A Fast Shrinking Suspicious Criminals System From the Voice
Chun-Hsi Shih' Chi-Shi Liu' Chun-Hsiao Lee So-Lin Yen2
Jung-Hsiang Liao2 Bor-Shenn Jeng'
1. Telecommunication Labs, Chung-Hwa Telecom Co., Ltd., Taiwan, ROC.
2. Ministry of Justice Investigation Bureau, Taiwan, ROC.
ABSTRACT
As science technique becomes newer, versatile
committiiig crime methods are coming. In Taiwan,
we usually find criminals use the telephone to
blackmail people. In this situation, only voice is the
cue to know who are doing that. However, at this
time if we have a large of pre-recorded voice files,
we need to search these large voice files to find
them, or predict them by experience if we do not
have any voice database. It should spend a lot of
time to do that, even for an experienced worker.
Computer can save that time if we can develop a
good method to slirink them to a smaller range of
possible criminals or directly find those men's
identity, and then let experienced workers more
quickly do the final judgement. This paper will
discuss this shrinking system and give a preliminary
result.
1. INTRODUCTION
As science technique and communication tools
become newer, versatile committing crime methods
are coming. To prevent more crime occurring and
quickly detect the place, the time, the man of
committing the crime, more efficient methods should
be developed. In Taiwan, voice is an auxiliary
witness in some criminal cases and some crime cases
are occurred by using the telephone to blackmail
people. In this situation, only voice is the cue to
know who are doing that. To attain the above
purpose, two organizations, Chung-Hwa
Telecommunication Labs, and Ministry of Justice
Investigation Bureau, are jointed together to
develop the speaker authentication research work. In
this paper, we will talk about the preliminary
0-7803-4535-5/98/$10.000 1998 IEEE
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studying result of a fast shrinking suspicious
criminals system. In this system, we provide an
environment to let operators easily to create the
criminal database. We use spectrum as system
feature set, which can be computed automatically
and left in the computer. To alleviate the channel and
noise influence, these derived feature set are
normalized and equalized. In the second step, a
statistical model is built up for each speaker by the
above feature sets. For a new coming speaker, the
same procedure is done.
As a criminal's voice is coming in face or from
remote transmission medium, our system can quickly
find some possible criminals to let experienced
workers do the final judgement. The number of
criminals to be finally judged can be defined by
operators. Operators can ask our system to pump out
a fixed number of criminals, whose voice
characteristics are closest to the true crime voice, or
set a confidence score to let our system
automatically pick up the possible criminals. It can
playback these selected speakers' voice and plot
their voice characteristics to let operators do the final
check.
The other sections of this paper are organized as
follows. Section 2 introduces the system architecture
for this system. Section 3 talks about how to use this
system and some demonstration is also given. In the
final section, A summary is given.
2. SYSTEM ARCHITECTURE
To develop a shrinking suspicious criminals system,
we plan it as the following architecture shown in
Fig. 1. This architecture contains the following
components: recording environment, high pass filter,
end point detection, bias removal, feature extraction,
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from t e l e p h c
Fig-1 system architecture
distance computation, candidates display and final
judgement. Below we will explain each step’s
function and the method to implement it.
2.1 Recording environment
The speaker’s voic;e could be obtained from tlie
internet, the telephone, the tape, the microphone and
other storage deviccs. According to the voice type,
we can divide it into two types, analog and digital.
As analog type, it sliould digitize it before storing in
the computer. If this analog voice comes from the
telephone, a telephone interface should be open to
receive this kind of voice. In our system, we use a
dialogic company’s telephone interface, which has a
telephone interface with an 8KHz sampling rate aiid
8 bit PCM encoder, to receive the analog voice.
Other analog voices, such as cassette tape, are also
recorded aiid digitized by a built-in sound blaster.
After digitizing processing, all data should be
transferred into a uniform data type, a 16-bit linear
PCM “.wav” format with sampling rate equal to
8KHz.
2.2 High pass filter
Some of the incoin ing digitized voice may contain
DC component. In order to remove this component, a
60Hz high pass filter is used.
2.3 End point detection
Since the incoming voice has soine
components before a true speech conies, it is
necessary to remove this part to reduce the
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computation loading and the side effect, which is
made due to non-speech part’s effect over the true
speech part’s. To achieve this purpose, we use the
features, zero crossing rate and normalized energy,
to detect the true speech. To avoid discarding tlie
true speech too much, we did not give a rigid bound.
2.4 Bias removal
The incoming speech may be from different
recording environments. To alleviate the recording
environment’s effect and reduce its effect on the final
performance, a simple but efficient way is used. We
take the average of the whole incoming speech’s log-
spectrum, called cepstrum, and then subtract it from
the whole speech’s cepstrum. This simple metliod
has proved to be efficient [l].
2.5 Feature extraction
The key point to develop an efficient speaker
authentication system is feature extraction. It is not
easy to directly find an efficient feature from the
time-domain waveform. The super-segmental
features, such as pitch, energy, zero crossing rate
can be used to improve the performance, but it is
limited and usually used an auxiliary feature for the
most speaker authentication systems. For this reason,
the segmental information, that is, spectrum, is taken
as the features for this shrinking system. LPC(Linear
Predictive Coding), LAR(Log Area Ratio), LSP(Line
silence Spectrum Pair), Ceptrurn are all good spectrum
features, In this paper, we choose the LPC-cepstrum
feature as our system feature.
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 from tape
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F i g 2 training algorithm

