IEICE TRANS. INF. & SYST., VOL.E83–D, NO.4 APRIL 2000

876

PAPER

A 16 kb/s Wideband CELP-Based Speech Coder
Using Mel-Generalized Cepstral Analysis
Kazuhito KOISHIDA†∗ , Member, Gou HIRABAYASHI†∗∗ , Nonmember, Keiichi TOKUDA†† ,and Takao KOBAYASHI† , Members

SUMMARY
We propose a wideband CELP-type speechcoder at 16 kb/s based on a mel-generalized cepstral (MGC) analysis technique. MGC analysis makes it possible to obtain a moreaccurate representation of spectral zeros compared to linear predictive (LP) analysis and take a perceptual frequency scale into
account. A major advantage of the proposed coder is that the
beneﬁts of MGC representation of speech spectra can be incorporated into the CELP coding process. Subjective tests show thatthe proposed coder at 16 kb/s achieves a signiﬁcant improvementin performance over a 16 kb/s conventional CELP coder under
the same coding framework and bit allocation. Moreover, theproposed coder is found to outperform the ITU-T G.722 standard at 64 kb/s.
key words: wideband speech coding, mel-generalized cepstral
analysis, mel-generalized cepstrum, CELP coding

1.

Introduction

Current telephone networks generally limit the bandwidth of the speech signals to between 300 to 3400 Hz.
This bandwidth limitation degrades a speech quality interms of naturalness, intelligibility and speaker recognition. However, future transmission networks can eliminate the bandwidth limitation of 300–3400 Hz since
they are digital end-to-end. Such networks are capableof delivering face-to-face communications quality by increasing the bandwidth of speech signals. Speciﬁcally,
wideband speech, whose bandwidth ranges from 50 to7000 Hz, is of increasing interest today because wideband speech oﬀers a much higher quality than telephone
speech. Extending the lower frequency range down to
50 Hz increases naturalness, and the higher frequency
range of 3400–7000 Hz improves intelligibility. While
wideband speech is primarily used for teleconferencing and videoteleconferencing today, it is expected thatwideband speech will be employed for a number of applications such as multimedia services and future ISDN
Manuscript received April 13, 1999.
The authors are with Interdisciplinary Graduate Schoolof Science and Engineering, Tokyo Institute of Technology,
Yokohama-shi, 226–8502 Japan.
††
The author is with the Department of Computer Science, Nagoya Institute of Technology, Nagoya-shi, 466–8555
Japan.
∗
Presently, with Signal Compression Laboratory, Department of Electrical and Computer Engineering, University of California, Santa Barbara, CA 93106-9560, USA.
∗∗
Presently, with Human Interface Laboratory, Research& Development Center, Toshiba Corporation, Kobe-shi,
658–0015 Japan.
†

voice communication. In most of these applications,wideband speech coding plays a key role in eﬃcienttransmission and storage of wideband speech signals.
In wideband speech signals, most of the important formants are typically located at low frequencies,
so that the energy in the high frequency region issmaller than that in the low frequency region. Transform and subband coding schemes [1], [2] obtain highquality speech by exploiting these characteristics. The
basic principle of transform and subband schemes is todecompose the speech signal into subbands and separately encode each band. The ITU-T G.722 [3], a twosubband ADPCM coder, is the current standard at 64,
56 and 48 kb/s for wideband applications. On the other
hand, code excited linear prediction (CELP) coding [4]has received much attention in recent years due to its
great success in the ﬁeld of telephone-band speech coding. CELP coding schemes achieve high performanceby utilizing a parametric model based on speech production and analysis-by-synthesis algorithms.
Wideband CELP coders are traditionally divided
into two classes; split-band CELP coding [5]–[8], andfullband CELP coding [9]–[13]. In split-band CELP
coding which belongs to the class of subband coders,each subband signal is encoded by a CELP coder. The
split-band CELP coders generally suﬀer from degradation of speech quality in the frequency region where theresponses of the ﬁlterbanks overlap. In contrast to splitband CELP coding, a fullband coding scheme directlyencodes wideband speech signals using only one CELP
coder. The fullband CELP coders usually suﬀer froma high frequency noise in the decoded speech. To increase the quality at high frequencies, several methods
have been reported, e.g., a modiﬁed perceptual weighting ﬁlter [11], a split-band excitation [12] and a multiband excitation [13].
One promising approach for improving the performance of fullband CELP coding is to incorporate afrequency-warping technique into short-term spectral
modeling. This makes it possible to obtain a perceptual frequency scale, in which the frequency resolution at high frequencies is less sharp than at low
ones. In addition, since the most energy of wideband
speech is concentrated in low frequencies as described
before, frequency warping also has an ability to takesuch properties into consideration. Although several

