-1-

Design and Evaluation of the Asynchronous Voice Meeting System AVM
Takuya NISHIMOTO, Hidehiro YUKI, Takehiko KAWAHARA, Masahiro ARAKI, andYasuhisa NIIMI

Abstract
A voice communication system for asynchronous meetings held in non-real time will attract a
broad range of users because it is convenient and easy to use with mobile devices. To realize such asystem, however, it will be necessary to make possible the natural input of the progressive process ofspeech at the same time as facilitating the quotation of audio information and outlining of others’references.
This study reports the design and evaluation of a new interface for effective asynchronous voice
meetings, the AVM (Asynchronous Voice Meeting System), a client-server type meeting system that uses
overlapping speech. Speeches were, in particular, displayed as text information on the client side for theeffective visualization of comments made during these voice meetings. The proposed system AVM wascompared with an electronic bulletin board system by discussing the same topic. Using the AVM, the total
number of speech utterances decreased, and the total amount of letters went down to 48 % compared tothe conventional electronic bulletin board system. Subjective evaluations also rated the AVM highly,
indicating the usefulness of the proposed system AVM.

Key Words: voice message, asynchronous non-real time media, interactive communication, overlapping
speech, barge-in

1 Introduction
Asynchronous, non-real time media, including electronic mail and bulletin board systems, help
people communicate whenever they want. These media have several advantages; they have no time
restrictions, all the messages are saved, people can take their time to make a reply, a message can be sent
to multiple recipients simultaneously, and so on. These advantages encourage many people with various
lifestyles to form communities. In addition, searching through the accumulated information makes it easy
for people to meet new friends (in the media). We can say that the community forming function is one of
the reasons that has made the Internet so popular.
On the other hand, Internet applications with digitized audio and video have been realized thesedays because there are many improvements in computer performance, multimedia encoding technology,
and so on.

Especially, when the applications of audio media on the Internet are classified as real time,

asynchronous, non-real time, one-way, or two-way, as in Table 1, in order to form communities using

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such technologies, asynchronous and bi-directional applications are required. However, thoughapplications such as real-time one-way communication (Internet radio), real-time two-way
communication (voice chat, Internet phone), and asynchronous one-way communication (radio archive,
music archive) have been proposed, applications for asynchronous two-way communication have notbeen developed.

Table 1 Applications of audio media on the internet.
1way
2way

Realtime
Internet Radio
Voice Chat
Internet Phone

AsynchronousRadio Archive
Music Archive(not available)

There are several applications on interactive services for asynchronous audio media, such as
HyperAudio [1]. Also, SMIL(Note 1) is an Internet standard for describing the layout of sound andanimation, synchronization attribute, and hyper-link information. However, though they make theselection of contents interactive, they do not make messages bi-directional.
If community formation is achieved by voice, it will provide user-friendly services that are suitable
for mobile devices. Because such services will attract a broad range of users, they will contribute tosupporting the social commitment of the elderly and the visually handicapped, and to overcoming thesocial digital divide (which is caused by the availability and literacy of using the Internet). Voicemessages also allow easy identification of the speaker, so it increases the reliability of its content.
Consequently, voice messages will give greater security than written messages.
In this study, we proposed a new interface for effective asynchronous voice meetings, and designed
and evaluated the client-server type voice meeting system, AVM.

We particularly investigated andevaluated the storing, recording and replaying of voice messages.
The organization of this study is as follows. The second section describes the progressive processof conversations. The third section explains our method for the bi-directional voice meeting system. The
forth section outlines the design of the proposed system. The fifth section describes the evaluation system.
The sixth section discusses the experiment to evaluate the performance of the system. The seventh section
states the summary and shows the problems that need to be solved.

