ulcers are usually due to voice mis-
use in the form of repeated sudden glottal attacks[47].
Cordectomy and frontolateral partial laryngectomy are
surgical treatments for glottic cancers [48]. The first is the
removal of a vocal fold or part of it. The second involves
resection of the tumorous vocal fold, the anterior commis-
sure, an anterior portion of the contralateral vocal fold and
an anterior fragment of the thyroid cartilage.1.
By analyzing the first row of Fig. 6, we can visually notice
that the spectrograms belonging to the two pathologies
cordectomy and frontolateral partial laryngectomy contain
more noise than those of the two pathologies pachydermia
laryngis and psychogenic dysphonia. Visual observations of
the first subset waveforms show similarity to noisy speech
signals, leading us to assess the impact of the use of speech
enhancement processing on the automatic classification of
voice pathologies of the first subset.
The Second SVD subset used in this study contains record-
ings of voice disorders based on an etiology related to organic-
structural, organic-neurogenic, and functional impairments.
Hence, the following pathologies have been selected: vocal
cord paresis, vocal fold polyp, chronic laryngitis and func-
tional dysphonia. This second subset has the advantage to
have a balanced number of files with respect to the gender
of speakers. The demographic information about the patients
is given in Table 2. We can note that the number of male and
female is balanced and the effect of age is neutralized since
all groups have the same age average.
3.3. Setup
The spectrogram is calculated with a Hamming window of
30 ms with a shift of 15 ms, resulting in a spectrogram of size
T  750, where 750 represents the frequency bins and T the
number of frames. Each spectrogram is divided into minis-
pectrograms of size 36  750, where 36 represents the number
of frames in the smallest voice signal. This division into min-
ispectrograms allows data augmentation. To classify a voice
signal, the mean of the predictions of its segments is
calculated.
Grids of predefined values are evaluated to obtain the opti-
mal learner parameters. The grid of filters is
1
Cordectomy and laryngectomy are not, strictly speaking, vocal
pathologies (they are acquired disorders) but they have been
considered as such in the SVD database. In this study, we kept the
same nomenclature as that used in SVD
470 diabetes research and clinical practice 4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0

Table 1 – Number of records for each pathology and speakers’ age.

Pathology Female records Average female agestd Male records Average male agestd Total

Cordectomy 2 51.50.71 33 61.366.99 35

Frontolateral partial laryngectomy 0 / 27 57.857.84 27
Pachydermia laryngis 1 610 30 56.3312.32 31
Psychogenic dysphonia 35 51.179.48 12 50.679.39 47

Fig. 5 – Comparison of the objective quality measures for the conventional and proposed MMSE-based enhancement
methods applied to each pathology: (a) short-time objective intelligibility (STOI), (b) signal to noise ratio (SNR), and (c)
perceptual evaluation of speech quality (PESQ).

½4; 8; 16; 32; 64; 128, the grid of the kernel size is ½2; 3; 4, the grid
of the pool size is ½2; 3; 4, the grid of LSTM units is
½50; 100; 150; 200, and the grid of the dropout rate is ½0:4; 0:5; 0:6.
After evaluation, the learner’s convolution layer consists of
32 filters of size 4 with a stride size of 1, allowing for the learn-
ing of 32 feature maps. The kernel size in the max pooling
layer is of 4. A hundred units constitute the LSTM layer, and
the dropout rate is 0.4. The learner is trained over 200 epochs
at a learning rate of 0.0001 using the Adam optimization algo-
rithm [49].
The experiments were implemented by using Python 3.5;
spectrograms were calculated using the Scipy 1.4.1 library;
and the network is constructed using Keras 2.1.6 library on
TensorFlow 2.3.0 platform.
3.4. Evaluation metrics
the number of correctly predicted spectrograms. The sum of
the diagonal elements divided by the total number of tested
spectrograms is the accuracy. The F1 score for each class is
the harmonic mean of the sensitivity and the precision ratios
Eq. (27). For a class B1 , the precision is the ratio of correctly clas-
sified spectrograms B1 and all spectrograms classified as B1 .
The sensitivity is the ratio of the correctly classified spectro-
grams B1 and all B1 tested spectrograms. The mean of the F1
scores of all classes is the macro F1.
precision  sensitivity
F1score ¼ 2 ð27Þ
precision þ sensitivity
Each system is evaluated ten times due to the network’s ran-
dom initialization. The average accuracy of the system over
the 10 iterations and its standard deviation, as well as the
average and standard deviation of the F1 score, are provided.

The confusion matrix, accuracy and macro F1 score are used to 3.5. Impact of the cross-validation design
evaluate the system performance. The rows and columns of
the confusion matrix indicate the true classes and predicted A n-fold cross-validation approach is used in our experi-
classes, respectively. The diagonal of this matrix represents ments. The files of each class are divided into n folds. For each
 diabetes research and clinical practice
Fig. 6 – Vowel/a/ spectrograms of four pathologies and their enhanced versions.
Table 2 – Number of records for each pathology and speakers’ age.
Pathology Female records Average female agestd
Vocal fold polyp 19 53.6815.46
Chronic laryngitis 19 49.9513.87
Functional dysphonia 19 51.4714.3
Vocal cord paresis 19 528.83
of the n iterations, n  1 folds from each class are used for
training, and the remaining fold is used for testing. During
the final iteration, all files are tested. To decide how to evalu-
ate the system, we tested two designs. The first is the 5-fold
cross-validation, motivated by maintaining speaker indepen-
dent experiences, since some speakers have multiple files.
The second is the 10-fold cross-validation. By studying the
details of the two configurations, we noted that the accuracy
of the 10-fold system is more than 8% better than the 5-fold
one. Further investigation has shown that this optimistic
10-fold accuracy is likely due to the presence of files from
the same speaker in the training and test data. Indeed, for
the 10-fold cross-validation, 90% of the voice recordings of
each class are used for training and as a result, some record-
ings of the same speaker may appear in the test set. There-
fore, to keep the experiences independent from the speaker
the 5-fold configuration is used.
3.6. Pathological voice enhancement
To explicitly show the benefits of using CHNR, a direct com-
parison of conventional MMSE (using Eq. (8)) and the modified
MMSE (using CHNR) is provided. The results obtained by using
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 471
Male records Average male agestd Total
19 49.5811.27 38
19 5011.23 38
19 49.2114.88 38
19 49.5812.67 38
standard speech enhancement evaluation metrics, such as
SNR, perceptual evaluation of speech quality (PESQ), and
short-time objective intelligibility (STOI) are reported in
Fig. 5. As shown in Fig. 5 the proposed MMSE-based configu-
ration performed better than the conventional MMSE what-
ever the evaluation metric.
In addition to the unsupervised log-MMSE enhancement
algorithm and for comparison purposes, spectral subtrac-
tion and Wiener filter enhancement methods are used, as
well as a deep speech enhancement generative adversarial
network (DSEGAN) and Large-Deep complex U-Net with
phone-fortified perceptual loss (LDCU–Net-PFPL), which is
a recent supervised deep learning-based enhancement
method [50].
Spectral subtraction is the simplest method of noise
reduction. It consists of subtracting the average power of
the noise from the instantaneous power spectrum of the
noisy signal. This generates musical noise. Hence, the
improvement, which subtracts an overestimation of the noise
[38].
Wiener filtering for speech enhancement applies an opti-
mal linear filter to minimize the mean square error (MSE)
between the clean signal and its estimate [38].
472 diabetes research and clinical practice 4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0