KOISHIDA et al.: A 16 KB/S WIDEBAND CELP-BASED SPEECH CODER USING MEL-GENERALIZED CEPSTRAL ANALYSIS

877

studies have shown that the use of frequency warping
enhances the perceptual performance of telephone-band
speech coders [14]–[17] and audio coders (for a bandwidth of 20.7 kHz) [18], [19], this approach has not yet
received much attention in the area of wideband speechcoding.
In this paper, we propose a wideband CELPtype speech coder based on a mel-generalized cepstral(MGC) analysis technique [20], called MGC-CELP. The
proposed MGC-CELP coder uses a framework of fullband CELP coding, and its distinguishing feature is to
exploit MGC analysis instead of linear predictive (LP)analysis. MGC analysis makes it possible to introduce afrequency-warping technique into CELP coding. Moreover, while LP analysis has limitations of representing
spectral zeros, MGC analysis can provide more accurate representation of spectral zeros. A major advantage of the MGC-CELP coder is that the beneﬁts of
MGC representation of speech spectra can be incorporated into the CELP coding process. We will showthat the MGC-CELP coder can generate high-quality
speech at 16 kb/s.
This paper is organized as follows: Sect. 2 gives an
overview of the wideband MGC-CELP algorithm. In
Sect. 3, a speciﬁc implementation of this algorithm at
16 kb/s is described. Subjective evaluation of the coder
is performed in Sect. 4. Finally, conclusions are given
in Sect. 5.
2.

Wideband MGC-CELP Coding

Figure 1 shows a simpliﬁed block diagram of wideband
MGC-CELP coding. The basic framework of MGCCELP coding is identical to that of conventional CELP
coding. However, MGC-CELP utilizes MGC analysis
instead of LP analysis and, as a result, the parts related
to MGC analysis are completely diﬀerent from the corresponding parts in conventional CELP. In this section,(a)

we focus the discussion on these parts.
2.1 MGC Analysis
In MGC analysis, a speech spectrum H(ejω ) is assumed
to be modeled by a set of MGC coeﬃcients c(m) as
H(z) = K · S(z)(1)

where K is a gain of H(z) and

1/γ
M




−m

c(m)z̃
, −1 ≤ γ < 0

 1+γ
m=0
S(z) =

M





exp
c(m)z̃ −m ,γ = 0.

m=0(2)
In the above equation, z̃ −1 is an all-pass transfer function deﬁned byz̃ −1 =z −1 − α,
1 − αz −1|α| < 1.

(3)

The parameter α controls the frequency warping. A
linear frequency scale is obtained for α = 0, and thefrequency resolution at low frequencies improves withincreasing α at the expense of decreasing the high frequency resolution. When α = 0.42, the phase characteristics of the all-pass system give a good approximation to the mel scale for a sampling frequency of 16 kHz.
The parameter γ controls the representation accuracyof poles and zeros. The function H(z) becomes all-polemodeling for γ = −1 and α = 0. As the value of γ approaches zero, the accuracy for spectral zeros increases
at the expense of formant accuracy. For γ = α = 0,
H(z) is identical to the cepstral representation in whichpoles and zeros are represented with equal weights.
The optimum set of MGC coeﬃcients, which minimizes the expectation value of the square of the prediction residual, can be obtained using an eﬃcient iterative algorithm based on the FFT and recursive formulas [20]. It has been proven that coeﬃcients obtainedwith this algorithm result in stable systems [20].
In wideband MGC-CELP coding, the parameter γ
is ﬁxed to be −1/2. This value gives a good representation of spectral poles and zeros as shown in Fig. 2, andalso oﬀers some implementation advantages which will
be described later. An appropriate value of α will be
determined through listening tests.
2.2 Synthesis Filter(b)Fig. 1 Simpliﬁed block diagram of MGC-CELP coding:(a) encoder, (b) decoder.

