INDIVIDUAL VOICE ACTIVITY DETECTION USING PERIODIC TO APERIODIC

COMPONENT RATIO BASED ACTIVITY DETECTION (PARADE) AND GAUSSIAN

MIXTURE SPEAKER MODELS

Naotoshi Seo

University of Maryland
ENEE632 Speech Processing
Final Project

ABSTRACT In section 3, several speaker identification methods are in-

This paper describes a study of individual voice activity detec-
tion (IVAD) which detect speech regions of an interest person
in audio. An IVAD method can be constructed by cascading
a voice activity detection (VAD) method and a speaker iden-
tification method. Therefore, various VAD and speaker iden-
tification methods were studied. Especially, a VAD method
called PARADE [10] and a speaker identification method based
on GMMs [21] are deeply investigated. The PARADE method
is a noise robust VAD method utilizing the periodic compo-
nent to aperiodic component ratio (PAR). The GMM based
speaker identification method is one of most widely used speaker processing wherein the presence or absence of human speech
identification methods which represents the feature space as a
mixture of Gaussian distributions. Finally, a semi-automatic
IVAD system is proposed which cascades the PARADE and
the GMM based speaker identification and the performance is
examined by experiments.
Index Terms Voice Activity Detection, Speaker Identi-
fication, Text Independent Speaker Identification, PARADE,
GMM
1. INTRODUCTION
Individual voice activity detection (IVAD) is an algorithm to
detect speech regions of an interest person in audio. In con-
trast, voice activity detection (VAD) is an algorithm wherein
the presence or absence of human speech is detected in re-
gions of audio. VAD is important to help speech processing
techniques such as speech enhancement [19], speech coding
[25], and automatic speech recognition [13]. IVAD is also im-
portant because it enables to enhance these speech processing
techniques by specializing to a person of interest.
An IVAD method can be constructed by 1) extracting voice [2]. The performance of the each part seriously influences
activity segments in audio and 2) classifying each segment
into speakers and picking segments of an interest person. There-
fore, VAD and speaker identification methods are studied. In
section 2 several VAD methods are introduced, and a noise
robut VAD method, PARADE [10], is deeply investigated.
troduced, and one of most widely used speaker identification
methods, the based speaker identification [21], is further ex-
amined. In section 4, an IVAD system is proposed and exam-
ined. It is constructed by cascading PARADE and the GMM
based speaker identification. In section 5, conclusion of this
study and outlines of future works are provided.
2. VOICE ACTIVITY DETECTION
Voice activity detection (VAD) is an algorithm used in speech
is detected in regions of audio. VAD is very important for
speech communication applications such as speech recogni-
tion, hands-free telephony and speech coding. When noise-
free speech is acquired, a proper threshold set in the signal
level allows relatively easy detection of the speech period.
However, real speech is distorted by background noise such as
computer-fans, air-conditioners and many other environment
sounds, especially in distant-talking situations. Inaccurate de-
tection of the speech period causes serious problems such as
degradation of recognition performance and deterioration of
speech quality. It is therefore highly desirable to develop a
robust and reliable VAD method.
In general, VAD consists of two parts: an acoustic fea-
ture extraction part, and a decision mechanism part. The
former extracts acoustic features that can appropriately indi-
cate the probability of target speech signals existing in ob-
served signals, which also include environmental sound sig-
nals. Based on these acoustic features, the latter part finally
decides whether the target speech signals are present in the
observed signals using, for example, a well-adjusted thresh-
old [16], the likelihood ratio [24], and hidden Markov models
VAD performance.
The short-term signal energy and zero-crossing rate [20]
have long been used as simple acoustic features for VAD.
However, they are easily degraded by environmental noise,
and environmental sounds also possess a similar energy and
zerocrossing rate to speech signals. To cope with this prob-
lem, various kinds of robust acoustic features have been pro-
posed for VAD such as autocorrelation function based fea-
tures [1], spectrum based features that utilize harmonicity [23][29],proposed by Sohn et al. [24] utilizes likelihoods derived from
pitch based features [26], the power in the band-limited re-
gion [16], mel-frequency cepstral coefficients [14], features
based on higher order statistics [15], and periodic to a peri-
odic component ratio (PAR) [9]. In this project, I examine
