• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.1067-1071
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• 2004

In this paper, the performance enhancement algorithm of channel allocation for voice and data transmission in cellular networks is proposed. The voice activity detection has been applied to dynamic channel allocation procedure to detect and separate the silence and speech among conversation periods. Hence a data user can use the silent period of an active voice channel to transmit its information. To control the selecting of channel allocation policies, the information of number of data in transmission waiting queue has been determined in order to accept the performance measurement. In the simulation results, the improvement of the performance shows via the quality of services, which are an average delay in queue, a blocking probability, and an impact of the proposed scheme is presented in the system.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.1
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• pp.86-91
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• 2012

To provide high voice quality of real-time voice communication based on telematics exposed to various noise environments, the noise reduction algorithm with low computing load is required to effectively remove the noise. In this paper, we propose a noise reduction algorithm based on Mel-Filter and illustrate the proposed algorithm comparing with conventional noise reduction algorithms. As a experimental result that evaluates the performance of the noise reduction algorithms under the car and babble noise environments, the proposed noise reduction algorithm has the lower computing load with the similar PESQ score compared to the conventional noise reduction algorithms. It proves that the proposed noise reduction algorithm can efficiently remove the noise in mobile telematics.

• English Language & Literature Teaching
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• v.12 no.2
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• pp.215-235
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• 2006

This paper explores the effects of two different methods of instruction for 106 low-level Korean learners of English at a college in learning request expressions. Both of the methods contained the focus-on-form and function characteristics, while the degree of explicitness for input enhancement was differentiated. Abundant email samples written by English native speakers for the input were provided and email writing practice for the output was proceeded for both groups of the students in the treatment sessions. The numbers of target forms used in pretest and posttest results were compared quantitatively: The tests included email writing and open-ended Discourse Completion Test (DCT). The results indicated that the target pragmatic features were slightly better learned under the condition of relatively high degree of explicit instruction with metapragmatic information, even though the difference was statistically insignificant. In addition, the students' use of request strategies both in email and DCT was affected positively by the treatment with email input and output. That is, the students applied the request strategies they learned through email into their oral production (open-ended DCT) as well as their email writing. Further study on the output effect of target features in advancing pragmatic competence is suggested.

• Journal of Technologic Dentistry
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• v.29 no.2
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• pp.105-117
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• 2007

As a dental technician, the aim of the present study on maxillofacial prosthesis was to research its relation with dental technology and further development aspects by looking into its history, kinds, production materials and process. Dental technicians are to expect a great potential to work as maxillofacial prosthetist if having an interest in education of maxillofacial prosthesis field, and developing and operating the education process by expanding the range of dental technology. This article is to present overall history of maxillofacial prosthesis and some background information on the materials which have been used from the past. The maxillofacial field plays essential functions of mastication and speech, as well as performs appearance, which evokes good or bad feelings as an instant and instinctive response. The use of maxillofacial prostheses is not merely the replacement of a missing part of the face, resulted from injuries, but a rehabilitation process to help individuals come back to society. Rehabilitation includes both patient's physical and psychological recovery, such as self-esteem and selfconfidence. There has been a rapid development in application potentials of maxillofacial prosthesis technology which include implant, which can penetrate skin, and new materials. In order to produce maxillofacial prosthesis, general procedures of maxillofacial laboratory work should be understood first. Maxillofacial prosthesis and the dental prosthesis have many similarities in its academic perspective and originality. Maxillofacial prosthesis should be added into the curriculum for dental technology to achieve co-enhancement of the two fields.

• The Journal of the Acoustical Society of Korea
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• v.21 no.7
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• pp.606-613
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• 2002

In this paper we applied a transmit beamforming technique to the underwater acoustic communication system for high rate data transmission. A prototype transmit system was designed and implemented with the general purpose DSP processor and multiple digital-to-analog converters. The performances of the implemented system were evaluated by the experiment in water tank. In order to simplify the procedure the channel coding and equalizer were omitted. And the simplest OOK (On-Off Keying) technique in digital communication methods was applied. The experimental result shows that the transmission data rate is higher about 3 times in the case of 5 hydrophone transmitting may than 1 hydrophone transmitter at bit error rate 10/sup -2/. We verified that the maximum data rate was 400 bps for speech signal transmission in water tank.