For γ = −1/2, the synthesis ﬁlter is realized by a rational transfer function of the form
S(z) =

1{C(z̃)}

2(4)

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(a)

(b)

(c)

Fig. 2 Example of spectral envelopes obtained by 20th-order analysis using a 32 ms
Hamming window: (a) LP analysis, (b) MGC analysis with α = 0 and γ = −1/2, (c)MGC analysis with α = 0.3 and γ = −1/2. Thin lines represent the FFT spectrum.

− +❢
− +❢
❅
✒ ❅
I
✒ ❅
I
❅
❅
❅
❅
✻
❆ ✁α ❅❆ ✁α ❅❆ ✁α ❅❅❅❅
Input✲+❢ z −1✲ +❄❢ r z −1 ✲r +❄❢ r z −1 ✲❢ r z −1r +❄− ✻Outputγ(1 − α2 )✟❍Fig. 3❆ ✁ b(1)❆ ✁ b(2)❄+❢✛❆ ✁ b(3)❄+❢✛an alternative representation of the MGC coeﬃcientsand is mathematically equivalent to the MGC coeﬃcient representation. The MGC-LSP representation offers a simple check of the ﬁlter stability and gives goodinterpolation and quantization performance [22].
The MGC-LSP is a frequency-domain representation of speech similar to the LSP, but is deﬁned on thewarped frequency scale. The procedure for obtaining
the MGC-LSP parameters is as follows: ﬁrst Eq. (5) is
modiﬁed asC(z̃) = (1 + γc(0))C1 (z̃)

Filter structure of 1/C(z̃) for M = 3.

(9)

where
where
C(z̃) = 1 + γ

M

C1 (z̃) = 1 + γ
c(m)z̃

−m.

To remove delay-free loops from S(z), we modify Eq. (5)as follows:C(z̃) = 1 + γ

Mb(m)Φ m (z)

(6)

m=1

where
(1 − α2 )z −1 −(m−1)Φm (z) =z̃
,
1 − αz −1

m≥1

(7)

and the ﬁlter coeﬃcients b(m) are obtained using a recursive formula given byc(M ),
m=Mb(m) =(8)c(m) − αb(m + 1), 0 ≤ m < M.
Note that, since the gain of S(z) is unity, the coeﬃcientb(0) is always zero [17]. The structure of 1/C(z̃) basedon Eq. (6) is shown in Fig. 3.
2.3 MGC-LSP Parameters
In MGC-CELP coding, MGC-based line spectrum pair
(MGC-LSP) parameters [21] are quantized and transmitted instead of MGC coeﬃcients. The MGC-LSP is

c1 (m)z̃ −m

(10)

m=1

(5)

m=0

M


and the coeﬃcients c1 (m) are calculated from the MGC
coeﬃcients using the relationshipc1 (m) =c(m),1 + γc(0)1 ≤ m ≤ M.

(11)

Next C1 (z̃) is decomposed into symmetric and antisymmetric polynomials:
CP (z̃) = C1 (z̃) + z̃ −(M +1) C1 (z̃ −1 )CQ (z̃) = C1 (z̃) − z̃−(M +1)C1 (z̃

−1

).

(12)
(13)

Finally, using the same technique as for the LSP case,
the MGC-LSP parameters can be obtained as the angular positions of the roots of CP (z̃) and CQ (z̃).
Histograms of the LSP and MGC-LSP parameterswith γ = −1/2 are shown in Fig. 4. It is noted that,in Fig. 4 (c), the MGC-LSP parameters are plotted on
the warped frequency scale. It can be seen that thevariance of the MGC-LSP parameters is smaller thanthat of the LSP parameters. The reason is that theﬁlter 1/C(z̃) has less sharp peaks since the synthesis
ﬁlter is realized by the two-stage cascade structure of
1/C(z̃) as shown in Eq. (4). Consequently, the MGCLSP parameters require less precision to avoid missing
roots of CP (z̃) and CQ (z̃) in their root search [21].

