• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8B
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• pp.739-746
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• 2004

An ATM/AAL2 is standardized to transmit delay sensitive application services, which has small size packet, efficiently. An AAL2 transmission scheme is used to deliver voice and data traffic on the lob interface between base station (Node-B) and Radio Network Controller (RNC) in UMTS network. To predict AAL2 performance, a detailed end-to-end UMTS network performance simulator was developed. We performed detailed simulation(cell packing density and bandwidth gain) for voice and data services in UTRAN. The results indicate that the maximum bandwidth gain in Node-B is about 17% and the bandwidth gain of AAL2 multiplexing in $I_{ub}$ for data services is less than that for voice service. Futhermore, the more offered load increase the more the bandwidth gain decreases in a concentrator.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.4
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• pp.227-236
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• 1986

In this paper, an integrated voice/data packet network with window flow control is modeled by a colsed multichain queueing system, and its performance is analyzed by the mean value analysis method. Particularly, for the analysis of a packet network having various kinds of messages with different priority classes, we introduce an approach based on the mean value analysis and the concept of effective capacity. By the mathematical analysis and computer simulation, we obtain the following network statistics in the steady state: Mean buffer occupancy at each node, utilization of link throughput of a virtual channel, and mean delay time of each message. Our iterative analysis method can predict the link data status in most cases within about 10 percent of accurady, and the statistics of voice messages and external data within 5 percent as compared to simulation results.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.12C
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• pp.1660-1668
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• 2004

This paper proposes a Voice Activity Detection (VAD) algorithm based on Radial Basis Function (RBF) network using dual threshold. The k-means clustering and Least Mean Square (LMS) algorithm are used to upade the RBF network to the underlying speech condition. The inputs for RBF are the three parameters in a Code Exited Linear Prediction (CELP) coder, which works stably under various background noise levels. Dual hangover threshold applies in BRF-VAD for reducing error, because threshold value has trade off effect in VAD decision. The experimental result show that the proposed VAD algorithm achieves better performance than G.729 Annex B at any noise level.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.61-64
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• 2003

GPRS is designed for transmitting packet data and supposed to take its radio resource form the pool of channels unused by GSM voice services. In this paper, The GPRS and GSM circuit switched services share the same radio resource. Whenever a channel is not used by circuit switched services, it may be utilized by GPRS. In this paper, the main aim is performance and comparison resource management policies with channel de-allocation in GPRS network. Three resource management policies is voice priority, R-reservation, dynamic reservation.

• International journal of advanced smart convergence
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• v.1 no.2
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• pp.34-38
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• 2012

The recent increase in use of the IEEE 802.15.4 standard for wireless connectivity in personal area networks makes of it an important technology for low-cost low-power wireless personal area networks. Studies showed that voice communications over IEEE 802.15.4 networks is feasible by Guaranteed Time Slot (GTS) allocation; but there are some constraints to accommodate voice transmission beyond two hops due to the excessive transmission delay. In this paper, we propose a GTS allocation scheme for bidirectional voice traffic in IEEE 802.15.4 multihop networks with the goal of achieving fairness and optimization of resource allocation. The proposed scheme uses a greedy algorithm to allocate GTSs to devices for successful completion of voice transmission with efficient use of bandwidth while considering closest devices with another factor for starvation avoidance. We analyze and validate the proposed scheme in terms of fairness and resource optimization through numeral analysis.

• Journal of the Korean Society of Industry Convergence
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• v.16 no.4
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• pp.127-132
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• 2013

We describe a research about remote control of mobile robot based on voice command in this paper. Through real-time remote control and wireless network capabilities of an unmanned remote-control experiments and Home Security / exercise with an unmanned robot, remote control and voice recognition and voice transmission are possible to transmit on a PC using a microphone to control a robot to pinpoint of the source. Speech recognition can be controlled robot by using a remote control. In this research, speech recognition speed and direction of self-driving robot were controlled by a wireless remote control in order to verify the performance of mobile robot with two drives.

• Journal of the Korean Society of Manufacturing Process Engineers
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• v.10 no.6
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• pp.33-38
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• 2011

We describe a research about remote control of mobile robot based on voice command in this paper. Through real-time remote control and wireless network capabilities of an unmanned remote-control experiments and Home Security / exercise with an unmanned robot, remote control and voice recognition and voice transmission are possible to transmit on a PC using a microphone to control a robot to pinpoint of the source. Speech recognition can be controlled robot by using a remote control. In this research, speech recognition speed and direction of self-driving robot were controlled by a wireless remote control in order to verify the performance of mobile robot with two drives.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.9
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• pp.373-383
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• 2003

In this paper we discuss the effect of the delay limit and the packet size related to the quality of service on a VoIP system using the Internet. We also provide a guideline to determining the optimal packet size of the voice data for a given delay limit. Empirical studies are done with two personal computers connected through the packet switched public IP network. The sender encodes the voice signal from the microphone to get PCM and ADPCM data and sends the data to the receiver using UDP packets. The receiver plays the reconstructed voice from the stream with lost and delayed packets. The quality of the reconstructed voice is evaluated offline by the MNB (Measuring Normal Block) method using the data acquired from the both sides. The result shows that under the delay limit of 100ms for 40Kbps, 32Kbps and l6Kbps of ADPCM data, the minimum packet size should be 300bytes, 400bytes and 600bytes respectively and the maximum packet size should be l200bytes commonly for the best quality of voice.

• Dementia and Neurocognitive Disorders
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• v.23 no.1
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• pp.1-10
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• 2024

Background and Purpose: Voice, reflecting cerebral functions, holds potential for analyzing and understanding brain function, especially in the context of cognitive impairment (CI) and Alzheimer's disease (AD). This study used voice data to distinguish between normal cognition and CI or Alzheimer's disease dementia (ADD). Methods: This study enrolled 3 groups of subjects: 1) 52 subjects with subjective cognitive decline; 2) 110 subjects with mild CI; and 3) 59 subjects with ADD. Voice features were extracted using Mel-frequency cepstral coefficients and Chroma. Results: A deep neural network (DNN) model showed promising performance, with an accuracy of roughly 81% in 10 trials in predicting ADD, which increased to an average value of about 82.0%±1.6% when evaluated against unseen test dataset. Conclusions: Although results did not demonstrate the level of accuracy necessary for a definitive clinical tool, they provided a compelling proof-of-concept for the potential use of voice data in cognitive status assessment. DNN algorithms using voice offer a promising approach to early detection of AD. They could improve the accuracy and accessibility of diagnosis, ultimately leading to better outcomes for patients.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.2
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• pp.108-121
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• 1986

In this paper, the development of a simulator for the performance estimation and parameter optimization of an integrates voice/data packet communication network is described. The simulator implemented is capable of simulating the integrated voice/data network that handles packet voice terminals as well as data terminals and hosts operating under standard CCITT protocols. Of the three descrete event simulation approaches presently known, the process interaction method has been chose. With this approach one can implement a simulator that is related most Closely with the real system. The simulator has been implemented in PL/I and GPSS simulation languages, resulting in a software package of about 4,000 lines. To reduce the computer run time of the simulator, we have used a method of reducing conditional events based on a GPSS LINK block. We describe various aspects of the simulation model developed. We then investigate the performance of a 7-node network using the simulator, and present the results. For validation of the simulator developed, we construct a simulation model for a simple voice/ data multiplexer, and compare the results of simulation with those of an analytical model.