a voice activity detection method using PAR features called
PARADE [10].
2.1. Periodic to Aperiodic Component Ratio based Activ-
ity Detection (PARADE)
Sound signals can be decomposed into their periodic and ape-
riodic components. For example, speech signals consist not
only of periodic signals such as steady parts of vowels and
voiced consonants, but also of non-periodic signals such as
fluctuations included in vowels, voiced consonants, stops, frica-
tives, and affricates. With regard to psychoacoustics, findings
derived from concurrent vowel identification experiments sug-
gest that the human auditory system can suppress harmonic
interferers and perceive the residual target signal [3]. This
finding suggests that the human auditory system may process
both the harmonic (periodic) component and the residue after
canceling the harmonicity (aperiodic) component, which de-
viates from the dominant periodicity. In terms of automatic
speech recognition, the word accuracy in noisy environments
can be improved by using the periodic and aperiodic compo-
nents of observed signals [11][9]. The PARADE [10] method
utilizes an acoustic feature that represents the power ratios of
the periodic and aperiodic components in observed signals.
This feature is referred to as the periodic component to aperi-
odic component ratio (PAR).
Now, let us define the problem that the PARADE method
solve. Observed signals are recorded monaurally, and there
is only one dominant sound, i.e. the target speech, which is
in the presence of background noise whose frequency spectra
distribute widely over all frequencies. There is no assumption
as regards the stationarity of the noise power, thus the power
of noise changes dynamically. In addition, there is no prior
knowledge about the kind of background noise.
To cope with this problem, the PARADE method decom-
poses observed signals into their periodic and aperiodic com-
ponents. In details, the term aperiodic component includes
both environmental noise and aperiodic components of speech
signals, and the term periodic component includes a domi-
nant harmonic component in the observed signal. Because
the PAR of noise is normally independent of the power of the
noise, voice activity detection based on the PAR is expected
to be insensitive to any dynamic change in the SNR. Section
2.2 describes the decomposition method performed using sub-
band signals in the frequency domain.
The VAD method then detects the existence of speech
based on the statistics of the PARs, which represent the differ-
ence between periods that contain only noise and periods that
contain both noise and speech. The statistical VAD method
posteriori SNRs, whereas the PARADE method calculates like-
lihoods derived from probability distributions of the PARs
without knowledge of the SNR. Section 2.3 describes the like-
lihood calculation in detail.
2.2. Decomposition of periodicity and aperiodicity
In this section, how to decompose sound signals into their
periodic and aperiodic components is explained. Let us define
the short time power of observed signal at frame n by
L
X
(n) = |s(t)g(t)|2 , (1)
t=0
that is, the 0th-order autocorrelation coefficient where s(t) is
the observed signal of length L at the frame n and g(t) is
the symmetric analysis window of length L such as hanning
window. Since the autocorrelation coefficients equal to the
inverse Fourier transform of the power spectrum, the short
time power can be also obtained as
M 1
1 X
(n) = |S(n, m)|2 (2)
M m=0
where S(n, m) is an STFT representation of s(t) at a fre-
quency bin of m [0, M 1] and a temporal frame n.
Let us assume that an observed signal s(t) can be de-
scribed as the sum of its periodic and aperiodic components,
sp (t) and sa (t), respectively.
s(t) = sp (t) + sa (t) (3)
Hereafter, let us denote STFT representations of the above
signals by S(n, m), Sp (n, m), and Sa (n, m), respectively,
and their short time power in a time frame by (n), p (n),
and a (n) respectively. Now, we assume the additivity on the
powers of the components as
|S(n, m)|2 = |Sp (n, m)|2 + |Sa (n, m)|2 (4)
Then the following equation can be derived.
(n) = p (n) + a (n) (5)
Let us denote the fundamental frequency (F0) of the periodic
component and the number of harmonics at frame n by f0 (n)
and (n), respectively, and an operator to transform the k-th
harmonic frequency kf0 (n) to an index of a frequency bin in
the corresponding frequency domain by [kf0 (n)]. Now, we
have (n) = b [f0M
(n)] c.
 Under assumption that harmonic components of observed When Hn = 0, i.e., there is no speech signal in the ob-
signal can be calculated as sum of power of pure tones, we served signal, assuming (n) = a (n), we can estimate the
can obtain the following equation error of an aperiodic component a(n) as

(n) a (n) = (n) a (n) = p (n). (12)