KOISHIDA et al.: A 16 KB/S WIDEBAND CELP-BASED SPEECH CODER USING MEL-GENERALIZED CEPSTRAL ANALYSIS

879

(a)

(b)

(c)

Fig. 4 Histograms of (a) LSP parameters; (b) MGC-LSP parameters with α = 0, γ =−1/2; (c) MGC-LSP parameters with α = 0.3, γ = −1/2. These parameters are obtainedby 20th-order analysis using a 32 ms Hamming window. Thin and thick lines represent
the parameters of even and odd order, respectively.

set to be zero [17]. Under such a constraint, µ is givenby

2.4 Perceptual Weighting Filterµ=The perceptual weighting ﬁlter is deﬁned by the MGC
coeﬃcients asC(z̃/β1 )C(z̃/β2 )Spw (z) =(14)

where z̃/β indicates a bandwidth expansion in the z̃plane. The ﬁlter C(z̃/β) can be realized using the samestructure as C(z̃) [17]; i.e.,C(z̃/β) = 1 + γMbβ (m)Φm (z).

(15)

m=1

The ﬁlter coeﬃcients bβ (m) are obtained from c(m) as
follows: ﬁrst, a set of intermediate coeﬃcients bβ (m) is
computed using
 M
β c(M ),m=Mbβ (m) =β m c(m) − αbβ (m + 1), 0 ≤ m < M(16)
and the coeﬃcients bβ (m) are then calculated bybβ (m) =bβ (m),1 + γbβ (0)1 ≤ m ≤ M.

(17)

2.5 Postﬁlter
The short-term postﬁlter is deﬁned by
Ssp (z) =C(z̃/β3 ).
C(z̃/β4 )(18)

The tilt compensation ﬁlter has a structure of the formStl (z) = (1 − µz −1 )n(19)where µ is a parameter that adaptively controls theglobal spectral tilt of the postﬁlter. The value of µ
is obtained in such a way that the ﬁrst mel-cepstralcoeﬃcient of the cascaded ﬁlter of Ssp (z) and Stl (z) is

3.

−γ(β4 − β3 )c1 (1).
−αγ(β4 − β3 )c1 (1) + n(1 − α2 )(20)

Wideband MGC-CELP Coder at 16 kb/s

This section describes a wideband MGC-CELP coder
at 16 kb/s. Its structure and bit allocation are shown in
Fig. 5 and Table 1, respectively. The coder operates on
speech frame of 10 ms corresponding to 160 samples at asampling rate of 16 kHz. The MGC coeﬃcients and thesignal power are extracted and encoded for every 10 ms
frame, while the excitation parameters are determined
once per subframe of 2.5 ms.
3.1 Encoder
Using a Hamming window of 32 ms duration centeredat the middle point of the last subframe, 20th-order
MGC analysis is performed once per 10 ms frame. The
MGC coeﬃcients are quantized in the MGC-LSP domain using a two-stage VQ with switched ﬁfth-ordermoving average (MA) prediction. The selection of MA
predictive coeﬃcients and the ﬁrst stage codebook require 1 bit and 8 bits, respectively. The input vector of
the second stage is split into a lower dimensional partand a higher dimensional part, and 6 bits are assigned
to each part. These codebooks are searched with the
Euclidean distance measure.
The power parameters are calculated on a twosubframe basis, i.e., a two-dimensional vector of thepower parameter is obtained for each frame. The vector is quantized into 7 bits in the logarithmic domain.
The excitation codebook 1 consists of an adaptive codebook and a ﬁxed codebook [23]. The adaptive
codebook represents the pitch periodicity, in which apitch delay is used with varying resolution. The resolution is 1/4 in the range 33–96 3/4, 1/2 in the range
97–160 1/2 and integers only in the range 161–224. The
ﬁxed codebook which contains 64 random codevectors

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Table 1 Bit allocation of wideband MGC-CELP coder at
16 kb/s. The frame of 10 ms is divided into four subframes.
Subframe
–
–
9
17
7
–

MGC-LSP parameters
Power
Excitation codebook 1
Excitation codebook 2
Gain codebook
Total

Table 2

Algebraic structure of excitation codebook 2.