As mentioned in subSection 2.1, the noise in these meth-

Table 3 – Accuracy of CNN-LSTM network using different
ods is estimated from the cepstral domain. voice signals.
The deep speech enhancement generative adversarial net-
work (DSEGAN) [51] is composed of one discriminator D and Signals Methods Accuracy F1 score
two generative networks Gj ; j 2 f1; 2g. The first generator G1, Original / 69.372.38 0.6720.0236
having as input the noisy signal x and a noise nj following a Enhanced MMSE 71.152.23 0.6950.0225
normal distribution Nð0; IÞ, generates an enhanced signal SPECSUB 70.542.18 0.6840.0215
Weiner 70.22.44 0.6840.0291
G1 ðn1 ; xÞ ¼ s^1 whose distribution approaches that of the origi-
SEGAN2 70.562.46 0.6940.0262
nal signal s after each update. The second generator G2
LDCU–Net-PFPL 65.162.88 0.6240.0388
receives as input the enhanced signal s^1 of its predecessor
and the noise n to improve the approximation to the original
signal distribution G2 ðn2 ; s^1 Þ ¼ s^2 . To ensure approximation to
the original signal distribution, the discriminator classifies with, l and m the densities of v and v ^, and g a function belong-
the pair ðs; xÞ as real and the pairs ðs^1 ; xÞ and ðs^2 ; xÞ as false. ing to a set G : fg : Rn ! Rn jkgðaÞ  gðbÞk 6 1ka  bk; 8a; b 2 Rn g
This informs the generators of their errors, which the gener- of all 1-Lipschitz functions.
ators try to reduce at each update. The mathematical expres- Fig. 6 shows spectrograms of the vowel/a/ as pronounced
sion summarizing this is given by by speakers with a cordectomy, pachydermia laryngis, fronto-
lateral partial laryngectomy and psychogenic dysphonia. The
1 spectrograms are of signals without preprocessing, namely,
minVðDÞ ¼ Es;x Pdata ðs;xÞ ðDðs; xÞ  1Þ2
D 2
‘Original’ and enhanced by different enhancement methods:
X2
1    2
þ En P ðnÞ;x Pdata ðxÞ D Gj nj ; ^sj1 ; x ð28Þ ‘MMSE’ for log MMSE, ‘SpecSub’ for spectral subtraction, ‘Wei-
j¼1
4 j n ner’ for Weiner filtering, ‘SEGAN’ for SEGAN2 and ‘LDCU–Net-
PFPL’ for LDCU–Net-PFPL.
X
2
1      2 Table 3 provides the accuracy and F1 score of the CNN-
minVðGÞ ¼ Enj Pn ðnÞ;x Pdata ðxÞ D Gj nj ; ^sj1 ; x  1
G 4 LSTM network using different speech enhancement methods.
j¼1
From the results, the spectral subtraction-based, the
X
2
 
þ kj jjGj nj ; ^sj1  sjj1 ð29Þ Wiener filter-based and the MMSE-based enhancements
j¼1 exhibited a performance improvement. The MMSE-based
enhancement achieved 71:15% accuracy, with an improve-
with E the expectation, ^s0 ¼ x; k1 ¼ 50 and k2 ¼ 100. Eqs. (28)
ment of 1.78%. On the other hand, for the deep learning-
and (29) are the least-square losses of the discriminator and
based enhancement methods, SEGAN2 achieved an accuracy
generators, respectively.
improvement of 1.19%. Whereas, the LDCU–Net-PFPL method,
Each generator has a fully convolutional encoder-decoder
decreased the accuracy by 4.21%.
architecture, and the discriminator is a fully convolutional
Since the best performance is achieved by the MMSE-
network.
based enhancement, and for a more in-depth look, this
DSEGAN is trained with the database used in [51]. The
enhancement is performed considering only the noise esti-
training of DSEGAN requires the availability of both clean
mated over the frequency ranges of 0–1.5 kHz, 0–2.5 kHz,
and noisy signals. This requirement is not verified for patho-
and 0–3.5 kHz. For simplicity, these ranges are denoted as
logical signals. Thus, we introduce a two-stage enhancement.
MMSE 15, MMSE 25 and MMSE 35, respectively. The limitation
The signals are first enhanced by MMSE and considered clean,
to these frequency bands is motivated by an investigation by
and the original signals are considered noisy. By adding the
[52], which concluded that a division into frequency sub-
original and MMSE-based enhanced signals to the training
bands, specifically the 0–3 kHz frequency band containing
data, the SEGAN2 network is obtained. In the test phase, the
the formants, is more suitable for classifying dysphonia com-
enhanced pathological signals are obtained by passing the
pared to the entire frequency band. The results in Table 4
original signals through the trained networks.
show an improved performance in the classification of voice
The Large-Deep complex U-Net with phone-fortified per-
pathologies. Compared to the original signals, the MMSE-
ceptual loss (LDCU–Net-PFPL) [50] uses the short-time Fourier
based enhancement in the 0–1.5 kHz, 0–2.5 kHz, and 0–
transform (complex spectrum) of the noisy signal to generate
3.5 kHz frequency bands yielded improvements of 1.57 %,
a complex ratio mask, which is multiplied by the complex
2.63 %, and 2.44 %, respectively.
spectrum of the noisy signal to obtain the complex spectrum
of the enhanced signal. The obtained spectrum is trans-
3.7. Effect of activation functions
formed into the time domain by the inverse short-time Four-
ier transform. The enhanced signal ^s as well as the clean
Activation functions are primordial blocks in a neural net-
signal s are then represented by latent representations v ^
work, and they allow for the learning of complex forms in
and v, respectively, rich in phonetic information resulting
the data. After a neuron is input, the functions determine
from the passage through a wav2vec encoder. These repre-
whether it should be activated and input to the next layer.
sentations allow to train the enhancement system taking into
To explore the effect of the activation functions on the classi-
consideration the phonetic information contained in the
fication of voice pathologies, and in addition to ReLU, several
speech signal using the perceptual loss, given by Eq. (30).
activation functions are employed: Gaussian error linear unit
Łðs; ^sÞ ¼ jjs  ^sjj1 þ supEl ½gðvÞ  Em ½gðv
^Þ ð30Þ (GELU) [53], considered a smoothed form of ReLU with posi-
 diabetes research and clinical practice 4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 473

Table 4 – Accuracy of CNN-LSTM network using different

voice signals.

