*Helmer Strik*

A^{2}RT, Dept. of Language & Speech,
University of Nijmegen

P.O. Box 9103, 6500 HD Nijmegen, The Netherlands

J. Acoust. Soc. Am., May 1998, Vol. 103, No. 5, pp. 2659-2669.

Automatic parametrization of differentiated glottal flow:
Comparing methods by means of synthetic flow pulses

Helmer Strik a)

University of Nijmegen, Department of Language and Speech, P.O. Box 9103, 6500 HD Nijmegen,
The Netherlands

(Received 14 July 1997; accepted for publication 14 January 1998)

The automatic parametrization of the first derivative of glottal flow is studied. Representatives of the
two types of methods used most often for parametrization were tested and compared. The chosen
representatives are all based on the Liljencrants± Fant model. As numerous tests were needed for a
detailed comparison of the methods, a novel evaluation procedure is used which consists of the
following stages: (1) use the Liljencrants± Fant model to generate synthetic flow pulses; (2) estimate
voice source parameters for these synthetic flow pulses; and (3) calculate the errors by comparing
the estimated values with the input values of the parameters. This evaluation procedure revealed that
in order to reduce the average error in the estimated voice source parameters, the estimation
methods should be able to estimate noninteger values of these parameters. The proposed evaluation
method was also used to study the influence of low-pass filtering on the estimated voice source
parameters. It turned out that low-pass filtering causes an error in all estimated voice source
parameters. On average, the smallest errors were found for a parametrization method in which a
voice source model is fitted to the flow derivative, and in which the voice source model is low-pass
filtered with the same filter as the flow derivative.

© 1998 Acoustical Society of America.

[S0001-4966(98)03204-4]

PACS numbers: 43.70.Aj, 43.72.Ar [AL]

Electronic mail: strik [at] let.ru.nl

**INTRODUCTION**

The technique of inverse filtering has been available for
a long time now. This technique, which was first described in
Miller (1959), can be used to decompose the speech signal
into two components: the voice source and the filter (the
vocal tract). In this way an estimate of the glottal volume
velocity waveform (U g ) or its first derivative (dU g ) is ob-tained.
For many applications, estimating a voice source sig-nal
(either U g or dU g ) is not enough and the glottal flow
signals have to be parametrized. Parametrization of the voice
source signals and evaluation of the parametrization methods
have received far less attention in the past. That is why we
focus on these aspects in this study.

Parametrization of U g or dU g can be done in several
ways. Often landmarks (like minima, maxima, zero cross-ings)
are detected in the signals (e.g., Sundberg and Gauffin,
1979; Gauffin and Sundberg, 1980, 1989; Alku, 1992; Alku
and Vilkman, 1995; Koreman, 1996). Because these land-marks
are estimated directly from the voice source signals,
these methods will be called direct estimation methods.
Voice source parameters are also calculated by fitting a
voice source model to the data (e.g., Ananthapadmanabha,
1984; Schoentgen, 1990; Karlsson, 1992; Strik and Boves,
1992; Fant, 1993; Milenkovic, 1993; Alku et al., 1997).
Many different voice source models have been proposed in
the literature (see, e.g., Rosenberg, 1971; Fant, 1979; Anan-thapadmanabha,
1984; Fant et al., 1985; Fujisaki and

Ljungqvist, 1986; Lobo and Ainsworth, 1992; Cummings
and Clements, 1995). Because in estimation methods of this
kind a model fitting procedure is used, they will be referred
to as ''fit estimation'' methods.

Estimation of voice source parameters can be useful for
many applications. Although speech synthesis is the applica-tion
most mentioned, the estimated voice source parameters
are also used for fundamental research on speech production
(e.g., Ni´ Chasaide and Gobl, 1993; Strik, 1994; Koreman,
1996). Other applications for which methods to measure
voice source behavior could be useful are clinical use, speech
analysis, speech coding, automatic speech recognition, and
automatic speaker verification and identification. Since most
of these applications require that the methods be fully auto-matic,
there is an increasing need for automatic parametriza-tion
methods (see, e.g., Fritzell, 1992; Fant, 1993; Ni´ Cha-saide
and Gobl, 1993).

The development of an automatic parametrization
method constitutes the long term goal of our research. Both
direct and fit estimation methods can be made completely
automatic. For this reason, and because they are the methods
used most often, a representative of the direct estimation
method will be compared with a representative of the fit
estimation method. The representatives chosen are described
in Secs. I E and I F.

The goals of the research reported on in this article are
to find out what the pros and cons of each method are, to get
a better understanding of the problems involved in estimat-ing
voice source parameters, and finally to determine which
method performs best. In order to make it easier to compare
the two methods, the same voice source model is used in
both methods. To this end we use the Liljencrants± Fant (LF)
model (Fant et al., 1985). The LF model and the reasons for
choosing it are described in Sec. I B. The evaluation method
and material are described in Secs. I C and I D, respectively.
Because we want to focus on the parametrization method, we
shall not evaluate inverse filtering in the current research.
The performance of the parametrization methods is assessed
in Secs. II and III. First, in Sec. II, it is studied how well the
estimation methods succeed in estimating noninteger values
of the parameters, which turned out to be a crucial property.
Second, we focus on low-pass filtering in Sec. III. In Sec. IV
the findings are discussed and some general conclusions are
drawn.

**I. GENERAL PROCEDURES**

In this article two estimation methods used to param-etrize
dU g are tested and compared. Before going on to de-scribe
these two methods (in Secs. I E and I F), we shall first
give some definitions in Sec. I A, discuss the LF model in
Sec. I B, and describe the method and material used for
evaluation in Secs. I C and I D, respectively.

A. Definitions

In the current article it will be assumed that dU g is a
digital signal. In order to avoid confusion later on, we shall
first define some terms related to sampling and quantization.
For all tests the sampling frequency F s 510 kHz, the
number of bits used for quantization B c 512 and the ampli-tude
range is [-2048,2047]. Consequently, the sampling
time T s 51/F s 51 ms and the step size d54096/2 B c 51.
Throughout this article a time parameter is said to have an
integer value if its value is precisely an integer multiple of
T s . Likewise, an amplitude parameter is said to have an
integer value if its value is exactly an integer multiple of d.