X
(p) = |Sp (n, [kf0 (n)])|2 (6)
k=1 Assuming that the error distribution follows a Gaussian distri-
PL1 PL1 bution whose mean and standard deviation are 0 and a (n)
where = 2 t=0 g(t)2 /( t=0 g(t))2 . In addition, we can with positive constant . Then, the likelihood of the observed
introduce the assumption that means of the average power of signal for non-speech periods can be modeled by
the aperiodic components at frequencies of the dominant har-
monic components is equal to that over the whole frequency 2 !
1 1 p (n)
rage, that is, p((n)|Hn = 0) = exp 2 .
2 a (n) 2 a (n)
M 1 (n) (13)
1 X 1 X In contrast, for the case Hn = 1, i.e. a speech signal
|Sa (n, m)|2 = |Sa (n, [kf0 (n)])|2 . (7)
M m=0 (n) exists in the observed signal, we can estimate the error of the
k=1
periodic component p (n) as
Under these assumptions, consequently, we can obtain the
following equations. p (n) = (n) a (n) p (n). (14)

M 1
1 X Because we cannot know the true value of a (n), we consider
(n) = |S(n, m)|2 (8) an ideal case of a (n) = 0. Here, p (n) becomes
M m=0

P(n) p (n) = (n) p (n) = a (n). (15)

k=1 |S(n, [kf0 (n)])|2 (n)(n)
p (n) = (9) Under this assumption, we again assume that the error distri-
1 (n)
bution follows a Gaussian distribution whose mean and stan-
a (n) = (n) p (n) (10) dard deviation are 0 and p (n) with positive constant .
Then, the likelihood of the observed signal for non-speech
where p (n) and a (n) indicate estimated values of p (n)
periods can be modeled by
and a (n).
This method requires F0 value, and so we will estimate 2 !
1 1 a (n)
it as the value that maximized the numerator of equation 9, p((n)|Hn = 1) = exp 2 .
that is, we determine the estimate f0 (n) by searching the fre- 2 p (n) 2 p (n)
quency range that includes the F0 of human speech (e.g. from (16)
50 to 500Hz) as follows: Finally, we calculate the likelihood ratio L(n) at frame n
as
(n)
X p((n)|Hn = 1)
L(n) = (17)
f0 (n) = arg max |S(n, [kf0 (n)])|2 (n)(n) p((n)|Hn = 0)
f0 (n)
k=1
If the likelihood is higher than a threshold decided by users
(11)
of this detector, we decide that a speech signal exists in the
This equation coincides with one adopted by a robust F0 es-
frame. The threshold would be determined as
timator known as REPS [18], therefore, this equation provide
us fairly reliable F0 estimates. p(Hn = 0)
= (18)
p(Hn = 1)
2.3. Likelihood Ratio Test
where p(Hn = 1) and p(Hn = 0) denotes the probability
If the decomposition can ideally estimate the powers of peri- of existence and non-existence of speech in the sound signal
odic components, we can detect speech signals based solely respectively. A user should be able to guess the probability of
on these estimates. However, the decomposition cannot com- existence of speech in the sound signal by looking a plot or
pletely avoid power estimation errors, therefore, statistical de- listening the sound signal. The guess is not necessary to be
tection method as [24] is required. accurate, rather, the user may adjust the probability parame-
Let us presume the state of the existence of speech signals ter, e.g., he may put larger probability of existence than his
(1 or 0) at frame n to be random variables, and denote them initial guess into the VAD detector when he want to decrease
by Hn . the probability of miss detection.
 0.5 0.5
(a)

(a)
0 0
0.5 0.5
1 2 3 4 5 6 1 2 3 4 5 6
4 4
x 10 x 10
0.5 0.5
(b)

(b)
0
0.5 0.5
0 1 2 3 4 5 6 0 1 2 3 4 5 6
4
x 10 4
x 10

Fig. 1. The first test signal (a) Speech waveform in the silence Fig. 3. The second test signal in the presence of real world
(b) Noisy speech waveform created by adding white noise at street noise (a) Speech waveform in the silence (b) Noisy
an SNR of 0dB speech waveform created by adding street noise at an SNR
of 0dB

Power
(a)

2
0 5
Power

(a)
0 10 20 30 40 50 60 70 80
0
Power of Periodic Part 0 20 40 60 80 100
(b)