2. The Progressive Process of Conversation and Barge-in
Speech synthesis software for electronic mail and web information and e-mail software with voicerecognition have already been commercialized. They are providing a better communication environment
for unskilled users and visually impaired people. However, in a system where the voice interface has been

-3-

added to a communication procedure involving written words, the para-linguistic information such astone and emotion of the original utterance will not be conveyed. That means the system does not succeed
to create a potential fulfillment of verbal communications. Consequently, in this study we designed the
proposed system to record and play the voice message itself, and used speech recognition only as a
subsidiary measure for browsing and searching.
The difference between the written language and spoken language should also be considered.
Spoken language for daily conversations and telephone talks has a progressive process. Speakers often
convey only the new information or utter fragments of information as they come to mind. This
progressive process allows the elimination of shared information between the speaker and the listener,
and uses cognitive expressions with simple syntactic structures and less complicated expressions inconversations [2]. In order to utilize voice messaging effectively in communication systems, it is crucial
to design it allowing these progressive processes of conversations.
To realize virtual real-time asynchronous conversations, consideration of proceeding progressive
speech is required.

Accordingly, we studied conversational phenomena such as overlapping utteranceswith other speakers and a type of barge-in response that helps a conversation go smoothly. Barge-in is a
short response given to the speaker; it allows speech to be interrupted freely at any moment, without
waiting for full sentences to be completed.
Previously, many dialog studies have treated barge-in as an exceptional phenomenon. However, we
examined the corpus of the RWC project and found out that the majority of the replies overlapped overthe speakers’ talks and that the overlaps saved about 13 % of the total conversation time [3]. Enomoto, M.
et al [4] reports that nearly 45% of the speech overlapped with each other in the Corpus of Map Task
Dialog. Kawaguchi, Y. et al [5] explains an aspect of overlaps that ‘the responder reasoned andunderstood what the speaker meant to say,’ according to Grice’s theory on the connotation of talks.
Several studies on speech overlap focused on the generation procedure of a barge-in. The simplest
example of barge-in generation occurs when the machine makes a barge-in after a certain period ofsilence [6][7]. There is also a proposal for a simulation model for barge-in generation [8] based on pitch
patterns. Regarding the role of the barge-in, some studies discussed its effects on the alternation andverification of the speakers [9,10,11]. Another study [12] constructed a spoken dialog system thatgenerates barge-in by processing dialog on real time.
Based on these preceding investigations, we modeled overlapping speech not as exceptional but ascommon phenomena, and also designed an interface from the standpoint that barge-in has an importantrole to play.

3 Inter-editing of Voice Messages

-4-Compared with written messages, voice messages have some advantages including being highly
realistic, containing various para-linguistic information, and allowing simple input without using
keyboards and etc. On the other hand, voice messages are difficult to skim through to search, quote, or
annotate previous messages for editing.

We present here two propositions to overcome these difficulties.

The first one is to utilize voice recognition for browsing and skimming voice messages. That is, thesystem appears only to record and replay the human voice, but at the same time it allows users to deal
with voice messages as if they were written words. By doing this, we intend to combine the various
advantages of voice messages and written words.
We also intend to prove that there is some practical application in our proposition, even if the voicerecognition is not perfect itself. This proposition is intended to provide users with the voice recognition
result as a means to select certain sections from the dialogs, but not as the ultimate goal. Errors inrecognition will not seriously damage its practical use, because users can understand the text by listening
to the original speech. A ready-to-use application system with existing technology enlightens system users
and opens a new market of voice systems.
The second proposition is the inter-editing of speeches that is different from text editing to provide
equal functions. This is from the viewpoint that bi-directional discussions with asynchronous
accumulating media cannot exist without the inter-editing of speeches. The users of e-mail and bulletinboard systems can read, excerpt, and annotate messages easily. They use two major functions: a)
indicating the message replied to, and b) indicating the part focused on in the message.

A tree view of

messages is an interface for function a), and the partial excerption of a message by adding letters such as‘>>’ to the utterance is an interface to realize function b). It can be considered an inter-editing of speeches.
This is what makes bi-directional discussion and chat possible with asynchronous media.
The simplest means to edit written messages are through computers text-editing functions such as
cut and paste.

For voice messages, however, a simple interface cannot deliver equivalent functions as

written ones. Therefore, based on the discussion in Section 2, we propose a new operating system thatprovides an active role to overlap speeches and their temporal information. It allows responses to barge inat any time, and uses the information on a) the target of the barge-in, and b) the focused part of the target
as the alternative for inter-editing. This operating system should be a reasonable interface in comparison
to our usual natural conversations. In addition, the system depends on the user for the message-relating
operation of the messages, so the system is not required to understand the contents. It is expected to be aninterface suitable for telephone and mobile equipments, because the system requires only an audio input
and output device.