Pulse

Amplitude

1
2
3
4

±1
±1
±1
±1

Table 3Structure of wideband MGC-CELP coder at 16 kb/s.

represents the nonperiodic excitation contributions.
The excitation codebook 2 is based on an algebraic
codebook structure. In this codebook, each codevector
contains four non-zero pulses. These pulses can assume
the amplitudes and positions given in Table 2 [24].
The gains of the excitation codebooks are vectorquantized jointly using a 7-bit codebook.
By means of informal listening experiments,
we have found that the satisfactory performance isachieved by setting the perceptual weighting parameters to β1 = 1.0 and β2 = 0.0, i.e., Spw (z) = C(z̃).
3.2 Decoder
In the decoder, the coder parameters are obtained fromtransmitted indices. The speech is reconstructed by ﬁltering the excitation through the synthesis ﬁlter, and isthen passed through the postﬁlter. Besides the shortterm postﬁlter and tilt compensation ﬁlter, the postﬁlter contains a pitch postﬁlter. We have found that
it is necessary to tune the postﬁlter parameters fordiﬀerent values of the parameter α. Informal listening tests showed that good performance is obtainedwith (β3 , β4 , n) = (0.8, 0.95, 2) for α = 0.0, 0.1, 0.2 and(0.8, 0.9, 2) for α = 0.3, 0.4, 0.5.
4.

Subjective Evaluations

4.1 Test Conditions
Table 3 summarizes input speech conditions. The input
signal was sampled at 16 kHz and ﬁltered by the sending
ﬁlter P.341 [25] with 50 to 7000 Hz bandwidth. The

Positions
0,
1,
2,
3,
4,

5,
6,
7,
8,
9,

10,
11,
12,
13,
14,

15,
16,
17,
18,
19,

20,
21,
22,
23,
24,

25,
26,
27,
28,
29,

30,
31,
32,
33,
34,

35
36
37
38
39

Input speech conditions for subjective evaluations.

Source Material
Fig. 5

Frame
21
7
9×4
17×4
7×4
160 bits

Sampling Frequency
Bandwidth
Word Length
Speech Level

Multi-lingual Speech Databasefor Telephonometry 1994 (NTT-AT)
16 kHz
50 to 7000 Hz
16 bits−26 dB

speech level was adjusted to −26 dB.
Subjective tests were carried out in a sound-proof
booth. Eight people took part in the tests. They listened to sixteen Japanese sentences spoken by 4 females
and 4 males.
Two types of subjective test were performed forevaluation: an absolute category rating (ACR) test anda degradation category rating (DCR) test. In the ACR
test, listeners were presented with the processed speechand gave a 5-point rating. In the DCR test, listeners
were presented with the original speech before listening
to the processed speech, and gave a 5-point rating according to the degradation. The average rating of ACR
and DCR tests for a particular system are referred toas mean opinion score (MOS) and degradation mean
opinion score (DMOS), respectively.
4.2 Eﬀectiveness of Frequency Warping
In order to determine an appropriate value of thefrequency-warping parameter α, a DCR test was conducted. Figure 6 shows DMOS scores of the 16 kb/s
MGC-CELP coder for several values of the parameter
α. It is obvious from the ﬁgure that frequency warping makes a large contribution to the improvement of
the subjective quality. Speciﬁcally, it is shown that frequency warping signiﬁcantly increases the performanceof female speech.
From these results, we decided to set the parameter
α to 0.3 in the following tests.

KOISHIDA et al.: A 16 KB/S WIDEBAND CELP-BASED SPEECH CODER USING MEL-GENERALIZED CEPSTRAL ANALYSIS

881

Fig. 6 Speech quality of 16 kb/s MGC-CELP coder as afunction of frequency-warping parameter α.

(a)4.3 Comparison with G.722 and Conventional CELP
ACR and DCR tests were carried out to evaluate the
16 kb/s MGC-CELP coder with α = 0.3. For comparison purpose, the ITU-T G.722 standard at 64, 56 and48 kb/s was included in the tests. A conventional CELP
coder at 16 kb/s was also included, whose framework
and bit allocation are the same as the 16 kb/s MGCCELP coder. The diﬀerences of the conventional CELP
coder from the MGC-CELP coder are listed below:• After the LP coeﬃcients a(m) are computed by
20th-order LP analysis, a bandwidth expansion of
20 Hz is performed.
• The LSP parameters are obtained from the LP coeﬃcients and quantized with a weighted Euclidean
distance measure [26].
• The synthesis ﬁlter is deﬁned by the LP coeﬃcientsa(m) asS(z) =

1
A(z)

(21)

−m
where A(z) = 20
and a(0) = 1.
m=0 a(m)z• The perceptual weighting ﬁlter is of the formSpw (z) =A(z/0.9).
A(z/0.6)(22)• The short-term postﬁlter is given by
Ssp (z) =A(z/0.65)
A(z/0.75)(23)

and the tilt compensation ﬁlter is deﬁned by a ﬁrst
order all-zero structure asStl (z) = 1 − 0.15k(1)z −1
where k(1) is the ﬁrst reﬂection coeﬃcient.