Signal Methods Accuracy Macro F1 score

Enhanced MMSE 15 70.942.18 0.6940.0254

MMSE 25 722.1 0.7050.022
MMSE 35 71.812.18 0.7010.0234

tive and negative values; scaled exponential linear unit

(SELU), introduced by [54] to ensure network self-
normalization; and Swish [55], which combines the input
and the sigmoid function. The aforementioned activation
functions, shown in Fig. 7, are compared to the SinRU that
we propose. SinRU is a combination of the ReLU function
and the periodic sinus function, resulting in a monotonic
function. By increasing the inputs, the activation increases Fig. 8 – Proposed activation function plot.
either rapidly in the intervals [0 þ 2np; p2 þ 2np] and
[3p
2
þ 2np; 2p þ 2np] or slowly in the interval [p2 þ 2np; 3p
2
þ 2np]
improvement in performance is observed using our proposed
with n 2 Z. For negative inputs, the function output is 0, as
SinRU activation function. The improvement in accuracy over
illustrated in Fig. 8. The activation functions are listed with
the ReLU function is 1.25%, 1.75% and 1.02% for the original,
their mathematical expressions in Table 5. These are imple-
MMSE 15 and MMSE 35 signals, respectively.
mented within the convolutional layer. Each system is then
Despite the imbalance between the number of male and
trained with different types of activation functions and differ-
female recordings in the first SVD dataset, we investigate
ent input signals, namely, ‘Original’, ‘MMSE 15’, ‘MMSE 25’,
the effect of gender on the systems’ accuracy. Table 7 shows
and ‘MMSE 35’. Table 6 summarizes the performance of the
that for the system with the SinRU activation function and
systems and highlights the best achieved performance.
enhanced signal input, the female overall accuracy is 87.36%
The results show that for each learner, an improvement in
and the male overall accuracy is 66.37%. Compared to the
the classification performance is reported for the enhanced
baseline system where the female and male overall accura-
signals compared to the original signals. This improvement
cies are, respectively, 84.74% and 62.06%, an improvement is
is achieved for all activation functions. The MMSE 25 signals
noticed for both genders.
improved the accuracy of the ReLU and SELU functions by
Tables 8 and 9 present the confusion matrices of the sys-
2.63% and 1.47% in accuracy and 3.3% and 2.3% in F1 score,
tems with the original and MMSE 35 enhanced input signals,
respectively. The MMSE 15 signals improved the accuracy
respectively. It is clear that the pathological voice enhance-
for the swish and GELU functions by 2.08% and 2.2%, respec-
ment improves the discrimination between the two classes
tively, in accuracy, and 2.3% in F1 score. The MMSE 35 signals
of cordectomy and frontolateral laryngectomy. As these two
improved the accuracy for the SinRU function by 2.21% in
classes have a large overlap of speakers, the voice recordings
accuracy and 2.3% in F1 score. When comparing the effect
of a cordectomized speaker are used for network training, and
of the activation functions, for each type of signal, an
the voice recordings of the same speaker who subsequently
underwent frontolateral laryngectomy are used for testing
and vice versa. Based on this observation, an additional
experiment, which consists of merging the data of the two
classes (cordectomy and frontolateral laryngectomy), was car-
ried out. Therefore, the recognition task consists of identify-
ing 3 classes. The results presented in Table 10 shows that
there is a significant improvement when the cordectomy
and frontolateral classes are combined. The accuracy and F1
score for the MMSE 35 system, when using SinRU activation
function, reach 87.38% and 0.854, respectively. These results
are 14.55% and 14.4% higher than the accuracy and F1 score,
respectively, achieved by the same system when the two
classes are distinct.

3.8. Performance with respect of gender

To extend our hypothesis on pathologies based on different

etiologies than those of the first subset while considering
the effect of the gender, additional experiments are per-
Fig. 7 – Activation functions plot. formed by using a second SVD dataset.
474 diabetes research and clinical practice 4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0

Table 5 – Activation functions. Table 8 – Confusion matrix of best-achieving performance

using SinRU activation function with original signals. Cord.:
Activation function Expression Cordectomy, F.l.: Frontolateral laryngectomy, P.l.: Pachyder-
 mia laryngis and P.d.: Psychogenic dysphonia.
ReLU x ifx > 0
ReLUðxÞ ¼
0 else: Predicted classes
Swish SwishðxÞ ¼ 1þexbx , with b=1
 Cord. F. l. P. l. P. d.
SELU bx ifx > 0
SELUðxÞ ¼
baðex  1Þ else; True classes Cord. 20 10 1 4
with b=1.05070098 and a=1.67326324 F. l. 7 19 1 0
 qffiffi 
GELU GELUðxÞ ¼ 2x 1 þ tanh 2
x þ 0:044715x3 P. l. 2 2 22 5
p
 P. d. 2 0 4 41
SinRU x þ sinðxÞ ifx > 0
SinRUðxÞ ¼
0 else:

Table 9 – Confusion matrix of best-achieving performance

using SinRU activation function with MMSE 35 signals.
Cord.: Cordectomy, F. l.: Frontolateral laryngectomy, P. l.:
The baseline system, with ReLU as activation function and
Pachydermia laryngis, and P. d.: Psychogenic dysphonia.
original input signal, is compared to the one with SinRU as
activation function and enhanced input signal. The systems Predicted classes
are evaluated for females exclusively, males exclusively, and Cord. F. l. P. l. P. d.
females and males together. The accuracies and F1 scores
True classes Cord. 22 7 2 4
are provided in Table 11. F. l. 5 20 1 1
From the obtained results, it can be observed that the P. l. 1 1 22 7
pathological voice enhancement as well as the SinRU func- P. d. 1 0 5 41
tion improve the performance of the pathological voice clas-
sification system. An improvement is achieved for the
female, male and gender mixed-based system. For the 0.86% is observed, while the F1 score improves by 1.5%. Nev-
females, the accuracy and F1 score improvements are of ertheless, this performance can be increased by searching for
3.04% and 4.2% respectively. For males, the accuracy and F1 an optimal CNN configuration since we reproduced the expe-
riences with the same system and conditions used with first
score improvements are of 5.36% and 3% respectively. When
subset. The pathological voices are best classified for women
the genders are mixed a little improvement of approximately

Table 6 – Voice disorder classification performance using several activation functions with original and enhanced signals.
The task consists of identifying 4 classes: Cordectomy, Frontolateral laryngectomy, Pachydermia laryngis and Psychogenic
dysphonia. Boldface indicates best performance.
Activation function ReLU Swish SELU GELU SinRU

Original
Accuracy 69.372.38 69.741.67 70.562.24 69.81.62 70.622.27
F1 score 0.6720.0236 0.6780.0189 0.6840.0242 0.680.0173 0.6870.0249
MMSE 15
Accuracy 70.942.18 71.821.99 71.81.66 722 72.691.8
F1 score 0.6940.0254 0.7010.0221 0.7020.0183 0.7030.0228 0.7090.017
MMSE 25
Accuracy 722.1 71.431.49 72.032.28 70.962.01 72.562.01
F1 score 0.7050.022 0.6970.0162 0.7070.0245 0.69400.0211 0.7080.0209
MMSE 35
Accuracy 71.812.18 70.841.97 71.651.36 71.072.04 72.831.14
F1 score 0.7010.0234 0.690.0219 0.6990.0187 0.6970.0205 0.710.0173

Table 7 – Voice disorder classification performance using ReLU and SinRU activation functions with original and enhanced
signals for female and male. Boldface indicates best performance.