4 The Design of the AVM System

-5-Based on the argument above, we designed the Asynchronous Voice Meeting system, AVM. In
this section, we describe the details of the system.
4.1 Server-Client Structures
AVM consists of a message server and a client. Users utilize the client to record their messages andthe server stores all of the recorded messages. The server also edits multiple messages and sends them asa serial file in real time upon the client’s request. The server carries out this editing, and stores eachoriginal message recorded by the client.

4.2 Operating Flow
The following is the operating flow from the view of the user:(1) Select a group to participate in, and get a message list.
(2) Choose a target message and command to play it.
(3) Record a reply to overlap the target message.
(4) Play back the reply, and to cancel it, go back to 3).
(5) Store the recorded reply in the server.

4.3 Communication Protocol and Data Structure
In the AVM system, AVML files, which are information files based on XML(Note 2), and voice files
are transferred between the server and the client. The voice file format is uncompressed (11.025 kHz 16
bit sampling, monophonic) at this point, but we plan to compress it in the future for more efficienttransferring.
(1) Communication Protocol
The file transmission protocols are GET and PUT methods of HTTP(Note 3) in WWW.

Group nameand data are expressed on the path element of the URL. For instance, to designate the AVML informationof a group name room1, the path is ‘/room1/text/avml.’ In order to designate the message ID, letters like‘? Index=1,2,3’ are added at the end of the path.

(2) AVML Information Generated by the Server
AVML information generated by the server is utilized by itself as a message list. Also, it is used asadditional information accompanying voice messages when an edited file is sent to the client.
An AVML file is composed of several segment entities. A segment indicates a part of a voice filedivided into sections along with temporal information. A segment entity indicates the beginning point andend point of the original message that generated the segment. Attributes of a segment entity include mesid(original message ID), sender (original message utterer), mestime (location of the segment’s head on the

-6-original message), playtime (location of the segment’s head on the edited file), length (segment length),and indent (layer level on the tree view). Segment entity also includes a text entity, which shows thecontents of messages and barge-in made by others to the message. Figure 1 shows an example of AVML
information generated by the server.

<?xml version="1.0" ?><avml><segment mesid="1" sender="nishi" playtime="0"
mestime="0" length="1.5" indent="0"><text mesid="1" begin="0.0" end="1.01">
Good morning!
</text></segment></avml>Fig.1 An example of AVML generated by the server.

(3) AVML Information of the Client
AVML information generated by the client is transmitted to the server to store an utterance, andeach segment of a message corresponds to a message entity. It is intended to provide the utterance with
the information of the parent message which was played while recording. The attributes of a message
entity include parent (parent message ID), reltime (location of the new message head on the parent
message), length (new message length), and overlap (if the new message is Aizuchi or not). The overlap
attribute is further described in section 4.5. An example of AVML information generated by the client isshown in Figure 2.

<?xml version="1.0" ?><avml sender="canny"><message parent="2" reltime="0.4" length="0.3"
overlap="1"><text begin="0" end="0.3">
Yes.
</text>-7-</message></avml>Fig.2 An example of AVML generated by the client.

4.4 Recording and Relating of Messages
The AVM client records a new message in full duplex while playing an existing one. The silent partof the recorded message is detected and trimmed automatically. The new message is marked with therelative time to the existing message based on the segment information. The segment information
indicates the location of the original message that corresponds to the specific segment edited by the server.
When the recorded message is stored by the server, the temporal relation between the original message
and the new one can be preserved. Table 2 shows the database structure of the message relations.
Table 2 Data structure of the message relations.
Field
mesid
lengthparentoffsetmemberid
wavefile
overlap
dateContent
Message ID
Length of the messages (sec)Parent message ID
Time between the parent message’s start and this message’s start
Member ID
Audio file name
Overlap flag
Date and time of recording