B. Liljencrants± Fant model

In the current research the voice source model used is
the LF model (see Fig. 1) because the LF model has the
following advantages:

(1) In previous research the LF model has often been
used to estimate voice source parameters, with manual or
(semi-)automatic methods. This research has shown that it is
a suitable model for description of the flow derivative (see,
e.g., Fujisaki and Ljungqvist, 1986; Karlsson, 1992; Strik
and Boves, 1992; Strik et al., 1992, 1993; Childers and Ahn,
1995).

(2) Fujisaki and Ljungqvist (1986) compared several
voice source models. Their results showed that the LF model
and their own FL-4 model performed best (i.e., had the
smallest prediction error).

(3) Previous research has also proven that the LF model
is suitable for speech synthesis (see e.g., Carlson et al.,
1989).

(4) Due to all research already performed, the model and
its behavior are well known.

The parameters shown in Fig. 1, in turn, can be used to
derive many other parameters. For instance, the speed quo-tient
is often calculated: SQ5(t p 2t 0 )/(t c 2t p ) (e.g., Alku
and Vilkman, 1995). However, in our opinion these derived
parameters are less suitable for evaluation of the parametri-zation
methods, because whenever there is a change in a
derived parameter, it is difficult to determine how this
change came about (Strik, 1996). An increase in SQ could be
the result of a larger t p , a smaller t 0 , a smaller t c ,ora
combination of any of these three changes. On the other
hand, whenever a derived parameter remains constant, this
does not necessarily imply that the underlying parameters
(i.e., the parameters which were used to calculate the derived
parameters) remain constant. It is always possible that
changes in these underlying parameters cancel each other
out. Therefore, we prefer to use the LF parameters specified
in Fig. 1 for the evaluation of estimation methods. Since the
parameters E e , t 0 , t p , t e , and T a give a complete descrip-tion
of an LF pulse, this set of parameters will be used in this
article.

C. Evaluation method

Estimates of voice source parameters can be influenced
by a large number of factors. So far, 11 of these factors have
been studied: sampling frequency, number of bits used for
quantization, position (shift) and amplitude (E e ) of the glot-tal
pulses, t c , T 0 (length of the fundamental period), signal-tonoise
ratio (i.e., the effect of additive noise), phase distor-tion
(which can be caused, e.g., by high-pass filtering), errors
in the estimates of formant and bandwidth values during in-verse
filtering (which will bring about formant ripple in the
estimated voice source signals), and low-pass filtering (Strik
and Boves, 1994). We have performed over 1000 model fits
for each of these 11 factors, making a total of much more
than 11 000 model fits. The fact that so many tests had to be
performed is the main reason for using the evaluation
method described below (other reasons can be found in Strik,
1997).

In our experiments we first synthesize flow pulses (see
Sec. I D). As we use the LF model for the fitting procedure,
it is obvious that we also used the LF model to synthesize the
flow pulses. Subsequently, the parametrization methods are
used to estimate the voice source parameters. Finally, the
estimated voice source parameters are compared with the
correct values (used to synthesize the flow pulses), and the
errors are calculated:

FIG. 1. Glottal flow (U g ) and glottal flow derivative (dU g ) with the param-eters
of the LF model: time of glottal opening (t 0 ); time (t p ) and value (U 0 )
of the maximum of U g ; time (t e ) and absolute value (Ee) of the minimum
of dU g ; T a describes the return phase, it is the length of the time interval
between t e and the projection of the tangent of dU g in t e ; and the time of
glottal closure (t c ).

ERR( X)5u X est 2X inpu / X inp , for X5E e
ERR( Y )5u Y est 2Y inpu , for Y5t 0 , t p , t e , and T a .
The experiments were carried out for a number (say N)
of test pulses. After calculating the errors in the estimates of
the five LF parameters for each test pulse, the errors had to
be averaged. This can be done in a number of ways. Gener-ally,
averaging was done by taking the median of the abso-lute
values of the errors. Absolute values were taken because
otherwise positive and negative errors could cancel each
other. The median was taken because (compared to the arith-metic
mean) it is less affected by outliers which are occa-sionally
present in the estimates. This method of averaging is
the default method in the current article. Whenever another
way of averaging was used, this is explicitly mentioned in
the text.

In all figures below, the errors are arranged in a similar
fashion (see, e.g., Fig. 2). In the upper left corner are the
errors for E e (in%), in the middle row are the errors for t 0
and t p and in the bottom row are the errors for t e and T a .
The errors in the time parameters t 0 , t p , t e , and T a are
expressed in ms.

D. Material

The estimation methods used in this study are pitch syn-chronous.
Among the parameters that have to be estimated
are t 0 and t c . Because these two parameters are not known
beforehand, the pitch period cannot be segmented exactly. In
practice, we first locate the main excitations (i.e., t e ) and
then use a window with a width larger than the length of the
longest (expected) pitch period. Generally, the pitch period
will be situated between two other pitch periods (except for
UV/V and V/UV transitions). Therefore, for each experiment
sequences of three equal LF pulses were used. Each time
voice source parameters were estimated for the (perturbed)
pulse in the middle. Another reason for not using a single
glottal pulse for evaluation is that the effects of perturbations
cannot always be studied by a single, isolated LF pulse.
Since the effect of a studied factor can depend on the
shape of a flow pulse, LF pulses with different shapes were
used. These pulses will be called the base pulses. The base
pulses were obtained by using the LF model for different
values of the LF parameters. The parameters of E e , T 0 , t 0 ,
and t c were kept constant at 1024, 10 ms, 10 ms, and 20 ms,
respectively. The values given for t 0 and t c are the values for
the second of the three pulses. For the first pulse one should
subtract 10 ms from the values of t 0 and t c , and for the last
pulse add 10 ms. T 0 and t c were kept constant because the
results of our experiments showed that varying these parameters
had very little effect on the estimations. The effects of
varying E e and shift (which is strongly related to t 0 ) were
studied separately (see Sec. II).

For defining the base pulses the values of t p , t e , and T a
were varied. Based on the data given in Carlson et al. (1989),
and the data from previous experiments (Strik and Boves,
1992; Strik et al., 1992, 1993; Strik, 1994) the 11 base pulses
shown in Table I were defined.

Subsequently, these 11 base pulses were used to gener-ate
the test pulses. For instance, to study the influence of the
factor low-pass filtering, the 11 base pulses were filtered with
M low-pass filters in order to generate M311 test pulses.
Calculation of the base pulses and the test pulses was first
done in floating point arithmetic. After the test pulses had
been created, the sample values were rounded towards the
nearest integer (as is done in straightforward A/D conver-sion).