LVQ training algorithm]

2.6 Distance computation
Before discussing the function of distance
computation, it is necessary to introduce the method
of creating speaker models. The steps of creating
speaker models are shown in Fig. 2. The steps,
recording environments, High pass filtering, end
point detection, bias removal and feature extraction,
are the same as the steps we just mentioned in
sections 2.1, 2.2, 2.3, 2.4, and 2.5. After these steps,
we use the cepstrum feature and VQ training
algorithm to create speaker models. Although there
Vector Quantization is a way to partition a vector
space into some of lion-overlap cells based on
predefined distortion. This vector space is then
represented by centroids of these non-overlap cells.
The centroid is called a codeword, and the set of
centroids is called a codebook. Design of such
codebook is by finding a set of codewords such that
the distortion DJQ is miniinurn. D-Q is defined
as
D,0 = cd ,
l 7
I
21))
O(x(L)>

are several methods wliicli can be used to create where

- -
speaker models, such as GMM(Guassian Mixture x ( i ) = a r g m i n d , (x(z),c ( j ) ) and d v Q ( x ( i ) , ~ ( iis) )
Model), HMM(Hidden MarKov Model), and C(/)

template model. However, the main point of this a distortion measurement between x(i) and x(i).
system is to quickly shrink the suspicious criminals LBG algorithin[2] can be used for the derivation of
and let the experts do tlie final judgement. Thus an codebook C. Encoding a test vector y(i) is to find a
efficient method but less computation time is used
for this purpose. Besides, we should also consider codeword j ( i ) from codebook C, such that
tlie incoming speech materials would be any kinds of
text combination. It would not be text-dependent like
d(y(i),y(i)) is minimum. Based on the above
algorithm, each speaker will obtained his own
system or a limited domain material system, so that
codebook.
as we choose the method to create speaker models,
we should consider this effect. Based on the above
(VQ recognition algorithnij
consideration, we discard the probabilistic model and
choose VQ(Vector Quantization) model as our
system model. In the following, we will introduce Given a sequence of feature
how to train the speaker models based on VQ vectors,X = {x(l),x(2);.+,x(T)}. Each element of
training algorithin and what VQ training algorithm X i s a feature vector obtained by signal analysis. The
is. object of source coder is to find a reproduction

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codebook such that the distortion between training
vectors X and reproduction codebook is minimum.
The total distortion for X is obtained by tlie
following equation.
[Beam search UlgorMzm]
To fast search the right person, we set up a
computation beam in which only the part of speakers
with smaller distanc;e are jointed. The input speech
distances in the computation beam speakers are
simultaneously computed frame by frame. After
computing several frames, we only leave the M
speakers in our computation beam. This step will
continuously proceed until tlie end of input speech.
At each step, some speakers with larger distance will
go out of this computation beam so that tlie
computation beam width will be reduced with time
going. This procedure will save the search time, but
tlie width should not reduced too much. Otherwise,
the target speaker will be lost in tlie search step.
final judgement
After tlie distance is computed, this system will list
the top N speakers whose distances are the smallest.
This value "N" is chosen by operators. Their distance
will also be shown to operators to let them show
their personal information further. Operators can
also use the distance score to decide which speaker
will show their personal database from the speaker
corpus pool.
3. OPERATION FLOW
As operators get the signals from tlie telephone, the
tape or other storage devices, they can recall the
recording program, like the menu shown in Fig. 3 ,
and select tlie record bar to record the signals. After
recording, they will call the recognition program to
recognize who speaks this voice from a speaker pool,
in which speaker templates are trained and stored by
tlie training procedure done by section 2. After
recognition, the shrinking system will show a list of
recognition scores i o tell operators which speakers
have the closest scores for the input voice. Operators
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then use these scores as a index to extract the related
information for those speakers. A typical information
is shown in Fig. 4. Here we only list some of
information for a speaker, such as his age, education,
address, self-picture, etc. Operators can define other
information that is good for them. They can also pop
out the speaker's voice characteristics, as shown in
Fig. 5. His waveform, sonogram or FFT-spectrum
can be displayed on tlie screen to let operators
compare his voice and spectrum characteristics with
the input speaker's voice. By tlie above procedure,
operators can reduced tlie human search time.
Especially when the suspicious criminals are large,
they do not need to do the full search speaker by
speaker. It is left to this system.
4. CONCLUSION
A first step to build up a shrinking system is
completed. By this system, operators can easily and
quickly find tlie set of suspicious criminals and
further analyze tlie voice characteristics of those
criminals. This will give them more evidence to do a
correct judgement.
In the next stage, we will collect and set up a
complete criminal database by tlie voice. This would
be a huge work, but it is important to find criminals.
Since the voice characteristic will change with time,
tlie new system should have tlie ability of updating
and easily re-building up the existing database. That
is, an adapting system should be developed. Finally,
We hope a user friendly and reliable system can be
coming soon.
5. REFERENCES
[ 11 S. Furui,"Cepstrum analysis technique for
automatic speaker verification", IEEE Tran.
Acoust., Speech, Signal Processing, vol.
ASSP-29, pp. 254-272, Apr., 1981.pp. 254-
272, Apr., 198 1.
Y. Linde and A. BUZOand R. M. Gray,"An
algorithm for vector quantization design"
IEEE Trans. Commun. ,vol. COM-28,pp. 84-
95, Jan., 1980.
 Fig. 3 Recording Menu Fig.5 analysis of voice characteristics: waveform and spectrum

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