International Journal of Speech Technology

https://doi.org/10.1007/s10772-017-9488-z

A novel whispered speaker identification system based on extreme

learning machine
J. Sangeetha1 · T. Jayasankar2

Received: 1 September 2017 / Accepted: 26 December 2017

© Springer Science+Business Media, LLC, part of Springer Nature 2018

Abstract
Whispered speech speaker identification system is one of the most demanding efforts in automatic speaker recognition appli-
cations. Due to the profound variations between neutral and whispered speech in acoustic characteristics, the performance
of conventional speaker identification systems applied on neutral speech degrades drastically when compared to whisper
speech. This work presents a novel speaker identification system using whispered speech based on an innovative learning
algorithm which is named as extreme learning machine (ELM). The features used in this proposed system are Instantaneous
frequency with probability density models. Parametric and nonparametric probability density estimation with ELM was
compared with the hybrid parametric and nonparametric probability density estimation with Extreme Learning Machine
(HPNP-ELM) for instantaneous frequency modeling. The experimental result shows the significant performance improve-
ment of the proposed whisper speech speaker identification system.

Keywords  Speaker identification · Whispered speech identification · MFCC · Extreme learning machine

1 Introduction concentrates on whispered speech since it is difficult to be

Nowadays biometrics facilitates the organization to vali-
date the user as divergent to a portion of information (like
a PIN number) or a section of apparatus (like a dedicated
telephone). Biometrics is defined as the statistical investiga-
tion of biological samples and human phenomena. Access
control methods based on human speech progressively turn
into the mainstream because authentication by means of PIN
codes and password is insufficient because it can be sub-
stituted easily. There are numerous usually employed bio-
metric methods, e.g., faces, irises, fingerprints and speech
(Wang et al. 2015; Jain et al. 2004). Speech has the benefits
of easy acquisition, non-intrusiveness, and proper ease of
use. Instead of taking normal speech, this proposed work
* J. Sangeetha
sangita.sudhakar@gmail.com
T. Jayasankar
jayasankar27681@gmail.com
1
Department of IT/SOC, SASTRA Deemed University,
Thanjavur 613409, India
2
Department of ECE, Anna University, BIT Campus,
Trichirappalli 620024, India
duplicated and has elevated protection.
Whispered speech is a precise form of speech that is an
innate alternative speech construction method in verbal
communication. It is most frequently used in public condi-
tions to keep away from being overhead or to keep private
information. For example, a narrator may choose whisper
speech when giving their date of birth, billing address and
credit card number, when making a travelling reservation
over the telephone. It is supposed to whisper in diverse con-
ditions, for example whenever we would like to compose a
discreet or a close atmosphere in discussion, in the library so
as not to bother other group, or while someone attempts to
hide some secret information from other people. Whispered
speech is frequently utilized in criminal activities, particu-
larly in phone conversations where criminals aim to cover up
their identity. Though, apart from the mindful production of
a whisper speech, It may happen due to health issues which
result after rhinitis and laryngitis, but for certain people it
seems like a chronic disease of larynx structures (Jovičić and
Šarić 2008); Ito et al. 2005).
By its temperament and method of making the whisper
speech is considerably dissimilar from standard speech.
There is a significant difference in the spectral domain of
the whisper and neutral speech which created by a loss in