E. Direct estimation method

In direct estimation methods, voice source parameters
are calculated directly from dU g or U g by means of simple
arithmetic operators like min, max, argmin, and argmax.
These arithmetic operators are used to detect landmarks in
the signals. Some examples of estimations used quite often
are: U 0 5max(U g ), t p 5argmax(U g ), E e 52min(dU g ), and
t e 5argmin(dU g ) (see, e.g., Sundberg and Gauffin, 1979;
Ananthapadmanabha, 1984; Gauffin and Sundberg, 1980,
1989; Alku, 1992; Alku and Vilkman, 1995; Koreman,
1996). Except for the value and the place of a maximum or
minimum, the place of a zero crossing is also used to esti-mate
parameters. For instance, in this way t 0 and t c can be
estimated (see Fig. 1).

One of the aims of the research reported in this article is
to compare the performance of a typical direct estimation
method with that of a fit estimation method. To this end we
chose the direct estimation method described in Alku and
Vilkman (1995), primarily because these authors provide a
fairly detailed description of their method (see especially
page 765 of their article), and because with this method it
was possible to estimate the LF parameters E e , t 0 , t p , and
t e (for which they use the terms A min , t 0 , t m , and t dm ,
respectively).

In their method Alku and Vilkman (1995) do not esti-mate
T a . They use the parameter t ret to describe the return
phase. Since T a cannot be derived from t ret and an LF model
is not complete without T a , another method had to be used
to estimate T a . For the current research all estimates were
made in the time domain. Because it is very difficult to esti-mate
T a in the time domain with a direct estimation method,
estimates of T a were obtained by fitting the LF model to the

TABLE I. Values of t p , t e , and T a (all in ms) for the 11 base pulses.

Base pulse

1 2 3 4 5 6 7 8 91011

t p 14.0 14.0 16.0 16.0 16.0 16.0 14.0 14.0 15.2 15.2 15.2
t e 15.2 15.2 17.2 17.2 18.8 18.8 16.0 16.0 17.2 17.2 17.2
T a 0.4 1.6 0.4 1.6 0.4 0.8 0.4 1.6 0.4 1.0 1.6

glottal pulse. More precisely, for given values of E e , t 0 , t p ,
and t e (made with the direct estimation method) the optimal
value of T a was estimated by fitting the LF model to the
data. Therefore, strictly speaking, only E e , t 0 , t p , and t e can
be said to be the result of the direct estimation method, while
T a is subsequently estimated with a fitting procedure. How-ever,
it is important to notice that the estimate of T a does
depend to a large extent on the estimates of E e , t 0 , t p , and
t e made before with the direct estimation method. Further-more,
estimating one parameter (here T a ) with a fitting pro-cedure,
is a relatively simple operation. Consequently, the
results showed that the error in the estimates of T a is mainly
the result of the errors in the estimates of E e , t 0 , t p , and t e
made with the direct estimation method. For instance, if es-timates
of E e and/or t e are too large, the resulting estimates
of T a will generally be too small.

F. Fit estimation methods

In our fit estimation method five LF parameters (E e , t 0 ,
t p , t e , and T a ) are estimated for each pitch period. The
method consists of three stages:

(1) initial estimate;

(2) simplex search algorithm;

(3) Levenberg± Marquardt algorithm (Marquardt, 1963).
The goal of the fit estimation method is to determine a
model fit which resembles the glottal pulse as much as pos-sible.
This resemblance is quantified by means of an error
function, which is calculated in the following way. The op-timization
procedure provides a set of LF parameters. These
LF parameters and the analytical expression of the LF model
are used to calculate a continuous LF pulse. The LF pulse is
then sampled and zeros are added before t 0 and after t c (until
the length of the fitted signal is equal to that of the glottal
pulse). These samples of the fitted signal together with the
samples of the glottal pulse constitute the input to the error
function that provides a measure of the difference between
these samples. The fitting procedure tries to minimize this
error.

We have experimented with several error functions
which were defined either in the time domain, the frequency
domain, or in both domains simultaneously. Defining a suit-able
error function in the frequency domain, for this auto-matic
fitting procedure, turned out to be problematic. Prob-ably
the main reason is that the spectrum contains some
details (e.g., the harmonics structure, the high-frequency
noise) which need not be fitted exactly. With simple error
measures, like, e.g., the root-mean-square (rms) error, we did
not succeed in obtaining a reasonable model fit. More so-phisticated
error functions are needed for this task. A suitable
error function should abstract away from the details which
are not important, and emphasize the important aspects (e.g.,
the slope of the spectrum).

In the time domain it is much easier to obtain a fairly
good model fit of dU g . Here a simple rms error does yield
plausible results. Still, also in the time domain some aspects
of dU g could be more important than others. It is likely that
more sophisticated error functions could be defined which
emphasize the relevant (e.g., perceptual) aspects. However,
what is relevant depends on the application. In the current
research we did not have a specific application in mind. The
goal of this research was to develop a method for which the
error in the estimated voice source parameters is small.
Therefore, an important property of the error function is that
it should decrease when the errors in the voice source param-eters
become smaller (this may sound trivial, but it is not).
The rms error (defined in the time domain) did have this
property and thus was suitable for this task, as our experi-ments
revealed.

For the fitting procedure different nonlinear optimization
techniques were tested: several gradient algorithms and some
versions of a nongradient algorithm, i.e., the simplex search
algorithm of Nelder and Mead (1964). Of the algorithms
tested the simplex search algorithms usually came closer to
the global minimum than the gradient algorithms. Owing to
discontinuities in the error function, gradient algorithms are
more likely to get stuck in local minima than simplex search
algorithms are. Therefore the best version of the simplex
search algorithm is used in the second stage of the fit esti-mation
method. However, in the neighborhood of a mini-mum,
the simplex algorithm may do worse (see Nelder and
Mead, 1964). As a final optimization, the Levenberg±
Marquardt algorithm (a gradient algorithm) is therefore used
in the third stage (Marquardt, 1963).