4.5 BISP Function
The preliminary testing of this system [13] revealed that when recording a new message while
playing an existing one, it is necessary to have a proper control on the client side in order to pause aplayback. In other words, recording and playing the messages simultaneously often causes difficulties forthe user to understand the contents of the playing message, and he or she is forced to replay the existing
message. However, pausing the existing message at every utterance even for barge-in will cause the user
to hesitate with barge-in even if occurring as a natural action.
Also, regarding editing by the server, it is difficult to understand the contents of long overlapping
messages. Therefore, if the message is long, it should be inserted into the parent message rather thanoverlapped. On the other hand, for barge-in and shorter messages, they should be overlapped onto theparent message. Hence, the reproduced conversation will become more natural and easier to understand.
Consequently, we designed the server to categorize an utterance according to the overlap attribute

-8-either as a mere barge-in (backchannel utterance) or as a usual conversation before storing it. Based on
this information, the server edits barge-in to overlap and usual conversation to barge into the parent
message in order to reproduce virtual conversations. The server realizes such processes by following thealgorithm below:(1) Find a root message in the ID list received from the client.
(2) Retrieve the child messages of the root message and select only dialog, non-barge-in messages,
and insert them without overlapping recursively into corresponding location of the parent
messages. At the same time, generate the corresponding segment information.
(3) After inserting all barge-in messages into the parent messages, overlap barge-in messages onto
the conversation. Choose the overlapping points for the barge-in messages by using thesegment information, as barge-in messages are managed in relative time to the parents.
In addition, to record a new utterance at the client side, whether it is a barge-in or not is determinedby the speech length. A short utterance is treated as a barge-in and recorded with no pausing of the replay
and for a speech longer than a certain time (currently 1.0 second), the existing utterance is paused until
the end of it, and is stored as a non-barge-in. This function is called BISP (Barge-in to Stop Playing). With
BISP, spontaneously uttered barge-in becomes reproducible and existing speeches are no longer missed
even if a long speech is given while playing. The client’s transition state with BISP is shown in Figure 3.

Fig.3 State transision chart of the client.

4.6 Managing Read/Unread Messages
When many messages accumulate in one group, it becomes difficult to understand which remarks
the user has already listened to. To eliminate this, the server is designed to manage the IDs of previously

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heard messages and to mark them respectively in the list for each user on the client software. This helps
users to distinguish the already heard messages in a group, after storing many messages.

5. Construction of the Evaluation System
Based on the design in the fourth section, we constructed a system for evaluation such as the one
shown below.
5.1 The Server
Voxer, the AVM server, is written with the Perl and C languages in order to make it portable, andprocesses HTTP requests from the client software. It has a database function to store information,
including voice files and relations between messages, and an editing function for reproduced speech.
Voxer runs on Windows for the following experiment, though it can be used on Linux. Speech recognition
is integrated to the server but was not used this time.

5.2 The Client
Voyager, the AVM client software, is implemented with Microsoft Visual C++, which has
full-duplex audio input-output. This software runs on Windows98. It has the functions of HTTP message
transmission, voice recording and playing including BISP function, and displaying messages as thetree-view. To prevent the playing message from being recorded in full duplex, a headset is used forrecording and playing. Figure 4 shows the screen image of the client. The left pane is tree-view formessage selection. The right pane displays the message contents based on text entity (see 4.3) generated
by speech recognition. The upper tool-bar has buttons for “play and record”, “play”, “stop”, etc. , and thelower gauge shows the level of input from the microphone.

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Fig.4 Screen image of the client.

6. Experiment
6.1 Experiment Method
The performance of the AVM system was evaluated against the conventional electronic bulletinboard system (BBS) by discussing the same topic: ‘Which sport has registered professional players ofclose to 1000 people: Auto racing, boat racing, horse racing (The Central Horse Racing Association), J1
league of soccer, snowboarding, V league of volleyball, boxing, bowling, or baseball?”
Usually with this type of topic, the participants can complement their own limited information
among themselves, while no one covers everything, and it can be expected that interactive discussion willlead the group closer to the right answer.