13
Vol.:(0123456789)

expressed excitation constitution and changing of formant
positions with lower frequency (Zhang and Hansen 2007;
Morris and Clements 2002; John 2007; Jovičić 1998; Fan
and Hansen 2011). It is described by the lack of glottal vibra-
tions and a noisy structure. As a result, whispering losses
the fundamental frequency of voice, intonation contours and
the other prosodic features (Huang et al. 2011; Haim et al.
2006). Added to that the whispered segment has a substan-
tially lower energy when compared to standard speech (Ito
et al. 2005). The discussed characters of whispered speech
are a considerable setback in speech technology, particu-
larly in speech recognition, speech synthesis as well as in
the recognition of the speaker. Thus, whispered speech is a
radiating topic in current research (Wang et al. 2007).
2 Related work
In the earlier decade, numerous speaker recognition systems
have been implemented in literature (Li 2001; Mak and
Kung 2000; Campbell et al. 2006a, b; Pellom and Hansen
1998; Poignant et al. 2014; Zhao et al. 2014; Sadjadi and
Hansen 2014; Xu and Zhao 2012; Gu and Zhao 2010; Wang
et al. 2015). Whispered speech segment has various attrib-
utes. It has lower signal-to-noise ratio (SNR) and much
lower energy profile. Besides, the minor frequency formants
are altered to higher frequencies in whispered speech (Jain
et al. 2004).
The remarkable variation between neutral and whispered
speech is shown in Fig. 1. The whispered voice segment has
lower amplitude and requires periodic segments (Wang et al.
2007). Major differences in the speech production method
between neutral and whispered speech are as follows: first,
Fig. 1  Waveforms of neutral
and whispered speech
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International Journal of Speech Technology
lack of harmonic structure or periodic excitation in whis-
pered speech. Second, the positions of lower frequency for-
mants are reallocated to higher frequencies in whispered
speech than that of neutral (Jain et al. 2004). Third, the whis-
pered speech segment’sspectral slope is flatter compared to
neutral speech and the period is longer for whispered speech.
Fourth, F1–F2 frequency space vowel regions boundaries
also differ in whispered speech from neutral speech. Finally,
whispered speech requires lower energy. For these reasons,
conventional neutral speaker identification systems degrade
drastically when experienced with whispered speech.
Related efforts on speaker recognition using whispered
speech have dealt with the setbacks using different methods.
To increase the performance of this work, Jin et al. proposed
(25) the utilization of frame-based score classification and
feature warping techniques. In Gu and Zhao (2010), Chinese
whispered speech speaker identification in the channel mis-
match circumstances has been proposed. To handle channel
variability issues, Factor Analysis (FA) is used. There is a
significantprogress in recognition rate andrequires less com-
putation at training and test time combined FA with SVM.
This is more important in real time identification situations.
An access control system (Wang et  al. 2015), which
is based on whispered speech, has been proposed using
instantaneous frequencies and an approximated probability
product kernel support vector machine (APPKSVM). In the
work, to estimate the probability product kernel, Riemann
sum is utilized.
In general, the learning pace of feed forward neural net-
works is time-consuming and it has been the most important
obstacle in their applicationsfor earlier decades. Two key fac-
torsat the back may be: (1) the learning algorithm which is
based on gradient are broadly used to train neural networks,
International Journal of Speech Technology
and (2) all the parameter are tuned iteratively by utilizing
such learning protocols. Apart from these traditional imple-
mentations, this paper utilizes a new learning algorithm called
extreme learning machine (ELM) which tends to offer good
generalization performance at enormously quick learning
speed.
A novel learning algorithm which is called ELM based
whispered speech speaker identification system has been
proposed in this work. It is dedicated for single-hidden layer
feed forward neural networks (SLFNs). The output weights
have been determined by randomly chosen hidden nodes.
This proposed system utilizes instantaneous frequencies for
feature extraction. Signal independent and signal dependent
filters are used to filter the input speech initially. Speech sig-
nal’s instantaneous frequencies are calculated by applying the
Hilbert transform. The investigated frequencies are continued
to be represented as probability density models. These models
are used as the feature in the speaker identification system. In
this work, we compare the results of MFCC, parametric and
nonparametric probability density models and hybrid models
with ELM. The experimental result reveals the supremacy of
the proposed whisper speech speaker identification system.
3 Proposed system overview
The general block diagram of the proposed identification
method is shown in Fig. 2. It comprises of the following blocks
via signal preprocessing, instantaneous frequency feature
extraction and recognition phase. In modern years for mod-
eling and characterizing speech with different technique have
been proposed. Of specific curiosity here is AM–FM signal
modeling. The AM–FM signal model is used to characterize
a speech resonance represented by.
⎛ t
⎞
⎜ ∫
⎜
ui (t) = ai (t)cos 2𝜋 fi (𝜏)d𝜏 ⎟
⎟
⎝ 0 ⎠
Fig. 2  Block diagram of the
proposed speaker identification
system
This model is mostly used to decompose a speech into
decorrelated band pass channels. In order to each of the sig-
nal characterized as instantaneous amplitude and frequency
with k resonance components.
For process viable speaker identification, execute a Gabor
filter and Emprical mode Decomposition (EMD) are utilized
to evacuate the on the instantaneous frequencies (IF)and
develop useful acoustical features for speaker identification-
from the enter talk signal.Gabor filters are preferred since
they are more compact and smooth in speech analysis in
terms of time and frequency domains. This filter character-
istic promises accurate estimate in instantaneous amplitude
and instantaneous frequency in demodulation phase.In addi-
tion, EMD as signal dependent filter which is adaptive and
appears to be appropriate for non-linear and non-stationary
signal analysis. It was carried in the time domain to form
the basis functions adaptively. The main advantage of EMD
basis functions can be directly derived from the signal itself.
The demodulation phase implemented is based on the Hil-
bert Transform. After works, separated IF is demonstratedas
parametric/nonparametric probability densities and after that
the yield can be utilized to prepare ELM for distinguishing
the whisper speaker.
4 Instantaneous frequency extractions
The significance of the IF stems from the information that
speech is a nonstationary with spectral features that vary
with time. In this work, we depict the initial two divisions
filtering is Gabor filtering and EMD utilized in extracting
the IF of an enter speech signal. Then how the IF signals is
demodulated from the filter utilizing the Hilbert transform
is showed up.
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 International Journal of Speech Technology