In order to start the simplex search algorithm of stage 2
an initial estimate is required, which is made in the first
stage. In principle, the best available direct estimation
method should be used to provide the initial estimate. In this
case the rms error for the fit estimation method can never be
larger, and will almost always be smaller than the rms error
for the direct estimation method used (because in stage 2 and
3 of our fit estimation method the rms error can never in-crease,
and usually decreases gradually). Consequently, the
errors in the voice source parameters estimated with the fit
estimation method would almost always be smaller than
those estimated with the direct estimation method used for
initial estimation. Therefore, if we had used the direct esti-mation
method described in the section above for initial es-timation,
the performance of this direct estimation method
would probably have been worse than that of the fit estima-tion
method. Because we considered this to be an unfair
starting point, we decided to apply for initial estimation the
routine used in our previous research (Strik et al., 1993).
In Sec. III we will introduce a second version of this fit
estimation method. This second version differs only slightly
from the version described here. Together with the direct
estimation method described in Sec. I E, the number of
methods studied amounts to three.

Above we already mentioned that so far 11 different
factors have been studied. In this article we shall confine
ourselves to the most important results, namely those con-cerning
the factors position (shift) and amplitude (E e ) (Sec.
II) and those of low-pass filtering (Sec. III).

**II. EXPERIMENT 1:SHIFT AND AMPLITUDE**

A. Introduction

Direct estimation methods try to locate (important)
events in the voice source signals. Thus the resulting esti-2662
mates are generally limited to the place or amplitude of
samples in the discrete signals, i.e., they are integers. Our
intention was to develop a fit estimation method that would
make it possible to estimate noninteger values too. Here we
shall test how well the fit estimation method succeeds in
estimating noninteger values of the voice source parameters,
and what the resulting errors are for the two estimation meth-ods
for different values of shift and amplitude.

B. Material

The definition of the 11 base pulses is such that all time
parameters have an integer value (see Sec. I D). In order to
create test pulses in which the time parameters did not have
integer values, the 11 base pulses were shifted in steps of
0.01 ms, from 0.0 up to 0.1 ms (11 values). This variable will
be called shift. For only two of the chosen 11 values of shift
(i.e., shift50.0 and 0.1), the time parameters will have an
integer value, while for the other 9 values of shift all time
parameters will have noninteger values.
In order to create test pulses in which the amplitude
(E e ) does not have integer values the amplitude E e was var-ied
from 1023 to 1025 in steps of 0.2 (11 values). This
makes a total of 1331 test pulses (11 base pulses311 shift
values311 E e values). Next, the direct estimation method
and the fit estimation method were used to estimate the voice
source parameters for these 1331 test pulses. The errors in
these estimations were then calculated.

C. Results of the direct estimation method

First, the results of the direct estimation method are pre-sented
in Figs. 2 and 3. Each error in Fig. 2 is the median of
121 errors (11 base pulses311 E e values), while each error
in Fig. 3 is the median of another set of 121 errors (11 base
pulses311 shift values).

Let us first look at the errors in Fig. 2. To estimate t 0 a
threshold function is used in the direct estimation method.
The consequence is that the estimate of t 0 is always much
too large (on average about 820 ms; see Fig. 3). For a shift of
0.03 ms the average error in t 0 is minimal, while for a shift
of 0.04 ms it suddenly becomes maximal. The reason is that
this extra shift of 0.01 ms causes the threshold to be ex-ceeded
one sample later in many test pulses, and thus the
average error in t 0 suddenly increases. The average errors of
the other parameters all behave as expected: the average er-rors
are zero for a shift of 0.0 and 0.1 ms and larger in
between.

The errors in the estimates for different values of E e are
shown in Fig. 3. The errors in the time parameters t 0 , t p ,
and t e obviously do not depend on the value of E e . There-fore,
the errors for these time parameters are constant. If a
large number of moments is randomly distributed, the aver-age
error (both the arithmetic mean and the median) due to
rounding toward the nearest sample would be T s /4525 ms.
The average errors of t p , t e , and T a do not deviate much
from this theoretical average. The reason why the error in t 0
is much larger was already explained above.
The average errors in the estimates of E e behave as was
expected: the average errors are minimal for integer values
of E e , and are larger in between. The median error in E e is
never zero, because it is obtained by averaging over different
values of shift, and for most values of shift the error in E e is
larger than zero. The estimate of T a depends on the estimates
of E e and t e , and thus is not constant as a function of E e .

D. Results of the fit estimation method

The resulting average errors for the fit estimation
method are shown in Figs. 4 and 5. In this case the errors
were averaged by taking the mean value. This was done for
two reasons: (1) since there are no outliers, median and mean
values do not differ much; (2) by taking the mean it is also
possible to calculate standard deviations. In turn, this makes
it possible to test whether there is a significant difference
between two mean values.

In this case for each value of shift the mean and standard
deviation of 121 errors (11 base pulses311 E e values) were
calculated. The results are shown in Fig. 4. Likewise, for
each value of E e the mean and standard deviation of 121
errors (11 base pulses311 shift values) were calculated. The
results are shown in Fig. 5.

In Figs. 4 and 5 one can observe that the mean errors do
not differ significantly from each other. Furthermore, no
trend can be observed in the errors. Put otherwise, the mag-nitude
of the error in all estimated parameters does not de-pend
on the value of the factors shift and E e . Furthermore,
all errors are very small, in general much smaller than the
FIG. 2. Results of the direct estimation method: median error for the esti-mated
parameters for different values of shift.
errors for the direct estimation method. Except of course for
the cases in which all the LF parameters have an integer
value. In the latter case the errors for the direct estimation
method are zero, which is smaller still than the tiny errors
found for the fit estimation method. However, it is clear that
in practice the voice source parameters will seldom have
exactly an integer value.

E. Conclusions

The conclusions that can be drawn from these tests are
the following. The errors obtained with the fit estimation
method are very small, in general much smaller than those
for the direct estimation method. With the fit estimation
method noninteger values can be estimated as accurately as
integer values. Therefore, the quality of the model fit does
not depend on the exact value of E e and the position of the
pulse (which is determined here by the variable shift). This
explains why t 0 and E e could be kept constant in the defini-tion
of the base pulses (see Sec. I D).

For the direct estimation method the average errors in t 0
are always larger than for the fit estimation method, because
in the former a threshold function is used to estimate t 0 .In
fact, the error in t 0 can be substantially reduced, simply by
subtracting a constant from its estimate. For the other param-eters
the estimation errors for the direct estimation method
are zero if the parameters have exactly an integer value.
Since in practice parameters rarely have an integer value, the
estimates of the parameters will almost always contain an
error due to this fact alone. These errors will be called the
intrinsic errors, because they are intrinsic to the estimation
methods. They will always be present, even if the glottal
pulses are perfectly clean glottal pulses, as was the case in
these tests. The results presented in this section make it pos-sible
to estimate what the average intrinsic errors are. For the
direct estimation method the average error in the time param-eters
(except t 0 ) is about T s /4525 ms, which is the theoret-ical
average for randomly distributed values, while for E e it
is about 1% (see Fig. 3). For the fit estimation method the
average error in the time parameters is less than 0.5 ms,
while the average error for E e is about 0.01% (see Figs. 4
and 5).