Female Male

Configuration ReLU & SinRU & ReLU & SinRU &

Original MMSE 35 Original MMSE 35
Accuracy 84.744.53 87.374.53 62.062.7 66.372.24
F1 score 0.4380.049 0.4540.0472 0.6070.029 0.6430.0248
 diabetes research and clinical practice 4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 475

Table 10 – Voice disorder classification performance using several activation functions with original and enhanced signals.
The task consists of identifying 3 classes: Pachydermia laryngis, Psychogenic dysphonia and a combined class composed of
Cordectomy and Frontolateral laryngectomy. Boldface indicates best performance.
Activation function ReLU Swish SELU GELU SinRU

Original
Accuracy 85.651.32 84.841.13 85.151.30 85.781.35 86.441.18
F1 score 0.830.0224 0.8220.16 0.8270.0205 0.8350.0169 0.8450.0157
MMSE 15
Accuracy 84.631.29 84.981.22 85.411.89 85.840.79 87.320.84
F1 score 0.82700.0168 0.8290.0164 0.8290.0284 0.8410.0104 0.8550.0128
MMSE 25
Accuracy 85.981.42 85.931.22 85.701.92 85.411.27 87.10.9
F1 score 0.840.0195 0.8390.0164 0.8340.0273 0.8330.0168 0.8530.0119
MMSE 35
Accuracy 86.261.56 85.861.24 85.401.25 85.421.56 87.380.64
F1 score 0.83900.0197 0.8380.0172 0.8290.0176 0.8330.0168 0.8540.0102

Table 11 – Voice disorder classification performance using ReLU and SinRU activation functions with original and enhanced
signals. The task consists of identifying 4 classes: Vocal fold polyp, Chronic laryngitis, Functional dysphonia and Vocal cord
paresis. Boldface indicates best performance.
Female and male Female Male

Configuration ReLU & SinRU & ReLU & SinRU & ReLU & SinRU &
Original MMSE 35 Original MMSE 35 Original MMSE 35
Accuracy 48.40.34 49.262.32 58.750.76 61.790.15 51.191.02 56.552.11
F1 score 0.4280.01 0.4430.018 0.5460.011 0.5880.013 0.4850.007 0.5150.019