Activity of the discussion can be evaluated by comparing theconclusion with the right answer, and also the system can be quantitatively analyzed by the number ofaccesses by the users and the number of utterances etc. in the discussion. This experiment was planned
from these viewpoints.
To encourage serious discussion, the subjects were notified that the team with the closest
conclusion to the correct answer was to be paid a higher amount. As the AVM system, the client softwareand the server system were implemented on the same computer with Windows98.

For the BBS software,

WebForum(Note 4) was selected. This software runs on the WWW server and can manage the written
utterances with a tree structure.
All ten subjects were male students in their twenties from university scientific laboratories(including speech research), and were well-experienced with keyboard operation. They were divided into
two groups of five, one for AVM and the other for BBS. The chairperson of each group gave the topic
through the system. The users were subsequently chosen randomly to use the system in turn.
Regarding the speech recognition of AVM, the operator repeated the utterance to ViaVoice98 (aspeech recognition software from IBM Japan) after every user finished, then he stored the recognized
sentences in the server. This allowed the speech recognition to maintain practical level of quality as wellas to include a certain level of misrecognition in the obtained text information.
When all the team members agreed on the conclusion, the chairperson orally answered the operator.
At the end, all subjects were asked to fill in a questionnaire.

6.2 Result
The winner was the AVM team, although second team reached close to the correct answer. The
quantitative analysis of the utterance is shown in Table 3, and the comparison of the lengths of the- 11 -

messages is in Figure 5. The lengths of the messages on AVM were counted using the manualtranscription of voice messages after the experiment. In regards to the utterances on BBS, the number ofcharacters were counted excluding the partial excerption.
ViaVoice98 showed 85.2 % word recognition rate for correctness and 82.5% for accuracy.
The average of the result of the questionnaire is shown in Table 4 (with a five-grade system,
including “do not think so at all” as one, and “think so very much ” as five.) The questions, (k) and (l),were asked only to the AVM users.

Table 3 Analysis of the messages on AVM and BBS.
AVM
Total count of using the system
Number of the messages
Number of the characters of all messages(excerptions are excluded)
Average number of characters of a message
Total time of messages (sec)
Average time of a message (sec)
Number of the overlapped messages
Number of the non-overlapped messages

BBS
20
71
2303

24
24
5657

32.4
501.3
7.1
12
59

235.7
-

Fig.5 Length of the messages on AVM and BBS.

Table 4 Result of questionnaire.

Items(a) I obtained satisfactory answers(b) I was able to adequately speak my opinions(c) Active discussion was able to be conducted(d) The discussion proceeded in a natural manner

AVM

BBS
4.4
4.0
3.8
4.0

3.6
4.4
3.4
4.0

- 12 -(e) The mood was very similar to when people actuallycongregate to conduct discussions(f) It was easy to expound the pros and cons(g) Easy to input messages(h) Easy to grasp the content of messages(i) Easy to view the discussions of the entire conference(j) The user interface was simple to understand(k) I used the text display more than the voice play(l) Using the text, including the misrecognitions, wasvaluable

2.2

2.8

4.4
3.4
3.4
3.4
3.8
4.0
4.6

3.8
4.2
4.0
3.8
4.0
-

6.3 Discussion
We shall discuss the result in view of the following hypotheses:(1) AVM realized an asynchronous voice meeting like spoken dialogs.
(2) AVM made use of the progressive process of spoken words.
(3) BISP function worked well and contributed to smooth conversation.
(4) The speech recognition result including misrecognitions was efficiently used.
(5) AVM functioned sufficiently as an alternative to BBS.