2
5
0 Power of Periodic Part

(b)
0 10 20 30 40 50 60 70 80
0
Power of Aperiodic Part 0 20 40 60 80 100
(c)

2
0 5
Power of Aperiodic Part

(c)
0 10 20 30 40 50 60 70 80
8 0
6 log likelihood ratio 0 20 40 60 80 100
(d)

4
2
0 8
6 log likelihood ratio
(d)

0 10 20 30 40 50 60 70 80 4
1.5 2
1 0
VAD Result 0 20 40 60 80 100
(e)

0.5
0 1.5
0.5 1 VAD Result
(e)

0 10 20 30 40 50 60 70 80 0.5
0
0.5
0 20 40 60 80 100

Fig. 2. VAD results for a speech signal in the presence of

white noise shown in Fig. 1(b). (a) Power of signal (b) Power
of periodic components (c) Power of aperiodic components
(d) Log likelihood ratios derived from the periodic and aperi-
odic components (e) VAD result
2.4. Experimental Results
The experimental results of the PARADE method are pre-
sented here.
The first test signal was created by adding white noise to a
speech data shown in Fig. 1(a) at an SNR of 0dB (Fig. 1(b)).
The result of decomposition of the periodic/aperiodic compo-
nents is shown in Fig. 2(a)-(c). It shows that the PARADE
method well decomposes periodic and a periodic components
well even at such a low SNR. In addition, the log likelihood
ratios is shown in Fig. 2(d). It correctly indicated the period
in which speech signals exist. The VAD result based on the
likelihood ratio test is shown in Fig. 2(e).
The second test signal was created by adding noise in the
real world. I added a street noise to a speech data at an SNR of
0dB. The street noise includes non-stationary sounds such as
sounds of passing cars. The generated signal is shown in Fig.
3(b). The result of decomposition of the periodic/aperiodic
components is shown in Fig. 4(a)-(c). The log likelihood ra-
Fig. 4. VAD results for a speech signal in the presence of real
world street noise shown in Fig. 3(b). (a) Power of signal (b)
Power of periodic components (c) Power of aperiodic compo-
nents (d) Log likelihood ratios derived from the periodic and
aperiodic components (e) VAD result
tios and VAD result based on the likelihood ratio test is shown
in Fig. 4(d)-(e). This result shows that the PARADE method
worked well in the presence of non-stationary noise.
3. SPEAKER RECOGNITION
The next step of the individual voice activity detection is to
pick speech frames of an interest person from voice activ-
ity region frames obtained by a VAD method. To perform
this task, we can use speaker recognition methods to identify
speakers of each frame.
Speaker recognition is the process of automatically rec-
ognizing a speaker by using speaker-specific information in-
cluded in speech waves [12]. This technique can be used to
verify the identity claimed by people accessing certain pro-
tected systems; that is, it enables access control of various ser-
vices by voice [7]. Speaker recognition can be classified into
two specific tasks: identification and verification. Speaker speech signal processing. The likelihood of the GMM is
identification is the process of determining which one of the
T
Y
voices known to the system best matches the input voice sam-
ple. Speaker verification is the process of accepting or re- p(X|) = p(xt |) (19)
t=1
jecting the identity claim of a speaker. Speaker recognition
methods can also be divided into text-dependent and text- Since the distribution of these vectors is unknown, it is ap-
independent. When the same text is used for both training proximately modeled by a mixture of Gaussian densities
and testing, the system is said to be text-dependent. For text-
c
X
independent operation, the text used to train and test the sys-
tem is completely unconstrained. p(xt |) = wi N (xt , i , i ) (20)
i=1
One of the early successful text-independent speaker iden-
tification method is based on vector quantization (VQ). In this where = {wi , i , i } denotes a set of model parameters,
method, VQ codebooks consisting of a small number of rep- wi and N (xt , i , i ), i = 1, . . . , c, are the mixture weights
resentative feature vectors are used as an efficient means of and the d-variate Gaussian component densities with mean
characterizing speaker-specific features. A speaker-specific vectors i and covariance matrices i , respectively.
codebook is generated by clustering the training feature vec- In training the GMM, these parameters are estimated such
tors of each speaker. In the recognition stage, an input utter- that in some sense, they best match the distribution of the
ance is vector-quantized using the codebook of each reference training vectors. The most widely used training method is
speaker and the VQ distortion accumulated over the entire in- the maximum likelihood (ML) estimation, where a new pa-
put utterance is used to make the recognition decision [5]. rameter model is found such that p(X|) p(X|). An
Another early successful method is ergodic HMM based auxiliary function Q is used for this task
method. The basic structure is the same as the VQ-method.
Over a long timescale, the temporal variation in speech signal c X
X T
=
Q(, ) p(i|xt , ) log[wi N (xt , i )]
i , (21)
parameters is represented by stochastic Markovian transitions
between states. This method uses a multiple-state ergodic i=1 t=1