• ETRI Journal
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• v.33 no.3
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• pp.423-433
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• 2011

This paper introduces an adaptive and integrated utterance verification (UV) framework using minimum verification error (MVE) training as a new set of solutions suitable for real applications. UV is traditionally considered an add-on procedure to automatic speech recognition (ASR) and thus treated separately from the ASR system model design. This traditional two-stage approach often fails to cope with a wide range of variations, such as a new speaker or a new environment which is not matched with the original speaker population or the original acoustic environment that the ASR system is trained on. In this paper, we propose an integrated solution to enhance the overall UV system performance in such real applications. The integration is accomplished by adapting and merging the target model for UV with the acoustic model for ASR based on the common MVE principle at each iteration in the recognition stage. The proposed iterative procedure for UV model adaptation also involves revision of the data segmentation and the decoded hypotheses. Under this new framework, remarkable enhancement in not only recognition performance, but also verification performance has been obtained.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1C
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• pp.54-60
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• 2005

In this Paper, an adaptive microphone array system with self-delay estimator is proposed. By showing that the adaptive blocking matrix (ABM) of the generalized sidelobe canceller (GSC) can estimate the relative time delay between each sensor, the proposed system utilizes the ABM not only for blocking target components in the blocked signal path, but also for estimating the relative time delay. Therefore, the proposed system requires only the GSC structure while maintaining the system performance similar to the conventional system using an additional time delay estimator as a preprocessor. Simulation results show that the performance of the proposed system is identical to the conventional system that uses an additional time delay estimation module.

• The Journal of the Acoustical Society of Korea
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• v.16 no.1
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• pp.81-87
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• 1997

In this paper, a new pre-processing method for the recognition of single vowels by use of spectrum envelope is presented. We use new extraction method of a spectrum envelope using the diffusion filter bank. By dividing analysis band of a diffusion filter bank into subbands, we decreased the number of diffusion process. And, by increasing the number of difference, we got higher selectivity. As a result of them, we reduced the total processing time, and got higher enhancement of discrimination. By getting 88.3% of average recognition rate for single vowels of natural voice through computer simulation. We confirmed it to be useful for speech recognition which use spectrum analysis of the voice signal to have many frequency components.

• The Korean Journal of Pain
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• v.20 no.2
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• pp.235-239
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• 2007

Spontaneous intracranial hypotension (SIH) is believed to be a benign disease. However, numerous studies have reported serious complications related to SIH, including subdural hematoma. In this case report, a 54-year-old male patient visited the emergency room with orthostatic headache. A brain magnetic resonance imaging (MRI) study showed diffuse mild thickening and enhancement of pachymeninges, with a suspicious minimal amount of subdural fluid collected in the left posterior parietal area. His orthostatic headache showed no improvement with conservative treatment; but his pain was almost completely relieved after two trials of cervical epidural blood patch. On the 74th day after the onset of his pain, the patient showed a drowsy mental status and slurred speech when he visited the pain clinic. Brain computerized tomography indicated a left subdural hemorrhage, and he underwent emergency operation to drain the SDH. In conclusion, pain clinicians should pay attention to abrupt changes in mental status as well as continuous headache, for the early diagnosis of SDH in SIH patients.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.5
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• pp.2574-2587
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• 2019

Blind Source Separation (BSS) is a technique used to separate supposed independent sources of signals from a given set of observations. In this paper, the High Exploration Particle Swarm Optimization (HEPSO) algorithm, which is an enhancement of the Particle Swarm Optimization (PSO) algorithm, has been used to separate a set of source signals. Compared to PSO algorithm, HEPSO algorithm depends on two additional operators. The first operator is based on the multi-crossover mechanism of the genetic algorithm while the second one relies on the bee colony mechanism. Both operators have been employed to update the velocity and the position of the particles respectively. Thus, they are used to find the optimal separating matrix. The proposed method enhances the overall efficiency of the standard PSO in terms of good exploration and performance. Based on many tests realized on speech and music signals supplied by the BSS demo, experimental results confirm the robustness and the accuracy of the introduced BSS technique.