We will first discuss Hypothesis 1. With AVM, the subjects mostly spoke with conversational andcasual expressions such as “Well, talking about soccer now… about 20 teams in a league. And to make1000 players, 50 for one team. There aren’t so many on a team, are there?” On the contrary, with BBS,
the participants mainly utilized written language, for example, “In my interpretation, as others are saying,
there should be more than 84 players (1000 ÷ 12 = 84) in the professional baseball league, including thefarm teams. However, there is also a limit for the number of registered players for one team, so …” Therelations among the team members cannot cause such a difference because all the subjects knew eachother well. Accordingly, the result supports Hypothesis 1.
Hypothesis 2 predicts the elimination of shared information, which causes the frequent utilizationof simple and fragmented expressions as stated in the Section 2. Several results supported this hypothesis,
including the fact that the total length of messages with AVM was 48 % of that of BBS (Table 3), that thelength of most messages were relatively short (Figure 5), and that many utterances depended on thecontext such as “Well, it’s close, as you said.”
Hypothesis 3 was only partially supported. Many of the overlapping utterances, which are 17 % of
the total speech, were fragmented because of unsuccessful detection of the speech segment. This also
caused the low rating of question (g) in Table 4, indicating insufficient processing of overlapping
speeches. However, the high rating of questions (a) to (d) in Table 4 suggests that smooth conversation forthis experiment was realized.

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Hypothesis 4 was supported by the results of the questionnaire, including high evaluation ofquestions (k), (l) in Table 4, subjects’ comments such as, ‘I confirmed the context roughly by the text andminutely by the replay,’ and, ‘The text information saved time from listening to various utterances.’
Hypothesis 5 was supported by the facts that stated in Table 4, AVM was not rated far lower than
BBS, and that both teams reached acceptable answers. However, rating of the question (e) for AVM was
as low as BBS, which indicates that AVM cannot replace face-to-face discussion yet.
Accordingly, the experiment results support hypothesis 1, 2, 4, and 5 completely, and hypothesis 3
partially.

7. Conclusion
In this report, we designed and evaluated AVM, an asynchronous, non-real time, and interactive
voice meeting system. We were able to confirm that AVM applies the advantages of voice messaging inthat it is highly expressive, easy to talk, and does not impair the merits of asynchronous communicationof electronic mail, bulletin board systems, etc.
In the future, we intend to advance the performance of speech recognition in the server particularly
for spoken language.

Improvements for smoother conversation are also required, including avoiding

redundant replaying of utterances already heard, and preventing the fragmenting of messages in the
middle of a word. In addition, large-scale operating tests with various users, such as people unfamiliar to
the keyboard, are necessary. We plan to further investigate the realization of an AVM application on
telephony services and handheld devices, to attract a wider range of users.

(Note 1): Synchronized Multimedia, http://www.w3.org/AudioVideo/(Note 2): Extensible Markup Language, http://www.w3.org/XML/(Note 3): Hypertext Transfer Protocol HTTP/1.1, RFC2068.
(Note 4): http://www.kent-web.com

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ProfilesTakuya Nishimoto
He received B.E. and M.E. from Waseda University, Japan in 1993 and 1995, respectively.
He is a Research Associate at Department of Electronics and Information Science, Kyoto Institute of Technology,
Japan.

His research interests include spoken dialogue systems and human-machine interfaces.

He is a member of Information Processing Society of Japan, the Acoustical Society of Japan, the Japan Society for
Artificial Intelligence, and Human Interface Society.

Hidehiro Yuki
He received B.E. and M.E. from Kyoto Institute of Technology, Japan in 1998 and 2000, respectively.

Takehiko Kawahara
He received B.E. and M.E. from Kyoto Institute of Technology, Japan in 1999 and 2001, respectively.

- 15 -

Masahiro Araki
He received M.E. in information science and D. Eng. from Kyoto University, Kyoto, Japan, in 1990 and 1998,
respectively.

He is an Associate Professor at Department of Electronics and Information Science, Kyoto Institute ofTechnology.

His research interests include spoken dialogue systems and natural language understanding.

He is a member of IPSJ, JSAI, ANLP, and ACL.

Yasuhisa Niimi
He received the B.E., the M.E. and the Ph.D. degrees from Kyoto University, Japan in 1962, 1964 and 1969,
respectively. From 1964 to 1969 he was working at Kyoto University. Since 1970 he has been working at the
Department of Electronics and Information Science of Kyoto Institute of Technology, where he is now a professor.
His current interests include speech information processing, natural language and artificial intelligence. He published
the bool “Speech Recognition” (in Japanese) in 1979. Dr. Niimi is a member of the Information Processing Society ofJapan, the Acoustical Society of Japan, and the Japan Society for Artificial Intelligence.