with an average accuracy of 61.79% and average F1 score of 4. Discussion

0.588.
3.9. Dysarthria severity levels’ classification
Dysarthria is characterized by multiple deviant speech
dimensions, such as perturbation of pitch and loudness,
reduced stress, inappropriate silence, variable rate, harsh
and breathy voice [56]. Both pitch and amplitude are per-
turbed which results to jitter2 and shimmer3 that can be con-
sidered as a modulation noise[39].
To assess the effect of our CHNR-based system on the clas-
sification of the severity levels of dysarthria, an experiment is
conducted with the same experimental conditions as those
deployed in a previous work [57], on Nemours [26] and Torgo
[27] databases. The performance of our CNN-LSTM system
with enhanced spectrograms and those of the systems of
Kadi K. L et al. [57] are presented in Table 12. As reported in
[57], the best accuracies of GMM and GMM-SVM systems are
achieved by using a fusion of auditory cues and MFCCs; the
SVM-based system performed the best by using only MFCCs.
As shown in Table 12, the proposed CNN-LSTM system
improved the performance by 6.14% compared to the best
conventional system, achieving less than 1% error rate in
classifying dysarthria severity levels.
2 (binary task) by a network composed of 2D-CNN and fully
Jitter is the fundamental frequency variation of the voice
signal from cycle to cycle.
3
Shimmer is the amplitude variation of the voice signal from
cycle to cycle.
The proposed CNN-LSTM using the new SinRU activation
function is an innovative architecture designed for voice
pathology classification that competes very well with state-
of-the-art systems. The proposed configuration, composed
of a 1D-CNN layer and an LSTM layer using spectrograms,
achieved an accuracy of 70.62% in the classification of four
voice pathologies (vocal cordectomy, psychogenic dysphonia,
pachydermia laryngis, frontolateral partial laryngectomy).
This can be considered as satisfactory accuracy since the
multiclass identification is a complicated task compared to
the detection (usually, binary) task. This statement was con-
firmed in [58], where four classes, three pathological and
one healthy, were identified using the VGGish network as a
feature extractor; the best accuracy was 41%, whereas when
the task was reduced to a pairwise classification (discriminat-
ing between pathological and healthy voice), the accuracy
increased to 80% for vocal cord paralysis detection using an
LSTM architecture and to 66% and 67% for dysphonia and
laryngitis detection, respectively, using 1D-CNN. Using the
SVD database, the system presented in [20] achieved an accu-
racy of 68.08% in voice pathology detection (binary task) using
a 1D-CNN-LSTM architecture with segmented raw signals as
inputs. [21] used spectrograms to detect organic dysphonia
connected layers pretrained by a convolutional deep belief
network, and achieved accuracies of 71% and 77% with and
without pretraining, respectively.
476 diabetes research and clinical practice
Table 12 – Comparison of the performance of our CNN-LSTM
system with those of Kadi K. L et al. in classifying three
levels of dysarthria severity.
Our system Kadi K. L et al. [57]
Configuration SinRU & SVM GMM GMM-SVM
MMSE 35
Accuracy 99.34 76.6 93.2 78.8
F1 score 0.995 - - - demonstrated the effectiveness of the CHNR-based system
One of prominent advantage of the LSTM component used
within our proposed system is its ability to learn the short
and long-term contextual dependencies by retaining relevant
information from previous state and forgetting irrelevant one.
The importance of contextual information in acoustic model-
ing was demonstrated in pathological speech recognition sys-
tems using DNNs. Usually, the contextual information is
provided by a fixed-size context window. For instance, in
the phonetic variation modeling and speaker adaptation for
dysarthric speech recognition system proposed in [56], the
context window size is 11. [59] opted for a context of size 15
using both acoustic and articulatory features. In [60], the
results showed that accuracies of stuttering detection in dys-
arthric and non-dysarthric speech are improved after using
the fixed-size context window. In contrast to these studies,
where the selection of the best size of the context window
was empirically determined, our approach does not use a
fixed-size context since it implicitly integrates the contextual
information with a changing size of the window thanks to the
cells and gates of LSTM.
To put these results into perspective, it is worthy to men-
tion the effectiveness of other competing techniques, such
as frequency and amplitude tremor measurements that have
been used to characterize utterances of speakers suffering
from Parkinson’s disease [61] and those with voice tremor
[62]. However, in automatic detection of organic voice
pathologies, features based on tremor measurements were
not in the top 3 most contributing parameters that improve
the performance [7]. Instead, using spectrograms seems to
have the advantage to cope with wider range of voice patholo-
gies since it is not restricted in terms of spectrum coverage.
The use of time–frequency representations seems to be an
effective approach for the identification and detection of
pathologies. Indeed, in [29], the authors found that architec-
tures based on DNNs perform better when scatter wavelet
features are used as inputs of Deep Convolutional Neural Net-
works. This latter system achieved 99.62% to classify normal
and pathological voices, which is a binary classification task.
On a more complex task, which consists of classifying three
dysarthia severity levels, our proposed system using time–fre-
quency spectrograms as input features achieved 99.34%
accuracy.
The utterances of the SVD database were recorded in a
sound-treated room. The lips to microphone distance was
assured by the headset condenser microphone. The sampling
frequency is above the minimum required as mentioned in
[63]. Knowing that recording conditions are the same for all
files, the new paradigm, which consists of using the patholog-
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0
ical voice enhancement as preprocessing prior to voice
pathology classification in a two-stage system, is found to
be effective when noise estimation is performed by the CHNR
in the cepstral domain. The CHNR is sensitive to both additive
noise and modulation noise since it decreases with increased
noise or jitter [39]. The CHNR feature related to the additive
noise seems to be effective in discriminating the SVD
pathologies investigated in this study. The results also
in the context of modulation noise, since it achieved the best
performance in classifying dysarthria severity levels.
The speech enhancement module improved the classifica-
tion of voice pathologies using conventional speech enhan-
cers, namely spectral subtraction, Weiner filtering and
MMSE. The MMSE-based enhancement, which use log-
magnitude spectra, gives the best performance using CHNR
to estimate noise. The effect of CHNR in enhancing voice is
measured by objective intelligibility metrics instead of subjec-
tive ones such as in language-dependent experiments carried
out in [64]. Since our experiments are independent of lan-
guage thanks to the use of sustained vocalization (vowel/a/),
the proposed approach based on CHNR seems to be adequate.
Therefore, our study confirms that using sustained vocaliza-
tion is recommended to perform voice pathology analysis as
demonstrated in studies that measured the effects of non-
invasive therapy on organic vocal pathologies [65], as well as
when performing voice tremor measurements [62]. Sustained
vocalization has also been successfully used by machine
learning algorithms to track improvements in voice quality
following a specific treatment based on botulinum injection,
where a significant correlation was found between dysphonia
clinical scores and ANN based scores [66]. These latest find-
ings are promising signs that machine learning algorithms
can be trained to be sensitive and accurate in monitoring
the rehabilitation process of people with voice disorders.
The noise estimation in limited bands increases the per-
