• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.573-578
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• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• IE interfaces
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• v.11 no.1
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• pp.155-163
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• 1998

In multiparty multimedia communication, each participant not only receives information from other participants but also generates real-time data streams to distribute to others and therefore the difference between source and destination is not clear in terms of data transmission. During a teleconference session, many sub-multicast trees may be generated to exchange information to specific members within the multicast group and if those sub-multicast trees use the same fixed multicast tree, the blocking probability will be high and it is hard to provide QoS for each sub-multicast group. Also, even though there exits some shortcuts between each sub-multicast group, fixed multicast tree does not allow to use those shortcuts. Thus to overcome these problems. We propose a network resource reservation protocol and show that its effectiveness in terms of blocking probability and network resources usage.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.173-176
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• 1999

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this system, it is recognized speaker location using microphone array and camera is directed to speaker location automatically. In this paper, it was described to be able to enhance the speech qualify through microphone array, decrease computational loads using IIR filter as inverse filter, and confirmed to implement hardware using DSP processor.

• Journal of Multimedia Information System
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• v.8 no.4
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• pp.295-300
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• 2021

Recent advances in Internet-of-Things and wearable computing have established computationally sufficient infrastructure to remove the barriers of physical locations. The recent COVID-19 pandemic has accelerated the use of remote collaboration, work-from-home, teleconferencing, online education, digital twin, and metaverse enriched with various configurations of augmented reality, virtual reality, mixed reality, and extended reality. Beyond specialized niche wearable applications in the medical and entertainment domains, we aim to identify the underlying characteristics of smartglass-enabled killer applications for everyday use. In this paper, we analyze intermediary roles in remote collaboration consisting of authoritative supervisors, synergetic collaborators, and speculative explorers. We also propose technical components for smartglass-assisted interactive remote collaboration composed of personal tracking, teleconference, and interaction components. More specifically, the application tasks of each intermediary role and objectives of three functional components are defined and elaborated.

• Electronics and Telecommunications Trends
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• v.37 no.3
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• pp.52-63
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• 2022

Immersive media, also known as spatial media, has become essential with the decrease in face-to-face activities in the COVID-19 pandemic era. Teleconference, metaverse, and digital twin have been developed with high expectations as immersive media services, and the demand for hyper-realistic media is increasing. Under these circumstances, MPEG-I Immersive Media is being standardized as a technologies of navigable virtual reality, which is expected to be launched in the first half of 2024, and the Audio Group is working to standardize the immersive audio technology. Following this trend, this article introduces the trend in MPEG-I immersive audio standardization. Further, it describes the features of the immersive audio rendering technology, focusing on the structure and function of the RM0 base technology, which was chosen after evaluating all the technologies proposed in the January 2022 "MPEG Audio Meeting."

• The Journal of Natural Sciences
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• v.5 no.2
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• pp.33-38
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• 1992

A new texture segmentation-based image coding technique which performs segmentation based on roughness of textural regions and properties of the human visual system (HVS) is presented for multime-dia teleconference. The segmentation is accomplished by thresholding the fractal dimension so that textural regions are classified into three texture classes; perceived constant intensity, smooth texture, and rough texture. An image coding system with high compression and good image quality is achieved by developing an efficient coding technique for each segment boundary and each texture class. We compare the coding efficiency of this technique with that of a well established technique (discrete cosine transform (DCT) image coding).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.6A
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• pp.497-502
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• 2012

In this paper, we present an immersive videoconferencing system that enables gaze correction between users in the internet protocol TV (IPTV) environment. The proposed system synthesizes the gaze corrected images using the depth estimation and the virtual view synthesis algorithms as one of the most important techniques of 3D video system. The conventional processes, however, causes several problems, especially temporal inconsistency of a depth video. This problem leads to flickering artifacts discomforting viewers. Therefore, in order to reduce the temporal inconsistency problem, we exploit the joint bilateral filter which is extended to the temporal domain. In addition, we apply an outlier reduction operation in the temporal domain. From experimental results, we have verified that the proposed system is sufficient to generate the natural gaze-corrected image and realize immersive videoconferencing.

• Journal of Institute of Control, Robotics and Systems
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• v.16 no.5
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• pp.415-419
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• 2010

A variety of methods for sound source localization have been developed and applied to several applications such as noise detection system, surveillance system, teleconference system, robot auditory system and so on. In the previous work, we proposed the sound source localization using the spatially mapped GCC functions based on TDOA for robot auditory system. Performance of the proposed one for the noise effect and estimation resolution was verified with the real environmental experiment under the single source assumption. However, since multi-talker case is general in human-robot interaction, multiple source localization approaches are necessary. In this paper, the proposed localization method under the single source assumption is modified to be suitable for multiple source localization. When there are two sources which are correlated, the spatially mapped GCC function for localization has three peaks at the real source locations and imaginary source location. However if two sources are uncorrelated, that has only two peaks at the real source positions. Using these characteristics, we modify the proposed localization method for the multiple source cases. Experiments with human speeches in the real environment are carried out to evaluate the performance of the proposed method for multiple source localization. In the experiments, mean value of estimation error is about $1.4^{\circ}$ and percentage of multiple source localization is about 62% on average.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.37-42
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• 1999

This paper was studied on change for Marine Administration of telecommunications. Analysis method was studied of Policy Delphi. The three kinds of telecommunications network influence on Marine administration is National Information Infrastructure Network, Integrated Services Digital Network and Local Area Network. The five kinds of telecommunications service influence on Marine administration is Telecommuting Service, Electronic Approval Service, Teleconference Service, Electronic Data Interchange Service, and Internet Service. The results of this as follow: First, all Marine agency must be innovation. Second, Marine Administration must be use Telecommunications. Marine Administration must be for customer.