**III. EXPERIMENT 2: LOW-PASS FILTERING**

A. Introduction

Before the glottal flow signals are parametrized, they are
low-pass filtered at least once in all methods, viz., before
A/D conversion. Often, they are low-pass filtered again after
A/D conversion, usually to cancel the effects of formants that
were not inverse filtered or to attenuate the noise component.
The latter operation seems very sensible for direct estimation
methods, because in these methods high-frequency distur-bances
can influence the estimated parameters to a large ex-tent.
Although parametrization of inverse filtered signals has
been done in many studies for almost 40 years now (i.e.,
FIG. 3. Results of the direct estimation method: median error for the esti-mated
parameters for different values of E e . FIG. 4. Results of the fit estimation method: mean and standard deviation of
the errors in the estimated parameters for different values of shift.
since Miller, 1959), it has only recently been noted that low-pass
filtering can influence the estimated voice source pa-rameters
(Strik et al., 1992, 1993; Perkell et al., 1994; Alku
and Vilkman, 1995; Strik, 1996; Koreman, 1996). Thus it
becomes very important to study what the effect of low-pass
filtering exactly is. This will be done in the present section.
An example of the distortion of a differentiated flow
pulse caused by low-pass filtering is given in Fig. 6. For
low-pass filtering a convolution with a 19-point Blackman
window was used. Shown are a base pulse before (solid) and
after (dashed) low-pass filtering, and a model fit on the low-pass
filtered pulse (dotted). Besides a picture of the three
signals for the whole pitch period, some details around im-portant
events are also provided.

One can see in Fig. 6 that low-pass filtering does influ-ence
the shape of the pulse. From this figure one can deduce
that the change in shape can have a large impact on the
estimates obtained by means of a direct estimation method.
This is most clear for the estimate of E e , which will gener-ally
be too small, but the estimates of the other parameters
will also be affected.

Low-pass filtering will also affect the estimates of a fit
estimation method. After low-pass filtering the shape of the
pulse is changed. The fitting procedure will try to find an LF
pulse that resembles the filtered pulse as closely as possible.
This is done by minimizing the rms error, which is a measure
of the difference between the test pulse and the fitted LF
pulse. The result is a fitted LF pulse that deviates from the
original base pulse (see Fig. 6).

The distortion of the differentiated glottal flow signals
depends on a number of factors, like, e.g., the type and the
bandwidth of the low-pass filter, the frequency contents of
the differentiated glottal flow signals, and the parametriza-tion
method used. We will study the effect of low-pass fil-tering
for two parametrization methods (i.e., the direct esti-mation
and the fit estimation method), for glottal pulses with
different frequency contents (i.e., the 11 base pulses), and for
different values of the bandwidth of the low-pass filter.
Low-pass filtering is done by means of a convolution
with a Blackman window.1 The bandwidth of this low-pass
filter is varied by changing the length of the Blackman win-dow
(the longer the window, the smaller the bandwidth).
This type of low-pass filtering was chosen because prelimi-nary
tests had shown that the error in the estimates induced
by this filter was smaller than that of other tested filters. In
part this can be explained by the fact that this low-pass filter
does not have a ripple in its impulse response, while a ripple
is present for many other low-pass filters. Therefore, for
most other low-pass filters (including the generally used
standard FIR filters) the estimation errors will be (much)
larger than the errors presented below (Strik, 1996).
In the example provided in Fig. 6 the test signal is low-pass
filtered. An LF model is then fitted to the low-pass
FIG. 5. Results of the fit estimation method: mean and standard deviation of
the errors in the estimated parameters for different values of E e .
FIG. 6. An example of a differentiated flow pulse before (solid) and after
(dashed) low-pass filtering, and a fit on the low-pass filtered pulse (dots in
the top panel, open circles in the lower four panels). Shown are the whole
pitch period, and some details around important events. For clarity, the zero
line (dashed-dotted) has been omitted in the top panel.
filtered test pulse. This seems the most obvious way to apply
the fit estimation method, and will be called the first version
of the fit estimation method. However, there is an alternative
(which will be called the second version of the fit estimation
method): apart from the test pulse one could also low-pass
filter the fitted LF pulse. In this case, the test pulse and fitted
LF pulse are altered in a similar fashion. In this way we hope
to achieve that the error in the estimated parameters (which
is due to low-pass filtering) will be smaller than when only
the test pulses are low-pass filtered. It is obvious that the
same procedure cannot be used in a direct estimation
method, because in this case the parameters are calculated
directly from the (low-pass filtered) signal.

B. Material

The 11 base pulses were low-pass filtered by means of a
convolution with a Blackman window. The length of the
window was varied from 3 to 19 samples in steps of 2
samples (9 lengths). For the resulting 99 test pulses (11 base
pulses39 window lengths) the parameters were estimated
with the direct estimation method and the fit estimation
method. For each length of the Blackman window the results
of the 11 base pulses were pooled and the median values of
the absolute errors were calculated. These median values are
shown in Figs. 7 and 8.

C. Results of the direct estimation method

In Fig. 6 one can see that low-pass filtering has most
effect on the amplitude of the signal (E e ) and the shape of
the return phase. Low-pass filtering causes the excitation
peak to be smoother, and thus the estimate of E e will be too
small. Low-pass filtering also makes the return phase less
steep, and therefore the estimate in T a too large. These ef-fects
are enhanced if the length of the Blackman window
increases (i.e., if the bandwidth of the low-pass filter is re-duced).
Therefore, the median errors of E e and T a increase
with increasing window length.

Low-pass filtering does not have much influence on t p
(5the position of the zero crossing in dU g ; see Fig. 6).
Therefore, in the majority of the cases the error in the esti-mates
remains within half a sample, and the median of the
errors is zero.

Usually, low-pass filtering causes the estimates of t e to
be too small (see Fig. 6). If the window length is 3 or 5, most
of the errors in t e remain within half a sample, and thus the
median error is zero. However, for larger window lengths the
errors in t e become larger. As a result the median error in-creases
too.