formance of voice pathology classification compared to its
estimation in the entire spectrum. Since these bands contain
the harmonics and formants of the vowel/a/, the improve-
ment in the performance can be explained by the wise choice
of estimating the noise in the bands where the harmonics are
corrupted by noise. Our approach confirms the importance of
selecting the adequate frequency bands on the detection and
classification of voice pathologies. The frequency band selec-
tion was also recommended by [8] where focusing on 1–8 kHz
band was found effective. Another experimental investigation
carried out in [67] found that for the Parkinson disease intel-
ligibility classification, the mean absolute deviation in the
cepstral separation difference between source and filter com-
puted between 0–1 kHz and the 4th MFCC, are strongly corre-
lated with the clinical scores. Similarly, the deep structure of
the proposed CNN-LSTM performs a blind selection of spec-
trograms by activating, thanks to SinRu function, relevant
feature maps for the classification task.
When the LDCU–Net-PFPL which is a supervised deep-
learning-based approach is used in the first stage for speech
enhancement, a performance degradation is noticed. This
may be explained by the fact that the system is not well suited
to intrinsic noises like those of the pathological voices, which
explains the appearance of horizontal lines on the enhanced
 diabetes research and clinical practice
spectrograms in Fig. 6. Indeed, the results confirm that the use
of pure deep learning approaches such as LDCU–Net-PFPL and
DSEGAN in the proposed scheme was not as effective as the
combination of conventional and deep-learning-based
enhancement techniques. Recently, this combination has
demonstrated its effectiveness in the case of external noises
[68,69]. In our study, where the noises are assumed to be of
intrinsic origin, the combination of the modified MMSE and
DSEGAN achieved better performance. In the context of patho-
logical signals where pure noisy zones and clean signals are not
available to train deep-learning-based methods, the support
provided by conventional techniques seems to be the best
strategy to face this unfavorable situation.
The experimental comparison between the different activa-
tion functions showed that the proposed SinRU function was
the most efficient. The SinRU activation function improved
the performance of the CNN-LSTM voice pathology classifier
with different enhancement techniques. This is probably due
to the sinusoidal component, which introduces additional
nonlinearity to the positive input part of the function. This
allows better learning of complex and strongly nonlinear pat-
terns from pathological spectrograms and may be the reason
for its efficiency which can lead to an accuracy of 87.38%.
In regard to the good performance achieved by the pro-
posed CNN-LSTM framework for the classification of dysarth-
ric severity levels, it becomes possible to consider its
integration within pathological speech recognition systems
to improve their effectiveness. Indeed, many recent
approaches of pathological speech recognition recommend
to design systems that are well-adapted to the severity level
of the speech disorder. For instance, in [70], a tempo adjust-
ment approach is proposed to perform robust personalized
dysarthric speech recognition. The results presented in this
latter study showed that the system based on phoneme-
based tempo adjustment performs the best for moderate
and severe cases. However, the authors also pointed out the
need of a robust mapping model of dysarthric speech dynam-
ics at the level of every single phoneme. This feature can be
provided by our proposed system that achieved less than 1%
error rate to distinguish between dysarthric severity levels.
The development of assistive technologies for the assess-
ment and therapeutic judgement of voice disorders opens
the way for new opportunities in the field of telemedicine
by providing screening tools that could be used in triage
and pre-diagnosis in a prospect of improving the current
clinical routine. Prospects are also oriented towards the
advantages provided by mobile health (m-health) systems
that can constitute a simple and rapid support for the detec-
tion of vocal pathologies. A flowchart of a possible m-health
system for voice health classification using machine learning
is presented in [14]. In the perspective of integrating the pro-
posed system in telemedicine applications, it would be impor-
tant to consider the aspects related to the signal compression
of the speech signal on the transmission channels. Indeed, as
demonstrated in [71], compression algorithms have notice-
able effects on speech intelligibility by modifying the vowels’
space area. Our system does not use data compression. How-
ever, in a telemedicine context, it is critical to be mindful of
these changes in intelligibility because our approach uses
sustained vowels as units of classification.
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 477
Recent advances in machine learning have provided an
opportunity to automate the diagnosis and assessment of
voice, paving the way to the development of clinically applica-
ble tools that could be used by voice specialists. However,
although the results demonstrate the general viability of
machine learning algorithms for the assistive diagnosis of
voice disorders, it is important to mention that the generaliza-
tion of the results to new patients is not necessarily achieved
nor measured properly because the training is usually per-
formed on limited or non-accessible datasets. Besides the
shortcomings of the existing datasets, the gaps in the adopted
evaluation methodologies and the lack of the standardization
of clinical assessment protocols constitute the main obstacles
to universal comparison and interpretation [72]. Recent
methodological developments show an awareness to create
favorable conditions by establishing guidelines for machine
learning algorithms such those presented in [63], to reach suf-
ficient maturity that will allow reaching full acceptability in
real-life healthcare settings and environments.
5. Conclusions
In this paper, a two-stage multiclass voice pathology classifi-
cation framework was proposed. The first stage carried out
pathological voice enhancement by estimating the noise in
the cepstral domain using CHNR. Supervised and unsuper-
vised speech enhancement algorithms were evaluated,
namely, spectral subtraction, Wiener filtering, log-MMSE,
LDCU–Net-PFPL and DSEGAN. The second stage learned rele-
vant feature maps from the spectrograms using a hybrid
CNN-LSTM neural network. To strengthen the learning of
complex patterns from spectrograms, a new activation func-
tion SinRU was proposed, and the results confirmed its effec-
tiveness. The classification experiments were carried out
considering different etiologies reflected in two subsets of
the SVD database, and dysarthria severity levels on Nemours
and Torgo databases.
The results showed that the proposed framework is
etiology-independent and therefore could be used to perform
a multiclass recognition of voice disorders, which is by far
more complex than the widely used pairwise methods. In
contrast to several approaches where fixed phonemic con-
texts are used, the deep structure of the proposed CNN-
LSTM layer implicitly learns contextual information from fea-
ture maps encompassing relevant spectrogram components.
Our study confirms that the use of sustained vocalization
is an effective way to perform voice pathology analysis as
demonstrated by several studies evaluating the effects of
non-invasive therapy as well as when performing measure-
ments of some pathological voice characteristics. The use of
sustained vocalization also has the advantage of providing
language-independent systems. However, the effect of speech
signal compression should be considered in transmission
channels because they modify the spacial area of vowels.
This work formulates the original hypothesis that a patho-
logical voice is intrinsically noisy. This assumption means
that the voice impairment induces a ‘‘noise effect” in sus-
tained vocalization. The hypothesis is experimentally vali-