4.1 Gabor filtering (3) Repeat the same process as steps 1 and 2, the normal
of the envelopes µıı (t) for ­hı (t) is computed and ­hıı (t)
The Tamil whispered speech samples for developed in is obtained by
recording session with very high quality recording envi-
ronment and apparatus. A little specimen of neutral and
h11 (t) = h1 (t) − 𝜇1 (t).
whispered speech was gathered from an aggregate of 15
subjects. In a quiet room, every speaker peruses sentences (4) The above steps are iteratively repeated until the thresh-
in typical and whispered discourse styles of articulation. old SD. The first IMF ­hıĸ (t) is then generated
For both articulation styles, there are 40 sentences includ-
SD ≥
ing 28 phonetically adjusted sentences and 12 sentences ∑
K
[h1K (t) − h1K −1(t)]∕h21K−1 (t).
from news articles. Discourse information was digitized k=2
utilizing a sample frequency of 16 kHz, with 16 bits for
every case.   Correspondingly, alternate IMFs can be made via
The extract the speech resonance using Gabor filters passing the staying signal rı(t) back to the EMD above
with 400 Hz constant bandwidth and the impulse response
r1 (t) = s(t) − h1K (t).
is given by
  The new speech signal is decayed into the mixture of
g(t) = exp (−𝛼 2 t2 )cos (2𝜋fct) many intrinsic mode functions.
The resultant frequency response is represented by
√ � � 2 � � 2 �� S(t) ≈ h1K (t) + h2K (t) + ⋯ + hNK (t).
2𝜋 −𝜋 (f − f )2 −𝜋 (f + f )2
G(f) = exp exp 4.3 Speech demodulation
2𝛼 𝛼2 𝛼2

where fc is the Center frequency with consistently spaced
on the Hertz scale, α is the Bandwidth control parameter.
4.2 EMD‑empirical mode decomposition
EMD disintegrates a signal into few intrinsic mode func-
tions (IMF). An IMF can be regarded as an AM–FM sepa-
rated component.The EMD decomposes the original signal
into a definable set of adaptive basis of functionscalled the
intrinsic mode functions.The disintegrated signal’s fun-
damental property can be introduced effectively by these
IMFs and this meets the imperative to figure exact IF uti-
lizing Hilbert transform. This work outlines the procedure
In order to calculate the IF of the input speech signals (t)
pass through filter bank and then each bandpass waveform is
demodulated using Hilbert transform or the energy operator.
This work implements to speech demodulation using Hilbert
transform because of less computationally expensive, lesser
error and smoother frequency evaluations. Combination of the
EMD and Hilbert transform are called as the Hilbert-Huang
transform (HHT). The HHT has been widely used in signal
analysis, mechanical fault diagnosis, medical pathology, etc.
The Hilbert transform H (t) is given as
∞
𝜋 ∫ t−𝜏
1 s(𝜏)
H(t) = P d𝜏
−∞