(24)

(b)
Fig. 7 Results of subjective evaluations: (a) ACR test,(b) DCR test.

Figure 7 shows the results of subjective tests. We
see from the ﬁgure that, in both tests, the MGC-CELP
coder obtains similar scores in terms of equivalent Q
values. It is also shown that the MGC-CELP coder at
16 kb/s outperforms the G.722 at 64 kb/s. Moreover,
it is clear that a signiﬁcant improvement is obtained
compared to the conventional CELP coder. The improvement is especially noticeable for female speech. As
a result, the MGC-CELP coder gives a much smaller
diﬀerence between male and female speakers than theconventional CELP coder.
5.

Conclusions

We have proposed a wideband CELP-type speech coder
at 16 kb/s. The proposed coder belongs to the classof fullband CELP coding, and its distinguishing feature is to exploit MGC analysis instead of LP anal-

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ysis. MGC analysis makes it possible to incorporate
a frequency-warping technique into CELP coding andobtain more accurate representation of spectral zeros.
Subjective tests showed that the performance of the
proposed coder at 16 kb/s is better than that of the
ITU-T G.722 standard at 64 kb/s. It was also foundthat the proposed coder achieves a signiﬁcant quality
improvement over a conventional CELP coder which
has the same coding framework and bit allocation as
the proposed coder.
Acknowledgment
The authors would like to thank Prof. Satoshi Imai,
Chiba Institute of Technology, for his valuable discussions.
This work was supported in part by Research Fellowships of the Japan Society for the Promotion of Science for Young Scientists, and in part by Support forInternational Communication Research of the International Communication Foundation.
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Kazuhito Koishidareceived the
B.E. degree in electrical and electronicengineering, and M.E., and Dr.Eng. degrees in intelligence science from Tokyo
Institute of Technology, Tokyo, Japan, in
1994, 1995 and 1998, respectively. Since
1996, he has been a research fellow of
the Japan Society for the Promotion of
Science. He is currently a post-doctoral
researcher with Signal Compression Laboratory, University of California, Santa
Barbara. His current research interests include speech coding
at medium and low bit rates and wideband speech/audio coding.
He is a recipient of the 1998 TELECOM System Technology Prize
for Student from the Telecommunications Advancement Foundation Award. He is a member of IEEE and ASJ.

KOISHIDA et al.: A 16 KB/S WIDEBAND CELP-BASED SPEECH CODER USING MEL-GENERALIZED CEPSTRAL ANALYSIS

883
Gou Hirabayashi
received the B.E.
and M.E. degrees from Tokyo Institute ofTechnology, Tokyo, Japan, in 1996 and1998, respectively. He joined Toshiba
Corporation in 1998. His research interests include speech coding and synthesis.

Keiichi Tokuda
was born in Nagoya,Japan, in 1960. He received the B.E. degree in electrical and electronic engineering from the Nagoya Institute of Technology, Nagoya, Japan, the M.E. and
Dr.Eng. degrees in information processing from the Tokyo. Institute of Technology, Tokyo, Japan, in 1984, 1986, and1989, respectively. From 1989 to 1996
he was a Research Associate at the Department of Electronic and Electric Engineering, Tokyo Institute of Technology. Since 1996 he has beenwith the Department of Computer Science, Nagoya Institute ofTechnology as Associate Professor. His research interests include
speech coding, speech synthesis and recognition and multimodal
signal processing. He is a member of IEEE, ASJ and JSAI.

Takao Kobayashi
received the B.E.
degree in electrical engineering, the M.E.
and Dr.Eng. degrees in information processing from Tokyo Institute of Technology, Tokyo, Japan, in 1977, 1979, and1982, respectively. In 1982, he joined
the Research Laboratory of Precision Machinery and Electronics, Tokyo Instituteof Technology as a Research Associate.
He became an Associate Professor at thesame Laboratory in 1989. He is currently a Professor of the Interdisciplinary Graduate School of Science and Engineering, Tokyo Institute of Technology, Yokohama,Japan. His research interests include speech analysis and synthesis, speech coding, speech recognition, and multimodal interface.
He is a member of IEEE, ESCA and ASJ.