dated through the improvement achieved when
478 diabetes research and clinical practice
enhancement of voice segments is used prior to voice pathol-
ogy classification. In the context of pathological signals,
where the noise is considered to be of intrinsic origin, our
study demonstrated the advantage of combining deep-
learning-based methods and conventional speech enhance-
ment techniques to achieve better classification performance.
The high classification accuracy of dysarthria severity
levels achieved by our system opens up many possibilities
in the perspective of its joint use with recent dysarthric
speech recognition systems. In the context of the need for a
robust mapping model of dysarthric speech dynamics that
many research studies have highlighted, our system can be
placed upstream of dysarthric speech recognizers to adapt
their configuration according to severity levels.
Finally, the experimental evidence of this study showed
that the proposed two-stage architecture is effective in per-
forming a multiclass identification of voice disorders. Never-
theless, in light of the thoughts shared by the authors in
[73], we can state that both the features and the conceptual
models used in this study are to a large extent relatively sim-
ple compared to the inexhaustible richness and variability of
voice, speech, and language pathologies. Therefore, to over-
come these limitations, it is necessary to deepen the multidis-
ciplinary collaboration between machine learning, behavioral
signal processing and clinical knowledge.
CRediT authorship contribution statement
Mounira Chaiani: Conceptualization, Writing - original draft,
Writing - review & editing, Visualization, Investigation,
Methodology. Sid Ahmed Selouani: Conceptualization,
Methodology, Software, Writing - review & editing, Supervi-
sion, Validation, Methodology. Malika Boudraa: Conceptual-
ization, Supervision. Mohammed Sidi Yakoub: Data
curation, Software, Validation, Resources.
Declaration of Competing Interest
The authors declare that they have no known competing
financial interests or personal relationships that could have
appeared to influence the work reported in this paper.
R E F E R E N C E S
[1] American Speech-Language-Hearing Association, Definitions
of communication disorders and variations [relevant paper],
available from www.asha.org/policy (1993). URL:https://www.
asha.org/policy/rp1993-00208/..
[2] Martins RHG, do Amaral HA, Tavares ELM, Martins MG,
Gonçalves TM, Dias NH, Voice disorders: etiology and
diagnosis, Journal of voice 30 (6) (2016) 761.e1–761.e9..
[3] American Speech-Language-Hearing Association, Voice
disorders (practice portal), Retrieved (December, 30, 2021).
URL:https://www.asha.org/Practice-Portal/Clinical-Topics/
Voice-Disorders/. .
[4] Baker J. The role of psychogenic and psychosocial factors in
the development of functional voice disorders. Int J Speech-
language Pathol 2008;10(4):210–30.
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0
[5] Chandrakala S, Rajeswari N. Representation learning based
speech assistive system for persons with dysarthria. IEEE
Trans Neural Syst Rehabil Eng 2017;25(9):1510–7.
[6] Muhammad G, Altuwaijri G, Alsulaiman M, Ali Z, Mesallam
TA, Farahat M, Malki KH, Al-nasheri A. Automatic voice
pathology detection and classification using vocal tract area
irregularity. Biocybern Biomed Eng 2016;36(2):309–17.
[7] Al-nasheri A, Muhammad G, Alsulaiman M, Ali Z, Mesallam
TA, Farahat M, Malki KH, Bencherif MA. An investigation of
multidimensional voice program parameters in three
different databases for voice pathology detection and
classification. J Voice 2017;31(1):113.e9–113.e18.
[8] Al-Nasheri A, Muhammad G, Alsulaiman M, Ali Z, Malki
KH, Mesallam TA, Ibrahim MF. Voice pathology detection
and classification using auto-correlation and entropy
features in different frequency regions. IEEE Access
2018;6:6961–74.
[9] Hammami I, Salhi L, Labidi S. Voice pathologies classification
and detection using emd-dwt analysis based on higher order
statistic features. IRBM 2020;41(3):161–71.
[10] Karan B, Sahu SS, Mahto K. Parkinson disease prediction
using intrinsic mode function based features from speech
signal, Biocybernetics and Biomedical. Engineering 2020;40
(1):249–64.
[11] Hossain MS, Muhammad G. Healthcare big data voice
pathology assessment framework. IEEE Access
2016;4:7806–15.
[12] Ali Z, Elamvazuthi I, Alsulaiman M, Muhammad G.
Automatic voice pathology detection with running speech by
using estimation of auditory spectrum and cepstral
coefficients based on the all-pole model. J Voice 2016;30
(6):757.e7–757.e19.
[13] Harar P, Galaz Z, Alonso-Hernandez JB, Mekyska J, Burget R,
Smekal Z, Towards robust voice pathology detection, Neural
Computing and Applications (2018) 1–11. .
[14] Verde L, Pietro GD, Sannino G. Voice disorder identification by
using machine learning techniques. IEEE Access
2018;6:16246–55.
[15] España-Bonet C, Fonollosa JAR. Automatic speech
recognition with deep neural networks for impaired speech.
In: Abad A, Ortega A, Teixeira A, Mateo CG, Hinarejos CDM,
Perdigão F, Batista F, Mamede N, editors. Advances in Speech
and Language Technologies for Iberian
Languages. Cham: Springer International Publishing; 2016. p.
97–107.
[16] Zaidi BF, Selouani SA, Boudraa M, Yakoub MS. Deep neural
network architectures for dysarthric speech analysis and
recognition. Neural Comput Appl 2021:1–20.
[17] Alhussein M, Muhammad G. Voice pathology detection using
deep learning on mobile healthcare framework. IEEE Access
2018;6:41034–41.
[18] Mohammed MA, Abdulkareem KH, Mostafa SA, Ghani MKA,
Maashi MS, Garcia-Zapirain B, Oleagordia I, Alhakami H, AL-
Dhief FT, Voice pathology detection and classification using
convolutional neural network model, Applied Sciences 10
(11) (2020) 3723..
[19] Alhussein M, Muhammad G. Automatic voice pathology
monitoring using parallel deep models for smart healthcare.
IEEE Access 2019;7:46474–9.
[20] Harar P, Alonso-Hernandezy JB, Mekyska J, Galaz Z, Burget
R, Smekal Z, Voice pathology detection using deep
learning: a preliminary study, in: 2017 International
Conference and Workshop on Bioinspired Intelligence
(IWOBI), 2017, pp. 1–4. .
[21] Wu H, Soraghan J, Lowit A, Di-Caterina G, A Deep Learning
Method for Pathological Voice Detection Using Convolutional
Deep Belief Networks, in: Proc. Interspeech 2018, 2018, pp.
446–450. .
 diabetes research and clinical practice
[22] Fang S-H, Tsao Y, Hsiao M-J, Chen J-Y, Lai Y-H, Lin F-C, Wang
C-T. Detection of pathological voice using cepstrum vectors:
A deep learning approach. J Voice 2019;33(5):634–41.
[23] Chen L, Chen J. Deep neural network for automatic
classification of pathological voice signals. J Voice 2020.
[24] Kim H, Jeon J, Han YJ, Joo Y, Lee J, Lee S, Im S. Convolutional
neural network classifies pathological voice change in
laryngeal cancer with high accuracy. J Clinical Med 2020;9
(11):3415.
[25] Pützer M, Barry WJ. Saarbrüecken Voice Database, publisher:
Institut für Phonetik. Universität des Saarlandes (May 2007.
URL:http://www.stimmdatenbank.coli.uni-saarland.de/
help_en.php4.
[26] Menendez-Pidal X, Polikoff JB, Peters SM, Leonzio JE, Bunnell
HT, The nemours database of dysarthric speech, in:
Proceeding of Fourth International Conference on Spoken
Language Processing. ICSLP’96, Vol. 3, IEEE, 1996, pp. 1962–
1965..
[27] Rudzicz F, Namasivayam AK, Wolff T. The TORGO database of
acoustic and articulatory speech from speakers with
dysarthria. Language Resour Eval 2012;46(4):523–41.
[28] Hsu Y.-T, Zhu Z, Wang C.-T, Fang S.-H, Rudzicz F, Tsao Y,
Robustness against the channel effect in pathological voice
detection, CoRR abs/1811.10376 (2018). arXiv:1811.10376. .
[29] Souli S, Amami R, Yahia S. B, A robust pathological voices
recognition system based on DCNN and scattering
transform. Appl Acoust 2021;177 107854.
[30] Bhat C, Das B, Vachhani B, Kopparapu SK, Dysarthric speech
recognition using time-delay neural network based denoising
autoencoder, in: Proc. Interspeech 2018, 2018, pp. 451–455. .
[31] Yakoub MS, Selouani SA, Zaidi B-F, Bouchair A. Improving
dysarthric speech recognition using empirical mode
decomposition and convolutional neural network. EURASIP J
Audio, Speech, Music Processing 2020;2020(1):1–7.