of EMD to the accompanying strides. where P is the Cauchy principle value of the singular inte-
gral, s(t) is the real input signal. The analytical signal func-
(1) Using the spline function the first speech signal, the tion can be calculated as ­Sa(t)
envelopes are enfolded to get u(t) and l(t).
√
(2) Gauge the normal of the envelopes acquired from Sa (t) = s(t) + H(t) = s2 (t) + H 2 (t)ej𝜃(t) ;
step 1.
phase 𝜃(t) = arc tan(H(t)∕s(t))

u(t) + l(t) The IF f (t) would then be able to be registered from the
𝜇1 (t) = phase:
2
1
h1 (t) = s(t) − 𝜇1 (t) f(t) = d𝜃(t)∕d(t)
2𝜋
where s(t) is the original speech signal, h (t) is the inter-
mediate signal, u(t) is the upper envelope, l(t) is the
lower envelope.

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International Journal of Speech Technology

4.4 Probability density modeling of instantaneous where Φ (.)-Kernel function, ɷĸ- IF instance which belongs
frequency to ­Wi; uι, σι -Weight and bandwidth of the ­lth kernel. In
the majority of the caseGaussian function utilized as kernel
This proposed work relates both EMD and multiband function, yielding the above condition:
Gabor filtering. Let Sĸ (t) states the ­kith IMF obtained by � �
the EMD or the signal which is behind leaving through the �
k×l
−‖𝜔 − 𝜔l‖2
F(𝜔) = u𝜄exp
kith band pass filtering thro the Gabor filtering. By means l=1
2𝜎 2
of Hilbert transform, the instantaneous frequency of Sĸ (t),
denoted as ɷĸ (t) is obtained. For the ith frame of filtered For the estimators, this work embraces a fixed bandwidth
signal Sĸ (t), its corresponding instantaneous frequencies kernel which gives the accompanying condition from:
set are described as the following equation:
� �
1 �
k×L
1 −‖𝜔 − 𝜔l‖2
Wi k = {wk[i − 1)L], wk[(i − 1)L + 1], … wk[(i − 1)L + (L1)]} f(𝜔) = √ u𝜄 exp −
k × L 2𝜋𝜎 l=1 2𝜎 2
L is the frame size. Frame I has k sets of frequencies that
is ­Wi = {Wi1, ­Wi2…Wik}. Each W ­ i is then represented as
the PDF by achieving the probability density estimation on 5 Extreme learning machine
it. Two methods used in this proposed work are parametric
density modeling (PDM) and nonparametric density mod- In this paper, a ELM is utilized as a training algorithm is
eling (NPDM). examined on the recognize to whispered speech and achieve
real speaker identification. ELM (Huang et al. 2006) is set
4.4.1 Parametric density modeling up for summed up single hiddenlayerfeed forward networks
(SLFNs) with a wide variability of hidden nodes. The gen-
GMM are frequently used to show the probability den- eral architecture of ELM as shown in Fig. 3. In principle,
sity of complex signal, for example, pictures and sound, this calculation has a tendency to give fast learning speed;
and yields astounding execution in numerous applications algorithm looks much simpler than compared with neural
(Haim et al. 2006). networks and SVM classifier and does not need any iterative
In this effort, GMMs are used to show the probability tuning or setting parameters such as learning rate, momen-
distribution of IF. Given an IF ɷ, the GMM is bidding in tum, etc.
follows:
5.1 Algorithm of ELM

Given a training set N={(xi, ­ti)|xi ∈ ­Rn, ­ti ∈ ­Rm, i = 1,...,N},

where νс is the Mixing parameter. The mixing parameters activation function g(x), and hidden node number N,
ought to fulfill the accompanying requirements:
1. Randomly assign input weight ­wi and bias ­bi, i = 1,...,N.
𝜈 ≥ 0 and
∑
C
𝜈=1 2. Compute the hidden layer output matrix H.
C=1 3. Find thee output weight 𝛽 = H × T,

To get the finest parameters of Gaussian probability where T = [t1, ­tn]T.

density functions, the EM algorithm is used.