Finally, the error in t 0 remains constant, at the value of
820 ms (see also Fig. 3). This can be explained with the help
FIG. 7. Median errors in the estimated voice source parameters due to
low-pass filtering by means of a convolution with a Blackman window. The
length of the Blackman window varies from 3 to 19 in steps of 2. Shown are
the errors for the direct estimation method (dashed) and for the first version
of the fit estimation method (solid).

FIG. 8. Median errors in the estimated voice source parameters due to
low-pass filtering by means of a convolution with a Blackman window. The
length of the Blackman window varies from 3 to 19 in steps of 2. Shown are
the errors for the first (solid) and the second (dashed) version of the fit
estimation method. Note that the vertical scales are different from those in
Fig. 7.

of Fig. 6. In this figure one can see that low-pass filtering has
a large effect on the signal in the direct neighborhood of t 0 ,
and that this effect diminishes away from t 0 . If the threshold
chosen is high enough (which is the case for the direct esti-mation
method used in the current research), low-pass filter-ing
will not have much influence on this estimate of t 0 .

D. Results of the fit estimation method

In Fig. 7 not only the errors of the direct estimation
method are presented, but also those of the first version of
the fit estimation method (i.e., the version in which only the
test pulses were low-pass filtered). If the median errors of the
fit estimation method are compared with those of the direct
estimation method, the following observations can be made:
(i) The median errors are larger for t p for all window
lengths, and for t e for windows with a length of 3 or 5.
(ii) In all other cases the errors of the first version of the
fit estimation method are smaller than those of the direct
estimation method.

The fact that in certain cases the error of the direct esti-mation
method is smaller than the error of the fit estimation
method can be explained quite easily. If the effect of a stud-ied
phenomenon (here low-pass filtering) on an event (here
t p or t e ) is such that the event is shifted by less than half a
sample, the error with the direct estimation method is zero,
while that of the fit estimation method is larger than zero.
However, one should keep in mind that this is only the case
for pulses in which all events coincide exactly with a sample
position, as is the case with the test pulses. Only in this case
does rounding towards the nearest sample position mean
rounding towards the correct value.

In Fig. 8 the results of the two versions of the fit esti-mation
method are compared, i.e., the first version, in which
only the test pulses are low-pass filtered (solid lines), and the
second version, in which both test pulses and fitted LF pulses
are low-pass filtered (dashed lines). Clearly, the errors for the
second version are much smaller. The errors are not zero, as
may seem to be the case from Fig. 8, but they are extremely
small. The largest error observed in the time parameters is 1
ms, and the errors in E e are always smaller than 0.03%.

E. Conclusions

From our research we can conclude that low-pass filter-ing
changes the shape of the flow pulses, and thus affects the
estimates of all voice source parameters. The error due to
low-pass filtering does depend on a lot of factors, e.g., the
shape of the flow derivative, the low-pass filter and the esti-mation
method used. So even for a given low-pass filter and
estimation method (i.e., within one experiment) the error is
not constant, because the shape of the glottal pulses is gen-erally
not constant. Furthermore, for a low-pass filter with a
ripple in its impulse response (like the often used standard
FIR filters) the average errors will be larger than for the
low-pass filter used in this study, i.e., a convolution with a
Blackman window (Strik, 1996).

Generally, the errors for the direct estimation method are
larger than those of the first version of the fit estimation
method. In turn, these errors are larger than the errors of the
second version of the fit estimation method. Therefore, the
conclusion is that the second version of the fit estimation
method is superior. Low-pass filtering both the test pulse and
the fitted voice source model seems to be a very good way to
reduce the error caused by low-pass filtering. Of course, it
cannot be used in a direct estimation method (as was already
noted above).

**IV. DISCUSSION AND GENERAL CONCLUSIONS**

Before we draw our conclusions regarding the compari-son
of the three estimation methods, we first discuss some
aspects of the fit estimation methods used in this study. The
first aspect is the voice source model used in the fit estima-tion
method, in our case the LF model. In the literature sev-eral
voice source models have been described (see, e.g.,
Rosenberg, 1971; Fant, 1979; Ananthapadmanabha, 1984;
Fant et al., 1985; Fujisaki and Ljungqvist, 1986; Lobo and
Ainsworth, 1992; Cummings and Clements, 1995). All voice
source models for which an analytical expression exists can
be used with the proposed fit estimation method to param-etrize
either U g or dU g . In our program there is a subroutine
which calculates the fitted signal. The model fit is now cal-culated
with the LF model, but this part can easily be re-placed
by the analytical expression of any voice source
model. Furthermore, any number of voice source parameters
can be used for parametrization. However, increasing the
number of parameters makes the optimization problem (i.e.,
the error space) more complex, thus increasing the probabil-ity
that the fitting procedure gets stuck in a local minimum.
Using a voice source model for parametrization has
some advantages, one of them being the possibility that the
estimated voice source parameters can subsequently be used
for speech synthesis. Of course, for fit estimation methods a
voice source model is mandatory. However, probably the
most important disadvantage of a voice source model used
for this purpose is that it cannot describe all the observed
glottal pulses. Although the LF model is capable of describ-ing
many different glottal pulse shapes, it cannot describe all
details. Whether a voice source model is suitable for a cer-tain
type of research depends on the goals of this research.
Above we explained that with our fit estimation method it is
possible to use many voice source models. The reasons for
choosing the LF model in this study are given in Sec. I B.
The second aspect of the fit estimation method we want
to discuss concerns the properties of the LF routine, which is
the routine used to calculate the LF pulses. The way in which
the LF routine is implemented turned out to be extremely
important. The first version of our LF routine was taken from
Lin (1990). Since in this version all input parameters are
rounded toward the nearest integer, the shapes of the result-ing
LF pulses do not change gradually but abruptly. The
consequence is that also the calculated rms error jumps from
one value to the next. Thus the error function has the shape
of a staircase, which is problematic for many optimization
algorithms: they often get stuck in a local minimum. This is
especially the case for gradient algorithms, because the gra-dient
is zero for each stair.

In the second version of the LF routine, oversampling
was used within the LF routine. For instance, we tried over-2667
sampling by a factor 10. Thus not only integer values can be
estimated, but also nine values between these integers. How-ever,
the error function still has the shape of a staircase.
Since the stairs are ten times smaller (compared to the first
version of the LF routine), the resulting estimates were bet-ter.
Still, the optimization often did not come close to the
global minimum.