HMM, that is, all possible transitions between states are al- where p(i|xt , ) is the a posteriori probability for acoustic
lowed, to classify speech segments into one of the broad pho- class i, i = 1, . . . , c and satisfies
netic categories corresponding to the HMM states. After the
classification, appropriate features are selected. In the train- c
X
ing phase, reference templates are generated and verification p(i|xt , ) = 1 (22)
thresholds are computed for each phonetic category. In the i=1
verification phase, after the phonetic categorization, a com-
to zero,
Setting derivatives of the Q function with respect to
parison with the reference template for each particular cate-
gory provides a verification score for that category. The fi- the following reestimation formulas are found
nal verification score is a weighted linear combination of the wi N (xt , i , i )
scores from each category [17]. p(i|xt , ) = Pc (23)
The GMM based speaker identification is a most widely k=1 wk N (xt , k , k )

used method. This method also works as VQ-method. By T

X T
X
representing the feature space as a mixture of Gaussian dis-
i = p(i|xt , )/ p(i|xt , ) (24)
tributions, from the training feature vector set, the GMM al- t=1 t=1
gorithm generates a model parameter set including mean vec- PT
tors, covariance matrices and mixture weights. Parameters are i = t=1 p(i|xt , )(xt i )T
i )(xt
PT (25)
trained in an unsupervised classification using the expectation t=1 p(i|xt , )
maximisation (EM) algorithm. This algorithm provides an T
iterative maximum likelihood estimation technique. Experi- 1X
w
i = p(i|xt , ) (26)
ments have shown that GMMs are effective models capable T t=1
of achieving high identification accuracy [21].
For speaker identification, let k , k = 1, . . . , N, denote speaker
In this project, the GMM based speaker identification method
models of N speakers, a classifier is designed to classify X
is studied.
into N speaker models by computing the log-likelihood of the
unknown X given each speaker model k and select speaker
3.1. Gaussian Mixture Models k as
XT
Let X = {x1 , x2 , . . . , xT } be a set of T vectors, each of k = arg max log p(xt |k ). (27)
1kN
which is a d-dimensional feature vector extracted by digital t=1
3.2. Experimental Results
The results of the GMM based speaker identification experi-
ments are presented here.
The King speech corpus [8] was used. The KING cor-
pus was created for research in the area of speaker identifica-
tion. It was collected partly in New Jersey and partly in San
Diego. There are twenty-six San Diego speakers (numbered
01 to 26) and twenty-five New Jersey speakers (numbered 27
to 60, with some gaps in the sequence). All speakers are male.
There are ten sessions for each speaker (numbered 01 to 10),
and each session was recorded in both a wide-band (wb) and
a narrow-band (nb) channel. Sessions were recorded a week
to a month apart.
The data were processed in 30 ms frames. Frames were
Hamming windowed and preemphasised with p = 0.9. For
each frame, 46 mel-spectral bands of a width of 110 me1
and 20 melfrequency cepstral coefficients (MFCC) were de-
termined and used as features [27]. In this project, I used the
first 26 speakers in the King corpose that were recorded in the
same place. The first 5 sessions of each speaker were used for
training, and rest 5 were used for testing.
The experimental results are presented in Table 1. The
GMMs for each speaker were trained with 16, 32 mixtures.
I stopped experiments to increase the number of mixtures at
32 because the classification rate was not improved any more.
I also experimented the effect of mean subtraction of feature
vectors. The mean subtraction will normalize feature vectors
so that they become independent with the recording trans-
fer function produced by microphone architecture, distance
to the microphone, and the room acoustics [6]. The experi-
ments show that the mean subtraction improves the classifi-
cation rate.
(Number of mixtures, Error Rate
Do mean subtraction)
(16,N) 0.1410
(16,Y) 0.0705
(32,Y) 0.1026
Table 1. Speaker identification error rate performed for 26
speakers using the King corpus database
4. INDIVIDUAL VOICE ACTIVITY DETECTION
Individual voice activity detection (IVAD) is an algorithm to
detect the speech regions of an interest person in audio. The
IVAD helps to enhance speech processing techniques such as
speech enhancement, speech coding, and automatic speech
recognition more than only VAD by enabling to specialize
these algorithm to an interest person. In this section, I propose
an IVAD method which is constructed by cascading of a VAD
method and a speaker identification method.
I propose a semi-automatic hybrid IVAD system of VAD
by the PARADE method and the speaker identification based
on GMMs. The system works as follows:
1. Use the PARADE method to obtain voice activity frames.
2. Label a few first voice activity frames to speakers by hand
to train GMMs.
3. Train GMMs and classify voice activity frames of the en-
tire signal.
Note that the PARADE method requires nothing to prepare
for users to run despite several VAD methods such as [22][4][28]
require users to prepare training sets. After VAD is performed,
users prepare training sets for speaker identification using frames
obtained by VAD. Note that users only have to label a few
first frames of individuals, and it is not so burdensome be-
cause segmentation is already processed by VAD. Finally, the
IVAD system trains GMMs using the prepared training sets
and classifies voice activity frames in the entire signal to each
speaker and pick voice activity frames of an interest person.
4.1. Experimental Results
The experimental results of the proposed method are presented
here.
The first test signal shown in Fig. 5(a) contains utterances
of two speakers recorded in silence. A VAD by the PARADE
method was applied to the signal and the result is shown in
Fig. 5(b). The ground truth is that the first 3 and the last 2
voice activity frames are utterances of person 1 and rest are ut-
terances of person 2. The training of GMMs were performed
using the first 5 voice activity frames in the signal where the
first 3 frames are utterances of person 1 and the last 2 frames
are utterances of person 2. After that, speaker identification
was performed to each voice activity frame in the entire signal
and the resulted voice activity regions of person 1 are shown
in 5(c). This result shows that this system worked well for a
speech signal under small noise.
The second test signal was created by adding white noise
at an SNR of 20db to the original of Fig. 5(a). The created
signal is shown in Fig. 6(a). The VAD worked well for this
signal as shown in Fig. 6(b). The same procedures were re-
peated and the result of the speaker identification can be seen
in Fig. 6(c). This shows that the GMM based speaker iden-
tification did not work well in the presence of noise, that is,
the GMM method is not so robust to noise as the PARADE
method. Use of noise robut speaker identification method in-
stead of the GMM based method is desired.
5. CONCLUSION
Various VAD methods, especially the PARADE method, were
studied. The PARADE [10] method was implemented and the
noise robustness of the method was verified by experiments.
Moreover, various speaker identification methods, especially
the GMM based speaker identification [21], were studied. I
 1 1
(a)