4.4.2 Non parametric density modeling

Kernel density estimation (KDE) is a non-parametric den-

sity modeling method that has been utilized to demonstrate
vital issue in different applications (Oyang et al. 2005).
Utilizing theKDE, an IFɷ is demonstrated as takes after:

∑
K×L
f (𝜔) = u𝜄Φ(𝜔;𝜔𝜄, σι)
l=1

Fig. 3  General structure of ELM

13

In this work, the identification performance is evaluated
by testing the following identification systems. (1) MFCC-
GMM based system, (2) PDM based fusion system, (3)
NPDM based fusion system, (4) Hybrid PNP based fusion
system, (5) PDM based ELM, (6) NPDM based ELM, (7)
Hybrid PNP based ELM.
6 Experimental results
The experiments have been carried out over a corpus which
consists of 10,000 Tamil speech samples. Initially voices
are recorded by using mobile phone and recorded audio files
are converted into wave extension file by using Gold wave
software. Sampling rate of speech signal is 44,100 Hz. The
channel is stereo type. The dataset is divided into the devel-
opment corpus and evaluation corpus. The development cor-
pus is utilized for training the structure and fine-tuning the
parameters which consists of 7000 samples. The evaluation
corpus is composed of 3000 speech which is for validation.
Metrics such as accuracy, precision, recall and average pro-
cessing Time are determined from the experiment.
6.1 Evaluation in terms of accuracy, precision, recall
and processing time
• Accuracy is defined as follows
Truepositive + Truenegative
Accuracy =
Truepositive + Truenegative + Forecastpositive + Forecastnegative
Fig. 4  Samples versus Accuracy
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International Journal of Speech Technology
• Precision is the fraction of retrieved instances that are
relevant. It is defined as follows.
 
True positive
Precision =
True positive + Forecast negative
• Recall is the fraction of relevant instances that are
retrieved. It is defined as
Truepositive
Recall =
Truepositive + Forecastnegative
Figures 4, 5, 6 and 7 respectively shows the accuracy,
precision, recall and processing time of the proposed
methods such as MFCC with GMM, Parametric Probabil-
ity Density Modeling, Non Parametric Probability Density
Modeling, HPNP, PDM with ELM, NPDM with ELM and
HPNP with Extreme Learning machine respectively. It is
clearly shows that HPNP with ELM achieves the higher
accuracy, precision; recall rate and lower processing time.
6.2 Overall performance analysis
From Fig.  8, our proposed system that is whispered
speaker identification using ELM has high accuracy, pre-
cision and recall.
From the Table 1, Accuracy, Precision, Recall and Aver-
age Processing Time for ELM is more when compared
International Journal of Speech Technology

Fig. 5  Samples versus Precision

Fig. 6  Samples versus Recall

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 International Journal of Speech Technology

Fig. 7  Samples versus Process-

ing Time

Fig. 8  System performance 100

90

80

70

60
Accuracy
50
Precision
40 Recall
30

20

10

0
MFCC- GMM HPNP PDM-ELM NPDM-ELM HPNP-ELM

Table 1  Comparison between MFCC- HPNP (%) PDM-ELM (%) NPDM-ELM (%) HPNP-ELM (%)
different systems GMM (%)

Accuracy 73.68 80.70 82.56 84.32 90.21

Precision 65.94 71.9 75.670 76.57 83.32
Recall 66.13 72.35 76.32 77.95 84.44

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International Journal of Speech Technology
with Parametric Probability Density Modeling, Non Para-
metric Probability Density Modeling and MFCC- GMM
based system.
7 Conclusion
This work presents an automatic speaker identification
system which is based on whispered speech. The testing
is carried out in the Tamil whispered speech corpus. Sig-
nals are filtered by means of Gabor filter. Then Empirical
Mode Decomposition is transformed utilizing the Hilbert
transform. Extreme Learning Machine is used as classifica-
tion method. We compared the performance of Parametric
Probability Density Modeling, Non-Parametric Probability
Density Modeling with Extreme Learning Machine. From
the experimental results the accuracy of Extreme Learning
Machine is 89.5% which is greater than the other methods
such as Parametric Probability Density Modeling, Non-Par-
ametric Probability Density Modelingbased speaker identi-
fication system. Then the Average processing time of ELM
is 1289.4 ms which is low and compared with other three
methods. The precision of proposed system gets improved by
10.99%. A possible future work is to enhance this framework
to online education for speaker adaptation.
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