Our conclusion is that oversampling can reduce the
width of the stairs in the error function, and thus improve the
estimates, but it can never take away the fundamental prob-lem
for optimization, i.e., that the error function is a stair-case.
That is why we tried to find an implementation of the
LF routine for which the error function changes smoothly.
This property will be called the ''smooth property.'' The
third version of the LF routine, which is described in Sec.
I F, did have this property. In this version the analytical ex-pression
of the LF model is used to calculate a continuous
LF pulse, which is then sampled. An enormous improvement
in the fit estimation method was observed when the third
version of the LF routine was used (compared to the first and
second version). The reason is that a smooth error function is
an enormous advantage for both simplex search and gradient
algorithms. All results presented in this article are obtained
with the third version of the LF routine.
The third aspect of the fit estimation method which will
be discussed is that no anti-aliasing low-pass filter is used. In
the LF routine a continuous LF pulse is first calculated and is
then sampled with the same sampling frequency (F s ) as the
flow derivative which has to be parametrized (here, 10 kHz).
We did not use an anti-alias low-pass filter here, because we
wanted to be able to study each factor in isolation. If we had
used an anti-alias low-pass filter, this factor (and its effect on
the estimated voice source parameters) would always have
been present, thus making it impossible to study it indepen-dently
of other factors.

If no anti-aliasing low-pass filter is used, aliasing effects
can be present in the digital signals. Careful inspection
showed that this was not the case for the LF pulses used in
this study. The dU g signals on average have a slope of
26 dB/oct. The first fundamental is at 100 Hz, so at 5 kHz
the attenuation is usually more than 30 dB. Using a F s of 10
kHz made it possible to study the effect of the factor low-pass
filtering independently of other factors (like, e.g., shift
and E e ).

If aliasing is a problem (e.g., because F s is smaller than
10 kHz), an anti-alias low-pass filter has to be used. The
most straightforward way to do this is to sample the continu-ous
LF signal first with a sampling frequency F s , and next
use a digital low-pass filter with a bandwidth smaller than
F s /2. However, in that case the smooth property is lost, and
the error function (which quantifies the difference between
the LF signal and the flow derivative) becomes a staircase.
The result is that the average error in the estimated voice
source parameters becomes larger, as mentioned above. A
somewhat better solution is to oversample the LF signal be-fore
digital low-pass filtering. By oversampling noninteger
values can also be estimated. Furthermore, the stairs of the
staircase become smaller. Consequently, the average error in
the estimated voice source parameters also becomes smaller.
Probably the best solution would be to use the analytic anti-alias
low-pass filter proposed by Milenkovic (1993), which
can be applied in continuous time. In this way the smooth
property is preserved, and the error function remains a func-tion
that changes smoothly (instead of being a staircase).
In the current study two factors were studied in detail.
As parameters rarely have an integer value, we first esti-mated
what the resulting intrinsic errors are for the two
methods. For the direct estimation method they turned out to
be much larger than for the fit estimation method.
Next, the effect of the factor low-pass filtering was stud-ied
independently, i.e., with all input parameters having an
integer value. For low-pass filtering we found that the errors
of the direct estimation method are sometimes smaller than
those of the fit estimation method. However, if the important
events had been positioned randomly, the errors of the fit
estimation method would have been slightly larger while
those of the direct estimation method would have been sub-stantially
larger. For a realistic comparison of the two meth-ods
the intrinsic errors should be added to the errors found
for low-pass filtering alone. If this is done the average errors
of the direct estimation method are always larger than those
of the first version of the fit estimation method, and these in
turn are larger than the average errors of the second version
of the fit estimation method.

The conclusion which can be drawn on the basis of the
tests presented in this article is that the second version of the
fit estimation method is superior. However, the effect of
more single factors and factors in combination should be
studied to get a more thorough understanding of the intrica-cies
of the various parametrization methods.
In order to test and compare the parametrization meth-ods
we have used a novel evaluation method in which syn-thetic
test material is generated by a production model. Sub-sequently,
the same production model is used to re-estimate
the synthesis parameters. This evaluation method turned out
to be useful for our research, e.g., it helped us find the im-portance
of the properties of the implementation of the LF
routine and the effects of the factor low-pass filtering. We
are convinced that with other evaluation methods this would
have been much more difficult or even impossible (see also
Strik, 1996).

Since in the present research we want to focus on the
estimation of voice source parameters from the flow deriva-tive,
without being distracted by the problems of inverse fil-tering,
we use a voice source model (the LF model) as the
production model. For other purposes a vocal tract model or
a complete synthesizer could be used.

A similar method was used by McGowan (1994) to
evaluate the estimation of vocal tract parameters. In our re-search,
just as in McGowan's work (1994), all details of the
generating procedure are explicitly known. We therefore
agree with him that these kinds of studies should be regarded
as best case studies which can be used to study the limita-tions
of estimation procedures and to optimize these estima-tion
procedures. There are two other reasons why the present
study is a best case study. First of all, because the test signals
are clean LF pulses, and besides the influence of low-pass
filtering contain none of the other disturbances that are gen-2668
erally present in natural speech. And second, because for a
standard FIR filter, which is used most often as a low-pass
filter, the resulting average errors are larger than for the low-pass
filter used in this study. Consequently, when estimation
methods are used to parametrize inverse filtered natural
speech signals, the errors in the resulting parameters will
generally be (much) larger.

The final topic we want to discuss is how the proposed
estimation methods can be used to estimate voice source
parameters for natural speech. The answer is straightforward:
first use inverse filtering to obtain estimates of the glottal
flow signals, and then apply the estimation methods. In Strik
and Boves (1992) and Strik et al. (1992) we showed that this
is possible for previous versions of the fit estimation method.
We only have to exchange the previous version of the fit
estimation method with the new improved version. The best
solution would be to take the second version of the fit esti-mation
method, and in the error routine use the same low-pass
filter as used during the inverse filter procedure.

**ACKNOWLEDGMENTS**

The research of Dr. H. Strik has been made possible by
a fellowship of the Royal Netherlands Academy of Arts and
Sciences. I would like to thank Loe Boves, Bert Cranen, and
Jacques Koreman for fruitful discussions. Furthermore, I am
grateful to Paavo Alku, Anders Lo¨ fqvist, and an anonymous
reviewer for their useful comments on a previous version of
this paper.

1 This idea was suggested to me by Bert Cranen.

Alku, P. (1992). ''An automatic method to estimate the time-based param-eters
of the glottal pulseform,'' Proceedings of the IEEE International
Conference on Acoustics, Speech, and Signal Processing, San Francisco,
CA, Vol. 2, 29± 32.