[32] Borrie SA, Baese-Berk M, Engen KV, Bent T. A relationship
between processing speech in noise and dysarthric speech. J
Acoust Soc Am 2017;141(6):4660–7.
[33] Stachler RJ, Francis DO, Schwartz SR, Damask CC, Digoy GP,
Krouse HJ, McCoy SJ, Ouellette DR, Patel RR, Reavis CCW,
Smith LJ, Smith M, Strode SW, Woo P, Nnacheta LC. Clinical
practice guideline: Hoarseness (dysphonia) (update).
Otolaryngology-Head Neck Surgery 2018;158(1_suppl):S1–S42.
[34] Gómez-Garcı́a J, Moro-Velázquez L, Arias-Londoño J, Godino-
Llorente J. On the design of automatic voice condition
analysis systems. part iii: review of acoustic modelling
strategies. Biomed Signal Process Control 2021;66 102049.
[35] Moers C, Möbius B, Rosanowski F, Nöth E, Eysholdt U,
Haderlein T. Vowel-and text-based cepstral analysis of
chronic hoarseness. J Voice 2012;26(4):416–24.
[36] Aronson AE, Bless D. Clinical Voice
Disorders. Thieme: Thieme Publishers Series; 2009.
[37] Ephraim Y, Malah D. Speech enhancement using a minimum
mean-square error log-spectral amplitude estimator. IEEE
Trans Acoust Speech Signal Process 1985;33(2):443–5.
[38] Loizou P. Speech Enhancement: Theory and Practice, Signal
processing and communications. Taylor & Francis; 2007.
[39] de Krom G. A cepstrum-based technique for determining a
harmonics-to-noise ratio in speech signals. J Speech,
Language, Hearing Res 1993;36(2):254–66.
[40] Nielsen JK, Jensen TL, Jensen JR, Christensen MG, Jensen SH.
Fast fundamental frequency estimation: Making a
statistically efficient estimator computationally efficient.
Signal Processing 2017;135:188–97.
[41] Jana GC, Sharma R, Agrawal A. A 1d-cnn-spectrogram based
approach for seizure detection from eeg signal. Procedia
Computer Science 2020;167:403–12.
[42] Nair V, Hinton GE, Rectified linear units improve restricted
boltzmann machines, in: ICML, 2010, pp. 807–814. .
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0 479
[43] Witten IH, Frank E, Hall MA, Pal CJ, Chapter 10 - deep
learning, in: Witten I. H, Frank E, Hall M. A, Pal C.J (Eds.), Data
Mining (Fourth Edition), fourth edition Edition, Morgan
Kaufmann, 2017, pp. 417–466. .
[44] Boureau Y-L, Ponce J, LeCun Y. A theoretical analysis of
feature pooling in visual recognition. In: Proceedings of the
27th international conference on machine learning (ICML-
10). p. 111–8.
[45] Srivastava N, Hinton G, Krizhevsky A, Sutskever I,
Salakhutdinov R. Dropout: A simple way to prevent neural
networks from overfitting. J Mach Learning Res 2014;15
(56):1929–58.
[46] Pützer M, Wokurek W. Electroglottographic and acoustic
parametrization of phonatory quality provide voice profiles
of pathological speakers. J Voice 2021.
[47] Sasaki CT, accessed: 2021-05-07. URL:https://www.
merckmanuals.com/fr-ca/professional/affections-de-l-
oreille,-du-nez-et-de-la-gorge/troubles-laryngiens/ulc%C3%
A8res-de-contact-du-larynx. .
[48] Mouawad F, Chevalier D, Santini L, Fakhry N, Bozec A,
Espitalier F. Chapitre 9 - traitement chirurgical par
cervicotomie et reconstruction laryngée. In: Barry B, Malard
O, Morinière S, editors. Cancers du Larynx. Paris: Elsevier
Masson; 2019. p. 89–115.
[49] Kingma DP, Ba J, Adam: A method for stochastic optimization
(2017). arXiv:1412.6980. .
[50] Hsieh T-A, Yu C, Fu S-W, Lu X, Tsao Y, Improving Perceptual
Quality by Phone-Fortified Perceptual Loss Using Wasserstein
Distance for Speech Enhancement, in: Proc. Interspeech
2021, 2021, pp. 196–200..
[51] Phan H, McLoughlin IV, Pham L, Chén OY, Koch P, Vos MD,
Mertins A. Improving gans for speech enhancement. IEEE
Signal Process Lett 2020;27:1700–4.
[52] Pouchoulin G, Fredouille C, Bonastre J-F, Ghio A, Giovanni A,
Frequency study for the characterization of the dysphonic
voices, in: Proc. Interspeech 2007, 2007, pp. 1198–1201..
[53] Hendrycks D, Gimpel K, Bridging nonlinearities and
stochastic regularizers with gaussian error linear units, CoRR
abs/1606.08415 (2016). arXiv:1606.08415. .
[54] Klambauer G, Unterthiner T, Mayr A, Hochreiter S, Self-
normalizing neural networks, in: Proceedings of the 31st
international conference on neural information processing
systems, 2017, pp. 972–981. .
[55] Ramachandran P, Zoph B, Le QV, Searching for
activation functions, CoRR abs/1710.05941 (2017).
arXiv:1710.05941. .
[56] Kim M, Kim Y, Yoo J, Wang J, Kim H. Regularized speaker
adaptation of kl-hmm for dysarthric speech recognition. IEEE
Trans Neural Syst Rehabil Eng 2017;25(9):1581–91.
[57] Kadi KL, Selouani SA, Boudraa B, Boudraa M. Fully automated
speaker identification and intelligibility assessment in
dysarthria disease using auditory knowledge. Biocybern
Biomed Eng 2016;36(1):233–47.
[58] Guedes V, Teixeira F, Oliveira A, Fernandes J, Silva L, Junior A,
Teixeira JP. Transfer learning with audioset to voice
pathologies identification in continuous speech. Procedia
Computer Science 2019;164:662–9.
[59] Yilmaz E, Mitra V, Bartels C, Franco H, Articulatory features
for ASR of pathological speech (2018). arXiv:1807.10948..
[60] Oue S, Marxer R, Rudzicz F, Automatic dysfluency detection
in dysarthric speech using deep belief networks, in:
Proceedings of SLPAT 2015: 6th Workshop on Speech and
Language Processing for Assistive Technologies, Association
for Computational Linguistics, Dresden, Germany, 2015, pp.
60–64..
[61] Brückl M, Ghio A, Viallet F. Measurement of tremor in the
voices of speakers with parkinson’s disease. Procedia
Computer Science 2018;128:47–54.
480 diabetes research and clinical practice
[62] Suppa A, Asci F, Saggio G, Leo PD, Zarezadeh Z, Ferrazzano G,
Ruoppolo G, Berardelli A, Costantini G. Voice analysis with
machine learning: One step closer to an objective diagnosis
of essential tremor. Mov Disord 2021;36(6):1401–10. [69]
[63] Rusz J, Tykalova T, Ramig LO, Tripoliti E. Guidelines for speech
recording and acoustic analyses in dysarthrias of movement
disorders. Mov Disord 2021;36(4):803–14.
[64] Nilsson C, Nyberg J, Strömbergsson S. How are speech sound [70]
disorders perceived among children? A qualitative content
analysis of focus group interviews with 1011-year-old
children. Child Language Teaching Therapy 2021;37
(2):163–75.
[65] Lin F-C, Chien H-Y, Kao Y-C, Wang C-T. Multi-dimensional [71]
investigation of the clinical effectiveness and prognostic
factors of voice therapy for benign voice disorders. J Formos
Med Assoc 2021.
[66] Suppa A, Asci F, Saggio G, Marsili L, Casali D, Zarezadeh Z, [72]
Ruoppolo G, Berardelli A, Costantini G. Voice analysis in
adductor spasmodic dysphonia: Objective diagnosis and
response to botulinum toxin. Parkinsonism Related
Disorders 2020;73:23–30.
[67] Khan T, Westin J, Dougherty M. Classification of speech
intelligibility in parkinson’s disease. Biocybern Biomed Eng [73]
2014;34(1):35–45.
[68] Zezario RE, Huang J-W, Lu X, Tsao Y, Hwang H-T, Wang H-M,
Deep denoising autoencoder based post filtering for speech
4 2 ( 2 0 2 2 ) 4 6 3 –4 8 0
enhancement, in: 2018 Asia-Pacific Signal and Information
Processing Association Annual Summit and Conference
(APSIPA ASC), 2018, pp. 373–377. .
Zhang Q, Nicolson A, Wang M, Paliwal KK, Wang C.
Deepmmse: A deep learning approach to mmse-based noise
power spectral density estimation. IEEE/ACM Trans Audio,
Speech, Language Processing 2020;28:1404–15.
Xiong F, Barker J, Christensen H. Phonetic analysis of
dysarthric speech tempo and applications to robust
personalised dysarthric speech recognition. In: ICASSP 2019–
2019 IEEE International Conference on Acoustics, Speech and
Signal Processing (ICASSP). p. 5836–40.
Utianski RL, Sandoval S, Berisha V, Lansford KL, Liss JM. The
effects of speech compression algorithms on the
intelligibility of two individuals with dysarthric speech. Am J
Speech-Language Pathology 2019;28(1):195–203.
Patel RR, Awan SN, Barkmeier-Kraemer J, Courey M, Deliyski
D, Eadie T, Paul D, vec JG, Hillman R. Recommended protocols
for instrumental assessment of voice: American speech-
language-hearing association expert panel to develop a
protocol for instrumental assessment of vocal function. Am J
Speech-Language Pathology 2018;27(3):887–905.
Corcoran C, Cecchi G. Using language processing and speech
analysis for the identification of psychosis and other
disorders. Biological Psychiatry: Cognitive Neuroscience
Neuroimaging 2020;5(8):770–9.