(a)
0 0

1 1
0 5 10 15 0 5 10 15
5
x 10 5
x 10
1.5 1.5
1 1
(b)

0.5

(b)
0.5
0
0
0.5
0 5 10 15 0.5
5
0 5 10 15
x 10 5
x 10
1.5
1.5
1
1
(c)

0.5

(c)
0.5
0
0.5 0
0 5 10 15 0.5
5 0 5 10 15
x 10 5
x 10

Fig. 5. The first signal at presence of two speakers in silence

Fig. 6. The second test signal created by adding white noise
(a) Speech waveform (b) VAD result. The first 3 frames and
at an SNR of 20db (a) Speech waveform (b) VAD result.
the last 2 frames are utterences of speaker 1 and rest are utter-
Again, the first 3 frames and the last 2 frames are utterences
ances of speaker 2. (c) IVAD result for an interest speaker 1
of speaker 1 and rest are utterances of speaker 2. (c) IVAD
(Error rate is 0)
result for an interest speaker (Error rate is 0.18)

constructed an semi-automatic individual voice activity de-
tection (IVAD) system by cascading VAD by the PARADE
method and speaker identification based on GMMs. How-
ever, the experiment showed that the GMM based speaker
identification method is not so robust to noise as the PARADE
method. In the future I will investigate noise robust speaker
identification methods and training-free speaker identification
methods to make a completely automatic IVAD system.
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[4] Z. Enqing, D. Guizhong L. Yatong and Z. Xiaodi, Ap-
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