Alku, P., and Vilkman, E. (1995). ''Effects of bandwidth on glottal airflow
waveforms estimated by inverse filtering,'' J. Acoust. Soc. Am. 98, 763±
767.

Alku, P., Strik, H., and Vilkman, E. (1997). ''Parabolic Spectral
ParameterÐ A new method for quantification of the glottal flow,'' Speech
Commun. 22, 67± 79.

Ananthapadmanabha, T. V. (1984). ''Acoustic analysis of voice source dy-namics,
'' Speech Transmiss. Lab. Q. Prog. Stat. Rep. 2-3, 1± 24.

Carlson, R., Fant, G., Gobl, C., Granstro¨m, B., Karlsson, I., and Lin, Q.
(1989). ''Voice source rules for text-to-speech synthesis,'' Proceedings of
the IEEE International Conference on Acoustic Speech Signal Process,
Glasgow, Scotland, Vol. 1, 223± 226.

Childers, D. G., and Ahn, C. (1995). ''Modeling the glottal volume-velocity
waveform for three voice types,'' J. Acoust. Soc. Am. 97, 505± 519.

Cummings, K. E., and Clements, M. A. (1995). ''Analysis of the glottal
excitation of emotionally styled and stressed speech,'' J. Acoust. Soc. Am.
98, 88± 98.

Fant, G. (1979). ''Glottal source and excitation analysis,'' Speech Trans-miss.
Lab. Q. Prog. Stat. Rep. 1, 70± 85.

Fant, G. (1993). ''Some problems in voice source analysis,'' Speech Com-mun.
13, 7± 22.

Fant, G., Liljencrants, J., and Lin, Q. (1985). ''A four-parameter model of
glottal flow,'' Speech Transmiss. Lab. Q. Prog. Stat. Rep. 4, 1± 13.

Fritzell, B. (1992). ''Inverse filtering,'' J. Voice 6, 111± 114.

Fujisaki, H., and Ljungqvist, M. (1986). ''Proposal and evaluation of mod-els
for the glottal source waveform,'' Proceedings of the IEEE Interna-tional
Conference on Acoustics, Speech, and Signal Processing, Tokyo,
Japan, Vol. 4, 1605± 1608.

Gauffin, J., and Sundberg, J. (1980). ''Data on the glottal voice source
behavior in vowel production,'' Speech Transmiss. Lab. Q. Prog. Stat.
Rep. 2-3, 61± 70.

Gauffin, J., and Sundberg, J. (1989). ''Spectral correlates of glottal voice
source waveform characteristics,'' J. Speech Hear. Res. 32, 556± 565.

Karlsson, I. (1992). ''Analysis and synthesis of different voices with em-phasis
on female speech,'' Ph.D. dissertation, KTH, Stockholm.

Koreman, J. (1996). ''Decoding linguistic information in the glottal air-flow,
'' Ph.D. dissertation, University of Nijmegen.

Lin, Q. (1990). ''Speech production theory and articulatory speech synthe-sis,
'' Ph.D. dissertation, KTH, Stockholm.

Lobo, A. P., and Ainsworth, W. A. (1992). ''Evaluation of a glottal ARMA
model of speech production,'' Proceedings of the IEEE International Con-ference
on Acoustics, Speech, and Signal Processing, San Francisco, CA,
Vol. 2, 13± 16.

Marquardt, D. (1963). ''An algorithm for least-squares estimation of non-linear
parameters,'' SIAM (Soc. Ind. Appl. Math.) J. Appl. Math. 11,
431± 441.

McGowan, R. (1994). ''Recovering articulatory movement from formant
frequency trajectories using task dynamics and a genetic algorithm: Pre-liminary
model tests,'' Speech Commun. 14, 19± 48.

Milenkovic, P. H. (1993). ''Voice source model for continuous control of
pitch period,'' J. Acoust. Soc. Am. 93, 1087± 1096.

Miller, R. L. (1959). ''Nature of the Vocal Cord Wave,'' J. Acoust. Soc.
Am. 31, 667± 677.

Nelder, J. A., and Mead, R. (1964). ''A simplex method for function mini-mization,
'' Comput. J. (Switzerland) 7, 308± 313.

Ni´ Chasaide, A., and Gobl, C. (1993). ''Contextual variation of the vowel
voice source as a function of adjacent consonants,'' Language and Speech
36, 303± 330.

Perkell, J. S., Hillman, R. E., and Holmberg, E. B. (1994). ''Group differ-ences
in measures of voice production and revised values of maximum
airflow declination rate,'' J. Acoust. Soc. Am. 96, 695± 698.

Rosenberg, A. E. (1971). ''Effect of glottal pulse shape on the quality of
natural vowels,'' J. Acoust. Soc. Am. 49, 583± 590.

Schoentgen, J. (1990). ''Non-linear signal representation and its application
to the modelling of the glottal waveform,'' Speech Commun. 9, 189± 201.

Strik, H. (1994). ''Physiological control and behavior of the voice source in
the production of prosody,'' Ph.D. dissertation, University of Nijmegen.

Strik, H. (1996). ''Comments on ''Effects of bandwidth on glottal airflow
waveforms estimated by inverse filtering'' [J. Acoust. Soc. Am. 98, 763±
767 (1995)],'' J. Acoust. Soc. Am. 100, 1246± 1249.

Strik, H. (1997). ''Automatic parametrization of voice source signals: A
novel evaluation procedure is used to compare methods and test the effects
of low-pass filtering,'' Internal Report, University of Nijmegen (available
at http://lands.let.ru.nl/~strik/).

Strik, H., and Boves, L. (1992). ''On the relation between voice source
parameters and prosodic features in connected speech,'' Speech Commun.
11, 167± 174.

Strik, H., and Boves, L. (1994). ''Automatic estimation of voice source
parameters,'' Proc. Int. Conf. Spoken Language Process., Yokohama, Ja-pan,
Vol. 1, 155± 158.

Strik, H., Cranen, B., and Boves, L. (1993). ''Fitting an LF-model to inverse
filter signals,'' Proc. of the 3rd European Conf. on Speech Technology,
Berlin, Germany, Vol. 1, 103± 106.

Strik, H., Jansen, J., and Boves, L. (1992). ''Comparing methods for auto-matic
extraction of voice source parameters from continuous speech,''
Proc. Int. Conf. Spoken Language Process., Banff, Canada, Vol. 1, 121±
124.